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Upgrade OpenSSL and A2Billing now
Click to view a printable version Thu, 17 Apr 2014 05:19:37 +1200

The A2Billing Team has posted a note asking people to upgrade their systems urgently:

It has been widely reported that a recently discovered bug in OpenSSL compromises the security of communications data. Any A2Billing administrator who has installed an SSL certificate to provide HTTPS encryption on their website should upgrade OpenSSL.

There is more information at http://heartbleed.com/. To check whether your site is affected, go to http://filippo.io/Heartbleed/ and enter the domain name to check.

We've also had a 3rd party security expert audit A2Billing in the interests of increasing security. They have found a major security issue. We've no evidence that this exploit is in the wild, but we advise in the strongest possible terms that you upgrade to A2Billing version 2.0.8 as a matter of urgency.

If you need assistance with upgrading your operating system or A2Billing or an installation with transfer of A2Billing data, then please contact us at sales@star2billing.com

Yours,
The A2Billing Team.
http://www.asterisk2billing.org/



JIRA, Commit Messages, and Smart Commits (oh my)
Click to view a printable version Fri, 11 Apr 2014 05:12:38 +1200

Matt Jordan has posted details of some changes to integration between JIRA and SVN:

Hey everyone -

Due to a security vulnerability in JIRA, we recently had to upgrade JIRA to version 6.2. While this was a good thing (tm), it did break the snot out of the Subversion plugin. That plugin does a number of things; one of those things is auto-closing issues based on the commit message.

Traditionally, the plugin used the following nomenclature to close an issue:

(closes issue JIRA_ISSUE)

Likewise, tagging a commit with the following linked the commit message to the issue:

(issue JIRA_ISSUE)

While I'd like to believe the Subversion plugin will get fixed at some point, our initial forays into getting it fixed haven't been fruitful. Part of this is reticence on our part to 'customize' the plugin or our JIRA instance further (more on that in a bit); another large part is Atlassian appears to be focussing on an alternate method for tying commit messages back to issues (more on that in a bit as well). Long story short, it doesn't appear like anyone is in a hurry to fix the old Subversion plugin.

Question: why not hack on the Subversion plugin and/or JIRA?

Answer: we could hack around on the Subversion plugin and/or JIRA (it isn't clear yet who is at fault), but the more customization we put into it the harder it is to upgrade the project's JIRA instance. This makes it harder to respond quickly to security vulnerabilities. We've done a lot of work over the past year or so minimizing the customizations so that we can respond quickly to said vulnerabilities in JIRA, and it's been very handy. Unless there's a giant outcry against what I'm about to propose, I'd hope that we don't have to try and modify the plugin ourselves.

Question: what is this alternate method that Atlassian has?

Answer: Atlassian Smart Commits.

Since we have Fisheye integration with JIRA, we can use a different commit message format that will trigger the smart commits in Fisheye, which will do all of the auto-closing for us in JIRA. In fact, as the name 'smart commit' implies, it can do a lot more than just close the issue, including leaving comments and other fanciness. It does mean however that our commit message format will have to change.

The biggest implication of changing the commit message format is making sure that patch submitters are properly attributed, such that the release summary scripts and other parts of the Asterisk project pick up who actually wrote the patch. Right now, I'm proposing something like the following:

ASTERISK-12345 #close
ASTERISK-12345 #comment patch my_awesome_fix.diff submitted by jdoe (license 12345)

You can see a commit that contains some smart commit tags in it here, which use a format similar to what is above.

What do you all think?

Matt

--
Matthew Jordan
Digium, Inc. | Engineering Manager



Call center satisfaction survey - results are out
Click to view a printable version Wed, 09 Apr 2014 04:27:27 +1200

Lenz from QueueMetrics has posted a link to their free eBook on Call Center satisfaction. Sorry for the delay in getting this up:

We are pleased to announce that our first call-centre customer satisfaction survey is online as a free eBook. We want to thank you all for supporting us in this research and answering to our questions.

According to our corporate philosophy that binds accurate data measurement to relevant business improvements, in 2013 we sent you a survey about the usage of Asterisk® and QueueMetrics™ for your daily needs in call-centers management, analysis and statistics. This was done to assess your perceived needs so that we could prioritize future improvements.

The results are quite relevant to all professionals who create or manage call-centers with Asterisk®. Too often, an Asterisk-based PBX is still perceived as a risky proposition by clients switching from other systems. But definitely Asterisk® proved to be a valid platform to build contact centres with, and grants its users an impressive level of customer satisfaction.

The survey 2014 is available for free download at http://queuemetrics.com/callcenter-survey.jsp

Do not hesitate to send us your feedbacks and suggestions. Your opinion is precious for us!

Warm regards,

Lorenzo Emilitri
Founder, Loway



Changes to Review Board Access
Click to view a printable version Fri, 07 Mar 2014 04:47:43 +1300

Matt Jordan has posted details of a change to review board access:

Hello everyone!

For some time now, the Asterisk project has used Review Board for performing code reviews on submitted patches. While having a patch be formally reviewed has never been a hard requirement for its inclusion, over time, the status quo in the project has evolved such that patches of any reasonable complexity are nearly always reviewed before being included. This is not a bad thing - code review is an invaluable tool in ensuring quality in the Asterisk project. However, some technical limitations with how accounts are configured in Review Board made it difficult for everyone to submit patches and participate in the review process. The situation today is that the vast majority of patches on the issue tracker are only looked at when they are first put up for code review. This has led to some high quality patches not being included in the Asterisk project as fast as they otherwise could.

This weekend, we finalized the integration of Review Board with Atlassian Crowd, the service that provides user identification and authentication for the rest of the Asterisk community services. This removes the bottleneck - namely, me! - that prevented any contributor from submitting patches for peer review.

For users who currently have an account in Review Board, if your username is the same as your Atlassian (JIRA) username, simply use your Atlassian password when logging in. Reviews that you currently have open will still be associated with you. If your existing username in Review Board is different than your Atlassian username, you will unfortunately need to re-open the reviews you have in Review Board. If that happens to be the case, contact me in #asterisk-dev or reply to this e-mail and we'll work it out.

For users who do not currently have an account in Review Board, if you signed a License Contributor Agreement in JIRA, this change opens Review Board up to you. Instructions for posting patches to Review Board can be found on the Asterisk wiki, as well as workflow guidelines for participating in code reviews. The wiki has other items to help you prepare a patch for review, including a check list of items to be aware of when performing a review or submitting a patch, as well as the project coding guidelines. Finally, because this change opens up Review Board to a much larger audience, the patch submission process has been clarified on the Asterisk wiki.

Please remember that this may greatly increase the volume of code reviews being submitted. Contributors are highly encouraged to participate in other code reviews as well, and to be patient with any submission they make. Patches that are accompanied by well written explanations, conform to the coding guidelines, and have accompanying unit tests and/or functional tests in the Asterisk Test Suite are always easier to review and will naturally move through the review process faster.

As always, thank-you for your support and participation in the Asterisk project - we hope that these changes make the process easier and more beneficial for everyone.

Matt

--
Matthew Jordan
Digium, Inc. | Engineering Manager



New Asterisk Developer - George Joseph
Click to view a printable version Fri, 07 Mar 2014 04:42:11 +1300

Matt Jordan has posted details of a new Asterisk Developer:

Hey everyone!

George Joseph has been doing a lot of great work on the new PJSIP stack for Asterisk 12. This includes adding the PJSIP_HEADER function, allowing users to add, modify, and remove headers from SIP messages; adding just about all of the CLI commands for the PJSIP modules; updating the Asterisk build system to support Lua 5.2 for pbx_lua; and fixing several bugs in the new PJSIP stack. Along the way George has done some serious work ironing out the kinks in the code that formats PJSIP sorcery-derived objects for various interfaces. We're thrilled to extend George commit access for the Asterisk project, and look forward to more of his contributions to the project and the new PJSIP stack.

Welcome George Joseph!

(And we're sorry you have to use subversion to commit things. Git... some day, some day.)

Matt

--
Matthew Jordan
Digium, Inc. | Engineering Manager



Asterisk 12.1.0 Now Available
Click to view a printable version Wed, 05 Mar 2014 04:40:27 +1300

The Asterisk Development Team has announced the release of Asterisk 12.1.0.

This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 12.1.0 resolves several issues reported by the community and would have not been possible without your participation.

Thank you!

The following are the issues resolved in this release:

New Features made in this release:

  • ASTERISK-23038 - Need config option to enable PJSIP logger at load time (Reported by Rusty Newton)

Bugs fixed in this release

  • ASTERISK-23051 - ARI: channel variables in JSON breaks passing parameters in JSON (Reported by Matt Jordan)
  • ASTERISK-22952 - res_pjsip_pubsub: crash when subscription_destructor is terminated from a non-PJSIP thread (Reported by Matt Jordan)
  • ASTERISK-22486 - ARI: TCP Reset after 204 response (Reported by David M. Lee)
  • ASTERISK-22854 - [patch] - Deadlock between cel_pgsql unload and core_event_dispatcher taskprocessor thread (Reported by Etienne Lessard)
  • ASTERISK-23074 - Crash in ChanIsAvail app (Reported by Kilburn)
  • ASTERISK-22910 - [patch] - REPLACE() calls strcpy on overlapping memory when is empty (Reported by Gareth Palmer)
  • ASTERISK-22871 - cel_pgsql module not loading after "reload" or "reload cel_pgsql.so" command (Reported by Matteo)
  • ASTERISK-23084 - [patch]rasterisk needlessly prints the AST-2013-007 warning (Reported by Tzafrir Cohen)
  • ASTERISK-23101 - pjsip: crash when parsing scheme from SIP URI (Reported by Matt Jordan)
  • ASTERISK-17138 - [patch] Asterisk not re-registering after it receives "Forbidden - wrong password on authentication" (Reported by Rudi)
  • ASTERISK-23011 - [patch]configure.ac and pbx_lua don't support lua 5.2 (Reported by George Joseph)
  • ASTERISK-23053 - The users of ao2_iterator_cleanup() are violating the ao2_iterator opacity. (Reported by Richard Mudgett)
  • ASTERISK-22924 - PJSIP MESSAGE support does not present the contact information on outbound messages (Reported by Anthony Messina)
  • ASTERISK-22884 - hangup_handler end with h extension: tests currently fail in Asterisk 12 + (Reported by Matt Jordan)
  • ASTERISK-23128 - res_ari: Memory leak on response headers and some JSON response messages (Reported by Joshua Colp)
  • ASTERISK-23081 - PJSip Tab Expansion erroring (Reported by xrobau)
  • ASTERISK-22946 - Local From tag regression with sipgate.de (Reported by Stephan Eisvogel)
  • ASTERISK-23065 - On Asterisk start, device state is INVALID for previously registered PJSIP endpoints, despite re-registrations (Reported by Rusty Newton)
  • ASTERISK-22790 - check_modem_rate() may return incorrect rate for V.27 (Reported by Paolo Compagnini)
  • ASTERISK-23034 - [patch] manager Originate doesn't abort on failed format_cap allocation (Reported by Corey Farrell)
  • ASTERISK-23062 - res_pjsip AOR config option qualify_frequency is inconsistently respected (Reported by Rusty Newton)
  • ASTERISK-23071 - pjsip: mailboxes documentation is lacking (Reported by Matt Jordan)
  • ASTERISK-23061 - [Patch] 'textsupport' setting not mentioned in sip.conf.sample (Reported by Eugene)
  • ASTERISK-23028 - [patch] Asterisk man pages contains unquoted minus signs (Reported by Jeremy Lainé)
  • ASTERISK-23046 - Custom CDR fields set during a GoSUB called from app_queue are not inserted (Reported by Denis Pantsyrev)
  • ASTERISK-23027 - [patch] Spelling typo "transfered" instead of "transferred" (Reported by Jeremy Lainé)
  • ASTERISK-23018 - PJSip 'allow=all' results in failed SDP negotiation (Reported by xrobau)
  • ASTERISK-23008 - Local channels loose CALLERID name when DAHDI channel connects (Reported by Michael Cargile)
  • ASTERISK-23051 - ARI: channel variables in JSON breaks passing parameters in JSON (Reported by Matt Jordan)
  • ASTERISK-23100 - [patch] In chan_mgcp the ident in transmitted request and request queue may differ - fix for locking (Reported by adomjan)
  • ASTERISK-22988 - [patch]T38 , SIP 488 after Rejecting image media offer due to invalid or unsupported syntax (Reported by adomjan)
  • ASTERISK-22861 - [patch]Specifying a null time as parameter to GotoIfTime or ExecIfTime causes segmentation fault (Reported by Sebastian Murray-Roberts)
  • ASTERISK-23177 - [patch] RealTime cant update sipbuddies table when registering or updating friend (Reported by Denis)
  • ASTERISK-23082 - Including g722 in pjsip codec configuration results in unexpected SDP offers (Reported by xrobau)
  • ASTERISK-17837 - extconfig.conf - Maximum Include level (1) exceeded (Reported by pz)
  • ASTERISK-23143 - ARI: subscribing to an already subscribed resource returns a 500 error (Reported by Matt Jordan)
  • ASTERISK-23056 - [patch]INFINITY and NAN undefined (Reported by capouch)
  • ASTERISK-23129 - segfault in res_pjsip_pubsub.so (Reported by Dan Jenkins)
  • ASTERISK-22662 - Documentation fix? - queues.conf says persistentmembers defaults to yes, it appears to lie (Reported by Rusty Newton)
  • ASTERISK-23134 - [patch] res_rtp_asterisk port selection cannot handle selinux port restrictions (Reported by Corey Farrell)
  • ASTERISK-23106 - pjsip: ACK to 200 OK sent to private IP address on outbound channel's INVITE request (Reported by Matt Jordan)
  • ASTERISK-23072 - MWI subscription from Cisco SPA fails with PJSIP (Reported by Bob M)
  • ASTERISK-23164 - CDRs: mid-call/pre-dial handlers perturb context/exten/app/data fields during Dial (Reported by Matt Jordan)
  • ASTERISK-23220 - STACK_PEEK function with no arguments causes crash/core dump (Reported by James Sharp)
  • ASTERISK-23249 - Skinny subchannel locking issues (Reported by Damien Wedhorn)
  • ASTERISK-19773 - Asterisk crash on issuing Asterisk-CLI 'reload' command multiple times on cli_aliases (Reported by Joel Vandal)
  • ASTERISK-22757 - segfault in res_clialiases.so on reload when mapping "module reload" command (Reported by Gareth Blades)
  • ASTERISK-23250 - CDR(start) function is broken due to sizeof dereference (Reported by snuffy)
  • ASTERISK-17727 - [patch] TLS doesn't get all certificate chain (Reported by LN)
  • ASTERISK-23168 - Overriding outbound_auth in a pjsip registration causes ERROR, assert failure. (Reported by George Joseph)
  • ASTERISK-23178 - devicestate.h: device state setting functions are documented with the wrong return values (Reported by Jonathan Rose)
  • ASTERISK-23231 - Since 405693 If we have res_fax.conf file set to minrate=2400, then res_fax refuse to load (Reported by David Brillert)

Improvements made in this release

  • ASTERISK-22919 - core show channeltypes slicing (Reported by outtolunc)
  • ASTERISK-22868 - chan_pjsip: 'setvar' should be supported on endpoints (Reported by Joshua Colp)
  • ASTERISK-22918 - dahdi show channels slices PRI channel dnid on output (Reported by outtolunc)
  • ASTERISK-21084 - New SIP Channel Driver - Path Support (Reported by Matt Jordan)
  • ASTERISK-23068 - http: Implement support for chunked Transfer-Encoding (Reported by Matt Jordan)
  • ASTERISK-22980 - [patch]Allow building cdr_radius and cel_radius against libfreeradius-client (Reported by Jeremy Lainé)
  • ASTERISK-22984 - ari: Transfer messages not being sent out ARI WebSocket (Reported by David M. Lee)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-12.1.0

Thank you for your continued support of Asterisk!



Asterisk 11.8.0 Now Available
Click to view a printable version Wed, 05 Mar 2014 04:33:15 +1300

The Asterisk Development Team has announced the release of Asterisk 11.8.0.

This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 11.8.0 resolves several issues reported by the community and would have not been possible without your participation.

Thank you!

The following are the issues resolved in this release:

Bugs fixed in this release:

  • ASTERISK-22544 - Italian prompt vm-options has advertisement in it (Reported by Rusty Newton)
  • ASTERISK-21383 - STUN Binding Requests Not Being Sent Back from Asterisk to Chrome (Reported by Shaun Clark)
  • ASTERISK-22478 - [patch]Can't use pound(hash) symbol for custom DTMF menus in ConfBridge (processed as directive) (Reported by Nicolas Tanski)
  • ASTERISK-12117 - chan_sip creates a new local tag (from-tag) for every register message (Reported by Pawel Pierscionek)
  • ASTERISK-20862 - Asterisk min and max member penalties not honored when set with 0 (Reported by Schmooze Com)
  • ASTERISK-22746 - [patch]Crash in chan_dahdi during caller id read (Reported by Michael Walton)
  • ASTERISK-22788 - [patch] main/translate.c: access to variable f after free in ast_translate() (Reported by Corey Farrell)
  • ASTERISK-21242 - Segfault when T.38 re-invite retransmission receives 200 OK (Reported by Ashley Winters)
  • ASTERISK-22590 - BufferOverflow in unpacksms16() when receiving 16 bit multipart SMS with app_sms (Reported by Jan Juergens)
  • ASTERISK-22905 - Prevent Asterisk functions that are 'dangerous' from being executed from external interfaces (Reported by Matt Jordan)
  • ASTERISK-23021 - Typos in code : "avaliable" instead of "available" (Reported by Jeremy Lainé)
  • ASTERISK-22970 - [patch]Documentation fix for QUOTE() (Reported by Gareth Palmer)
  • ASTERISK-21960 - ooh323 channels stuck (Reported by Dmitry Melekhov)
  • ASTERISK-22350 - DUNDI - core dump on shutdown - segfault in sqlite3_reset from /usr/lib/libsqlite3.so.0 (Reported by Birger "WIMPy" Harzenetter)
  • ASTERISK-22942 - [patch] - Asterisk crashed after Set(FAXOPT(faxdetect)=t38) (Reported by adomjan)
  • ASTERISK-22856 - [patch]SayUnixTime in polish reads minutes instead of seconds (Reported by Robert Mordec)
  • ASTERISK-22854 - [patch] - Deadlock between cel_pgsql unload and core_event_dispatcher taskprocessor thread (Reported by Etienne Lessard)
  • ASTERISK-22910 - [patch] - REPLACE() calls strcpy on overlapping memory when is empty (Reported by Gareth Palmer)
  • ASTERISK-22871 - cel_pgsql module not loading after "reload" or "reload cel_pgsql.so" command (Reported by Matteo)
  • ASTERISK-23084 - [patch]rasterisk needlessly prints the AST-2013-007 warning (Reported by Tzafrir Cohen)
  • ASTERISK-17138 - [patch] Asterisk not re-registering after it receives "Forbidden - wrong password on authentication" (Reported by Rudi)
  • ASTERISK-23011 - [patch]configure.ac and pbx_lua don't support lua 5.2 (Reported by George Joseph)
  • ASTERISK-22834 - Parking by blind transfer when lot full orphans channels (Reported by rsw686)
  • ASTERISK-23047 - Orphaned (stuck) channel occurs during a failed SIP transfer to parking space (Reported by Tommy Thompson)
  • ASTERISK-22946 - Local From tag regression with sipgate.de (Reported by Stephan Eisvogel)
  • ASTERISK-23010 - No BYE message sent when sip INVITE is received (Reported by Ryan Tilton)
  • ASTERISK-23135 - Crash - segfault in ast_channel_hangupcause_set - probably introduced in 11.7.0 (Reported by OK)

Improvements made in this release:

  • ASTERISK-22728 - [patch] Improve Understanding Of 'Forcerport' When Running "sip show peers" (Reported by Michael L. Young)
  • ASTERISK-22659 - Make a new core and extra sounds release (Reported by Rusty Newton)
  • ASTERISK-22919 - core show channeltypes slicing (Reported by outtolunc)
  • ASTERISK-22918 - dahdi show channels slices PRI channel dnid on output (Reported by outtolunc)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.8.0

Thank you for your continued support of Asterisk!



Asterisk 1.8.26.0 Now Available
Click to view a printable version Wed, 05 Mar 2014 04:24:11 +1300

The Asterisk Development Team has announced the release of Asterisk 1.8.26.0.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 1.8.26.0 resolves several issues reported by the community and would have not been possible without your participation.

Thank you!

The following are the issues resolved in this release:

Bugs fixed in this release:

  • ASTERISK-22544 - Italian prompt vm-options has advertisement in it (Reported by Rusty Newton)
  • ASTERISK-12117 - chan_sip creates a new local tag (from-tag) for every register message (Reported by Pawel Pierscionek)
  • ASTERISK-20862 - Asterisk min and max member penalties not honored when set with 0 (Reported by Schmooze Com)
  • ASTERISK-22746 - [patch]Crash in chan_dahdi during caller id read (Reported by Michael Walton)
  • ASTERISK-22788 - [patch] main/translate.c: access to variable f after free in ast_translate() (Reported by Corey Farrell)
  • ASTERISK-21242 - Segfault when T.38 re-invite retransmission receives 200 OK (Reported by Ashley Winters)
  • ASTERISK-22590 - BufferOverflow in unpacksms16() when receiving 16 bit multipart SMS with app_sms (Reported by Jan Juergens)
  • ASTERISK-22905 - Prevent Asterisk functions that are 'dangerous' from being executed from external interfaces (Reported by Matt Jordan)
  • ASTERISK-23021 - Typos in code : "avaliable" instead of "available" (Reported by Jeremy Lainé)
  • ASTERISK-22970 - [patch]Documentation fix for QUOTE() (Reported by Gareth Palmer)
  • ASTERISK-22856 - [patch]SayUnixTime in polish reads minutes instead of seconds (Reported by Robert Mordec)
  • ASTERISK-22854 - [patch] - Deadlock between cel_pgsql unload and core_event_dispatcher taskprocessor thread (Reported by Etienne Lessard)
  • ASTERISK-22910 - [patch] - REPLACE() calls strcpy on overlapping memory when is empty (Reported by Gareth Palmer)
  • ASTERISK-22871 - cel_pgsql module not loading after "reload" or "reload cel_pgsql.so" command (Reported by Matteo)
  • ASTERISK-23084 - [patch]rasterisk needlessly prints the AST-2013-007 warning (Reported by Tzafrir Cohen)
  • ASTERISK-17138 - [patch] Asterisk not re-registering after it receives "Forbidden - wrong password on authentication" (Reported by Rudi)
  • ASTERISK-23011 - [patch]configure.ac and pbx_lua don't support lua 5.2 (Reported by George Joseph)
  • ASTERISK-22834 - Parking by blind transfer when lot full orphans channels (Reported by rsw686)
  • ASTERISK-23047 - Orphaned (stuck) channel occurs during a failed SIP transfer to parking space (Reported by Tommy Thompson)
  • ASTERISK-22946 - Local From tag regression with sipgate.de (Reported by Stephan Eisvogel)
  • ASTERISK-23010 - No BYE message sent when sip INVITE is received (Reported by Ryan Tilton)

Improvements made in this release:

  • ASTERISK-22659 - Make a new core and extra sounds release (Reported by Rusty Newton)
  • ASTERISK-22918 - dahdi show channels slices PRI channel dnid on output (Reported by outtolunc)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.26.0

Thank you for your continued support of Asterisk!



DAHDI-Linux and DAHDI-Tools 2.9.0 Now Available
Click to view a printable version Tue, 25 Feb 2014 06:15:31 +1300

The Asterisk Development Team has announced the releases of:
DAHDI-Linux-v2.9.0
DAHDI-Tools-v2.9.0
dahdi-linux-complete-2.9.0+2.9.0

This release is available for immediate download at:
http://downloads.asterisk.org/pub/telephony/dahdi-linux
http://downloads.asterisk.org/pub/telephony/dahdi-tools
http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete

  • Introduces support for Digium's new TE131 and TE132 products.
  • Updates firmware for existing TE133 and TE134 products.
  • New documentation and support tool improvements for configurable span/channel numbering
    • Currently, span/channel ordering is determined by module load order
    • Work arounds are used to specify channel assignment order by blacklisting all modules and then loading them in a specific order to preserve channel assignments.
    • We have been driving towards moving span/chan assignments out of kernel space and into user space.
    • This is a much more robust solution which allows for:
      • hotplugging, surprise device removal and installation while maintaining channel ordering
      • parallel module loading (much faster booting on dense systems)
      • discrete control over span and channel ordering via configuration files
      • "sticky" channel assignments which can be tied to specific hardware ids or pci slots


    • This new system is enabled by setting the module parameter of dahdi auto_assign_spans=0
    • More info here: http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.user/278656/match=auto_assigned_spans



Shortlog of dahdi-linux changes since v2.8.0.1:
Oron Peled (3):
xpp: deprecate dahdi_autoreg
xpp: continue xpp.dahdi_autoreg deprecation
sysfs: new device attribute: registration_time

Russ Meyerriecks (6):
wcte13xp: wcaxx: Fix broken devicetype attributes
wcte13xp: Update firmware to 0x780017
wcte13xp: Add support for te131 and te132 products
Revert "dahdi: Change auto_assign_spans default from 1 to 0."
wcte13xp: wcaxx: wcte43x: Remove VPM_SUPPORT compile option.
wcte13xp: wcxb: Add delayed reset firmware feature

Shaun Ruffell (10):
wctdm24xxp: Reset module specific type information on probe.
dahdi: Move clearing of DAHDI_ALARM_NOTOPEN to __dahdi_assign_span().
dahdi: Change auto_assign_spans default from 1 to 0.
wcaxx, wcte13xp, wcte43x: Honor max_latency module parameter.
wcte13xp: Export max_latency module parameter.
wcte43x, wcte13xp: Use MSI interrupts if possible.
dahdi: Do not access invalid memory if invalid local span number is passed to spantype attribute.
wcte43x: Trivial drop of unnecessary local variables.
wct4xxp: Trivial drop of unnecessary local variables.
wcte43x, wcte13xp, wcaxx: Bump irqmisses counter when there are DMA underruns.

Tzafrir Cohen (4):
README: xpp.dahdi_autoreg is deprecated
README: the new registration_time device attribute
README: The sysfs class now includes no channels
sysfs: registration_time: use ktime_get_ts


Shortlog of dahdi-tools changes since v2.8.0:
Oron Peled (6):
Makefile: do install all man-pages
hotplug modularization: move sources to a subdir
hotplug modularization: split logic to scriptlets
new "dahdi_waitfor_span_assignments" tool
dahdi_span_types: allow defaults + overrides
Change span-type.conf generation policy

Russ Meyerriecks (2):
wcte13xp: Teach tools about te131 te132 products
dahdi.init: Don't exit on lack of /etc/dahdi/system.conf

Shaun Ruffell (8):
dahdi_cfg: Wait for all spans to be assigned.
dahdi_span_config: Do not run auto span configuration if spans are auto assigned.
dahdi_handle_device, dahdi_span_config: Check for auto_assign_spans only when ACTION is add.
dahdi_genconf: Add 'modules', 'spantypes', and 'assignedspans' to list of available generators.
dahdi_span_types: Show location of configuration file in help message.
dahdi_handle_device: Auto assign only the device being added.
dahdi_cfg: Add semaphore to prevent parallel execution.
dahdi_cfg: Allow dynamic spans to handle udev based span assignment.

Tzafrir Cohen (16):
dahdi.rules: Replace SYSFS with ATTRS
dahdi.rules: use += for RUN
.gitignore: more generated files
README: indentation level for config samples
README: document initialization
README: Update the install targets
span_types/assignments: no * in device list
dahdi_genconf: don't generate spantypes by default
dahdi_span_assignments.8: s/register/assign/
dahdi_span_types: hush warning of missing attribute
programmable bash completion for some commands
dahdi_perl: fix regression with an AB with no modules
bash_completion: fix dahdi_genconf
hyphen/minus fixes in man pages
hotplug: document asterisk scriptlet
README: udev hooks run scripts from directories


The diffstat from the dahdi-linux v2.8.0.1 release:


README | 26 +++++----
drivers/dahdi/dahdi-base.c | 23 +++++---
drivers/dahdi/dahdi-sysfs.c | 36 +++++++++---
drivers/dahdi/firmware/Makefile | 4 +-
drivers/dahdi/wcaxx-base.c | 28 ++++-----
drivers/dahdi/wct4xxp/base.c | 9 +--
drivers/dahdi/wctdm24xxp/base.c | 5 +-
drivers/dahdi/wcte13xp-base.c | 119 ++++++++++++++++++++-------------------
drivers/dahdi/wcte43x-base.c | 38 ++++---------
drivers/dahdi/wcxb.c | 92 ++++++++++++++++++++++++------
drivers/dahdi/wcxb.h | 10 +++-
drivers/dahdi/xpp/xbus-core.c | 10 +++-
include/dahdi/kernel.h | 2 +
13 files changed, 245 insertions(+), 157 deletions(-)

The diffstat from the dahdi-tools v2.8.0 release:
.gitignore | 14 ++
Makefile | 28 +++-
README | 148 ++++++++++++++++--
dahdi-bash-completion | 133 ++++++++++++++++
dahdi.init | 5 -
dahdi.rules | 8 +-
dahdi_cfg.c | 193 +++++++++++++++++++++---
dahdi_handle_device | 80 ----------
dahdi_span_assignments | 2 +-
dahdi_span_config | 99 ------------
dahdi_span_types | 175 ++++++++++++++-------
dahdi_waitfor_span_assignments | 73 +++++++++
doc/dahdi_cfg.8 | 2 +-
doc/dahdi_maint.8 | 4 +-
doc/dahdi_monitor.8 | 24 +--
doc/dahdi_span_assignments.8 | 113 ++++++++------
doc/dahdi_span_types.8 | 107 +++++++++----
doc/dahdi_waitfor_span_assignments.8 | 49 ++++++
hotplug/dahdi_handle_device | 85 +++++++++++
hotplug/dahdi_span_config | 83 ++++++++++
hotplug/handle_device.d/10-span-types | 5 +
hotplug/handle_device.d/20-span-assignments | 8 +
hotplug/span_config.d/10-dahdi-cfg | 28 ++++
hotplug/span_config.d/20-fxotune | 12 ++
hotplug/span_config.d/50-asterisk | 14 ++
modules.sample | 2 +
system.conf.sample | 14 +-
xpp/dahdi_genconf | 59 +++++++-
xpp/perl_modules/Dahdi/Config/Gen/Spantypes.pm | 22 ++-
xpp/perl_modules/Dahdi/Hardware/PCI.pm | 4 +-
xpp/perl_modules/Dahdi/Span.pm | 6 +-
xpp/perl_modules/Dahdi/Xpp/Xbus.pm | 4 +-
32 files changed, 1216 insertions(+), 387 deletions(-)



For a full list of changes in these releases, please see the shortlog at:
http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=shortlog;h=refs/tags/v2.9.0
http://git.asterisk.org/gitweb/?p=dahdi/tools.git;a=shortlog;h=refs/tags/v2.9.0

Issues found in this release can be reported in the DAHDI-Linux and DAHDI-Tools projects at https://issues.asterisk.org/jira

Thank you for your continued support of Asterisk!



QueueMetrics 13.12 released
Click to view a printable version Tue, 24 Dec 2013 07:15:46 +1300

Lenz has posted details of the release of QueueMetrics version 13.12:

We are happy to announce the release of QueueMetrics version 13.12!

Release 13.12 addresses usability and bug fixes in a number of areas.The main new features included are:

  • Externally-generated report pages
  • Improved QA
  • Easier integration to external CRMs
  • Ready for Asterisk 12
  • A long set of small improvements on all pages

In total, over 70 issues were fixed/added in this release.

The full release notes - with all the fixed bugs and new features - can be found here: Release notes for QueueMetrics 13.12

A broader overview of the newly released features can be found here: What's new in QueueMetrics - release 13.12



Asterisk 12 released!
Click to view a printable version Tue, 24 Dec 2013 07:08:45 +1300

The Asterisk Development Team is pleased to announce the release of Asterisk 12.0.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases

Asterisk 12 is the next major release series of Asterisk. It is a Standard release, similar to Asterisk 10. For more information about support time lines for Asterisk releases, see the Asterisk versions page:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

For important information regarding upgrading to Asterisk 12, please see the Asterisk wiki:

https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+12

As a Standard Release, Asterisk 12 contains many new major architectural improvements and features. A short list of some of these features includes:

  • A new SIP channel driver and accompanying SIP stack named chan_pjsip has been added. This new channel driver is based on the PJSIP SIP stack by Teluu. It includes support for the vast majority of features currently in chan_sip, as well as numerous architectural improvements that alleviate pain points present in the legacy SIP channel driver. Users who wish to use the new SIP channel driver are encouraged to read the instructions on installing and configuring PJSIP for Asterisk on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/J4GLAQ. Detailed instructions on configuring the new SIP stack in Asterisk can be found on the Asterisk wiki as well, at https://wiki.asterisk.org/wiki/x/hYCLAQ.
  • The Asterisk REST Interface (ARI) has been added. This interface lets external systems harness the telephony primitives within Asterisk to develop their own communications applications. Communication with Asterisk is done through a RESTful interface, while asynchronous events from Asterisk are encoded in JSON and sent via a WebSocket. More information on ARI can be found at https://wiki.asterisk.org/wiki/x/lYBbAQ
  • Major standardization of the Asterisk Manager Interface and its events have occurred within this version. In particular, the names of Asterisk channels no longer change and are stable throughout the lifetime of the channel. More information on the changes in AMI can be seen in the AMI v2 Specification at https://wiki.asterisk.org/wiki/x/dAFRAQ
  • All bridging within Asterisk is now performed using the Asterisk Bridging API, which previously was only used by the ConfBridge application. This affords Asterisk users greater stability, and has resulted in the abstraction of channel masquerades, renaming, and other internal implementation details. It also allows for the seamless transition between two-party and multi-party bridges using core features.


And much more!

Please note that Asterisk 12 went through both an alpha and a beta testing process. During this time, many bugs were fixed, features enhanced, and improvements made. If you participated during the alpha and beta testing process, thank you! Please note that Asterisk 12 has changed as a result of the testing, and the UPGRADE and CHANGES notes should still be reviewed.

Information about the new features and changes in Asterisk 12 can be found on the Asterisk wiki:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Documentation

A full list of all new features can also be found in the CHANGES file:

http://svnview.digium.com/svn/asterisk/branches/12/CHANGES

For a full list of changes in the current release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.0.0

Thank you for your continued support of Asterisk!



New mailing list - asterisk-app-dev
Click to view a printable version Wed, 04 Dec 2013 10:53:38 +1300

Ok, so I am really behind! I missed this one. There is a new mailing list for Asterisk App Development. Matt Jordan wrote a post about it:

Hey all -

After much discussion at AstriCon, it became clear that the Asterisk project could use a mailing list dedicated specifically to application development. This new mailing list should be used specifically for discussions regarding the development of applications using AMI, AGI, or ARI - or any other interface exposed by Asterisk in the future. Today, we're pleased to announce the asterisk-app-dev mailing list, now available on lists.digium.com:

http://lists.digium.com/cgi-bin/mailman/listinfo/asterisk-app-dev

This new list is not a replacement for asterisk-dev or asterisk-users. The asterisk-dev list should continue to be used for discussions regarding development directly in the Asterisk project itself or the Asterisk Test Suite. The asterisk-users list should continue to be used for discussions regarding deployment and usage of Asterisk itself. As we expect the amount of questions and discussions regarding application development and API usage to grow, it makes sense to provide a dedicated forum for those discussions outside of -users and -dev, and this new mailing list serves that purpose.

Of course, there's bound to be some cross-over between mailing lists, and sometimes it might not be clear where to post your question. As always, please do not cross post between lists. If your question on asterisk-app-dev is more appropriate for asterisk-users or asterisk-dev, someone will kindly ask you to direct your question to the other mailing lists, where a wider audience may be available to assist you.

Thanks - and we all look forward to lots of productive discussions on the new mailing list about building applications that use Asterisk as their communications engine!

Matt

--
Matthew Jordan
Digium, Inc. | Engineering Manager



IOS iBat Apps Suite to control Asterisk servers
Click to view a printable version Wed, 04 Dec 2013 10:42:40 +1300

Aldo Bergamini has posted details of four new iOS apps for Asterisk:

Four newly released applications are available in the Apple App Store to drive your customer's Asterisk PBX. iBat Pro, iBat User, iBat Access and the free iBat Setup run on any IOS device, see more at: http://www.pxc.biz.

They cover a wide range of basic needs, always making it easy for users to do what they need:

  • to act as their convenient 'BLF sidecar' to originate calls: without limits to the number of targeted extensions.
  • to let the user get the called party's phone number from the iPhone's contact list, choose the outgoing line and originate calls with ease from his/her Asterisk extension
  • to keep track of calls originated to an external destination (a quick way to keep a lively one touch phone numbers list)
  • to monitor their own extension, even when away from the desk
  • to monitor a centrally managed, per user defined list of extensions, with far more information about the state than a colored LED
  • they offer a clean and simple call transfer interface, far beyond what even the most advanced phones can do - perform a transfer to external destinations choosing the outgoing line in a flash

Further, both iBat Pro as well as iBat User offer a mix of advanced ways to originate calls:

The Operator Access function lets users bypass the PBX operator to get connected to any Asterisk extension, saving time and with ease of knowing when to call (the allowed extensions state is displayed in real time, thus avoiding to call somebody who's already on the phone). [iBat Pro] [iBat User]

The BLF Callback function acts as the Operator Access, but lets Asterisk call the user's iPhone, as opposed to the automated call from the iPhone to any phone line served by Asterisk. [iBat Pro]

Again iBat Pro offers a way ('Callback Dialer') to place double originated calls to any external destination; Asterisk thus bridges the connection to the iPhone on one line to the destination over a second channel. This can be used for many purposes (from keeping the user's mobile phone number private, to saving money using convenient operators from the PBX).

Finally the purposely inexpensive iBat Access app makes the Operator Access functionality available to external calling partners, people not directly served by Asterisk (they have no extension, but do just call the PBX from the public network). This is a way to bring preferred business partners closer to a company, by letting them see when the desidered callee is available and thus getting a quick connection bypassing the operator with ease from their iPhone.

Come have a look at the iBat apps suite directly on the Apple App Store:
iBat Setup https://itunes.apple.com/us/app/ibat-setup
iBat Pro https://itunes.apple.com/us/app/ibat-pro
iBat User https://itunes.apple.com/us/app/ibat-user
iBat Access https://itunes.apple.com/us/app/ibat-access
Or come see more information on our website:

http://www.pxc.biz

Aldo Bergamini
PXC.BIZ



DAHDI-Linux and DAHDI-Tools 2.7.0.2 Now Available
Click to view a printable version Wed, 04 Dec 2013 09:00:53 +1300

The Asterisk Development Team has announced the releases of DAHDI-Linux-v2.7.0.2, DAHDI-Tools-v2.7.0.2 and dahdi-linux-complete-2.7.0.2+2.7.0.2

This release is available for immediate download at:
http://downloads.asterisk.org/pub/telephony/dahdi-linux
http://downloads.asterisk.org/pub/telephony/dahdi-tools
http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete

*** THIS RELEASE FIXES CENTOS 6.5 "PDE_DATA" build issue ***

dahdi-linux-complete tarballs now include all firmware necessary to build without an internet connection.

Issues closed in this release:
https://issues.asterisk.org/jira/browse/DAHLIN-331

Shortlog of changes since v2.7.0.1:
Shaun Ruffell (2):
dahdi: CentOS 6.5 backported PDE_DATA definition.
dahdi: Fix previous CentOS 6.5 commit.


The diffstat from the v2.7.0.1 release:

include/dahdi/kernel.h |   12 ++++++++++++
1 file changed, 12 insertions(+)



For a full list of changes in these releases, please see the shortlog at:
http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=shortlog;h=refs/tags/v2.7.0.2
http://git.asterisk.org/gitweb/?p=dahdi/tools.git;a=shortlog;h=refs/tags/v2.7.0.2

Issues found in this release can be reported in the DAHDI-Linux and DAHDI-Tools projects at https://issues.asterisk.org/jira

Thank you for your continued support of Asterisk!



DAHDI-Linux and DAHDI-Tools 2.8.0-rc4 Now Available
Click to view a printable version Wed, 04 Dec 2013 05:40:25 +1300

The Asterisk Development Team has announced the releases of DAHDI-Linux-v2.8.0-rc4, DAHDI-Tools-v2.8.0-rc4, dahdi-linux-complete-2.8.0-rc4+2.8.0-rc4

This release is available for immediate download at:
http://downloads.asterisk.org/pub/telephony/dahdi-linux
http://downloads.asterisk.org/pub/telephony/dahdi-tools
http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete

*** THIS RC FIXES CENTOS 6.5 "PDE_DATA" build issue ***

In version 2.8 we have introduced two new drivers:
wcte43x - For Digium's new line of 2/4 span T1/E1 cards
wcaxx - For Digium's new line of analog fxs/fxo cards

We introduced a common library called "wcxb" which ties the previous two drivers, plus the recently introduced wcte13xp driver, together into one common base.

dahdi-linux-complete tarballs now include all firmware necessary to build without an internet connection.

Shortlog of changes since v2.7.0.1:
Oron Peled (16):
xpp: Serialize dahdi registration
xpp: refactor FXS ring settings
xpp: FXS: ring/mwi settings: a sysfs interface
xpp: ring/mwi settings: add to FXS init script
add a 'location' attribute to sysfs (dahdi_device):
dahdi: add "tools_rootdir" module parameter
Also send DAHDI_TOOLS_ROOTDIR with device events
live_dahdi: load "dahdi" with tools_rootdir=$DESTDIR
remove udev rules: moved to dahdi-tools
sysfs: create symlink "ddev" to device of span
dahdi: Rename span 'master' as 'master_span'
.gitignore: *.ko.unsigned
Rename "pinned spans" to "assigned spans"
xpp: automatic dahdi_registration by default
sysfs: new driver attribute: master_span
Makefile: new 'make-dist' target

Russ Meyerriecks (3):
wcte13xp: Migrate to wcxb library
wcte13xp: Hold framer in reset to stop xmit on modprobe -r
wcte13xp: Improve maintenance functions and error counters

Shaun Ruffell (28):
dahdi_config: Remove unused NO_DCDC definition.
dahdi: Clear DAHDI_ALARM_NOTOPEN when spans are re-initialized.
dahdi: Fix placement of '/' in output of /proc/dahdi/x
dahdi: Work around missing KBUILD_MODNAME
dahdi: Backport try_wait_for_completion() and list_first_entry()
wct4xxp: Print warning in dmesg if span priority is not set correctly.
wct4xxp: Fix bipolar error insertion test mode.
wct4xxp: VPM companding switch print is now debug only.
wct4xxp: If linemode changed via sysfs, reset the complete part.
wct4xxp, wcte13xp: Move the octasic DSP code into separate module.
wcaxx: New driver for A4A/A4B/A8A/A8B analog cards.
wcaxx: Update A4B firmware to version 0b0017
wcxb: Update the firmware meta block during flash update.
wcte43x: Do not grab reglock in handle_transmit/handle_receive.
wcte43x: Remove 'dcxo' debug attribute.
oct612x: Make dependent on dahdi.ko
dahdi_dynamic: Create a span type for dynamic spans.
wcaxx: Use startup/shutdown callbacks to protect access to channel registers.
wctdm24xxp: Remove assigned callback.
dahdi: Remove "ddev" symlink before unregistering the span device.
dahdi: CentOS 6.5 backported PDE_DATA definition.
wcxb: is_pcie -> pci_is_pcie()
wcxb: Do not access cur_transfer/cur_msg outside of lock.
wcaxx: Add extra dummy read when checking for single fxs modules.
wcte43x: Update firmware to version e0017.
dahdi: Replace drv_attr with drv_groups on kernels > 3.12.
Revert "wcaxx: Use startup/shutdown callbacks to protect access to channel registers."
dahdi: Fix previous CentOS 6.5 commit.

Shaun RuffellL (1):
wcaxx: Remove some left over debugging trace statements.

Tzafrir Cohen (6):
xpp: Firmware for Astribanks 2.02
xpp: Firmware for Astribanks 2.02: Makefile
xpp: USB_FW.202.hex: provide as a symlink
xpp: mark an AB as failed if it gives bad desc
xpp: Fail loading if no module on first slot
Ignore some more firmware files

Wendell Thompson (2):
wcte13xp: Use interrupts for Falc alarms and signaling
wcte43x: Add driver for TE435/TE235 digital cards.



The diffstat from the v2.7.0.1 release:


.gitignore | 9 +
Makefile | 28 +-
README | 17 +-
build_tools/genudevrules | 40 -
build_tools/live_dahdi | 2 +-
build_tools/make_dist | 26 +
drivers/dahdi/Kbuild | 21 +-
drivers/dahdi/dahdi-base.c | 74 +-
drivers/dahdi/dahdi-sysfs.c | 150 +-
drivers/dahdi/dahdi_dynamic.c | 4 +
drivers/dahdi/firmware/Makefile | 71 +-
drivers/dahdi/oct612x/Kbuild | 32 +
drivers/dahdi/oct612x/oct612x-user.c | 200 +
drivers/dahdi/oct612x/oct612x.h | 49 +
drivers/dahdi/wcaxx-base.c | 4544 ++++++
drivers/dahdi/wct4xxp/Kbuild | 4 +-
drivers/dahdi/wct4xxp/base.c | 159 +-
drivers/dahdi/wct4xxp/vpm450m.c | 139 +-
drivers/dahdi/wct4xxp/vpm450m.h | 8 +-
drivers/dahdi/wctdm24xxp/base.c | 31 +-
drivers/dahdi/wcte13xp-base.c | 2294 ++-
drivers/dahdi/wcte43x-base.c | 3591 +++++
drivers/dahdi/wcxb.c | 951 ++
drivers/dahdi/wcxb.h | 184 +
drivers/dahdi/wcxb_flash.c | 170 +
drivers/dahdi/wcxb_flash.h | 34 +
drivers/dahdi/wcxb_spi.c | 386 +
drivers/dahdi/wcxb_spi.h | 116 +
drivers/dahdi/xpp/card_fxs.c | 295 +-
drivers/dahdi/xpp/card_global.c | 6 +
drivers/dahdi/xpp/firmwares/FPGA_1161.202.hex |20517 +++++++++++++++++++++++++
drivers/dahdi/xpp/firmwares/Makefile | 5 +-
drivers/dahdi/xpp/firmwares/USB_FW.202.hex | 1 +
drivers/dahdi/xpp/init_card_1_30 | 22 +-
drivers/dahdi/xpp/xbus-core.c | 24 +-
drivers/dahdi/xpp/xbus-sysfs.c | 11 +-
drivers/dahdi/xpp/xpp.rules | 11 -
include/dahdi/dahdi_config.h | 3 +-
include/dahdi/kernel.h | 55 +-
39 files changed, 32537 insertions(+), 1747 deletions(-)



For a full list of changes in these releases, please see the shortlog at:
http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=shortlog;h=refs/tags/v2.8.0-rc4
http://git.asterisk.org/gitweb/?p=dahdi/tools.git;a=shortlog;h=refs/tags/v2.8.0-rc4

Issues found in this release can be reported in the DAHDI-Linux and
DAHDI-Tools projects at https://issues.asterisk.org/jira

Thank you for your continued support of Asterisk!



Astricon started
Click to view a printable version Wed, 09 Oct 2013 02:48:59 +1300

Hi all!

Astricon has started!

I'm sitting in the first presentation at Astricon - one of the surprising things is that around 70% of the people here are here for the first time.

There are a great collection of regulars too and it's great to catch up with everyone again.

If you're not here you're missing out :-)



New Asterisk Developer - Scott Griepentrog
Click to view a printable version Wed, 02 Oct 2013 07:43:20 +1300

Matt Jordan has posted info on a new Asterisk Developer at Digium:

Hey all -

Scott Griepentrog (sgriepentrog) has recently joined the Asterisk development team here at Digium. Scott has a long and storied history with both Asterisk and FreePBX, having developed and used both Asterisk and FreePBX since the early days of 1.2 (or thereabouts). Scott will be participating in all aspects of Asterisk development - and we're thrilled that he joined the team here at Digium.

Feel free to say hi to Scott in #asterisk-dev or on the mailing lists - or if you make it - at AstriCon this year!

Matt

--
Matthew Jordan
Digium, Inc. | Engineering Manager



AstLinux 1.1.3 Released
Click to view a printable version Wed, 02 Oct 2013 07:37:09 +1300

The AstLinux Team has released AstLinux 1.1.3. All current users are encouraged to upgrade as this release addresses several security and bugfix issues.

AstLinux 1.1.3 adds:

  • An Asterisk Operator Panel (FOP2) via a new Add-On Package facility
  • Logrotate automatically configured to control logging
  • The latest Prosody which now works with Asterisk 1.8 res_jabber and supports XMPP PubSub
  • Web Interface enhancements
  • Package upgrades providing security and bug fixes

A full changelog can be viewed in the release pages:

http://www.astlinux.org/release/113-asterisk-1151

http://www.astlinux.org/release/113-asterisk-18231

New AstLinux Documentation Topics:

Asterisk Flash Operator Panel 2

http://doc.astlinux.org/userdoc:tt_asterisk-fop2

Distribute Asterisk Events using XMPP PubSub

http://doc.astlinux.org/userdoc:tt_distribute_events_xmpp_pubsub

External Music on Hold Source

http://doc.astlinux.org/userdoc:tt_external_moh_source

--

The AstLinux Team



QueueWiz Call Center Simulator
Click to view a printable version Wed, 02 Oct 2013 07:32:13 +1300

Lenz from QueueMetrics has posted details of a new free app to simulate call centers and aid in sizing cost and revenue simulation:

http://queuewiz.queuemetrics.com



Asterisk Videos on New YouTube Channel
Click to view a printable version Fri, 02 Aug 2013 04:48:53 +1200

Billy Chia has written a blog post on the Digium site with links to video recordings of talks from Astricon last year including mine.

I'll be speaking again at this year's Astricon and it'll be a big one - the 10th anniversary!

Excerpt from the post:

To celebrate the 10th anniversary of Astricon we’ll be adding videos to a new Asterisk YouTube channel each week leading up to this year’s Astricon in Atlanta. To kick off the fun we’ve created an Asterisk Integration playlist with tips and advice for using APIs, Adhearson and Dialplan scripting. Featuring experts in the Asterisk community like Ben Klang and Matt Riddell these videos are worth a view!

Read More...



Asterisk 11.5.0 Now Available
Click to view a printable version Fri, 19 Jul 2013 06:03:59 +1200

The Asterisk Development Team has announced the release of Asterisk 11.5.0.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 11.5.0 resolves several issues reported by the community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

  • Fix Segfault In app_queue When "persistentmembers" Is Enabled And Using Realtime
    (Closes issue ASTERISK-21738. Reported by JoshE)
  • IAX2: fix race condition with nativebridge transfers.
    (Closes issue ASTERISK-21409. Reported by alecdavis)
  • Fix The Payload Being Set On CN Packets And Do Not Set Marker Bit
    (Closes issue ASTERISK-21246. Reported by Peter Katzmann)
  • Fix One-Way Audio With auto_* NAT Settings When SIP Calls Initiated By PBX
    (Closes issue ASTERISK-21374. Reported by Michael L. Young)
  • chan_sip: NOTIFYs for BLF start queuing up and fail to be sent out after retries fail
    (Closes issue ASTERISK-21677. Reported by Dan Martens)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.5.0

Thank you for your continued support of Asterisk!



Asterisk 1.8.23.0 Now Available
Click to view a printable version Fri, 19 Jul 2013 06:00:51 +1200

The Asterisk Development Team has announced the release of Asterisk 1.8.23.0.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 1.8.23.0 resolves several issues reported by the community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

  • Fix a memory copying bug in slinfactory which was causing mixmonitor issues.
    (Closes issue ASTERISK-21799. Reported by Michael Walton)
  • IAX2: fix race condition with nativebridge transfers.
    (Closes issue ASTERISK-21409. Reported by alecdavis)
  • Fix crash in chan_sip when a core initiated op occurs at the same time as a BYE
    (Closes issue ASTERISK-20225. Reported by Jeff Hoppe)
  • Fix The Payload Being Set On CN Packets And Do Not Set Marker Bit
    (Closes issue ASTERISK-21246. Reported by Peter Katzmann)
  • chan_sip: Session-Expires: Set timer to correctly expire at (~2/3) of the interval when not the refresher
    (Closes issue ASTERISK-21742. Reported by alecdavis)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.23.0

Thank you for your continued support of Asterisk!



Asterisk 12 pjproject installation testing needed
Click to view a printable version Tue, 16 Jul 2013 04:42:14 +1200

Matt Jordan has posted a request for people to test the installation and building of pjproject as a part of Asterisk 12:

Hey all!

As you know, Asterisk 12 includes a new SIP stack built on pjproject. As part of the work for Asterisk 12, we've pulled pjproject out of Asterisk and modified pjproject's build system to produce shared object libraries. The modified version of pjproject is available on github.

While we've done a fair amount of testing of the build system - predominately on CentOS 6 (for which we've produced packages of the modified pjproject) and on our local development machines - which range from Fedora to Ubuntu to OS X - we really need some additional testing of the build system before releasing Asterisk 12.

Because pjproject itself embeds a number of third party libraries, getting pjproject properly configured for a distribution/environment can be a bit tricky. We've taken the current findings and issues that people have run into and put together a page on the Asterisk wiki here:

https://wiki.asterisk.org/wiki/display/AST/Installing+pjproject

Note that the target audience of that page are Asterisk users, so we want to get as much useful information on there as possible.

Ensuring that building and installing pjproject is as painless as possible will be critical to the success of Asterisk 12. No one is going to try out the new SIP functionality if they can't get its dependencies to build and install :-)

If you'd like to help us test it out, please reply to this e-mail with whatever distro you're willing to test it on.

If you find any problems, please also reply to this e-mail and we can work out what configure options may be needed, what needs to be changed in the build system, updated on the wiki page, etc.

In particular, we'd like to make any last changes to the build system before we push the changes back up stream to Teluu. So this is our opportunity to get it right!

Thanks!

Matt

--
Matthew Jordan
Digium, Inc. | Engineering Manager



Asterisk 12: ARI update
Click to view a printable version Tue, 16 Jul 2013 04:31:35 +1200

David Lee has posted an update on the Asterisk Rest Interface:

Hey everybody!

There's been a few people following the progress of the new Asterisk REST Interface (ARI), so I thought I'd give an update. I also merged in some patches this morning that change the ARI interface, so a heads up is warranted.

If you're not familiar with ARI, please check out its project page on the Asterisk wiki. There's also detailed documentation of the API's themselves. We should soon have a getting started page to offer a more complete introduction to the interface.

For those who are daring, quite a bit of the API is now implemented. You can send calls to an ARI application; answer; play media; dial; create bridges; bridge channels together. The code is still very new, but early feedback is welcome!

If you've been playing around with ARI, today there are many changes that will affect you.

  • The prefix for URL's was changes from /stasis to /ari.
  • The WebSocket for events was moved from /ws to /stasis/events.
  • The interface now supports (and requires) authentication.
    • Users are configured in ari.conf
    • In addition HTTP Basic authentication, ARI allows you to authenticate by passing ?api_key={username}:{password} as a parameter to your request. This allows swagger-ui to used for API documentation.

  • If building with DEV_MODE enabled, responses from ARI are validated against the API docs. If the response provided does not match the documented response, warnings are logged and the response is replaced with a 500 Internal Server Error.
  • If you have changes to api-docs in a branch, you'll probably have conflicts in the generated files (res/res_stasis_http_*.c, res/res_stasis_json_*.c, and res/stasis_json/resource_*.h). Since they're generated, the best strategy is to simply revert the files and re-run 'make ari-stubs'.
  • Oh yeah, 'make stasis-stubs' was changed to 'make ari-stubs'.

    --
    David M. Lee
    Digium, Inc. | Software Developer



    Asterisk 12 Feature Freeze Reminder
    Click to view a printable version Tue, 09 Jul 2013 06:43:24 +1200

    Matt Jordan has posted a note about the (rapidly) approaching feature freeze date for Asterisk 12:

    Hey everyone -

    This is a friendly reminder that the feature freeze deadline for Asterisk 12 is rapidly approaching. Per the Asterisk wiki, feature freeze for Asterisk 12 is scheduled for the 3rd Wednesday of July - or July 17th. Given the scale of Asterisk 12 (and the fact that this reminder is a tad late), feature free for Asterisk 12 will occur on July 31st. That gives us a few extra weeks to get everything on the Asterisk 12 roadmap cleaned up and merged in.

    Please try to have all patches to Asterisk up on Review Board by that date. Patches do not have to be merged, but they should be in "good shape" and posted for code review. If you're wondering what "good shape" means, patches should adhere to the coding guidelines and be well tested.

    Shortly after the feature freeze is hit, we will create the Asterisk 12 branch and being preparing the first release from that branch for public testing.

    Thanks!

    Matt

    --
    Matthew Jordan
    Digium, Inc. | Engineering Manager



    Still alive
    Click to view a printable version Tue, 09 Jul 2013 06:31:43 +1200

    I thought I would write a quick post explaining the break in posts.

    Over the last year I have been moving round quite a bit. Near the end of last year I moved from my native Dunedin, New Zealand to Orlando, Florida. I'd been living there pretty much full time until I started visiting Panama City, Panama (Central America).

    A few weeks ago I decided it was getting too expensive to keep staying in hotels every time I came to Panama so I got hold of an apartment.

    On the first day I moved in they told me that the gas was turned off because of a leak and that I would have to spend the first weekend with no cooking or hot water but that I wouldn't have to pay rent.

    I went to sleep that night and was woken by a cracking sound that resulted in the floor tiles lifting right across the middle of the bedroom!

    On the Monday they came to check the gas and said there was no problem and so turned it on. I spent the next few days with a gas leak before they finally came to fix it.

    They had also agreed to put up curtains (floor to ceiling windows over two floors was a bit much at dawn) and so that took a few days too.

    I'm sitting downstairs at the moment while they not only remove all of the floor tiles upstairs but all of the cement beneath them using hammers and chisels.

    Needless to say it's been rather difficult to work and has been driving me crazy.

    I'm starting to get back on top of my emails again regardless and so the Asterisk News should start flowing again :-)

    The gas is thankfully fixed, the curtains are up and all I need now is a floor.

    If you ever get a chance to visit Panama you should - it's an amazing place - just slightly more red tape than I would normally like and a somewhat difficult language (seeing as I don't speak it).

    Anyway, sorry to everyone for the delays in my posts and I'll try and get back on top of it now :-) Hope you understand :-)

    Cheers,

    Matt Riddell



    DAHDI-Linux and DAHDI-Tools 2.7.0 Now Available
    Click to view a printable version Tue, 09 Jul 2013 06:22:17 +1200

    The Asterisk Development Team has announced the releases of:

    DAHDI-Linux-v2.7.0
    DAHDI-Tools-v2.7.0
    dahdi-linux-complete-2.7.0+2.7.0

    This release is available for immediate download at:
    http://downloads.asterisk.org/pub/telephony/dahdi-linux
    http://downloads.asterisk.org/pub/telephony/dahdi-tools
    http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete

    In this release:

    • Driving closer towards sysfs configuration of dahdi devices
    • Experimental support to "pin" specific span and channel numbers to specific device/local spans
    • New wcte13xp base driver

    Shortlog for dahdi-linux:
    Tzafrir Cohen:
    xpp: FXO: fix firmware pol-rev detection
    README: xpp: xpd_fxo param use_polrev_firmware
    README: xpp: xpd_fxo: new value of caller_id_style
    Copy xpp module docs from README.Astribank
    README: subsections for module parameters docs
    Only use bus and no class for channel devices
    Document new channel sysfs interface
    How to get OSLEC from dahdi-linux-extra
    README fixes: DKMS indentation and such
    xpp: also install the new .201 fixrmware files
    README: note on DKMS
    dev_set_name(): remove unneeded ';'
    xpp: style: fix an improper line break.
    xpp: fix "non-const" index, right header, indent

    Matthew Fredrickson:
    wcb4xxp: Support for when network side deactivates layer1.

    Russ Meyerriecks:
    wcte13xp: New driver for digium's te13x product range
    dahdi: Do not define trace_printk if CONFIG_TRACING is not defined.
    dahdi: Fix unused variable compile warning
    Kbuild: Fix OSLEC build error
    Merge tag 'review-sysfs-chan' of http://git.tzafrir.org.il/git/dahdi-linux into for-trunk
    dahdi: Remove 'getlin_lastchunk' from struct dahdi_chan.
    dahdi-base: Minor maint mode error

    Oron Peled:
    xpp: FXO: add a "squelch_polrev" parameter
    xpp: FXO: common function for POLREV reporting
    xpp: FXO: new CID style -- passthrough
    xpp: FXO: in-firmware polarity-reversal detection
    sysfs: new channel attr (ec_factory, ec_state)
    sysfs: stringify channels 'sigcap' attribute
    dahdi: sysfs: add channel attributes
    dahdi: sysfs: use dynamically allocated chrdev's
    dahdi: sysfs: chrdev region (not usefull yet)
    dahdi: sysfs: a channel bus (not usefull yet)
    xpp: FXS: improve fxs_info output layout
    xpp: BRI: make it always SYNC_MODE_AB (like PRI)
    xpp: PRI/BRI: fix channels opening/closing:
    xpp: init_card: better variable naming
    Remove support for kernels < 2.6.18
    sysfs channels: cleanup device files handling
    sysfs channels: dahdi-sysfs-chan.c
    sysfs channels: refactor compat macros
    sysfs: add 'lineconfig' attribute to span
    dahdi: build fix for Kernels < 2.6.16
    code cleanup: remove unused debug_printk()
    code cleanup - refactor module_printk()
    fix class_create() return value test
    xpp: FXO: improve ring debounce notice
    show Master change to/from core timer
    A channel-less span should not crash dahdi
    better chan_printk() output
    dahdi: style - checkpatch clean dahdi-sysfs.c
    xpp: style - remove extra braces
    xpp: style - kfree() is NULL safe
    xpp: style - add const to file_operations
    xpp: style - manual whitespace/line-breaks cleanup
    xpp: style - one macro cleanup
    xpp: style - no more typedef byte
    xpp: style - place EXPORT_SYMBOL() (manually)
    xpp: style - clean many long lines (manually)
    xpp: style - Run Lindent
    xpp: style - convert typedef of byte to __u8
    xpp: style - add space after comma
    xpp: style - insert space afer if/while/for/switch
    xpp: style - Remove space before tabs
    xpp: style - Remove 0/NULL static initializers
    xpp: style: convert __FUNCTION__ to __func__
    xpp: style - remove eolspace

    Shaun Ruffell:
    wctc4xxp: Ensure the descriptors are zeroed out on start.
    xpp: Don't use create_proc_read_entry()
    dahdi_dynamic_ethmf: Don't use create_proc_read_entry()
    dahdi: Replace create_proc_entry() with proc_create_data()
    oct612x: Fix confusing compile error when kernel source is not present
    wcte12xp: Reset all the framer registers when switching linemodes.
    wcte12xp: Allow non-interrupting cards to unload faster.
    dahdi: Completely stop spans when unassigning.
    dahdi: Prevent potential error when only switching spantype of single span.
    wcte12xp: Look for multiple loopup codes before setting looping up the framer.
    wct4xxp: Allow vpm450m.c to compile against vanilla 2.6.18.
    wctdm24xxp: Fix FXO failure to detect battery CO disconnects.
    dahdi: Prevent memory corruption on device unload.
    wct4xxp: Companding on VPM needs to be changed when switching linemodes.
    oct612x: Break the oct612x out into a separate library.
    dahdi: Save the current maintstat in the span before calling into the drivers.
    build_tools/make_version: Only strip 'v' if followed by a digit.
    dahdi: Tear down conference links when conferences are emptied out.
    dahdi: Restore DAHDI_CONFLINK functionality as compile time option.
    dahdi: Give timers their own file_operations
    dahdi: Decrease dahdi_timer_lock contention.
    dahdi: Remove call to lock_kernel when calling unlocked_ioctl.
    dahdi: Initialize the channels cdev structure.
    sysfs: Remove signed one-bit fields.
    dahdi: Trivial change of '__u32' -> 'u32' in struct dahdi_count.
    dahdi: Move 'timingslips' in with the other maintenance counters.
    dahdi: Only watch transitions of ABIT when using E&M signalling.
    Revert "dahdi_dynamic_eth: Move tx packet flushing to process context."
    dahdi_dynamic: Use a tasklet for flushing dynamic drivers.
    dahdi: Running without the Big Kernel Lock (BKL) is no longer experimental.
    dahdi: Remove unused 'rxbufpolicy' and 'rxdisable' from dahdi_chan.
    wctdm24xxp: Set dahdi_span.spantype to SPANTYPE_ANALOG_MIXED.
    wct4xxp: Ensure all spans are configured by default.
    dahdi: Filter 'HWEC' from DAHDI_GETVERSION results if hwec is really not present.
    dahdi: Increase the number of conference buffers to eight.
    dahdi_ioctl_spanstat() backward compat hack
    sysfs: refactor lineconfig string representation
    sysfs: add a linecompat span attribute
    convert span->spantype to enumerated type
    wcb4xxp: Fix typo in last commit that broke compilation.
    Remove Makefiles that are only needed on kernels < 2.6.9
    Update Digium copyright on files changed since beginning of the year.
    dahdi: Remove __exit annotation from dahdi_sysfs_exit().
    dahdi: Use monotonic clock for coretimer.

    The diffstat from the v2.6.2 release for dahdi-linux:


    Makefile | 10 +- README | 316 +- build_tools/make_version | 2 +- drivers/dahdi/Kbuild | 12 +- drivers/dahdi/Makefile | 6 - drivers/dahdi/dahdi-base.c | 785 +++-- drivers/dahdi/dahdi-sysfs-chan.c | 483 +++ drivers/dahdi/dahdi-sysfs.c | 473 +-- drivers/dahdi/dahdi-sysfs.h | 55 + drivers/dahdi/dahdi.h | 1 + drivers/dahdi/dahdi_dummy.c | 2 +- drivers/dahdi/dahdi_dynamic.c | 41 +- drivers/dahdi/dahdi_dynamic_eth.c | 51 +- drivers/dahdi/dahdi_dynamic_ethmf.c | 89 +- drivers/dahdi/dahdi_echocan_jpah.c | 5 +- drivers/dahdi/dahdi_echocan_kb1.c | 5 +- drivers/dahdi/dahdi_echocan_mg2.c | 5 +- drivers/dahdi/dahdi_echocan_oslec.c | 2 - drivers/dahdi/dahdi_echocan_sec.c | 3 - drivers/dahdi/dahdi_echocan_sec2.c | 3 - drivers/dahdi/firmware/Makefile | 50 +- drivers/dahdi/hpec/dahdi_echocan_hpec.c | 14 +- drivers/dahdi/oct612x/Makefile | 66 +- drivers/dahdi/tor2.c | 4 +- drivers/dahdi/voicebus/Makefile | 6 - .../vpmadt032_loader/dahdi_vpmadt032_loader.c | 11 +- drivers/dahdi/wcb4xxp/Makefile | 7 - drivers/dahdi/wcb4xxp/base.c | 122 +- drivers/dahdi/wcb4xxp/wcb4xxp.h | 5 +- drivers/dahdi/wct1xxp.c | 4 +- drivers/dahdi/wct4xxp/Kbuild | 2 +- drivers/dahdi/wct4xxp/Makefile | 8 - drivers/dahdi/wct4xxp/base.c | 187 +- drivers/dahdi/wct4xxp/vpm450m.c | 73 +- drivers/dahdi/wct4xxp/vpm450m.h | 2 + drivers/dahdi/wctc4xxp/Makefile | 7 - drivers/dahdi/wctc4xxp/base.c | 13 +- drivers/dahdi/wctdm24xxp/Makefile | 6 - drivers/dahdi/wctdm24xxp/base.c | 39 +- drivers/dahdi/wctdm24xxp/wctdm24xxp.h | 2 + drivers/dahdi/wctdm24xxp/xhfc.c | 4 +- drivers/dahdi/wcte11xp.c | 4 +- drivers/dahdi/wcte12xp/Makefile | 6 - drivers/dahdi/wcte12xp/base.c | 147 +- drivers/dahdi/wcte12xp/wcte12xp.h | 1 + drivers/dahdi/wcte13xp-base.c | 3243 ++++++++++++++++++++ drivers/dahdi/xpp/card_bri.c | 1287 ++++---- drivers/dahdi/xpp/card_bri.h | 4 +- drivers/dahdi/xpp/card_echo.c | 116 +- drivers/dahdi/xpp/card_echo.h | 6 +- drivers/dahdi/xpp/card_fxo.c | 1201 +++++--- drivers/dahdi/xpp/card_fxo.h | 17 +- drivers/dahdi/xpp/card_fxs.c | 1618 +++++----- drivers/dahdi/xpp/card_fxs.h | 18 +- drivers/dahdi/xpp/card_global.c | 561 ++-- drivers/dahdi/xpp/card_global.h | 101 +- drivers/dahdi/xpp/card_pri.c | 1879 ++++++------ drivers/dahdi/xpp/card_pri.h | 5 +- drivers/dahdi/xpp/dahdi_debug.c | 48 +- drivers/dahdi/xpp/dahdi_debug.h | 290 +- drivers/dahdi/xpp/firmwares/PIC_TYPE_2.hex | 676 ++-- drivers/dahdi/xpp/init_card_1_30 | 8 +- drivers/dahdi/xpp/init_card_2_30 | 3 + drivers/dahdi/xpp/mmapbus.c | 12 +- drivers/dahdi/xpp/mmapdrv.c | 222 +- drivers/dahdi/xpp/parport_debug.c | 40 +- drivers/dahdi/xpp/parport_debug.h | 2 +- drivers/dahdi/xpp/print_fxo_modes.c | 31 +- drivers/dahdi/xpp/xbus-core.c | 1263 ++++---- drivers/dahdi/xpp/xbus-core.h | 326 +- drivers/dahdi/xpp/xbus-pcm.c | 827 ++--- drivers/dahdi/xpp/xbus-pcm.h | 111 +- drivers/dahdi/xpp/xbus-sysfs.c | 573 ++-- drivers/dahdi/xpp/xdefs.h | 100 +- drivers/dahdi/xpp/xframe_queue.c | 187 +- drivers/dahdi/xpp/xframe_queue.h | 29 +- drivers/dahdi/xpp/xpd.h | 203 +- drivers/dahdi/xpp/xpp_dahdi.c | 714 ++--- drivers/dahdi/xpp/xpp_dahdi.h | 22 +- drivers/dahdi/xpp/xpp_usb.c | 822 +++-- drivers/dahdi/xpp/xproto.c | 392 +-- drivers/dahdi/xpp/xproto.h | 244 +- include/dahdi/dahdi_config.h | 11 +- include/dahdi/kernel.h | 115 +- include/dahdi/user.h | 5 + 85 files changed, 12798 insertions(+), 7673 deletions(-)


    For a full list of changes in these releases, please see the shortlog at:
    http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=shortlog;h=refs/tags/v2.7.0-rc1
    http://git.asterisk.org/gitweb/?p=dahdi/tools.git;a=shortlog;h=refs/tags/v2.7.0-rc1

    Issues found in this release can be reported in the DAHDI-Linux and DAHDI-Tools projects at https://issues.asterisk.org/jira

    Thank you for your continued support of Asterisk!



    DAHDI-Linux and DAHDI-Tools 2.7.0-rc1 Now Available
    Click to view a printable version Thu, 30 May 2013 09:07:36 +1200

    The Asterisk Development Team has announced the releases of:
    DAHDI-Linux-v2.7.0-rc1
    DAHDI-Tools-v2.7.0-rc1
    dahdi-linux-complete-2.7.0-rc1+2.7.0-rc1

    This release is available for immediate download at:
    http://downloads.asterisk.org/pub/telephony/dahdi-linux
    http://downloads.asterisk.org/pub/telephony/dahdi-tools
    http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete

    In this release:
    * Driving closer towards sysfs configuration of dahdi devices
    * New wcte13xp base driver

    Tzafrir Cohen:
    xpp: FXO: fix firmware pol-rev detection
    README: xpp: xpd_fxo param use_polrev_firmware
    README: xpp: xpd_fxo: new value of caller_id_style
    Copy xpp module docs from README.Astribank
    README: subsections for module parameters docs
    Only use bus and no class for channel devices
    Document new channel sysfs interface
    How to get OSLEC from dahdi-linux-extra
    README fixes: DKMS indentation and such
    xpp: also install the new .201 fixrmware files
    README: note on DKMS
    dev_set_name(): remove unneeded ';'
    xpp: style: fix an improper line break.
    xpp: fix "non-const" index, right header, indent

    Matthew Fredrickson:
    wcb4xxp: Support for when network side deactivates layer1.

    Russ Meyerriecks:
    wcte13xp: New driver for digium's te13x product range
    dahdi: Do not define trace_printk if CONFIG_TRACING is not defined.
    dahdi: Fix unused variable compile warning
    Kbuild: Fix OSLEC build error
    Merge tag 'review-sysfs-chan' of http://git.tzafrir.org.il/git/dahdi-linux into for-trunk
    dahdi: Remove 'getlin_lastchunk' from struct dahdi_chan.
    dahdi-base: Minor maint mode error

    Oron Peled:
    xpp: FXO: add a "squelch_polrev" parameter
    xpp: FXO: common function for POLREV reporting
    xpp: FXO: new CID style -- passthrough
    xpp: FXO: in-firmware polarity-reversal detection
    sysfs: new channel attr (ec_factory, ec_state)
    sysfs: stringify channels 'sigcap' attribute
    dahdi: sysfs: add channel attributes
    dahdi: sysfs: use dynamically allocated chrdev's
    dahdi: sysfs: chrdev region (not usefull yet)
    dahdi: sysfs: a channel bus (not usefull yet)
    xpp: FXS: improve fxs_info output layout
    xpp: BRI: make it always SYNC_MODE_AB (like PRI)
    xpp: PRI/BRI: fix channels opening/closing:
    xpp: init_card: better variable naming
    Remove support for kernels < 2.6.18
    sysfs channels: cleanup device files handling
    sysfs channels: dahdi-sysfs-chan.c
    sysfs channels: refactor compat macros
    sysfs: add 'lineconfig' attribute to span
    dahdi: build fix for Kernels < 2.6.16
    code cleanup: remove unused debug_printk()
    code cleanup - refactor module_printk()
    fix class_create() return value test
    xpp: FXO: improve ring debounce notice
    show Master change to/from core timer
    A channel-less span should not crash dahdi
    better chan_printk() output
    dahdi: style - checkpatch clean dahdi-sysfs.c
    xpp: style - remove extra braces
    xpp: style - kfree() is NULL safe
    xpp: style - add const to file_operations
    xpp: style - manual whitespace/line-breaks cleanup
    xpp: style - one macro cleanup
    xpp: style - no more typedef byte
    xpp: style - place EXPORT_SYMBOL() (manually)
    xpp: style - clean many long lines (manually)
    xpp: style - Run Lindent
    xpp: style - convert typedef of byte to __u8
    xpp: style - add space after comma
    xpp: style - insert space afer if/while/for/switch
    xpp: style - Remove space before tabs
    xpp: style - Remove 0/NULL static initializers
    xpp: style: convert __FUNCTION__ to __func__
    xpp: style - remove eolspace

    Shaun Ruffell:
    xpp: Don't use create_proc_read_entry()
    dahdi_dynamic_ethmf: Don't use create_proc_read_entry()
    dahdi: Replace create_proc_entry() with proc_create_data()
    oct612x: Fix confusing compile error when kernel source is not present
    wcte12xp: Reset all the framer registers when switching linemodes.
    wcte12xp: Allow non-interrupting cards to unload faster.
    dahdi: Completely stop spans when unassigning.
    dahdi: Prevent potential error when only switching spantype of single span.
    wcte12xp: Look for multiple loopup codes before setting looping up the framer.
    wct4xxp: Allow vpm450m.c to compile against vanilla 2.6.18.
    wctdm24xxp: Fix FXO failure to detect battery CO disconnects.
    dahdi: Prevent memory corruption on device unload.
    wct4xxp: Companding on VPM needs to be changed when switching linemodes.
    oct612x: Break the oct612x out into a separate library.
    dahdi: Save the current maintstat in the span before calling into the drivers.
    build_tools/make_version: Only strip 'v' if followed by a digit.
    dahdi: Tear down conference links when conferences are emptied out.
    dahdi: Restore DAHDI_CONFLINK functionality as compile time option.
    dahdi: Give timers their own file_operations
    dahdi: Decrease dahdi_timer_lock contention.
    dahdi: Remove call to lock_kernel when calling unlocked_ioctl.
    dahdi: Initialize the channels cdev structure.
    sysfs: Remove signed one-bit fields.
    dahdi: Trivial change of '__u32' -> 'u32' in struct dahdi_count.
    dahdi: Move 'timingslips' in with the other maintenance counters.
    dahdi: Only watch transitions of ABIT when using E&M signalling.
    Revert "dahdi_dynamic_eth: Move tx packet flushing to process context."
    dahdi_dynamic: Use a tasklet for flushing dynamic drivers.
    dahdi: Running without the Big Kernel Lock (BKL) is no longer experimental.
    dahdi: Remove unused 'rxbufpolicy' and 'rxdisable' from dahdi_chan.
    wctdm24xxp: Set dahdi_span.spantype to SPANTYPE_ANALOG_MIXED.
    wct4xxp: Ensure all spans are configured by default.
    dahdi: Filter 'HWEC' from DAHDI_GETVERSION results if hwec is really not present.
    dahdi: Increase the number of conference buffers to eight.
    dahdi_ioctl_spanstat() backward compat hack
    sysfs: refactor lineconfig string representation
    sysfs: add a linecompat span attribute
    convert span->spantype to enumerated type
    wcb4xxp: Fix typo in last commit that broke compilation.
    Remove Makefiles that are only needed on kernels < 2.6.9
    Update Digium copyright on files changed since beginning of the year.
    dahdi: Remove __exit annotation from dahdi_sysfs_exit().
    dahdi: Use monotonic clock for coretimer.

    The diffstat from the v2.6.2 release:


    Makefile | 10 +-
    README | 316 +-
    build_tools/make_version | 2 +-
    drivers/dahdi/Kbuild | 12 +-
    drivers/dahdi/Makefile | 6 -
    drivers/dahdi/dahdi-base.c | 785 +++--
    drivers/dahdi/dahdi-sysfs-chan.c | 483 +++
    drivers/dahdi/dahdi-sysfs.c | 473 +--
    drivers/dahdi/dahdi-sysfs.h | 55 +
    drivers/dahdi/dahdi.h | 1 +
    drivers/dahdi/dahdi_dummy.c | 2 +-
    drivers/dahdi/dahdi_dynamic.c | 41 +-
    drivers/dahdi/dahdi_dynamic_eth.c | 51 +-
    drivers/dahdi/dahdi_dynamic_ethmf.c | 89 +-
    drivers/dahdi/dahdi_echocan_jpah.c | 5 +-
    drivers/dahdi/dahdi_echocan_kb1.c | 5 +-
    drivers/dahdi/dahdi_echocan_mg2.c | 5 +-
    drivers/dahdi/dahdi_echocan_oslec.c | 2 -
    drivers/dahdi/dahdi_echocan_sec.c | 3 -
    drivers/dahdi/dahdi_echocan_sec2.c | 3 -
    drivers/dahdi/firmware/Makefile | 50 +-
    drivers/dahdi/hpec/dahdi_echocan_hpec.c | 14 +-
    drivers/dahdi/oct612x/Makefile | 66 +-
    drivers/dahdi/tor2.c | 4 +-
    drivers/dahdi/voicebus/Makefile | 6 -
    .../vpmadt032_loader/dahdi_vpmadt032_loader.c | 11 +-
    drivers/dahdi/wcb4xxp/Makefile | 7 -
    drivers/dahdi/wcb4xxp/base.c | 122 +-
    drivers/dahdi/wcb4xxp/wcb4xxp.h | 5 +-
    drivers/dahdi/wct1xxp.c | 4 +-
    drivers/dahdi/wct4xxp/Kbuild | 2 +-
    drivers/dahdi/wct4xxp/Makefile | 8 -
    drivers/dahdi/wct4xxp/base.c | 187 +-
    drivers/dahdi/wct4xxp/vpm450m.c | 73 +-
    drivers/dahdi/wct4xxp/vpm450m.h | 2 +
    drivers/dahdi/wctc4xxp/Makefile | 7 -
    drivers/dahdi/wctc4xxp/base.c | 13 +-
    drivers/dahdi/wctdm24xxp/Makefile | 6 -
    drivers/dahdi/wctdm24xxp/base.c | 39 +-
    drivers/dahdi/wctdm24xxp/wctdm24xxp.h | 2 +
    drivers/dahdi/wctdm24xxp/xhfc.c | 4 +-
    drivers/dahdi/wcte11xp.c | 4 +-
    drivers/dahdi/wcte12xp/Makefile | 6 -
    drivers/dahdi/wcte12xp/base.c | 147 +-
    drivers/dahdi/wcte12xp/wcte12xp.h | 1 +
    drivers/dahdi/wcte13xp-base.c | 3243 ++++++++++++++++++++
    drivers/dahdi/xpp/card_bri.c | 1287 ++++----
    drivers/dahdi/xpp/card_bri.h | 4 +-
    drivers/dahdi/xpp/card_echo.c | 116 +-
    drivers/dahdi/xpp/card_echo.h | 6 +-
    drivers/dahdi/xpp/card_fxo.c | 1201 +++++---
    drivers/dahdi/xpp/card_fxo.h | 17 +-
    drivers/dahdi/xpp/card_fxs.c | 1618 +++++-----
    drivers/dahdi/xpp/card_fxs.h | 18 +-
    drivers/dahdi/xpp/card_global.c | 561 ++--
    drivers/dahdi/xpp/card_global.h | 101 +-
    drivers/dahdi/xpp/card_pri.c | 1879 ++++++------
    drivers/dahdi/xpp/card_pri.h | 5 +-
    drivers/dahdi/xpp/dahdi_debug.c | 48 +-
    drivers/dahdi/xpp/dahdi_debug.h | 290 +-
    drivers/dahdi/xpp/firmwares/PIC_TYPE_2.hex | 676 ++--
    drivers/dahdi/xpp/init_card_1_30 | 8 +-
    drivers/dahdi/xpp/init_card_2_30 | 3 +
    drivers/dahdi/xpp/mmapbus.c | 12 +-
    drivers/dahdi/xpp/mmapdrv.c | 222 +-
    drivers/dahdi/xpp/parport_debug.c | 40 +-
    drivers/dahdi/xpp/parport_debug.h | 2 +-
    drivers/dahdi/xpp/print_fxo_modes.c | 31 +-
    drivers/dahdi/xpp/xbus-core.c | 1263 ++++----
    drivers/dahdi/xpp/xbus-core.h | 326 +-
    drivers/dahdi/xpp/xbus-pcm.c | 827 ++---
    drivers/dahdi/xpp/xbus-pcm.h | 111 +-
    drivers/dahdi/xpp/xbus-sysfs.c | 573 ++--
    drivers/dahdi/xpp/xdefs.h | 100 +-
    drivers/dahdi/xpp/xframe_queue.c | 187 +-
    drivers/dahdi/xpp/xframe_queue.h | 29 +-
    drivers/dahdi/xpp/xpd.h | 203 +-
    drivers/dahdi/xpp/xpp_dahdi.c | 714 ++---
    drivers/dahdi/xpp/xpp_dahdi.h | 22 +-
    drivers/dahdi/xpp/xpp_usb.c | 822 +++--
    drivers/dahdi/xpp/xproto.c | 392 +--
    drivers/dahdi/xpp/xproto.h | 244 +-
    include/dahdi/dahdi_config.h | 11 +-
    include/dahdi/kernel.h | 115 +-
    include/dahdi/user.h | 5 +
    85 files changed, 12798 insertions(+), 7673 deletions(-)



    For a full list of changes in these releases, please see the shortlog at:
    http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=shortlog;h=refs/tags/v2.7.0-rc1
    http://git.asterisk.org/gitweb/?p=dahdi/tools.git;a=shortlog;h=refs/tags/v2.7.0-rc1

    Issues found in this release can be reported in the DAHDI-Linux and
    DAHDI-Tools projects at https://issues.asterisk.org/jira

    Thank you for your continued support of Asterisk!

    --
    Russ Meyerriecks
    Digium, Inc. | Linux Kernel Developer



    New Asterisk Test Suite Developer
    Click to view a printable version Tue, 14 May 2013 02:59:07 +1200

    Matt Jordan has posted details of a new test suite developer at Digium:

    Hey all -

    We have a new addition to the Asterisk development team here at Digium who will be working on tests for the Asterisk Test Suite, specifically to help support Asterisk 12 - John Bigelow! While relatively new on the Asterisk development team, John has been working with Asterisk and Digium for a long time, and brings a lot of knowledge to the test process on how Asterisk is used and deployed.

    You will see some reviews to the Asterisk Test Suite by John already up on Review Board, and - in the very near future - commits as well. Please join me in welcoming John!

    Matt

    --
    Matthew Jordan
    Digium, Inc. | Engineering Manager



    Asterisk 12 Project Update
    Click to view a printable version Wed, 08 May 2013 04:05:31 +1200

    Hey everyone!

    It's been awhile since the last project update, and since we're heading into the home stretch on Asterisk 12, it felt like it was time for another project update. As a general overview of the state of things, all of the work for Asterisk 12 is now in the issue tracker and represented on the various projects' wiki pages. There's still a lot of things to get done, and lots of opportunities for participation and collaboration. If you're interested in any of the work, don't hesitate to ask in #asterisk-dev or on the asterisk-dev mailing list for ideas.

    We have tasks for every kind of participation, many of which require different levels of effort - so if you're wondering if there's something you could contribute towards, don't worry - there probably is!

    New SIP Channel Driver

    As Mark noted earlier, the new SIP channel driver is now in trunk. There's still a ways to go however, and new work is being merged into the team/group/pimp_my_sip branch as it is being completed. Recently, this included some work on being able to negotiate media in a fine grained fashion, call forwarding and diversion header support, SDES SRTP support, and out of call messaging support. WebSocket support is well on it's way, as is initial support for device state. snuffy has also started configuration documentation - a hugely needed effort to make all of this usable! You may note that the configuration documentation is actually in source as opposed to in a .conf file - due to some recent patches in trunk (work originally done by Terry Wilson aka otherwiseguy), configuration for some modules can be defined as XML documentation. That means the documentation will be up to date on the wiki, and the sample configuration files can actually be used for sample configuration - as opposed to just documentation.

    I'd be remiss if I didn't point out that we're currently having a discussion on what to name the new SIP channel driver - pop over to that thread and vote if you haven't already. If you don't vote, you can't complain about the results of the election!

    API Work

    Stasis-Core has now been in Asterisk trunk for awhile, and we continue to refactor AMI, CDRs, CEL (and whatever else we can get our hands on) on top of it. There's a lot of power in Stasis-Core: not only is the vast majority of Asterisk state now available in a pub/sub architecture, but you can aggregate this state into your own topic and route only the messages you care about to your module. David Lee has written design documentation for Stasis-Core on the wiki, and a new 'demo' module, res_statsd, for showing how to use Stasis-Core has been added to Asterisk. (Unfortunately, refactoring res_snmp over to Stasis-Core is a bit larger than a 'demo' task!)

    An initial cut of CDRs using Stasis-Core is up on Review Board, and we're working through a lot of the various AMI events now to get them onto Stasis-Core. CEL is coming up next!

    Stasis-HTTP continues to develop as the infrastructure in Asterisk expands to support the concepts it needs. This includes having the ability to treat endpoints as first class citizens within Asterisk, such that endpoints can have state associated with them that can be queried from resource modules. We're also currently hard at work on getting media playback up and running - expect a "Hello World" Stasis-HTTP sample in the relatively near future.

    Bridging Framework

    The bridging core continues to be developed in the team/group/bridge_construction branch and is starting to reach a point where it's incorporating more consumers of bridging within Asterisk.
    This includes:
    * Local channel optimization - this now occurs completely within the bridging framework (and appears to be just a tad bit faster!)
    * Transfers - initial support for externally initiated blind transfers is starting to go in. This includes chan_iax2, but also chan_sip.
    Expect chan_gulp and the other channel drivers to start getting some attention real soon!

    There's a lot to say about the power of the new bridging framework, and a lot of it is difficult to explain in just a single e-mail. Let's just say the white boards here are filled with diagrams trying to cover all the interesting corner cases that a dialplan writer may create (how do you park a multi-party bridge?) There's a lot of fun things to play with, now that channels can move between two-party and multi-party bridges seamlessly.

    Speaking of channels, if you're a maintainer of one of Asterisk's channel drivers, you may get an e-mail from me soon asking for some help converting your channel driver over to the new bridging framework. Not to fear - it's far less painful than you might think, and you'll like the number of lines of code you get to delete!


    Phew. I'm sure I'm forgetting things, and as you can see there's a lot of work going on and a lot left to do. As I mentioned previously, collaboration and help is always appreciated - whether you're testing, developing, documenting, or just providing input. It's coming along well, but we've still got a ways to go, and the more collaboration we get, the better Asterisk 12 will be.

    Thanks!

    Matt

    --
    Matthew Jordan
    Digium, Inc. | Engineering Manager



    Pimp My SIP merge
    Click to view a printable version Sat, 27 Apr 2013 03:43:09 +1200

    Mark Michelson has posted information about the merging of the Pimp My SIP branch to trunk:

    Hi!

    Those of you who watch the commits list have probably seen that the pimp_my_sip branch has been merged to trunk. The reason for this is that, with the exception of an API for handling incoming PUBLISH requests, the API for new SIP work has reached a stable point. There may still be forthcoming changes, but they will not be major.

    So does this mean that SIP development for Asterisk 12 is complete? Not by a long shot!

    For those of you brave enough to give what's in Asterisk trunk a whirl, here's a brief list of what you can do:

    • Basic calls (inbound and outbound)
      • Audio and video support
      • DTMF support for RFC 4733, inband, and INFO
      • Caller ID and limited Connected Line support
      • Session timers
      • PRACK
      • RFC 3326 (Reason header) supportAuthentication (inbound and outbound)
      • Direct media

    • Registration (inbound and outbound)
    • Call forwarding
    • Sending OPTIONs outbound
    • NAT traversal (including ICE support)
    • MWI (Just NOTIFY support, no SUBSCRIBE support)
    • SIP debugging
    • Configuration for the following items:
      • Endpoints ("peers" in chan_sip terminology)
      • Addresses of record and their contacts
      • Domains
      • Authentication
      • Transports (to include support for multiple transports)

    Here's a brief list of items that are currently in development and/or up for review:

    MESSAGE support (both in-call and out-of-call)
    A media negotiation dialplan function to explicitly set codecs on outbound calls
    SDES SRTP support
    Diversion header support

    Other upcoming tasks can be found on the SIP project page's JIRA issues section.

    Documentation for how to configure the new SIP work is slim for now. If you have questions or would like to improve documentation, please feel free to speak up. Currently, Brad Latus has a review up adding XML documentation for configuration items. It's a good first step towards making the new work more user-friendly.

    This merge is a milestone, of sorts, mostly due to the API stability. Developers interested in adding new features should continue working either in the pimp_my_sip SVN branch or in a branch based off of pimp_my_sip. We're not sure yet when the next batch of code will make it into trunk, but the next batch will in all likelihood be much smaller.

    Thanks for the support,
    Mark Michelson



    AstriCon 2013 10th Anniversary special
    Click to view a printable version Thu, 25 Apr 2013 07:14:30 +1200

    David Duffett has posted details of the 10th anniversary of AstriCon this year:

    I’m having to whisper, because I’ve got all the details about AstriCon 2013 – and I’m spilling them just for you, before anyone else has them. The official release will happen later today – so you really are the first to know!

    I can reveal that AstriCon, the only ‘must-attend’ Asterisk user event in the world, will be at the at the Renaissance Atlanta Waverly Hotel & Convention Center in Atlanta, GA (as its name suggests) from October 8-10 this year.

    We’re expecting more Asterisk enthusiasts, telephony geeks, smart business types that are in the know and gifted developers than ever before.

    Not only that, but Asterisk 12 *may* be released around that time, and we’ll have the Asterisk developers on hand to give you the lowdown on the most fundamental changes made to Asterisk for years.

    Read More...



    jasmin-remote-hold-1.8
    Click to view a printable version Tue, 23 Apr 2013 04:14:40 +1200

    Olle has done some work on remote hold for SIP for Asterisk:

    Friends,

    I've been working with a small piece of code that makes it possible for Asterisk to put a SIP phone on hold.

    Previously, if one party in a call puts the other party on hold, Asterisk plays music. With this code enabled,
    Asterisk will put the call on the other side of the bridge on hold too. This is beneficial if you have multiple
    Asterisks or use Asterisk as an endpoint, a phone.

    The code is ready for you to test. Provide feedback on this mailing list or any other channel.

    http://svn.digium.com/svn/asterisk/team/oej/jasmin-remote-hold-1.8

    Looking forward to your feedback!

    /O



    MWI expectations
    Click to view a printable version Fri, 12 Apr 2013 00:58:21 +1200

    Mark Michelson is looking for feedback on MWI while he works on SIP subscribe and notify events:

    Hi devs,

    I'm in the process of defining an API in the new SIP work for SIP SUBSCRIBE/NOTIFY events. As a proof of concept, I'm implementing MWI support. I'm writing to the list right now because I'm interested in what expectations are with regards to MWI.

    I'll begin by describing the behavior in chan_sip.

    In chan_sip, a peer configures mailboxes using the "mailbox" option in configuration. The option is a comma-separated list of mailboxes defined in voicemail.conf. In addition, there is an option called "subscribemwi" that determines whether we send MWI unsolicited or whether we require a SUBSCRIBE to be sent from the peer before we start sending MWI NOTIFYs.

    For unsolicited MWI, any time one of the configured mailboxes changes state, we create an MWI NOTIFY to send out. This MWI NOTIFY contains the combined number of new and old messages for all of the configured mailboxes for the SIP peer. What this means is that there is a many-to-one relationship between configured mailboxes and SIP MWI subscriptions.

    If using solicited MWI, then we first require a SUBSCRIBE for the "message-summary" event to arrive from the peer. When such a SUBSCRIBE arrives, we then exhibit the same behavior as for unsolicited MWI. That is, we send a NOTIFY with combined mailbox state of all configured mailboxes when any of the configured mailboxes changes state. If we receive a subsequent SUBSCRIBE from the same peer intending to start a new dialog, then we end the old dialog and use the new SUBSCRIBE as the basis for sending MWI NOTIFYs to the peer.

    This process works for many scenarios, but there are some limitations:

    * It assumes that the peer in question is a phone with a single MWI indicator. Combining all mailboxes' states into a single MWI NOTIFY makes sense when communicating with such a device, but it may not make as much sense for other types of endpoints.
    * If a SUBSCRIBE request arrives, there is no attempt to look at the request URI in order to try to determine if there is a specific resource that is desired. It is just assumed that configured mailboxes for the peer are what is wanted.
    * If multiple SUBSCRIBEs arrive, then the subsequent ones replace the previous ones as opposed to allowing multiple subscriptions to be active.

    For MWI in the new SIP work, I'd like to know what expectations are for its behavior. If the current chan_sip behavior is fine, then I can mirror its behavior. I suspect, though, that this is not what is desired, so I have a tentative idea for how MWI will function:

    Like with peers in chan_sip, we will allow for endpoints to configure mailboxes. Any mailboxes listed on an endpoint will be assumed to be unsolicited MWI. The directive on whether to combine the mailbox states into a single NOTIFY will be determined by a configuration option (e.g. "aggregate_mwi"). Thus if you had an endpoint configured like so:

    [alice]
    type=endpoint
    mailboxes=alice@vmcontext,tech-support@vmcontext
    aggregate_mwi=yes

    Then alice will be sent a NOTIFY with the aggregated voicemail state of alice@vmcontext and tech-support@vmcontext any time that either of the mailboxes changes state. If the aggregate_mwi option had been set to "no" then alice would be sent individual NOTIFYs with each individual mailbox's state when the mailbox in question changes state.

    For solicited MWI, the request-URI of the SUBSCRIBE will determine what mailbox(es) to subscribe to. The request-URI will map to a configured address of record in res_sip.conf. Mailbox options on that AOR will determine what mailboxes' states are reported. If you had an aor configured like so:

    [sales]
    type=aor
    mailboxes=jim@vmcontext,bob@vmcontext,lucius@vmcontext

    Then if a SUBSCRIBE arrives that wishes to subscribe to "sales" then the result would be a subscription that relays the combined state of the three configured mailboxes in the "sales" aor. Unlike with endpoints, there is no need for an aggregate_mwi option since the mailboxes have been explicitly grouped under the banner of a single aor. If non-aggregated state were desired, then the mailboxes can be placed into separate aors, and individual subscriptions can be made to each aor in order to subscribe to the individual state of each mailbox.

    With this method in place, it allows for an endpoint to use a combination of solicited and unsolicited MWI if desired, and it allows for multiple MWI subscriptions, which is currently missing from chan_sip. It also allows for central definitions of arbitrary groups of related mailboxes.

    Let me know what you think. Is the current scheme used by chan_sip exactly what should be done for the new SIP work? Are my ideas for configuration in the new SIP work great/okay/awful or in need of further explanation? Are there any shortcomings of MWI support in chan_sip that I have not touched upon that should be addressed in the new SIP work?

    Thanks in advance for your feedback,
    Mark Michelson



    Pinefool: Poor mans PLC
    Click to view a printable version Wed, 10 Apr 2013 07:25:12 +1200

    Olle has been doing some work in the RTP channel on Packet Loss Concealment:

    Hi!

    We see a lot of issues in the RTP streams when bridging two sip calls in Asterisk. If there's packet loss coming in, it's the same going out - but in this case the sequence numbers doesn't indicate packet loss. This causes issues with recordings sounding bad and possibly other issues since we have quite long phone calls.

    Investigating the PLC in Asterisk, according to Malcolm's wiki article there has to be a core bridge with transcoding and a jitter buffer. That won't work for us.

    Looking into the RTP channel, where I've spent some time recently, I notice that we don't bother checking that packets are sequential, so the RTP channel doesn't bother with packet loss much. That needed to change.

    I created a very simple PLC in the RTP channel for the incoming stream, following these principles:

    - If there is packet loss, resend the previous packet. We discover this at the next packet, so we introduce two frames at once. In most cases with small packet loss, the receiving end will sort this out in the jitter buffer.
    - If there is packet reordering, a jitter causing a packet to arrive too late, we ignore that packet since we've already replaced it
    with another one.

    Normally PLC would work hand-in-hand with a jitter buffer so we first reorder incoming RTP packets, then add the missing ones. In this case, I can't afford introducing extra delay caused by a jitter buffer, so I needed a poor man's implementation. It will add a bit of CPU, copying frames that wasn't done previously, but it is not a massive amount of operations - unless you have a really poor connection. In that case, audio will be lost anyway and there's nothing much we can do to save the situation.

    How does it work?
    ================
    - If we receive packets 2,3,5 - we will send 2,3,3,5 into the bridge, making sure that we get no skew.
    - If we receive 2,3,5,4 we will send 2,3,3,5 and throw away #4.

    In both cases, the extra "3" will be sent at the same time as the "5" into the bridge.

    Looking at wireshark on the other side of the call, the outbound stream now looks better. Of course, there's jitter but no packet loss in the stream, we still have all expected frames - for Alaw 50 RTP frames per second. Without this PLC we could get 45 and the playback of a recording would sound "fast" - Donald Duckish... :-)

    It is a small hack compared with other RTP hacks I've worked with recently- the DTMF length, Comfort Noise and others. But it does make the audio better in case of bad incoming streams.

    Please test!

    http://svn.digium.com/svn/asterisk/team/group/pinefool-poor-mans-plc-1.4
    http://svn.digium.com/svn/asterisk/team/oej/pinefool-poor-mans-plc-1.8

    There's also a soup of the day branch with a larger set of patches combined:
    - PLC
    - Comfort Noise support
    - DTMF
    - PRACK
    - RTCP improvements

    http://svn.digium.com/svn/asterisk/team/oej/pinetestedition-1.8
    Look at the patches and the README files in the /patches directory of that branch for more information.

    Thank you for your tests and feedback!
    I could have missed some serious issues or a cool trick we could add while working with this.

    /O



    AST-2013-003: Username disclosure in SIP channel driver
    Click to view a printable version Wed, 10 Apr 2013 04:35:42 +1200

                  Asterisk Project Security Advisory - AST-2013-003
    
             Product         Asterisk                                            
             Summary         Username disclosure in SIP
                             channel driver           
        Nature of Advisory   Unauthorized data disclosure                        
          Susceptibility     Remote Unauthenticated Sessions                     
             Severity        Moderate                                            
          Exploits Known     No                                                  
           Reported On       January 30, 2013                                    
           Reported By       Walter Doekes, OSSO B.V.                            
            Posted On        February 21, 2013                                   
         Last Updated On     March 27, 2013                                      
         Advisory Contact    Kinsey Moore                     
             CVE Name        CVE-2013-2264                                       
    
       Description  When authenticating via SIP with alwaysauthreject enabled,   
                    allowguest disabled, and autocreatepeer disabled, Asterisk   
                    discloses whether a user exists for INVITE, SUBSCRIBE, and   
                    REGISTER transactions in multiple ways.                      
    
                    This information was disclosed:                              
    
                    * when a "407 Proxy Authentication Required" response was    
                    sent instead of "401 Unauthorized" response.                 
    
                    * due to the presence or absence of additional tags at the   
                    end of "403 Forbidden" such as "(Bad auth)".                 
    
                    * when a "401 Unauthorized" response was sent instead of     
                    "403 Forbidden" response after a retransmission.             
    
                    * when retransmissions were sent when a matching peer did    
                    not exist, but were not when a matching peer did exist.      
    
       Resolution  This issue can only be mitigated by upgrading to versions of  
                   Asterisk that contain the patch or applying the patch.        
    
                                  Affected Versions
                   Product                Release Series    
             Asterisk Open Source              1.8.x        All Versions         
             Asterisk Open Source              10.x         All Versions         
             Asterisk Open Source              11.x         All Versions         
              Certified Asterisk              1.8.15        All Versions         
          Asterisk Business Edition            C.3.x        All Versions         
            Asterisk Digiumphones        10.x-digiumphones  All Versions         
    
                                     Corrected In
                     Product                              Release                
              Asterisk Open Source               1.8.20.2, 10.12.2, 11.2.2       
              Asterisk Digiumphones                10.12.2-digiumphones          
               Certified Asterisk                      1.8.15-cert2              
            Asterisk Business Edition                     C.3.8.1                
    
                                        Patches                             
                                   SVN URL                                  Revision  
    http://downloads.asterisk.org/pub/security/AST-2013-003-1.8.diff         Asterisk  
                                                                            1.8       
    http://downloads.asterisk.org/pub/security/AST-2013-003-10.diff          Asterisk  
                                                                            10        
    http://downloads.asterisk.org/pub/security/AST-2013-003-11.diff          Asterisk  
                                                                            11        
    http://downloads.asterisk.org/pub/security/AST-2013-003-1.8.15-cert.diff Certified 
                                                                            Asterisk  
                                                                            1.8.15    
    http://downloads.asterisk.org/pub/security/AST-2013-003-C.3.diff         Asterisk  
                                                                            BE C.3    
    
          Links    ASTERISK-21013       
    
       Asterisk Project Security Advisories are posted at                        
       http://www.asterisk.org/security                                          
    
       This document may be superseded by later versions; if so, the latest      
       version will be posted at                                                 
       http://downloads.digium.com/pub/security/AST-2013-003.pdf and             
       http://downloads.digium.com/pub/security/AST-2013-003.html                
    
                                   Revision History
           Date          Editor                    Revisions Made                
       2013-02-20    Kinsey Moore    Initial revision.                           
       2013-02-27    Kinsey Moore    Added Asterisk BE patch information.        
       2013-02-27    Kinsey Moore    Corrected open source Asterisk versions.    
    
                  Asterisk Project Security Advisory - AST-2013-003
                 Copyright (c) 2013 Digium, Inc. All Rights Reserved.
     Permission is hereby granted to distribute and publish this advisory in its
                              original, unaltered form.





    AST-2013-002: Denial of Service in HTTP server
    Click to view a printable version Wed, 10 Apr 2013 04:30:10 +1200

                  Asterisk Project Security Advisory - AST-2013-002
    
             Product         Asterisk                                            
             Summary         Denial of Service in HTTP server                    
        Nature of Advisory   Denial of Service                                   
          Susceptibility     Remote Unauthenticated Sessions                     
             Severity        Major                                               
          Exploits Known     None                                                
           Reported On       January 21, 2013                                    
           Reported By       Christoph Hebeisen, TELUS Security Labs             
            Posted On        March 27, 2013                                      
         Last Updated On     March 27, 2013                                      
         Advisory Contact    Mark Michelson        
             CVE Name        CVE-2013-2686                                       
    
      Description AST-2012-014, fixed in January of this year, contained a   
                  fix for Asterisk's HTTP server since it was susceptible to a   
                  remotely-triggered crash.                                      
    
                  The fix put in place fixed the possibility for the crash to be 
                  triggered, but a possible denial of service still exists if an 
                  attacker sends one or more HTTP POST requests with very large  
                  Content-Length values.                                         
    
       Resolution  Content-Length is now capped at a maximum value of 1024       
                   bytes. Any attempt to send an HTTP POST with content-length   
                   greater than this cap will not result in any memory           
                   allocated. The POST will be responded to with an HTTP 413     
                   "Request Entity Too Large" response.                          
    
                                  Affected Versions
              Product          Release Series    
       Asterisk Open Source         1.8.x        1.8.19.1, 1.8.20.0, 1.8.20.1    
       Asterisk Open Source         10.x         10.11.1, 10.12.0, 10.12.1       
       Asterisk Open Source         11.x         11.1.2, 11.2.0, 11.2.1          
        Certified Asterisk         1.8.15        1.8.15-cert1                    
       Asterisk Digiumphones  10.x-digiumphones  10.11.1-digiumphones,           
                                                 10.12.0-digiumphones,           
                                                 10.12.1-digiumphones            
    
                                     Corrected In
                    Product                              Release                 
             Asterisk Open Source               1.8.20.2, 10.12.2, 11.2.2        
              Certified Asterisk                      1.8.15-cert2               
             Asterisk Digiumphones                10.12.2-digiumphones           
    
                                        Patches                             
                                   SVN URL                                  Revision  
    http://downloads.asterisk.org/pub/security/AST-2012-014-1.8.diff         Asterisk  
                                                                            1.8       
    http://downloads.asterisk.org/pub/security/AST-2012-014-10.diff          Asterisk  
                                                                            10        
    http://downloads.asterisk.org/pub/security/AST-2012-014-11.diff          Asterisk  
                                                                            11        
    http://downloads.asterisk.org/pub/security/AST-2012-014-1.8.15-cert.diff Certified 
                                                                            Asterisk  
                                                                            1.8.15    
    
      +------------------------------------------------------------------------+
      |  Links   | ASTERISK-20967                                              |
      |          | http://telussecuritylabs.com/threats/show/TSL20130327-01    |
      +------------------------------------------------------------------------+
    
       Asterisk Project Security Advisories are posted at                        
       http://www.asterisk.org/security
    
       This document may be superseded by later versions; if so, the latest      
       version will be posted at                                                 
       http://downloads.digium.com/pub/security/AST-2013-002.pdf and             
       http://downloads.digium.com/pub/security/AST-2013-002.html                
    
                                   Revision History
               Date                  Editor               Revisions Made         
       February 12, 2013      Mark Michelson        Initial Draft                
       March 27, 2013         Matt Jordan           Updated CVE                  
    
                  Asterisk Project Security Advisory - AST-2013-002
                 Copyright (c) 2013 Digium, Inc. All Rights Reserved.
     Permission is hereby granted to distribute and publish this advisory in its
                              original, unaltered form.





    AST-2013-001: Buffer Overflow Exploit Through SIP SDP Header
    Click to view a printable version Wed, 10 Apr 2013 04:04:40 +1200

                  Asterisk Project Security Advisory - AST-2013-001
    
             Product         Asterisk                                            
             Summary         Buffer Overflow Exploit Through 
                             SIP SDP Header      
        Nature of Advisory   Exploitable Stack Buffer 
                             Overflow                   
          Susceptibility     Remote Unauthenticated Sessions                     
             Severity        Major                                               
          Exploits Known     No                                                  
           Reported On       6 January, 2013                                     
           Reported By       Ulf Ha:rnhammar                                     
            Posted On        27 March, 2013                                      
         Last Updated On     March 27, 2013                                      
         Advisory Contact    Jonathan Rose              
             CVE Name        CVE-2013-2685                                       
    
       Description  The format attribute resource for h264 video performs an     
                    unsafe read against a media attribute when parsing the SDP.  
                    The vulnerable parameter can be received as strings of an    
                    arbitrary length and Asterisk attempts to read them into     
                    limited buffer spaces without applying a limit to the        
                    number of characters read. If a message is formed            
                    improperly, this could lead to an attacker being able to     
                    execute arbitrary code remotely.                             
    
       Resolution  Attempts to read string data into the buffers noted are now   
                   explicitly limited by the size of the buffers.                
    
                                  Affected Versions
                   Product              Release Series  
            Asterisk Open Source             11.x       All Versions             
    
                                     Corrected In  
                        Product                              Release             
                  Asterisk Open Source                        11.2.2             
    
                                       Patches                        
                                  SVN URL                              Revision  
      Http://downloads.asterisk.org/pub/security/AST-2013-001-11.diff Asterisk   
                                                                      11         
    
          Links ASTERISK-20901
    
       Asterisk Project Security Advisories are posted at                        
       http://www.asterisk.org/security                                          
    
       This document may be superseded by later versions; if so, the latest      
       version will be posted at                                                 
       http://downloads.digium.com/pub/security/AST-2013-001.pdf and             
       http://downloads.digium.com/pub/security/AST-2013-001.html                
    
                                   Revision History
               Date                  Editor               Revisions Made         
       February 11, 2013      Jonathan Rose         Initial Draft                
       March 27, 2013         Matt Jordan           CVE Added                    
    
                  Asterisk Project Security Advisory - AST-2013-001
                 Copyright (c) 2013 Digium, Inc. All Rights Reserved.
     Permission is hereby granted to distribute and publish this advisory in its
                              original, unaltered form.
    





    QueueMetrics 13.04 released
    Click to view a printable version Sun, 07 Apr 2013 09:49:23 +1200

    Lenz has posted details of the latest release of Queue Metrics:

    QueueMetrics version 13.04 is available for download today. This version includes an important number of changes and new features that make QueueMetrics significantly better.

    First of all, the long awaited report export is present. You can export complete reports to PDF or XLS files. This works for the report you are seeing in front of you and can be set up to work automatically, sending the report to a list of recipients over e-mail.

    The QA section was improved by adding non-scoring questions and comments. This works especially well from the agent's page - this means your agents can now gather structured and free-text information on the current call right from their own screen, using specially crafted QA forms.

    The agent's page itself was improved - now it leverages the caching features added in 12.10 in order to refresh automatically when a call comes in. This works beautifully for inbound and can easily be leveraged for outbound when running WombatDialer.

    A new kind of report was added - IVR traversals. This lets you track in-deep IVR activity and understand what happens even before the call hits the queue. reached goals and attrition points can easily be displayed. A compatible module was added to FreePBX 2.11 to make integration easier.

    Another feature that was often asked for is the detailed tracking of attempts on each call - in fact now all events happening on a call (like MOH, attempts or IVR events) are visible in detail. No more wondering why a call took so long to be answered!

    Last but not least, QueueMetrics is nowadays often deployed as a cloud service. The new QlogSplitter that ships with Qloaderd lets you break the relationship between a single Asterisk server and a QueueMetrics instance - now you can have multiple clients sharing the same Asterisk instances having separate QueueMetrics instances, with a no-strings-attached pricing thanks to the Keyring.

    The new release also includes a number of lesser changes and bugs fixed - in total almost 100 different issues were addressed.

    Please rest assured that this release works with the same activation keys you used for your current version of QueueMetrics.

    Enjoy the update,
    Lorenzo Emilitri
    Managing Director, Loway



    Please test Pinefrog-rtcp-1.8
    Click to view a printable version Thu, 14 Mar 2013 06:30:00 +1300

    Olle has posted a request for people to test his RTCP branch:

    Hi!

    I've spent a few days porting my RTCP work for Asterisk 1.4 to Asterisk 1.8, which has a different RTP subsystem. When this works and we're satisfied with these changes, it will be much more simple to port it to 11 and trunk than starting with 1.4.

    I haven't changed the actual data in RTCP - number of packets, jitter, loss, latency - but I've changed the way we handle RTCP and exchange messages as well as added extended reporting. Asterisk 1.8 already had manager events, but I've added Call Quality Records.

    Earlier discussions about CDRs on this mailing list concluded that there's no way possible to store this data for every call scenario in the CDRs. That's why I decided to create CQRs. The Asterisk SIP channel now stores records for every RTP stream in a database, with pointers to both SIP call IDs and Asterisk channel IDs (uniquieid and channel name). Optionally we can log them to the "cqr" logging channel.

    Many of these changes is done in the RTP core engine and the Asterisk RTP subsystem, making it available for other channels that use RTP too.


    A few GUI developers seems to have taken the challenge to do something useful with this data, including Edgar Landivar in the Elastix project. On the list of my ideas:

    - Monitoring of SIP trunks. What's the quality of the latest 10 calls, the latest 100? Is it getting worse or better? Mark the sip trunk useless in worst case and fail over to another one in the dialplan.
    - Monitoring of long calls in call centers - use the manager reports and monitor call quality. If an agent fills the ethernet cable with file transfers and cool youtube videos, don't send any new calls
    - Checking call quality for individual peers, reporting issues
    - Nagios/Icinga integration


    I'm not fully done, still looking for some issues. The branch is fully useful and ready for testing. If you want to help testing and give feedback the time is now. Read the README.pinefrog and send me feedback.

    Thanks to NORDICOM Norway AS that has funded this work.

    Instructions
    =========
    svn checkout http://svn.digium.com/svn/asterisk/team/oej/pinefrog-rtcp-1.8

    configure, build and install like any other Asterisk branch!

    I'm looking forward to your feedback!

    Best regards,
    /O



    DAHDI-Linux and DAHDI-Tools 2.6.2 Now Available
    Click to view a printable version Tue, 12 Mar 2013 09:57:49 +1300

    The Asterisk Development Team has announced the release of:

    DAHDI-Linux 2.6.2
    DAHDI-Tools 2.6.2
    DAHDI-Linux-Complete 2.6.2+2.6.2

    This release is available for immediate download at:
    http://downloads.asterisk.org/pub/telephony/dahdi-linux
    http://downloads.asterisk.org/pub/telephony/dahdi-tools
    http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete

    2.6.2 is a bugfix release of which the most noteable changes are:
    - Fix compile error on RHEL 5.2 / Centos 5.9 and later
    - Development switch from svn to git

    Issues closed in this release:
    DAHLIN-314 wcb4xxp: kernel oopses when debug st state is enabled
    DAHLIN-302 "Failed to apply echo can changes on channel 243 00107001!" in +/var/log/syslog.
    DAHLIN-298 dahdi-linux 2.6.1 fails to detect ringing
    DAHLIN-312 Error: conflicting types for 'bool' when compiled after CentOS upgraded +to 5.9, kernel 2.6.18-348.el5
    DAHLIN-313 When I upgraded to CENTOS 5 V
    DAHLIN-315 Error while asterisk to upgrade to centos 5.9

    Shortlog of changes since v2.6.1:
    Doug Bailey (1):
    Assign NULL values to pointers to insure that future kfree calls do not cause errors.

    Oron Peled (1):
    xpp: usermode_helper() bugfix for kernels >= 3.3.0

    Shaun Ruffell (12):
    wcte12xp: Destroy the cache if the linemode is not recognized.
    wcte12xp: Allow default_linemode to be set to j1.
    wcte12xp: Fix pulse digit detection when set for FXO signalling modes.
    wcte12xp: Fix stack corruption when checking T1 RBS states.
    dahdi: pci-aspm.h was included in 2.6.26 not 2.6.25.
    xpp: Do not typedef bool on RHEL 5.2 or later.
    wctdm24xxp: Only two polarity reversals are needed to validate RING on FXO ports.
    wct4xxp: EC channel calculation in TONEDETECT assumes TE820.
    wct4xxp: t4_serial_setup() was called more often than necessary.
    wcb4xxp: Allocate memory in hfc_decode_st_state() with GFP_ATOMIC.
    wctdm24xxp: Use framecount and not jiffies when looking for battery present.
    wctdm24xxp: Eliminate chance for channel to be stuck in RED alarm.

    Tzafrir Cohen (4):
    xpp: pre/post_unregister: not for the EC
    Add .gitignore file
    gitignore: Add README.html to git ignore list
    Redefine the removed __dev* for now

    The diffstat from the v2.6.1 release:

    .gitignore                        |  41 +++
    .version                          |   1 -
    ChangeLog                         | 555 --------------------------------------
    drivers/dahdi/wcb4xxp/base.c      |   3 +-
    drivers/dahdi/wct4xxp/base.c      |  12 +-
    drivers/dahdi/wctdm24xxp/base.c   |  43 +--
    drivers/dahdi/wcte12xp/base.c     | 238 +++++++++++++---
    drivers/dahdi/wcte12xp/wcte12xp.h |   1 +
    drivers/dahdi/xpp/card_global.c   |  12 +-
    drivers/dahdi/xpp/xdefs.h         |  14 +-
    drivers/dahdi/xpp/xpp_dahdi.c     |   4 +-
    include/dahdi/kernel.h            |  10 +-
    12 files changed, 300 insertions(+), 634 deletions(-)

    For a full list of changes in these releases, please see the shortlog at:
    http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=shortlog;h=refs/tags/v2.6.2
    http://git.asterisk.org/gitweb/?p=dahdi/tools.git;a=shortlog;h=refs/tags/v2.6.2

    Issues found in this release can be reported in the DAHDI-Linux and DAHDI-Tools projects at https://issues.asterisk.org/jira

    Thank you for your continued support of Asterisk!

    --
    Russ Meyerriecks
    Digium, Inc. | Linux Kernel Developer



    AppKonference 2.3
    Click to view a printable version Wed, 27 Feb 2013 04:51:44 +1300

    Paul Albrecht has posted details of the latest release of AppKonference:

    Hi,

    I have released an updated AppKonference that compiles with Asterisk 10 and 11.

    You can download the latest source from sourceforge: sourceforge.net/projects/appkonference

    --
    Paul Albrecht



    Asterisk 12 Update - Bridge Work
    Click to view a printable version Wed, 13 Feb 2013 10:40:39 +1300

    Matthew Jordan has posted details about the work that has been going on in the bridging code for Asterisk - another great advance:

    Hello!

    Some folks have noticed that we've been doing a lot of work on the bridging functionality in Asterisk. The motivation for this came out of discussions at AstriDevCon, where many people noted the difficulties external applications have when attempting to maintain the state of who is talking to whom. These problems often start with Asterisk's masquerade operation, which itself is a symptom of problems in the way that Asterisk has historically bridged channels.

    Luckily, Asterisk already has a solution to this (and has since 1.6 days) - Joshua Colp's Bridging API. The API is already used by ConfBridge and Page, but hasn't been deployed outside of those applications. Since we have a good layer of abstraction for bridging already implemented, and since it provides a model of bridging that lends itself well to the needs of external APIs (and Asterisk users!), we started refactoring Asterisk to use the Bridging API for all of its bridging.

    Up until now, the work for this has been tracked under the API project, but the scope of the work being done really deserves its own page.
    Thus, there is now a project page for this up on the wiki. This provides a dedicated place to document project planning, design, tests, and other useful information for people who want to participate and contribute to the project.

    So what's been done/being done?

    If we're going to tear into the guts of such a fundamental concept, we have to have some assurance that we can stitch it all back together! So for some time, we've been working on a large set of tests that cover the currently supported bridging features. You can find the current bridging tests in the Asterisk Test Suite, and the test plan that they are implementing on a sub page of the project page. There's more to be done here, however, so expect additional tests to pop up (and more reviews on Review Board for the tests).

    In addition to testing, we've started the work on expanding usage of the Bridge API to the various consumers that initiate bridging, starting with the Bridge application. Richard has created an integration branch for this work, and is currently working through the various ways in which channels can be moved between bridges. That includes things like transfers, but also Local channel optimization, merging bridges, swapping between two party and multi-party bridge technologies, and a whole host of other cases.

    In parallel, Jonathan has been working on implementing timed features. Previously, the Bridging API provided initial implementations of DTMF triggered features, such as blind and attended transfers and hangup. However, the Bridge application (as well as Dial) allows for features that occur on a timed or periodic basis, such as hanging up a channel after n seconds. Luckily, Joshua had already started on this work as well, so Jonathan brought that up to trunk and has started to merge it into Richard's work.

    Some of the next steps on the bridging work are going to entail:
    * Finish up the API cleanup and threading model changes
    * Get the other consumers of bridging moved over to the new Bridging API
    * Implement an API that allows external entities to initiate a transfer operation on channels in a bridge
    * Refactor the Park family of applications to use a new bridging technology

    As always, if you're interested in any aspect of this work, feel free to reply to this e-mail, start a new discussion, or talk about it in #asterisk-dev. We also accept message via pigeon, but given how fast things are moving, I'd recommend an electronic form of communication.

    tl;dr: We're going to use the Bridging API for lots of stuff.
    Collaboration is always welcome :-)

    --
    Matthew Jordan
    Digium, Inc. | Engineering Manager



    Specific SIP packets can cause ethernet controller reset
    Click to view a printable version Wed, 13 Feb 2013 08:20:22 +1300

    Kristian Kielhofner has posted a blog entry about packets that can cause your ethernet controller to reset.

    The original article now contains links to updates from Intel too.

    Here's an excerpt from the article:

    Packets of death. I started calling them that because that’s exactly what they are.

    Star2Star has a hardware OEM that has built the last two versions of our on-premise customer appliance. I’ll get more into this appliance and the magic it provides in another post. For now let’s focus on these killer packets.

    About a year ago we released a refresh of this on-premise equipment. It started off simple enough, pretty much just standard Moore’s Law stuff. Bigger, better, faster, cheaper. The new hardware was 64-bit capable, had 8X as much RAM, could accommodate additional local storage, and had four Intel (my preferred ethernet controller vendor) gigabit ethernet ports. We had (and have) all kinds of ideas for these four ports. All in all it was pretty exciting.

    This new hardware flew through performance and functionality testing. The speed was there and the reliability was there. Perfect. After this extensive testing we slowly rolled the hardware out to a few beta sites. Sure enough, problems started to appear.

    All it takes is a quick Google search to see that the Intel 82574L ethernet controller has had at least a few problems. Including, but not necessarily limited to, EEPROM issues, ASPM bugs, MSI-X quirks, etc. We spent several months dealing with each and every one of these. We thought we were done.

    We weren’t. It was only going to get worse.

    I thought I had the perfect software image (and BIOS) developed and deployed. However, that’s not what the field was telling us. Units kept failing. Sometimes a reboot would bring the unit back, usually it wouldn’t. When the unit was shipped back, however, it would work when tested.

    Wow. Things just got weird.

    Read the rest of the article



    VoIPGMap: Graphing active Asterisk calls on Google Maps
    Click to view a printable version Sat, 09 Feb 2013 08:44:57 +1300

    Markus has posted details of a script he has created to plot calls on a Google map:

    Hi list,

    I've always wanted to graph my active SIP calls on a map somehow, and now I've finally taken the time to do it. My script is called VoIPGMap and it displays active calls on Google Maps. Logic-wise it's designed for a callthrough or calling card scenario, where calls will be displayed to originate from a single location, but you can surely adjust it to instead show caller and callee location connected.

    I think the cool part is that it really tries to get the most accurate coordinates for every destination, down to the city level, by using text (country code, ISO, lat/long) and MySQL (ITU self-compiled DB + MaxMind public DB) based databases, which are included.

    Here are some demos:

    http://truemetal.org/universe/voipgmap/live_demo1.html
    http://truemetal.org/universe/voipgmap/live_demo2.html
    http://truemetal.org/universe/voipgmap/live_demo3.html
    http://truemetal.org/universe/voipgmap/live_demo4.html

    First I thought about calling it AstGMap but then realized you can simply feed it any form of E.164 numbers so I chose VoIPGMap instead :)
    If you feel like rewriting it to PHP or something like that for better performance, please do so. Unfortunately I'm not a programmer or I wouldn't have chosen shellscript. :)

    You can get it here:
    http://sourceforge.net/projects/voipgmap/files/

    And if you like it and decide to use the output on the web somewhere, please let me know the URL so that I can check it out, thanks!

    Regards
    Markus



    New Asterisk Developer - Kevin Harwell
    Click to view a printable version Thu, 07 Feb 2013 10:01:29 +1300

    Matt Jordan has posted details of a new developer at Digium:

    Hello everyone!

    Kevin Harwell (kharwell) has recently joined the Asterisk development team here at Digium, and will be contributing to all areas of the Asterisk project. Kevin has a diverse software development background, including voice recognition, video applications, database integration, and more.

    Please join me in welcoming Kevin!

    --
    Matthew Jordan
    Digium, Inc. | Engineering Manager



    Automatic queue recalls with WombatDialer and QueueMetrics
    Click to view a printable version Wed, 30 Jan 2013 03:50:02 +1300

    Lenz has posted a blog article on requeueing calls in Asterisk.

    Excerpt from his article:

    If you run a call center, serving clients in a timely way is often very complex, as it requires having enough people available to handle traffic spikes. The number of callers that disconnect because they have been waiting too long in a queue is then an important driver of the quality of your work, and these frustrated callers are the focus of much attention and scheduling/planning efforts in all call centers. This is because in a traditional setting doing inbound calling you basically had no other way of servicing the client but waiting for the person to call in.

    Read More



    24 Jan 2013 SIP project update
    Click to view a printable version Fri, 25 Jan 2013 11:12:23 +1300

    Mark Michelson has posted an update on the SIP redevelopment project:

    Hi everyone,

    It's been a while since I sent out an update about what's going on with the new SIP project in Asterisk. Here's what's currently going on:

    Joshua Colp and I have recently completed a milestone of being able to run a SIP call through the new SIP service in Asterisk. You can find a review of this work on reviewboard. If you have objections to design decisions that have been made so far or can foresee problematic areas, now would be a great time to make it known on that review.

    On the front of making PJProject packageable, David Lee recently created a git repo where the externally-located PJProject will be located once it is no longer bundled with Asterisk. Jason Parker is working on a branch right now where he is making improvements to the PJProject code and build system in an effort to pave the way to make it possible to create shared libraries for the different PJProject components. This work will be provided to Teluu, the maintainers of PJProject.

    On the lower end, I recently merged threadpool support into Asterisk. While the new SIP code is not using this at the moment, it eventually will. Other portions of Asterisk may also see benefit since the threadpool is written to be generic, not with SIP in mind. More exciting is Joshua Colp's recent work on a data access layer for Asterisk, nicknamed "sorcery". With this, we will streamline data access so that applications will not have to have special code sections depending on how configuration is stored (e.g. there won't be separate realtime and static configuration code). This will also help in facilitating easy backwards-compatibility of classic chan_sip's configuration. The sorcery work is in its final stages and should be integrated very soon.

    As far as future development is concerned, you can watch the "Project Planning" section of the project page on the wiki to see all related upcoming work. Priority right now is being placed on improving the calling experience with SIP at the moment. After sorcery is committed to trunk, you can expect to see it used heavily in the SIP work. The first place you will see it used is for endpoint location and identification. In addition, expect to see some work going into request authentication soon.

    Mark Michelson




    Somethings going on with Asterisk. And its good. Very good.
    Click to view a printable version Thu, 24 Jan 2013 07:00:07 +1300

    Olle has written a blog entry on changes that are afoot in Asterisk.

    Excerpt from his article:

    I must say that I’m impressed with the new project leader for Asterisk, Matt Jordan. There are big changes happening under his leadership and I don’t think everyone has understood the huge tasks he’s taken on with his team. I’m not sure I’ve understood all of it either.

    Here’s a list of things I’ve seen on the mailing lists:

    • Dialog with the community is much improved. Brainstorms happen in the open and things are developed in the open. This is a big change. I just hope I had more time to engage, but that’s my problem. The community seems a bit surprised too. I expected much more participation. We need to relearn how to contribute and discuss, now that we’re allowed to do that again.

      Digium had to re-learn how to interact with the community – but that goes two ways. The community needs to stand up to the challenge. Get involved.
    • Old issues that we wanted to forget are attacked and fixed. Like Masquerade and bridging – it’s all about to change. When I worked with Terry Wilson on the SIP transfer code in 1.2 we learned to hate Masquerade. That’s years ago and the masquerade is still around, annoying all Asterisk developers. It’s a complex operation that, well, causes a lot of issues. Time to get rid of it and get a modern multichannel bridge active in every call.
    • Manager is being changed dramatically. Manager is getting an overhaul into something more logical for the current design of Asterisk and languages and methods used to build AMI applications. Yes, this will affect everyone’s 3rd party applications. Applications needs to move on, sometimes we just can’t be backwards compatible. Most applications doesn’t support punch cards and output exceed 80 characters per line quite often… Asterisk is ten years old and it’s time for some new designs. Asterisk 11 is a LTS that will live for a long time, giving developers time to adopt to the new Asterisk.
    • The SIP Channel is rewritten. This is a gigantic project and has been needed for almost ten years. I never thought we could get funding for a complete rewrite, so I opted for remaking the current code in my old Codename Pineapple project that never got fully funded at the time. I have tons of opinions about a new SIP channel, but have no resources to really participate, just add a comment here and there like a grumpy old grandpa…

      I wish the developers follow my SIP2012 efforts (soon to be renamed SIP2014) and learn modern SIP and have the new SIP functions as part of the design. I do hope that they DO NOT base it on the current SIP channel that lacks transaction support, has poor ideas on SIP branching and forking and… You know. The new SIP implementation is required to work in larger SIP networks with proxies, unlike the old SIP channel. We need domain support, we need SIP URI support, we need security that works. That means that the developers need a better understanding of SIP than has been the tradition in the Asterisk development team. They’re always free to contact me with questions, which would be a positive change too.
    • New APIs are developed. Manager and AGI and ExternIVR are things that have happened and evolved but was never part of a consolidated effort to create a good unified API to Asterisk. We’ve discussed this at many Astridevcons under different names – PineMango was one of them. For Asterisk to survive, we need a modern API.
      Realtime is changing. Josh Colp is working on a new object handling system, again something discussed many times that finally happens. We do need a replacement of the poor realtime architecture that has been extended beyond control. We need to be able to use an API to manage in-memory objects in real time. It seems to me that this is exactly what File is working on.

    Read More...



    Asterisk 1.8.20.1, 10.12.1 and 11.2.1 Now Available
    Click to view a printable version Wed, 23 Jan 2013 11:55:40 +1300

    The Asterisk Development Team has announced the release of Asterisk 1.8.20.1, 10.12.1 and 11.2.1.

    This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

    The release of Asterisk 1.8.20.1, 10.12.1 and 11.2.1 resolves several issues reported by the community and would have not been possible without your participation.
    Thank you!

    The following are the issues resolved in this release:

    * --- Fix astcanary startup problem due to wrong pid value from before daemon call
    (Closes issue ASTERISK-20947. Reported by Jakob Hirsch)

    * --- Update init.d scripts to handle stderr; readd splash screen for remote consoles
    (Closes issue ASTERISK-20945. Reported by Warren Selby)

    * --- Reset RTP timestamp; sequence number on SSRC change
    (Closes issue ASTERISK-20906. Reported by Eelco Brolman)

    For a full list of changes in this release, please see the ChangeLogs:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.20.1
    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.12.1
    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.2.1

    Thank you for your continued support of Asterisk!



    Asterisk Manager Changes
    Click to view a printable version Wed, 23 Jan 2013 10:55:51 +1300

    Hi everyone, if you're using the Asterisk Manager in your software you might want to have a look at the reformatting that is being done and will be pretty extensive.

    There's a description available here and discussion on the Asterisk Developers Mailing List:

    https://wiki.asterisk.org/wiki/display/AST/AMI+1.4+Specification

    This is mostly just a link for me so that I can read through it later on :-)



    Digium needs more people
    Click to view a printable version Wed, 23 Jan 2013 08:52:13 +1300

    David Duffett has posted a note looking for new staff at Digium:

    As we continue to grow and focus on getting the Asterisk word out there, we've come to a point where we need more people!

    Right now we have a number of vacancies, including those for a technical trainer and a training manager, both based at our headquarters, in Huntsville, AL. The Technical Trainer is basically responsible for developing and delivering our Asterisk training curriculum which covers our all of open source Asterisk solutions and all of our Digium products. As you might expect, the Training Manager is responsible for managing and growing our overall Asterisk training programs - it's a pretty exciting role, if I say so myself.

    We have some terrific trainers, and our classes are always well received. That's why it's important for us to fill these positions with truly talented people. We're looking for people who not only can get the job done well, but who are also as passionate as the rest of us about the growing Asterisk community.

    You can see more information about these and other positions online at http://www.digium.com/en/company/careers/

    If you think you can help us take Asterisk further, and have the relevant skills, go ahead and apply now. What are you waiting for?

    Don't waste any time in getting in touch!

    David Duffett
    Digium, Inc. · Director, Worldwide Asterisk Community



    Asterisk 1.8.11-cert10, 1.8.19.1, 10.11.1, 10.11.1-digiumphones, 11.1.1 Now Available (Security Release)
    Click to view a printable version Wed, 23 Jan 2013 07:21:41 +1300

    The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.11 and Asterisk 1.8, 10, and 11. The available security releases are released as versions 1.8.11-cert10, 1.8.19.1, 10.11.1, 10.11.1-digiumphones, and 11.1.1.

    These releases are available for immediate download at
    http://downloads.asterisk.org/pub/telephony/asterisk/releases

    The release of these versions resolve the following two issues:

    * Stack overflows that occur in some portions of Asterisk that manage a TCP connection. In SIP, this is exploitable via a remote unauthenticated session; in XMPP and HTTP connections, this is exploitable via remote authenticated sessions.

    * A denial of service vulnerability through exploitation of the device state cache. Anonymous calls had the capability to create devices in Asterisk that would never be disposed of.

    These issues and their resolutions are described in the security advisories.

    For more information about the details of these vulnerabilities, please read security advisories AST-2012-014 and AST-2012-015, which were released at the same time as this announcement.

    For a full list of changes in the current releases, please see the ChangeLogs:

    http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.11-cert10
    http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.19.1
    http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.11.1
    http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.11.1-digiumphones
    http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.1.1

    The security advisories are available at:

    AST-2012-014.pdf
    AST-2012-015.pdf

    Thank you for your continued support of Asterisk!



    Asterisk 11.2.0 Now Available
    Click to view a printable version Wed, 23 Jan 2013 06:37:27 +1300

    The Asterisk Development Team has announced the release of Asterisk 11.2.0.

    This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

    The release of Asterisk 11.2.0 resolves several issues reported by the community and would have not been possible without your participation.
    Thank you!

    The following is a sample of the issues resolved in this release:

    * --- app_meetme: Fix channels lingering when hung up under certain conditions
    (Closes issue ASTERISK-20486. Reported by Michael Cargile)

    * --- Fix stuck DTMF when bridge is broken.
    (Closes issue ASTERISK-20492. Reported by Jeremiah Gowdy)

    * --- Add missing support for "who hung up" to chan_motif.
    (Closes issue ASTERISK-20671. Reported by Matt Jordan)

    * --- Remove a fixed size limitation for producing SDP and change how ICE support is disabled by default.
    (Closes issue ASTERISK-20643. Reported by coopvr)

    * --- Fix chan_sip websocket payload handling
    (Closes issue ASTERISK-20745. Reported by Iñaki Baz Castillo)

    * --- Fix pjproject compilation in certain circumstances
    (Closes issue ASTERISK-20681. Reported by Dinesh Ramjuttun)

    For a full list of changes in this release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.2.0

    Thank you for your continued support of Asterisk!



    Asterisk 1.8.20.0 Now Available
    Click to view a printable version Wed, 23 Jan 2013 06:35:29 +1300

    The Asterisk Development Team has announced the release of Asterisk 1.8.20.0.
    This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

    The release of Asterisk 1.8.20.0 resolves several issues reported by the community and would have not been possible without your participation.

    Thank you!

    The following is a sample of the issues resolved in this release:

    * --- app_meetme: Fix channels lingering when hung up under certain conditions
    (Closes issue ASTERISK-20486. Reported by Michael Cargile)

    * --- Fix stuck DTMF when bridge is broken.
    (Closes issue ASTERISK-20492. Reported by Jeremiah Gowdy)

    * --- Improve Code Readability And Fix Setting natdetected Flag
    (Closes issue ASTERISK-20724. Reported by Michael L. Young)

    * --- Fix extension matching with the '-' char.
    (Closes issue ASTERISK-19205. Reported by Philippe Lindheimer, Birger "WIMPy" Harzenetter)

    * --- Fix call files when astspooldir is relative.
    (Closes issue ASTERISK-20593. Reported by James Le Cuirot)

    For a full list of changes in this release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.20.0

    Thank you for your continued support of Asterisk!



    Asterisk 10.12.0 Now Available
    Click to view a printable version Wed, 23 Jan 2013 06:33:15 +1300

    The Asterisk Development Team has announced the release of Asterisk 10.12.0.

    This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

    The release of Asterisk 10.12.0 resolves several issues reported by the community and would have not been possible without your participation.
    Thank you!

    The following is a sample of the issues resolved in this release:

    * --- app_meetme: Fix channels lingering when hung up under certain conditions
    (Closes issue ASTERISK-20486. Reported by Michael Cargile)

    * --- Fix stuck DTMF when bridge is broken.
    (Closes issue ASTERISK-20492. Reported by Jeremiah Gowdy)

    * --- Improve Code Readability And Fix Setting natdetected Flag
    (Closes issue ASTERISK-20724. Reported by Michael L. Young)

    * --- Fix extension matching with the '-' char.
    (Closes issue ASTERISK-19205. Reported by Philippe Lindheimer, Birger "WIMPy" Harzenetter)

    * --- Fix call files when astspooldir is relative.
    (Closes issue ASTERISK-20593. Reported by James Le Cuirot)

    For a full list of changes in this release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.12.0

    Thank you for your continued support of Asterisk!



    Certified Asterisk 1.8.15-cert1 Now Available
    Click to view a printable version Wed, 23 Jan 2013 06:30:28 +1300

    The Asterisk Development Team is pleased to announce the first release of Certified Asterisk 1.8.15. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases

    Certified Asterisk 1.8.15 is the next release of Certified Asterisk 1.8. Users who are currently on Certified Asterisk 1.8.11 are encouraged to switch to Certified Asterisk 1.8.15 at their earliest convenience.

    For a full list of changes in this release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-1.8.15-cert1

    Thank you for your continued support of Asterisk!



    libpri 1.4.14 Now Available
    Click to view a printable version Fri, 21 Dec 2012 13:45:15 +1300

    The Asterisk Development Team has announced the release of libpri 1.4.14.
    This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri

    The release of libpri 1.4.14 resolves several issues reported by the community and would have not been possible without your participation.
    Thank you!

    The following are the issues resolved in this release:

    * --- Fix compiler warning in pritest.c.
    (Closes issue PRI-145. Reported by Tzafrir Cohen)

    * --- Q.SIG: Allow PROGRESS when in the Active state.
    (Closes issue PRI-147. Reported by Nick Merrett)

    * --- Handle optional Recommendation octet 3a in Cause IE.
    (Closes issue PRI-151. Reported by Tzafrir Cohen)

    For a full list of changes in this release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/libpri/ChangeLog-1.4.14

    Thank you for your continued support of Asterisk!



    SIP Stack - Update
    Click to view a printable version Thu, 13 Dec 2012 09:11:22 +1300

    Matt Jordan has posted an update on the new SIP stack discussions with a decision on which one to use:

    Greetings!

    Recently, we've had a lot of discussion about which SIP stack to use for a new SIP channel driver. Josh's initial research resulted in three candidates: Sofia-sip, reSIProcate, and pjproject. All three SIP stacks have various advantages and disadvantages, which the community discussed in depth. While there are many factors to consider in the choice of a SIP stack, the discussion primarily landed on whether or not the SIP stack could be available as a package outside of the Asterisk source.

    Of the three candidates, both Sofia-sip and reSIProcate are available as packages. However, it was determined that Sofia-sip is no longer actively maintained, and the largest project that utilizes it currently embeds it within their source. While there may be options there, the fact that there is no active upstream maintainer of the Sofia-sip stack means that we would become the SIP stack maintainers ourselves, which would be a very limited improvement from the current state of SIP in Asterisk.

    This leaves reSIProcate and pjproject. reSIProcate is a fantastic SIP stack that is available as a package on some distros; it is, however, written in C++ and does not currently declare an API compatible with C linkage. The pjproject SIP stack is also a great SIP stack and one that Digium is very familiar with; however, it is not available as a package and its build system is oriented toward embedding in a project. As both have similar SIP capabilities, this leads us to the following two questions:

    1) If a C++ SIP stack were chosen, what effect would that have on the SIP channel driver that uses it, and on the Asterisk project as a whole?
    2) Is it possible to package pjproject?

    The second was easier to answer. While not exactly a trivial effort, it is possible to modify the pjproject build system to produce shared object libraries suitable for packaging. Teluu has agreed to support such an effort, although the work would have to be started by the Asterisk project. The initial project to create such a package is outlined on the Asterisk wiki. At a minimum, Digium will work to create:

    * A Git repository with a modified build system that produces shared object libraries and install targets of the libraries needed by Asterisk.
    * Tarballs on downloads.asterisk.org.
    * A package for CentOS.

    Ideally, the entire repository would end up being pushed up stream, but this would work in the interim to pull pjproject outside of the Asterisk source. The results of this effort will be available for developers who are interested in using them as the basis for creating and maintaining packages for other distributions.

    The first question was harder: what impact would a C++ SIP stack have on the Asterisk project? There are two facets to this:

    1) The technical impact on using C++ libraries in C
    2) The impact on the Asterisk community.

    The first poses significant problems – for a full write up, see [5]. To summarize that page, the C++ model of memory management, error handling, and building has major implications that would fundamentally change the Asterisk project. That leads to the second issue - given that Asterisk's developer community have traditionally written modules in C, a C++ SIP stack would potentially create a barrier for entry in development for the current community. Based on those two problems, using a SIP stack written in C++ would have to be a last resort for the Asterisk project.

    Given all of the above, we feel that going with pjproject as a SIP stack is feasible and the best option for Asterisk. If Asterisk were written in C++ I'm certain we would have chosen reSIProcate, but as it is, the only major issue with pjproject is its lack of a package, which I believe we can address. Thanks to everyone who has participated in the discussion – with your input, I think we have what is the best path forward for Asterisk.

    --
    Matthew Jordan
    Digium, Inc. | Engineering Manager



    Asterisk 11.1.0 Now Available
    Click to view a printable version Tue, 11 Dec 2012 15:29:59 +1300

    The Asterisk Development Team has announced the release of Asterisk 11.1.0:

    This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

    The release of Asterisk 11.1.0 resolves several issues reported by the community and would have not been possible without your participation.

    Thank you!

    The following is a sample of the issues resolved in this release:

    * --- Fix execution of 'i' extension due to uninitialized variable.
    (Closes issue ASTERISK-20455. Reported by Richard Miller)

    * --- Prevent resetting of NATted realtime peer address on reload.
    (Closes issue ASTERISK-18203. Reported by daren ferreira)

    * --- Fix ConfBridge crash if no timing module loaded.
    (Closes issue ASTERISK-19448. Reported by feyfre)

    * --- Fix the Park 'r' option when a channel parks itself.
    (Closes issue ASTERISK-19382. Reported by James Stocks)

    * --- Fix an issue where outgoing calls would fail to establish audio due to ICE negotiation failures.
    (Closes issue ASTERISK-20554. Reported by mmichelson)

    For a full list of changes in this release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.1.0

    Thank you for your continued support of Asterisk!



    Asterisk 10.11.0 Now Available
    Click to view a printable version Tue, 11 Dec 2012 15:27:59 +1300

    The Asterisk Development Team has announced the release of Asterisk 10.11.0:

    This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

    The release of Asterisk 10.11.0 resolves several issues reported by the community and would have not been possible without your participation.

    Thank you!

    The following is a sample of the issues resolved in this release:

    * --- Prevent resetting of NATted realtime peer address on reload.
    (Closes issue ASTERISK-18203. Reported by daren ferreira)

    * --- Do not use a FILE handle when doing SIP TCP reads.
    (Closes issue ASTERISK-20212. Reported by Phil Ciccone)

    * --- Fix ConfBridge crash if no timing module loaded.
    (Closes issue ASTERISK-19448. Reported by feyfre)

    * --- confbridge: Fix a bug which made conferences not record with AMI/CLI commands
    (Closes issue ASTERISK-20601. Reported by Vilius)

    * --- Fix execution of 'i' extension due to uninitialized variable.
    (Closes issue ASTERISK-20455. Reported by Richard Miller)

    For a full list of changes in this release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.11.0

    Thank you for your continued support of Asterisk!



    Asterisk 1.8.19.0 Now Available
    Click to view a printable version Tue, 11 Dec 2012 15:25:34 +1300

    The Asterisk Development Team has announced the release of Asterisk 1.8.19.0:

    This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

    The release of Asterisk 1.8.19.0 resolves several issues reported by the community and would have not been possible without your participation.

    Thank you!

    The following is a sample of the issues resolved in this release:

    * --- Prevent resetting of NATted realtime peer address on reload.
    (Closes issue ASTERISK-18203. Reported by daren ferreira)

    * --- Do not use a FILE handle when doing SIP TCP reads.
    (Closes issue ASTERISK-20212. Reported by Phil Ciccone)

    * --- Fix execution of 'i' extension due to uninitialized variable.
    (Closes issue ASTERISK-20455. Reported by Richard Miller)

    * --- Ensure that the Queue application tracks busy members in off nominal situations
    (Closes issue ASTERISK-20623. Reported by Bryan Walters)

    * --- Properly extract the Body information of an EWS calendar item
    (Closes issue ASTERISK-19738. Reported by Dmitry Burilov)

    For a full list of changes in this release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.19.0

    Thank you for your continued support of Asterisk!



    Asterisk 1.8.18.1, 10.10.1 and 11.0.2 now available
    Click to view a printable version Fri, 07 Dec 2012 11:46:14 +1300

    The Asterisk Development Team has announced the release of Asterisk 1.8.18.1, 10.10.1 and 11.0.2:

    These releases are available for immediate download at

    http://downloads.asterisk.org/pub/telephony/asterisk

    The release of Asterisk 1.8.18.1, 10.10.1 and 11.0.2 resolve an issue reported by the community and would have not been possible without your participation.
    Thank you!

    The following is the issue resolved in this release:

    * --- chan_local: Fix local_pvt ref leak in local_devicestate().
    (Closes issue ASTERISK-20769. Reported by rmudgett)

    For a full list of changes in this release, please see the ChangeLogs:

    ChangeLog-1.8.18.1
    ChangeLog-10.10.1
    ChangeLog-11.0.2

    Thank you for your continued support of Asterisk!



    Elastix queue call-backs with WombatDialer
    Click to view a printable version Tue, 04 Dec 2012 11:10:48 +1300

    Lenz from QueueMetrics has written a blog post about using WombatDialer to queue call-backs.

    Excerpt from the post:

    You are called in to a client site; they seem to have a problem. They run a small (10 agents) inbound call centre, and when you join everybody else in the meeting room, there is a large and colourful graph in the middle of the table. The graph shows the call wait times during the week and boy, it’s not a good sight. Their main inbound activity is to offer client support for a company selling sport bikes, and everybody seems to be calling on Monday morning. It looks like people go riding on weekends and whatever problem they have, they call on Monday morning. Wait times peak, abandon rates spike, and nobody is happy. The call center manager is mostly concerned of having to hire and train some temp people in order to handle the load that only happens one day a week. They ask you if you have any better idea on what can be done. And yes, you have some.

    You can program an Asterisk queue so that when people tire of waiting, they press a digit and get to a menu where they can leave their number. Then the system queues their call and attempts to call them at a convenient time.

    Read More...



    Introducing: Earl Grey : sip2cause.conf
    Click to view a printable version Sat, 01 Dec 2012 11:05:23 +1300

    Olle has posted details of a new branch he has created that allows you to 'manipulate the SIP-ISDN cause/response code tables':

    Friends,
    I just opened up a new branch for a small hack. There is a need in some very strange and unusual installations to be able to manipulate the sip-ISDN cause/response code tables. These are the ones used to convert from SIP codes to ISDN cause codes at hangup.

    The Earl Grey branch aims to make these configurable in a separate configuration file, sip2cause.conf, that will be needed in just a few cases, but for these cases, it's important. The branch will be created based on 1.8 LTS, but will hopefully also be ported to trunk.

    If you have any input on these conversions and how you like to be able to use them, now is the time to speak up!

    Cheers,
    /O

    http://svn.digium.com/svn/asterisk/team/oej/earl-grey-sip2cause-configurable-1.8/



    Introducing ToRELP
    Click to view a printable version Wed, 28 Nov 2012 15:43:30 +1300

    Shane Spencer has posted details of a 'quick and dirty way to push notifications away from Asterisk to a Python Tornado process.':

    Heya everybody.

    I work on a lot of AGI/AMI/AJAM/etc.. projects and recently discovered RELP (available via rsyslog) which is defined here:
    http://www.librelp.com/relp.html

    I've been pimping out (yes.. pimping) the Log dialplan application to quickly emit a message to my local syslog which is then delivered to rsyslog. I have rsyslog configured to use RELP between itself and a remote RELP server (ToRELP) in a reliable way.

    Messages are queued and redelivered if for any reason the remote RELP server dies.

    After doing some testing I've found that setting the core debug and verbosity to 0 does not negatively impact messages going to syslog.. which is wonderful.

    I personally use this to send JSON commands off to a worker process that helps with some screenpop processes.

    https://github.com/whardier/torelp

    My dialplan has a line like the following in it:

    Log(NOTICE, JSON:{'i': 'havealovelybunchofcoconuts', 'dee': 'dledee'})

    Have fun.. always looking for contributions to my projects. Hope somebody out there can utilize this.



    Testers wanted for for Comfort Noise labs
    Click to view a printable version Fri, 23 Nov 2012 16:32:32 +1300

    Olle is looking for testers of his new Comfort Noise work:

    Friends,

    My work with comfort noise support in Asterisk took a big leap forward today. I had a version that replaced the lack of frames with music on hold, but today it was swapped out and I now use the noise generator contributed by cmantunes earlier.

    This only applies to the core bridge (at least that's what I tested). I will investigate what's needed for app_queue and possibly if needed, the rtp bridge.

    Asterisk will properly negotiate CN support in SDP and will take incoming CNG frames and either forward over the bridge (if the other side of the call supports CN) or generate noise until audio frames are received.

    I have not added a silence generator, so asterisk will not generate it's own CN frames and suppress silence. This is a coming step. At least we can now bridge a call with full CN support.

    If you want to help me test, check out this branch
    http://svn.digium.com/svn/asterisk/team/oej/roibos-cng-support-1.8

    Check the README.roibos-cng.txt file for more information.

    A big thank you to Matt Jordan and Joshua Colp for guidelines and ideas on this work.

    /O



    New SIP channel driver project page
    Click to view a printable version Fri, 23 Nov 2012 15:34:55 +1300

    Mark Michelson has posted details of a wiki page that has been created to guide the development of a new SIP channel driver for Asterisk:

    Hi everyone.

    One of the resolutions of Astridevcon was that we should write a new SIP channel driver to be included in Asterisk 12. I have created a project page here: https://wiki.asterisk.org/wiki/display/AST/New+SIP+channel+driver

    The page is pretty devoid of content at the moment, but it is very much a "living document". As the SIP project progresses, the page will be updated accordingly. If you are interested in following the progress of the new SIP channel driver, then I recommend watching this page and its subpages.

    I will be making an effort to send out regular e-mails to the developer mailing list on the progress of the project as well as plans for the upcoming weeks. I will also highlight ways that you can get involved in the project.

    During this period, my goal was mainly to get the project page I linked to started. Joshua Colp has been working on researching and evaluating SIP stacks to potentially use on the project.

    I will be posting a review on the review board shortly to evaluate the project page as it currently is. If you want to provide feedback on the page's structure or things that should be added, feel free to comment on the review once it gets posted.

    I will be sending an e-mail out either Friday or Monday with the next set of items we'll be working on.

    Mark Michelson

    Update from Josh Colp:

    Hola,

    (Apologies for the semi-long email)

    As Mark mentioned in his earlier email I've been working on researching viable SIP stacks for the new SIP channel driver. I've scoured the far reaches of the internet looking for new stacks that may have come into existence since my previous Asterisk SCF research but in the end the three that came to the top were Sofia-sip, Resiprocate, and pjsip.

    You can find my research on them at https://wiki.asterisk.org/wiki/display/AST/SIP+Stack+Research

    I've detailed what I looked at and the impressions I got for each. This is by no means a deep deep deep analysis as that would take considerably more time. What it does cover though is the community for each project, how easy it is to use, general impression on features, interoperability, and (in my opinion) most importantly documentation.

    Through doing all of this research I have learned that there is no perfect SIP stack, but all of the ones I mentioned will do what we need. No matter which one you choose though you are effectively giving up something you could have gotten with another one so we have to find a balance. I think our balance should be towards a stack that has great documentation, is easy to use (and expand), provides the features we want, and still has an active community.

    After examining the three options I've ordered them as follows according to my personal opinion against the above:

    1. pjsip
    2. Resiprocate
    3. Sofia-sip

    But Josh, why did you order it as such?!?
    Good question, Josh.

    The documentation for pjsip is great and makes it easy to learn exactly how to use it and what is going on. The general architecture of it makes extending it *extremely* easy and you can inject your own modules at many different layers. The project itself is still being actively developed and maintained. The time required to get up to speed and use pjsip is also very low since there are people in the Asterisk community who are familiar with it and have used it.

    Resiprocate came second because the documentation is not as complete or expansive as pjsip, 'nor is the higher level (APIs that reduce the amount of code we have to write) feature count. It would also be more difficult to use it with Asterisk as it is C++ based, and this would also decrease the number of people who could help with that integration.

    Sofia-sip came last because it did not have as good of documentation as resiprocate or pjsip and the project itself seems to have become stagnant. It also lacks as many higher level features as pjsip.

    The purpose of this email is to bring about discussion on this subject as a whole and try to reach a conclusion on which SIP stack would be the best in the eyes of the community.

    Thanks for reading this long email and I look forward to the discussion.

    Cheers,

    --
    Joshua Colp
    Digium, Inc. | Senior Software Developer

    An update from Mark

    Hi folks,

    This is just a periodic update on the current planned work for the new SIP channel driver for the next couple of weeks. I had meant to send this out yesterday, but I didn't because I suck and am a terrible person :)

    The hot topic on the mailing list at the moment is the proposed method for Asterisk's use of a SIP stack. On that front, we've opened up dialogs with the PJSIP and ReSIProcate developers to find out what our role can be in their development and find out the reality of being able to provide or use packaged versions of their SIP stacks.

    Things you can expect to see on the wiki soon:

    * Brent Eagles is doing an investigation into exactly what would be required in order to write a C++ SIP channel driver.
    * I will soon be starting on a page to diagram a proposed stack-agnostic SIP architecture for the SIP channel driver.
    * We're working to start creating issues in JIRA dealing with the tasks that need to be done. These will make their way onto the main SIP project page once they are created.

    I'll send an e-mail out in a couple of weeks with what was accomplished and what will be worked on next.

    Mark Michelson

    An update from Olle (author of the current SIP stack in Asterisk)
    Friends,

    A few years ago, after 1.4 release my opinion was that the stack in Asterisk chan_sip wasn't the big issue, tests showed that it was better than I thought at the time ;-)

    The big issue was the code - one large chunk based on an old chan_iax - and the internals. Peers and users that makes no sense from a SIP point of view. No transaction support.

    After 1.4 with the additions of IPv6 as well as TCP and TLS - things are much worse and I agree that we need to restart or do a massive clean up, which in fact would be a restart. The current code base costs too much to maintain. I do agree with the decision to start a new project, something that I've been waiting for many years to happen.

    I find it hard to depend on an upstream provider of a library for such a core feature, but also recognize the fact that using an existing SIP stack would speed up the process. This discussion is critical for the future of Asterisk and I do appreciate very much that it happens outside the corridors in Huntsville.

    I've reported bugs to SIP software and got as an answer that I have to report it to PJSIP directly - as a user. That's no good. A project needs to be able to deliver a quality product and have some level of control. PJsip doesn't seem to make it possible to control quality without a fork, something that I don't understand. As stated before, I have not developed or interacted with PJsip directly so I have no personal experience. This is just based on observations of other that develop with the library.

    The Asterisk project need to always be able to handle our own bugs and support our users. If the upstream developer group makes it hard for us to do so, forking and managing our own fork is the only option and we should not be afraid of it, even though it's of course the very last option. If we realize from start that this is what we have to do with a SIP stack, we've selected a bad option. In the end, it may be the only one.

    Unfortunately that's what I see with both PJsip, Resiprocate and Sofia users - many private "forks". I have no insight into why this is the state of the SIP stack "industry". The discussion I've had with developers indicates low interest, a limit in resources to manage incoming patches or different opinions about the need for them - or combinations. Like us in Asterisk, the other projects are probably in a state where lack of funding makes it hard to spend time on general housekeeping and just have to focus on short term revenue-related work. They are all good guys, but everyone gotta get a living, and with the financial climate we have today, resources are very limited.

    When I looked at 3rd party stacks a few years ago there was a lot of functionality missing from Asterisk's point of view. They all had high-level interfaces for people who wants to create SIP phones, but some operations in a b2bua are very low-level. Look at the stuff we need to do in order to support call transfers, where we fake some SIP messages to be sip-compliant, just because we can't get proper information from the other end of the call. Most of the SIP stack developers I talked with then would not let us do that low-level stuff and break their elegant API's and phone abstractions.

    SIP is such a core function in Asterisk so this is a huge decision. If we end up managing a fork mostly by ourselves, then we are in the same situation that we're in today, but with an updated code base with an oppurtunity to improve things easier than today. And an oppurtunity to do things right, from the core out to sip.conf.

    I would like to see another discussion soon - how we should change the core PBX in order to be able to build a proper SIP stack. If Digium and the community invest in this development, I would like to see an upgrade of the core so that we can do things right. In my installations, I have almost only SIP. I do believe a majority of Asterisk channels today are SIP.

    That wasn't the state when the core was designed, which gives us translations to ISDN cause codes, caller IDs that doesn't support domains and much more that makes it complicated to implement SIP properly. Don't you think it's time we took a look at this and the architecture of a potential new SIP channel driver so we don't start with peers/users and confuse everyone again... :-)

    Let's continue the SIP stack discussion and I'll try to back off and let people with experience of developing with these stacks hash this out.

    Cheers,
    /O



    Asterisk 10.10.0 Now Available
    Click to view a printable version Thu, 08 Nov 2012 05:40:11 +1300

    The Asterisk Development Team has announced the release of Asterisk 10.10.0.

    This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

    The release of Asterisk 10.10.0 resolves several issues reported by the community and would have not been possible without your participation.
    Thank you!

    The following is a sample of the issues resolved in this release:

    * --- Resolve issues in ConfBridge regarding marked, waitmarked, and unmarked users
    (Closes issue ASTERISK-19562. Reported by flan)

    * --- dsp.c User Configurable DTMF_HITS_TO_BEGIN and DTMF_MISSES_TO_END
    (Closes issue ASTERISK-17493. Reported by alecdavis)

    * --- Fix error where improper IMAP greetings would be deleted.
    (Closes issue ASTERISK-20435. Reported by fhackenberger)

    * --- iax2-provision: Fix improper return on failed cache retrieval
    (Closes issue ASTERISK-20337. Reported by John Covert)

    * --- Fix T.38 support when used with chan_local in between.
    (Closes issue ASTERISK-20229. Reported by wdoekes)

    For a full list of changes in this release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.10.0

    Thank you for your continued support of Asterisk!



    Asterisk 1.8.18.0 Now Available
    Click to view a printable version Thu, 08 Nov 2012 05:37:46 +1300

    The Asterisk Development Team has announced the release of Asterisk 1.8.18.0.

    This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

    The release of Asterisk 1.8.18.0 resolves several issues reported by the community and would have not been possible without your participation.
    Thank you!

    The following is a sample of the issues resolved in this release:

    * --- dsp.c User Configurable DTMF_HITS_TO_BEGIN and DTMF_MISSES_TO_END
    (Closes issue ASTERISK-17493. Reported by alecdavis)

    * --- Fix error where improper IMAP greetings would be deleted.
    (Closes issue ASTERISK-20435. Reported by fhackenberger)

    * --- iax2-provision: Fix improper return on failed cache retrieval
    (Closes issue ASTERISK-20337. Reported by John Covert)

    * --- Fix T.38 support when used with chan_local in between.
    (Closes issue ASTERISK-20229. Reported by wdoekes)

    * --- Fix an issue where media would not flow for situations where the legacy STUN code is in use.
    (Closes issue ASTERISK-20415. Reported by Michele Cicciotti)

    For a full list of changes in this release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.18.0

    Thank you for your continued support of Asterisk!



    Asterisk 11.0.1 Now Available
    Click to view a printable version Wed, 07 Nov 2012 05:12:37 +1300

    The Asterisk Development Team has announced the release of Asterisk 11.0.1.
    This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

    The release of Asterisk 11.0.1 resolves several issues reported by the community and would have not been possible without your participation.
    Thank you!

    The following are the issues resolved in this release:

    * --- chan_sip: Fix a bug causing SIP reloads to remove all entries from the registry
    (Closes issue ASTERISK-20611. Reported by Alisher)

    * --- confbridge: Fix a bug which made conferences not record with AMI/CLI commands
    (Closes issue ASTERISK-20601. Reported by Vilius)

    * --- Fix an issue with res_http_websocket where the chan_sip WebSocket handler could not be registered.
    (Closes issue ASTERISK-20631. Reported by danjenkins)

    For a full list of changes in this release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.0.1

    Thank you for your continued support of Asterisk!



    QueueMetrics 12 Released
    Click to view a printable version Fri, 02 Nov 2012 07:28:28 +1300

    Lenz has posted details of the release of QueueMetrics version 12:

    QueueMetrics version 12.10 is now available, offering a variety of new features, performance improvements and bug fixes that we are sure you will find extremely useful!

    The first point we want to highlight is the work that has been done in improving QueueMetrics' memory and caching performance, which will mainly benefit larger contact centers.

    We have worked on the database access caching, we have limited the total memory footprint, minimized the creation of a large number of temporary objects and we have created a new string cache that outperforms the native Java implementation used in previous versions of QueueMetrics by two orders of magnitude.
    Furthermore, we now have an even stronger caching system that keeps pre-processed objects in memory and avoids hitting the database for most real-time queries. This is an optional feature that only affects the Real-Time and the Agent's Page when running with SQL or CLUSTER storage.

    The results obtained through this change have demonstrated a 10x - 20x performance improvement on page generation times!

    We have also provided a new monitoring page, which is accessible from the DBTEST page, that allows to query the status of the new caches in real-time and reset them, as required.

    The guide to fine-tuning QueueMetrics memory settings in order to get the best performance can be found in the QueueMetrics Advanced Configuration manual and it is a must-read for system administrators.

    Given the frequent need to access specific administrative tools, we have simplified how administrators can reach two main areas of QueueMetrics: the configuration.properties file and the database Test page. These are now available directly from the Home page, within the ’Administrative Tools’ listing.

    The next major addition is a new functionality that allows agents to run specific reports directly from the agent page. These new features are already enabled for the included sample agents.

    Furthermore, the Agent page now includes a new ’Home’ button that allows an agent to get back to his/her main web page at any stage and various reporting options are selectable by queue, report type and durations. An agent can run a report based only on the queues and reports assigned to him/her. An administrator can also assign the maximum amount of time that an agent can have available, when viewing historical reports from the agent web page. When an agent runs a report, the agent does not have a ’Search’ button enabled, limiting access to any kind of Custom Reports.

    Last but not least, QueueMetrics now includes a new agent page URL feature. Each time the call history changes, and if there is a URL associated to the call shown on top of the list, a new window will automatically be opened pointing to the specific URL.

    The new release also includes a number of bug fixes, as listed within the 'Minor changes'.

    Please rest assured that this release works with the same activation keys you used for your current version of QueueMetrics.

    Enjoy the update,

    Lorenzo Emilitri
    Managing Director, Loway



    Asterisk 11.0.0 Now Available
    Click to view a printable version Wed, 31 Oct 2012 05:48:10 +1300

    Ok, this one is a little controversial. This was actually released at Astricon but this is the formal announcement.

    Basically you should have been at Astricon :-P

    This is an awesome release and is the next Long Term Support version. The first since Asterisk 1.8. There are some great new features and everyone should start using it :-)

    So without further ado, here's the official announcement.

    The Asterisk Development Team is pleased to announce the release of Asterisk 11.0.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases

    Asterisk 11 is the next major release series of Asterisk. It is a Long Term Support (LTS) release, similar to Asterisk 1.8. For more information about support time lines for Asterisk releases, see the Asterisk versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

    For important information regarding upgrading to Asterisk 11, please see the Asterisk wiki:

    https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11

    A short list of new features includes:

    * A new channel driver named chan_motif has been added which provides support for Google Talk and Jingle in a single channel driver. This new channel driver includes support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk, hold, unhold, and ringing notification. It is also compliant with the current Jingle specification, current Google Jingle specification, and the original Google Talk protocol.

    * Support for the WebSocket transport for chan_sip.

    * SIP peers can now be configured to support negotiation of ICE candidates.

    * The app_page application now no longer depends on DAHDI or app_meetme. It has been re-architected to use app_confbridge internally.

    * Hangup handlers can be attached to channels using the CHANNEL() function. Hangup handlers will run when the channel is hung up similar to the h extension; however, unlike an h extension, a hangup handler is associated with the actual channel and will execute anytime that channel is hung up, regardless of where it is in the dialplan.

    * Added pre-dial handlers for the Dial and Follow-Me applications. Pre-dial allows you to execute a dialplan subroutine on a channel before a call is placed but after the application performing a dial action is invoked. This means that the handlers are executed after the creation of the callee channels, but before any actions have been taken to actually dial the callee channels.

    * Log messages can now be easily associated with a certain call by looking at a new unique identifier, "Call Id". Call ids are attached to log messages for just about any case where it can be determined that the message is related to a particular call.

    * Introduced Named ACLs as a new way to define Access Control Lists (ACLs) in Asterisk. Unlike traditional ACLs defined in specific module configuration files, Named ACLs can be shared across multiple modules.

    * The Hangup Cause family of functions and dialplan applications allow for inspection of the hangup cause codes for each channel involved in a call. This allows a dialplan writer to determine, for each channel, who hung up and for what reason(s).

    * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE() lets you set some of the configuration options from the general section of features.conf on a per-channel basis. FEATUREMAP() lets you customize the key sequence used to activate built-in features, such as blindxfer, and automon.

    * Support for DTLS-SRTP in chan_sip.

    * Support for named pickupgroups/callgroups, allowing any number of pickupgroups
    and callgroups to be defined for several channel drivers.

    * IPv6 Support for AMI, AGI, ExternalIVR, and the SIP Security Event Framework.

    More information about the new features can be found on the Asterisk wiki:

    https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Documentation

    A full list of all new features can also be found in the CHANGES file.

    http://svnview.digium.com/svn/asterisk/branches/11/CHANGES

    For a full list of changes in the current release, please see the ChangeLog.

    http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.0.0

    Thank you for your continued support of Asterisk!



    AstriDevCon 2012 Notes Available
    Click to view a printable version Tue, 30 Oct 2012 08:59:12 +1300

    Matt Jordan has posted details of the AstriDevCon meeting:

    Hello!

    I'd like to say thanks again to everyone who showed up in Atlanta and/or participated in #astridevcon for AstriDevCon this year. We had a very active discussion about the state of Asterisk development, and I'm looking forward to some exciting projects in the upcoming year. Expect a number of e-mails over the next few weeks announcing documented policies and project proposals.

    For those who are interested in the topics discussed, some fleshed out notes are available on the Asterisk wiki:

    https://wiki.asterisk.org/wiki/display/AST/AstriDevCon+2012

    Also, for those of you who did attend or participated via #astridevcon, I apologize if I missed your name or company on the list of participants! I had a handwritten list during the event, but unfortunately it got left behind in a stack of other papers that the cleaning crew absorbed, so the list on the wiki was reconstructed from memory (which, after a week at AstriCon, is a little fuzzy). If I missed something, please contact me so that I can get the information corrected.

    Again, thanks to everyone who participated, and I look forward to a year of developing Asterisk with you all!

    Matt

    --
    Matthew Jordan
    Digium, Inc. | Engineering Manager



    AstriDevCon 2012 - Are you ready
    Click to view a printable version Thu, 18 Oct 2012 09:17:24 +1300

    Josh Colp has posted details about a stream for AstriDevCon on Monday:

    Hola,

    As many of you are aware the AstriDevCon conference is coming up next week. To all those who are coming it's going to be a great time but for those who can't make it we understand. New this year is a listen-only conference bridge that will be accessible by SIP and IAX2 using VoIP or as an audio stream using VLC. As this will be listen-only we invite you to join the #astridevcon IRC channel on chat.freenode.net during the conference to participate in a real time manner. We will make as much effort as we can to make sure your opinions and comments are heard by everyone. If you've been on the fence about physically attending and don't think an audio stream will suffice we are still accepting signups. You can find additional details at https://wiki.asterisk.org/wiki/display/AST/AstriDevCon.

    For those who have specific items they wish to discuss feel free to respond to this email with your thoughts. This offer is open to any Asterisk developer who is willing to participate and if previous years are any indication there will be an assortment of topics: APIs, policies, tools, and SIP will all no doubt be brought up. Don't worry developers of other channel drivers as SIP has not yet completely taken over the world! We'd love to discuss your channel driver as well.

    Audio conference details:

    SIP: sip:31337@astridevcon.asterisk.org
    IAX2: IAX2/guest@astridevcon.asterisk.org/31337

    VLC: http://astridevcon.asterisk.org:8000/astridevcon.ogg

    These streams will go live shortly before AstriDevCon officially begins.

    Cheers,

    --
    Joshua Colp
    Digium, Inc. | Senior Software Developer



    New Asterisk Developers
    Click to view a printable version Thu, 18 Oct 2012 04:46:57 +1300

    Matthew Jordan has posted details of new developers in the Asterisk community:

    Hey all -

    We've been a bit remiss in introducing some recent additions to the Asterisk developer community. A few of you have seen commits from them recently, and its only fit and proper that you know the people behind the commit messages!

    Pedro Kiefer (pkiefer) has fixed all sorts of bugs in app_alarmreceiver, and has introduced a number of new features to the application as well. As such, he's agreed to help the Asterisk project by being the primary maintainer of the app_alarmreceiver module - which will be a huge benefit to all users of that module.

    David Lee (dlee) is one of those developers who has done a bit of everything. He's written software for telecommunications, database security, and data center infrastructure management, embedded C++, web applications in Java and Scala. Most recently, Dave has spent the last two years working on our Asterisk SCF project.

    Brent Eagles (beagles) has been with Digium for over two years and also comes to us from the Asterisk SCF team. He is working on all elements of Asterisk, with a focus on improving APIs and the SIP channel driver.

    Please join me in welcoming Pedro, David, and Brent!

    --
    Matthew Jordan
    Digium, Inc. | Engineering Manager



    LibPRI 1.4.13 Now Available
    Click to view a printable version Thu, 11 Oct 2012 09:35:51 +1300

    The Asterisk Development Team has announced the release of libpri 1.4.13.

    This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri

    The release of libpri 1.4.13 resolves several issues reported by the community and would have not been possible without your participation.
    Thank you!

    The following are the issues resolved in this release:

    * --- Outgoing BRI calls fail when using Asterisk 1.8 with HA8, HB8, and B410P cards.
    (Issue AST-598. Reported by Trey Blancher)

    * --- Implement handling a multi-channel RESTART request.
    (Closes issue PRI-93. Reported by Marcin Kowalczyk)

    * --- Removed MDL/TEI management configuration warning message.
    (Closes issue PRI-137. Reported by Bart Coninckx)

    * --- Allow passing compiler flags (CFLAGS, LDFLAGS)
    (Closes issue PRI-144. Reported by Tzafrir Cohen)

    For a full list of changes in this release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/libpri/ChangeLog-1.4.13

    Thank you for your continued support of Asterisk!



    Asterisk 11.0.0-rc1 Now Available
    Click to view a printable version Wed, 10 Oct 2012 07:36:11 +1300

    While I don't normally write about release candidates this one is the first release candidate for the next Long Term Support version of Asterisk. The last was 1.8 and the one before that was 1.4. If you're going to test any release candidate this one would be a good idea!

    Anyway, here's the release announcement:

    The Asterisk Development Team is pleased to announce the first release candidate of Asterisk 11.0.0. This release is available for immediate download at
    http://downloads.asterisk.org/pub/telephony/asterisk/releases

    All interested users of Asterisk are encouraged to participate in the Asterisk 11 testing process. Please report any issues found to the issue tracker, https://issues.asterisk.org/jira. It is also very useful to see successful test reports. Please post those to the asterisk-dev mailing list. All Asterisk users are invited to participate in the #asterisk-testing channel on IRC to work together in testing the many parts of Asterisk.

    Asterisk 11 is the next major release series of Asterisk. It will be a Long Term Support (LTS) release, similar to Asterisk 1.8. For more information about support time lines for Asterisk releases, see the Asterisk versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

    For important information regarding upgrading to Asterisk 11, please see the Asterisk wiki:

    https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11

    A short list of new features includes:

    * A new channel driver named chan_motif has been added which provides support for Google Talk and Jingle in a single channel driver. This new channel driver includes support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk, hold, unhold, and ringing notification. It is also compliant with the current Jingle specification, current Google Jingle specification, and the original Google Talk protocol.

    * Support for the WebSocket transport for chan_sip.

    * SIP peers can now be configured to support negotiation of ICE candidates.

    * The app_page application now no longer depends on DAHDI or app_meetme. It has been re-architected to use app_confbridge internally.

    * Hangup handlers can be attached to channels using the CHANNEL() function. Hangup handlers will run when the channel is hung up similar to the h extension; however, unlike an h extension, a hangup handler is associated with the actual channel and will execute anytime that channel is hung up, regardless of where it is in the dialplan.

    * Added pre-dial handlers for the Dial and Follow-Me applications. Pre-dial allows you to execute a dialplan subroutine on a channel before a call is placed but after the application performing a dial action is invoked. This means that the handlers are executed after the creation of the callee channels, but before any actions have been taken to actually dial the callee channels.

    * Log messages can now be easily associated with a certain call by looking at a new unique identifier, "Call Id". Call ids are attached to log messages for just about any case where it can be determined that the message is related to a particular call.

    * Introduced Named ACLs as a new way to define Access Control Lists (ACLs) in Asterisk. Unlike traditional ACLs defined in specific module configuration files, Named ACLs can be shared across multiple modules.

    * The Hangup Cause family of functions and dialplan applications allow for inspection of the hangup cause codes for each channel involved in a call. This allows a dialplan writer to determine, for each channel, who hung up and for what reason(s).

    * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE() lets you set some of the configuration options from the general section of features.conf on a per-channel basis. FEATUREMAP() lets you customize the key sequence used to activate built-in features, such as blindxfer, and automon.

    * Support for DTLS-SRTP in chan_sip.

    * Support for named pickupgroups/callgroups, allowing any number of pickupgroups and callgroups to be defined for several channel drivers.

    * IPv6 Support for AMI, AGI, ExternalIVR, and the SIP Security Event Framework.

    More information about the new features can be found on the Asterisk wiki:

    https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Documentation

    A full list of all new features can also be found in the CHANGES file.


    Don't forget you can get 20% off all tickets by using the discount code: AC12MATT

    Astricon is run between the 23rd and the 25th of October at the Sheraton in Atlanta.

    I'm going to be doing a tutorial on using the Asterisk Manager with PHP. Basically how to write back end and front end software that you can use to expose Asterisk services.

    I'm going to be speaking on the 24th at 10:00am.

    If you're coming along be sure to pop in and say hi either before or after my tutorial. I'm pretty easy to recognize although this year I don't have a goatee :-)



    Asterisk 10.8.0 Now Available
    Click to view a printable version Fri, 14 Sep 2012 09:59:57 +1200

    The Asterisk Development Team has announced the release of Asterisk 10.8.0.

    This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

    The release of Asterisk 10.8.0 resolves several issues reported by the community and would have not been possible without your participation.
    Thank you!

    The following is a sample of the issues resolved in this release:

    * --- AST-2012-012: Resolve AMI User Unauthorized Shell Access through ExternalIVR
    (Closes issue ASTERISK-20132. Reported by Zubair Ashraf of IBM X-Force Research)

    * --- AST-2012-013: Resolve ACL rules being ignored during calls by some IAX2 peers
    (Closes issue ASTERISK-20186. Reported by Alan Frisch)

    * --- Handle extremely out of order RFC 2833 DTMF
    (Closes issue ASTERISK-18404. Reported by Stephane Chazelas)

    * --- Resolve severe memory leak in CEL logging modules.
    (Closes issue AST-916. Reported by Thomas Arimont)

    * --- Only re-create an SRTP session when needed
    (Issue ASTERISK-20194. Reported by Nicolo Mazzon)

    For a full list of changes in this release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.8.0

    Thank you for your continued support of Asterisk!



    Asterisk 1.8.16.0 Now Available
    Click to view a printable version Fri, 14 Sep 2012 09:56:25 +1200

    The Asterisk Development Team has announced the release of Asterisk 1.8.16.0.

    This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

    The release of Asterisk 1.8.16.0 resolves several issues reported by the community and would have not been possible without your participation.
    Thank you!

    The following is a sample of the issues resolved in this release:

    * --- AST-2012-012: Resolve AMI User Unauthorized Shell Access through ExternalIVR
    (Closes issue ASTERISK-20132. Reported by Zubair Ashraf of IBM X-Force Research)

    * --- AST-2012-013: Resolve ACL rules being ignored during calls by some IAX2 peers
    (Closes issue ASTERISK-20186. Reported by Alan Frisch)

    * --- Handle extremely out of order RFC 2833 DTMF
    (Closes issue ASTERISK-18404. Reported by Stephane Chazelas)

    * --- Resolve severe memory leak in CEL logging modules.
    (Closes issue AST-916. Reported by Thomas Arimont)

    * --- Only re-create an SRTP session when needed; respond with correct crypto policy
    (Issue ASTERISK-20194. Reported by Nicolo Mazzon)

    For a full list of changes in this release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.16.0

    Thank you for your continued support of Asterisk!



    Support for Asterisk-SCF dropped
    Click to view a printable version Fri, 14 Sep 2012 03:08:08 +1200

    Digium has dropped support for Asterisk-SCF. While this is somewhat of a surprise, I always thought that the splitting of development resources was something that would end up being a bad idea.

    The other problem was that setting up an environment to compile and run SCF was always difficult.

    The features that SCF had though were awesome. I just hope that people figure out how to cross port some of them into Asterisk.

    The other thing is that there was basically a core rewrite - looking at the task that needed to be achieved and rewriting code from scratch to achieve it.

    That's something that's unlikely to happen in a wholesale manner in Asterisk.

    Update: Thanks to Kirk Bateman for pointing out I don't discuss what Asterisk-SCF is. So here's a brief overview:

    The Asterisk SCF Mission Statement

    Asterisk Scalable Communications Framework (SCF) is a highly-scalable, distributed, extensible open-source communications platform and application suite.

    Asterisk SCF Goals

    Scalability
    High Availability and Fault Tolerance
    Extensibility
    Performance

    Asterisk SCF Is

    Asterisk SCF is a collection of fine-grained software components that focus on very specific functional tasks using a message-oriented and distributed architecture. Asterisk SCF provides a flexible deployment model, taking advantage of modern multi-core processors and operating across servers in multiple locations. Asterisk SCF provides a set of building blocks for communications applications, interfaces and services.

    Anyway, here's a post that was sent to the Asterisk-SCF committee - Thanks to Andrew Latham for posting it to the Asterisk-Dev list (originally written by David Duffett):

    As you know, work has been on-going in this project for the last couple of years, and while SCF was never intended to be a replacement for Asterisk, it certainly was designed to solve some of the architectural issues relevant to Asterisk. Currently there is a working version of the code that is publicly available.

    During this time period, Asterisk has also evolved: performance has been improved, many new features have been added, and architectural improvements, such as improved media handling, have been incorporated into recent versions. While this clearly does not satisfy all the goals of Asterisk SCF, it certainly represents a significant step in improving the experience for Asterisk users.

    Although the community has provided assistance in guiding the Asterisk SCF project, Digium has provided almost all of the software development effort. Due to the sheer complexity, the Asterisk SCF project has yet to progress to a point that enables significant development contribution from the community, and the level of resource investment required to adequately sponsor and maintain both Asterisk and Asterisk SCF is simply stretching the company too thinly.

    Therefore, the decision has been made to suspend our development work on the Asterisk SCF open source project, in order to more fully support the needs of the Asterisk community.

    There may well be some community developers that would like to work with the current version of Asterisk SCF so we will continue to make the code publicly available, although Digium will not be supporting the software in the same manner to which Asterisk community members are accustomed.

    Valuable lessons have been learned in the process of working on Asterisk SCF. Some of these are directly applicable to Asterisk, and will be beneficial during the development of Asterisk 12 and, indeed, all future releases.

    For now, we would like to thank everyone who has participated in the development of Asterisk SCF for their efforts. If you are interested in discussing how some of the goals and experience from Asterisk SCF can be applied to Asterisk, please join us for the DevCon event at Astricon.



    111 Useful and/or Funny New Prompts For Asterisk, Courtesy of Allison Smith
    Click to view a printable version Wed, 12 Sep 2012 10:22:40 +1200

    Wow ok, so I totally missed this one - Allison has recorded a whole lot of new prompts for Asterisk. Check them out:

    Greetings Asterisk Users / Biz,

    Allison Smith, the voice of Asterisk, has released the first installment of her riotous (and absolutely free) "Asterisk Prompts I Wish I Had" series, and boy does it include some amazing stuff. Here are a few of the funnier entries:

    "I think you may have me confused with Siri."

    "Thank you for calling. Self destruct has been initiated will occur in thirty seconds. If you believe you have reached this destination in error please enter the twelve digit abort code now or hang up and seek shelter immediately."

    "It's not you, it's me. Let's find an agent."

    "For quality control purp.. (laughs) for quality control purposes, this call (laughing) may be..."

    "Awkward!"

    That's only a tiny sample of the 111 new prompts, all available for immediate download from the downloads page at Asterisk.org.

    Editor's note:
    You can actually download them directly from here:

    Allison Add-Ons: July 2012 Zip Archive

    and read the list of them here:

    Prompt List / Script



    Remember that you can request new prompts at any time from the online form. The prompts can be funny or practical. If they're practical, they need to be generic / universally (i.e. don't ask her to record your automated attendant messages - that's how she earns her living). If they're funny they should be in reasonably good taste. Allison will issue a new set every few months.

    As always, if you need custom prompts you can order them from Allison on the IVR prompts page on Digium.com.

    Thanks,

    -S

    Steve Sokol
    Asterisk Marketing Director | Digium, Inc.



    HOWTO - Lumicall/Android with an Asterisk PBX
    Click to view a printable version Tue, 11 Sep 2012 05:53:22 +1200

    Daniel from Lumicall has posted details of how to set things up with Asterisk and Lumicall:

    After a lot of questions from Asterisk users, I've now created a dedicated page with settings for Lumicall and Asterisk to work together

    http://www.lumicall.org/configuration/asterisk

    As of the 1.8.x releases of Lumicall, it is much more consistent at remaining registered on devices that frequently sleep. This is mainly based on the addition of SIP outbound keep-alive over TLS/TCP, together with improved code for background processing when Android sleeps.

    Lumicall is also available from F-Droid now, so it is easier than ever to be certain that you are running a binary built from the published source code, and no Google account is needed to install with F-Droid:

    http://f-droid.org/repository/browse/?fdfilter=lumicall&fdid=org.lumicall.android&fdpage=1



    AdhearsionConf 2012 - Voice Application Development Conference
    Click to view a printable version Tue, 11 Sep 2012 04:51:14 +1200

    Ben Klang has posted details of AdhearsionConf coming up just before Astricon:

    Hello Asterisk people,

    I'm writing to let you know about a conference that is happening in Palo Alto this October 20 - 21, AdhearsionConf 2012. Now in its third year, AdhearsionConf is the annual summit for voice application developers who work with Ruby. This year we are meeting at the spaces of the AT&T Foundry, which is their innovation center in downtown Palo Alto. As with the past two years we will have two days of presentations from the community (videos from last year can be found at http://vimeo.com/channels/AdhearsionConf2011). New this year will be the full day of training on Friday October 19th. The training will cover everything from setting up the telephony engine (Asterisk, FreeSWITCH or PRISM) to getting a development environment set up (Adhearsion) and include all kinds of best practices.

    I realize that the dates this year may be tight if you're planning on going to AstriCon like I am (scheduling conferences is a remarkably challenging task). Regardless, I invite you to join us in Palo Alto. All information on the conference can be found on the AdhearsionConf website: http://adhearsionconf.com.

    If you have any questions please feel free to contact me off-list.

    Thanks! I hope to see some of you there.
    /BAK/



    Asterisk 1.8.11-cert7, 1.8.15.1, 10.7.1, 10.7.1-digiumphones Now Available (Security Release)
    Click to view a printable version Fri, 31 Aug 2012 08:53:45 +1200

    The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.11 and Asterisk 1.8 and 10. The available security releases are released as versions 1.8.11-cert7, 1.8.15.1, 10.7.1, and 10.7.1-digiumphones.

    These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases

    The release of Asterisk 1.8.11-cert7, 1.8.15.1, 10.7.1, and 10.7.1-digiumphonesresolve the following two issues:

    * A permission escalation vulnerability in Asterisk Manager Interface. This would potentially allow remote authenticated users the ability to execute commands on the system shell with the privileges of the user running the Asterisk application. Please note that the README-SERIOUSLY.bestpractices.txt file delivered with Asterisk has been updated due to this and other related vulnerabilities fixed in previous versions of Asterisk.

    * When an IAX2 call is made using the credentials of a peer defined in a dynamic Asterisk Realtime Architecture (ARA) backend, the ACL rules for that peer are not applied to the call attempt. This allows for a remote attacker who is aware of a peer's credentials to bypass the ACL rules set for that peer.

    These issues and their resolutions are described in the security advisories.

    For more information about the details of these vulnerabilities, please read security advisories AST-2012-012 and AST-2012-013, which were released at the same time as this announcement.

    For a full list of changes in the current releases, please see the ChangeLogs:

    http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.11-cert7
    http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.15.1
    http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.7.1
    http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.7.1-digiumphones

    The security advisories are available at:

    * http://downloads.asterisk.org/pub/security/AST-2012-012.pdf
    * http://downloads.asterisk.org/pub/security/AST-2012-013.pdf

    Thank you for your continued support of Asterisk!



    Discount for Astricon 2012
    Click to view a printable version Fri, 31 Aug 2012 04:01:32 +1200

    Astricon 2012 is rapidly approaching and will be held in Atlanta between October 23 and 25. The Daily Asterisk News has secured a 20% discount on all tickets if you use the discount code below:

    AC12MATT

    I will be speaking at the conference, so would love to catch up with everyone.

    I'll be doing a tutorial on how to work with the Asterisk Manager Interface using PHP. I've got a library that I'll be providing that will help you write applications that get events via a MySQL database.

    Anyway, I look forward to seeing you all there and don't forget to use the code above to get your 20% discount off your tickets.

    I'll post about this a few times before Astricon arrives.



    Launching: The NEW Edvina SIP Masterclass in Stockholm and Miami
    Click to view a printable version Tue, 28 Aug 2012 07:10:09 +1200

    Olle has posted details of some new SIP MasterClass training sessions he will be running - well worth attending!

    Friends,

    Yesterday Edvina, my company, launched the new and updated SIP Masterclass training. It is scheduled for Stockholm in October 2012 and Miami, Florida in December 2012 (thanks to Redfone Communication).

    The new SIP Masterclass is partly based on the old Asterisk SIP Masterclass, but focuses more on Kamailio - the leading Open Source SIP server - and the SIP protocol. We've added a lot of Kamailio information and new labs, as well as information about SIP updates like Outbound, GRUU, ICE, Turn and much more.

    * A HIGH LEVEL TRAINING IN SIP AND BUILDING SCALABLE REALTIME PLATFORMS

    The Edvina SIP Masterclass is aimed at people who work professionally with Asterisk, FreeSwitch or other systems and wants to learn more about how to operate a SIP proxy - Kamailio - and get scalability, new services and failover for their platforms. The class also gives a thorough understanding of the SIP protocol, from the basics to advanced functionality like dialog states, SIP transfers and NAT traversal.

    • The SIP Protocol
    • Kamailio – the SIP server
    • SIP call flows: Call transfers
    • SIP: Forking and routing
    • Kamailio – transactions and forking
    • SIP Media: RTP, RTCP and QoS issues
    • SIP NAT traversal: Stun, Turn, Outbound
    • SIP presence infrastructure: SUBSCRIBE, NOTIFY, PUBLISH
    • SIP Dialogs, dialog states, blinking lamps
    • SIP messaging and presence: SIMPLE and MSRP
    • Kamailio messaging and presence
    • Building SIP services with Kamailio and a media server (Asterisk, FreeSwitch)
    • SIP load balancing and failover, DNS
    • Kamailio: DNS, failover with Dispatcher
    • SIP security: TLS, S/MIME, SRTP, SIP identity

    * A NEW CLASS BASED ON MANY YEARS OF EXPERIENCE

    The old class has been running for many years with good feedback from the students. The classes are based on years of experience on building scalable platforms for enterprises, the public sector, call centers, service providers and universities. The class starts on a level where basic Elastix, FreePBX, Asterisk and FreeSwitch trainings leave the student.

    Read more about the class here:
    - http://edvina.net/blog/2012/08/new-sip-masterclass/
    - http://edvina.net/training/new-sip-masterclass/

    * DISCOUNT ON THE STOCKHOLM CLASS!

    We have a *SPECIAL OFFER* for the first class in Stockholm, since this is the first class with the new material and the new labs. New students get a 20% discount and students that have attended Edvina trainings before get 30% discount! Contact us today via e-mail to reserve a seat!

    If you have any questions, please don't hesitate to contact me!

    Regards,
    /Olle



    Asterisk community services - Old Mantis instance to be shutdown on Aug 28th, 2012
    Click to view a printable version Tue, 28 Aug 2012 06:59:08 +1200

    Digium has posted details on the killing off of Mantis (at last) and moving permanently to Jira. To be fair the Mantis has been pretty much disabled for over a year anyway:

    On June 5, 2011, we migrated from Mantis to Jira as the issue tracker for Asterisk. We temporarily left Mantis running in read-only mode to smooth the transition. At 15 months, temporary has turned into semi-permanent. As a part of other infrastructure changes we are making to the community services, we will finally shut down Mantis for good.

    We will update our DNS servers on the morning of Tuesday, August 28th, however it may take a few hours for those changes to propagate.

    We have done our best to put redirects in place so that old links to Mantis will still work. If you find a link that does not redirect as expected, or have any other problems you think may be caused by the Mantis shutdown, please report them in the "JIRA Help" project. If you would rather report your issue via email, you may contact us at asteriskteam@digium.com.

    Digium's Asterisk Development Team



    AstLinux 1.0.4 Released
    Click to view a printable version Fri, 17 Aug 2012 03:59:35 +1200

    The AstLinux Team is happy to announce the release of 1.0.4.

    New in this release:

    • Asterisk 1.4.44 and 1.8.14.1
    • DAHDI, dahdi-linux 2.6.1 and dahdi-tools 2.6.1
    • wanpipe, version bump to 3.5.27
    • rhino, version bump to 0.99.6b2. Support is now enabled again by default.
    • libPRI, upstream patch to add layer 2 persistence option to customize the layer 2 behavior on BRI PTMP lines. (Thanks to Michael Keuter)
    • PHP version bump to 5.3.14 to address security issues.
    • Security fixes for OpenSSL
    • miniupnpd added (disabled by default) to support Universal Plug and Play. (Many thanks to David Kerr)
    • mtr added. Network diagnostic tool that combines ping and traceroute.
    • Updates to the web interface including the addition of a MeetMe tab, firewall enhancements and UPnP support.

    For the complete changelog and to download the install images go to the following pages:

    http://www.astlinux.org/release/104-asterisk-18141
    http://www.astlinux.org/release/104-asterisk-1444

    The AstLinux Team



    Asterisk 11.0.0-beta1 Now Available
    Click to view a printable version Sat, 11 Aug 2012 14:37:09 +1200

    The Asterisk Development Team is pleased to announce the first beta release of Asterisk 11.0.0 (the first Long Term Support release since 1.8). This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases

    All interested users of Asterisk are encouraged to participate in the Asterisk 11 testing process. Please report any issues found to the issue tracker, https://issues.asterisk.org/jira. It is also very useful to see successful test reports. Please post those to the asterisk-dev mailing list. All Asterisk users are invited to participate in the #asterisk-testing channel on IRC to work together in testing the many parts of Asterisk.

    Asterisk 11 is the next major release series of Asterisk. It will be a Long Term Support (LTS) release, similar to Asterisk 1.8. For more information about support time lines for Asterisk releases, see the Asterisk versions page:
    https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

    For important information regarding upgrading to Asterisk 11, please see the Asterisk wiki:

    https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11

    A short list of new features includes:

    * A new channel driver named chan_motif has been added which provides support for Google Talk and Jingle in a single channel driver. This new channel driver includes support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk, hold, unhold, and ringing notification. It is also compliant with the current Jingle specification, current Google Jingle specification, and the original Google Talk protocol.

    * Support for the WebSocket transport for chan_sip.

    * SIP peers can now be configured to support negotiation of ICE candidates.

    * The app_page application now no longer depends on DAHDI or app_meetme. It has been re-architected to use app_confbridge internally.

    * Hangup handlers can be attached to channels using the CHANNEL() function. Hangup handlers will run when the channel is hung up similar to the h extension; however, unlike an h extension, a hangup handler is associated with the actual channel and will execute anytime that channel is hung up, regardless of where it is in the dialplan.

    * Added pre-dial handlers for the Dial and Follow-Me applications. Pre-dial allows you to execute a dialplan subroutine on a channel before a call is placed but after the application performing a dial action is invoked. This means that the handlers are executed after the creation of the caller/callee channels, but before any actions have been taken to actually dial the callee channels.

    * Log messages can now be easily associated with a certain call by looking at a new unique identifier, "Call Id". Call ids are attached to log messages for just about any case where it can be determined that the message is related to a particular call.

    * Introduced Named ACLs as a new way to define Access Control Lists (ACLs) in Asterisk. Unlike traditional ACLs defined in specific module configuration files, Named ACLs can be shared across multiple modules.

    * The Hangup Cause family of functions and dialplan applications allow for inspection of the hangup cause codes for each channel involved in a call. This allows a dialplan writer to determine, for each channel, who hung up and for what reason(s).

    * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE() lets you set some of the configuration options from the general section of features.conf on a per-channel basis. FEATUREMAP() lets you customize the key sequence used to activate built-in features, such as blindxfer, and automon.

    * Support for named pickupgroups/callgroups, allowing any number of pickupgroups and callgroups to be defined for several channel drivers.

    * IPv6 Support for AMI, AGI, ExternalIVR, and the SIP Security Event Framework.

    More information about the new features can be found on the Asterisk wiki:

    https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Documentation

    A full list of all new features can also be found in the CHANGES file.

    http://svnview.digium.com/svn/asterisk/branches/11/CHANGES

    For a full list of changes in the current release, please see the ChangeLog.

    http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.0.0-beta1

    Thank you for your continued support of Asterisk!



    So long, and thanks for all the fish!
    Click to view a printable version Tue, 07 Aug 2012 12:42:27 +1200

    Kevin Fleming is moving on! Sorry it has taken me so long to post this article as I have been preparing to move from New Zealand to Orlando, Florida for 3 months. The fact that Kevin is departing will be a great loss to the Asterisk community.

    I remember when Kevin first appeared on the scene with Asterisk, pointing out a multitude of problems with the way things were done in Asterisk. At first everyone was pretty pissed that someone would come in and complain so much, but Kevin followed his complaints with solutions. Something so rarely done in an Open Source project.

    He will be dearly missed by the Asterisk community for having brought so much professionalism to the Asterisk project and for having given Digium a strong direction with software development.

    Here's his post sent to the Asterisk Developer's mailing list:

    I've been with Digium for just over seven years, and it's been an incredible experience that I wouldn't have traded for anything. When Mark Spencer invited me to visit Digium (and Huntsville) in early 2005, I could not have dreamed that I'd end up working for such an exciting, innovative company, finding a wife, and meeting hundreds of people (many of whom are now friends) around the world. It's been a time of tremendous personal and career growth, and my wonderful colleagues at Digium and in the Asterisk open source community have been directly responsible for most of that.

    Recently, though, I've been presented an opportunity to take on a new challenge and this has resulted in my acceptance of a new job, in a new industry. In the middle of September, I'll start working for Bloomberg, L.P., in the Office of the CTO, helping to lead their nascent open source initiative. I'll be working to bring the power of open source software, open standards, and community building to the financial market data services industry, where it is sorely needed (and overdue). Michelle and I will be relocating to the greater New York City area, but Michelle will continue in her role as Digium's in-house counsel. Because of our need to relocate, I'll only be at Digium until August 8th, although I'll be in Huntsville until around Labor Day.

    This is yet another incredibly exciting, career changing opportunity in my life, and I can't wait to see what it will bring. I'll be forever thankful for the opportunity that Digium and the Asterisk community provided me to learn, grow and find the place where my skills and experience are the most valuable (to both myself and my employer).

    --
    Kevin P. Fleming
    Digium, Inc. | Director of Software Technologies

    David Duffet has posted on the subject too

    It's amazing what you can learn in a few days...

    Having just found out that Queen Elizabeth has a great sense of humor, it has now emerged that Kevin Fleming - a man who (both with and without his moustache) has been an amazing contributor and influencer in the Asterisk project is set to move on to a new challenge outside the project - but still within the realms of Open Source.

    Kevin has been involved with Asterisk for 7+ years, and has been both a thought leader and a powerful voice in the Asterisk world during that time. I first met Kevin at a TMC event called VoIP Developer in California (old school, well before the days of IT Expo), where he was speaking about Asterisk as well as helping to man the Digium booth at the event.

    I've also followed Kevin around Berlin looking for great gelato during the AstriCon Europe 2006 tour - and it was well worth it, that man knows his gelato!

    I'd like to take this opportunity to say thanks to Kevin for his enormous contribution to the Asterisk Project. Without his efforts, Asterisk would not be the success it is today... Anyway, back to main theme - when someone in a senior role like Kevin moves on, it is important that others are there to pick up his responsibilities and move the project forward.

    As it turns out, we've already been working on this, and have some very talented people that will be taking up the key responsibilities of the project going forward.

    Some of them have been involved with Asterisk for several years, and some are recent additions, but together they form a great team to lead Asterisk into the future.

    Matt Jordan has assumed the project leader role for Asterisk, and is responsible for managing the releases of Asterisk, as well as all of the development efforts within Digium.

    Mark Michelson is serving as the Technical Lead for the project, responsible for architecture and design direction.

    We have also recently created the role of Community Support Manager, which Rusty Newton has filled. Rusty is a long time Digium employee with many years supporting Asterisk and Digium products, and will be the day to day interface for community technical issues.

    As you know, I recently joined Digium to look after the interests of the worldwide Open Source Asterisk community and I will therefore also be working alongside the good people identified above, especially Rusty.

    So while we wish Kevin all the best as he moves on, we are also confident that the good work he and the rest of the team have done continues to be in the best hands going forwards.

    To the future...

    David


    David Duffett
    Digium, Inc. · Director, Worldwide Asterisk Community



    AstriDevCon on Monday, Oct 22nd, 2012 - Asterisk developers invited
    Click to view a printable version Fri, 27 Jul 2012 13:51:44 +1200

    Digium has written a post about the AstriDevCon that happens before AstriCon:

    Digium invites developers who are interested in the future of Asterisk to join us at AstriDevCon.

    AstriDevCon is an annual event, held alongside AstriCon - it includes an in-depth review of progress made in the past year and an open discussion about the future direction of the project. The event is free and open for any active Asterisk community developer to apply. It's a great opportunity to meet the core development team in person, pitch ideas for new features and functions and coordinate efforts with others.

    Capacity for this event is limited. Preference will be made for those who are active in and currently take part in the Asterisk development community by contributing, reviewing, testing and so forth. Otherwise it's first come first serve, so apply for registration soon.

    For details on the event and registration: https://wiki.asterisk.org/wiki/display/AST/AstriDevCon



    AppKonference 2.1
    Click to view a printable version Thu, 19 Jul 2012 11:34:05 +1200

    Paul Albrecht has posted details of an update to AppKonference:

    Hi,

    I have released an updated AppKonference today. This release includes the following changes:

    Asterikast was updated to use the speaker scoreboard which is now by default enabled. If you want to revert to using conference state events to track speakers you can enable it in the Makefile with a compiler flag.

    Kick member, mute member, and unmute member cli commands have been deprecated. You should use the corresponding channel commands: kick channel, mute channel, and unmute channel.

    Changed the incoming and outgoing frame queues to use the asterisk frame header as the queue entry because there's no reason to encapsulate the frames in another header.

    Added timers to replace gettimeofday/usleep in the conference thread. For linux, periodic timing can be provided using timerfd_create/read, and for *bsd the kqueue can provide essentially the same functionality. To enable the timers set the appropriate compiler flag in the Makefile.

    For *bsd, note that the kqueue periodic timer is broken, that is, runs slow because of a bug so you should make sure you update your kernel with the fix before you run the module using kqueue.

    You can download the latest source from source forge:

    sourceforge.net/projects/appkonference

    --
    Paul Albrecht



    Asterisk 10.6.1 and 1.8.14.1 Now Available
    Click to view a printable version Tue, 17 Jul 2012 11:11:55 +1200

    The Asterisk Development Team has announced the release of Asterisk 10.6.1.
    This release is available for immediate download at
    http://downloads.asterisk.org/pub/telephony/asterisk

    The release of Asterisk 10.6.1 and 1.8.14.1 resolves an issue reported by the community and would have not been possible without your participation.
    Thank you!

    The following is the issue resolved in this release:

    * --- Remove a superfluous and dangerous freeing of an SSL_CTX.
    (Closes issue ASTERISK-20074. Reported by Trevor Helmsley)

    For a full list of changes in this release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.6.1

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.14.1

    Thank you for your continued support of Asterisk!



    Request for documenters - AMI Events
    Click to view a printable version Wed, 11 Jul 2012 18:06:53 +1200

    Matt Jordan has posted a request for help documenting Asterisk Manager Event:

    Hello everyone!

    As you may be aware, we've added documentation for AMI Events to Asterisk 11 (project page on the wiki here: https://wiki.asterisk.org/wiki/display/AST/AMI+Event+Documentation). This may come as no surprise, but there happen to be a lot of AMI events in Asterisk - and as such, we could use some help! If you're interested in helping us document some AMI events, the following could use some TLC:

    * chan_sip/chan_iax2
    * All resource modules
    * All extended support modules

    Some of the events in 'main' have received only brief documentation blurbs as well - enough so that their name, a brief synopsis and their fields will show up in the CLI commands/on the wiki. However, some additional meat to their field descriptions (as opposed to just their names, which is inferred from the *manager_event* macros) would probably be helpful as well.

    If you have any questions about how to document AMI Events, gripes, or generally just want to chat about it, feel free to ping me in #asterisk-dev.

    Thanks!
    --
    Matthew Jordan
    Digium, Inc. | Software Developer



    Asterisk 1.8.14.0 and 10.6.0 Now Available
    Click to view a printable version Wed, 11 Jul 2012 18:00:12 +1200

    The Asterisk Development Team has announced the release of Asterisk 1.8.14.0 and 10.6.0.

    Asterisk 1.8.14.0:

    The Asterisk Development Team has announced the release of Asterisk 1.8.14.0.
    This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

    The release of Asterisk 1.8.14.0 resolves several issues reported by the community and would have not been possible without your participation.
    Thank you!

    The following is a sample of the issues resolved in this release:

    * --- format_mp3: Fix a possible crash in mp3_read().
    (Closes issue ASTERISK-19761. Reported by Chris Maciejewsk)

    * --- Fix local channel chains optimizing themselves out of a call.
    (Closes issue ASTERISK-16711. Reported by Alec Davis)

    * --- Update a peer's LastMsgsSent when the peer is notified of waiting messages
    (Closes issue ASTERISK-17866. Reported by Steve Davies)

    * --- Prevent sip_pvt refleak when an ast_channel outlasts its corresponding sip_pvt.
    (Closes issue ASTERISK-19425. Reported by David Cunningham)

    * --- Send more accurate identification information in dialog-info SIP NOTIFYs.
    (Closes issue ASTERISK-16735. Reported by Maciej Krajewski)

    For a full list of changes in this release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.14.0

    Thank you for your continued support of Asterisk!

    Asterisk 10.6.0:

    The Asterisk Development Team has announced the release of Asterisk 10.6.0.
    This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

    The release of Asterisk 10.6.0 resolves several issues reported by the community and would have not been possible without your participation.
    Thank you!

    The following is a sample of the issues resolved in this release:

    * --- format_mp3: Fix a possible crash in mp3_read().
    (Closes issue ASTERISK-19761. Reported by Chris Maciejewsk)

    * --- Fix local channel chains optimizing themselves out of a call.
    (Closes issue ASTERISK-16711. Reported by Alec Davis)

    * --- Re-add LastMsgsSent value for SIP peers
    (Closes issue ASTERISK-17866. Reported by Steve Davies)

    * --- Prevent sip_pvt refleak when an ast_channel outlasts its corresponding sip_pvt.
    (Closes issue ASTERISK-19425. Reported by David Cunningham)

    * --- Send more accurate identification information in dialog-info SIP NOTIFYs.
    (Closes issue ASTERISK-16735. Reported by Maciej Krajewski)

    For a full list of changes in this release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.6.0

    Thank you for your continued support of Asterisk!



    Testers wanted - PRACK for 1.8
    Click to view a printable version Tue, 10 Jul 2012 15:43:29 +1200

    Olle has written a post asking for testers of the PRACK support for Asterisk 1.8:

    http://svn.digium.com/svn/asterisk/team/oej/darjeeling-prack-1.8

    As far as I can see from tests with various phones this implementation of SIP PRACK - or "100rel" seems to work now. All you have to do is to add prack=yes to sip.conf (general or per-device) and Asterisk will start playing the PRACK game.

    Feedback from tests is always welcome!

    Regards,
    /Olle

    ========== README.darjeeling ===============================

    Edvina AB
    Olle E. Johansson

    Project started: 2012-06-15

    Darjeeling-prack-1.8
    --------------------

    This branch will implement PRACK in the SIP stack of Asterisk.

    PRACK stands for reliable unreliable provisional responses.

    In SIP, the provisional responses are often sent over UDP, which means that they can get lost. Some of them are retransmitted every minute (to get at least one through once during a three minute period following RFC 3261). For some messages, it's important that they get through immediately. Like if you want to play a message to the customer that his call will be cancelled due to lack of funds in his account.

    PRACK adds a retransmit and ACK mechanism to the 1xx messages excluding 100 (since it's transmitted hop-by-hop).

    Configuration
    =============

    Add prack=yes in the [general] section of sip.conf or in device configurations.

    Asterisk will now add 100rel to the list of supported options. If the other device supports PRACK Asterisk will activate it or support it if the other side requires it.

    There's currently no support for requiring PRACK in a call.

    Technical details
    ==================

    If the INVITE contains
    Supported: 100rel

    then the 1xx answer can add
    Require: 100rel
    Rseq: 42

    The Rseq is the PRACK sequence number
    The caller then needs to confirm the message with a new request, during the INVITE transaction
    PRACK
    RAck: 42 <cseq #> <cseq method>

    And the callee confirms this (and close the PRACK transaction) with a 200 OK.

    If the PRACK is not received in time, the 1xx response will be retransmitted.

    There can only be ONE outstanding PRACK, which makes it easier to integrate in the Asterisk SIP stack that unfortunately lacks a transaction layer.

    PRACK is documented in RFC 3262.

    Retransmission works like the retransmission of responses to INVITE (like the 200 OK). It starts with T1 and doubles for each retransmission. Unlike INVITE responses, the retransmission timer does not cap at T2.

    If retransmission times out, a 5xx message should terminate the INVITE transaction.

    A PRACK received that does not match existing SIP_PVT is responded to with 481.

    * When to stop?
    ---------------
    From the RFC:

    " The UAS MAY send a final response to the initial request before having received PRACKs for all unacknowledged reliable provisional responses, unless the final response is 2xx and any of the unacknowledged reliable provisional responses contained a session description. In that case, it MUST NOT send a final response until those provisional responses are acknowledged. If the UAS does send a final response when reliable responses are still unacknowledged, it SHOULD NOT continue to retransmit the unacknowledged reliable provisional responses, but it MUST be prepared to process PRACK requests for those outstanding responses. A UAS MUST NOT send new reliable provisional responses (as opposed to retransmissions of unacknowledged ones) after sending a final response to a request"

    * Receive in order until final response
    ---------------------------------------
    "Handling of subsequent reliable provisional responses for the same initial request follows the same rules as above, with the following difference: reliable provisional responses are guaranteed to be in order. As a result, if the UAC receives another reliable provisional response to the same request, and its RSeq value is not one higher than the value of the sequence number, that response MUST NOT be acknowledged with a PRACK, and MUST NOT be processed further by the UAC. An implementation MAY discard the response, or MAY cache the response in the hopes of receiving the missing responses.
    The UAC MAY acknowledge reliable provisional responses received after the final response or MAY discard them."

    * When does the call begin?
    -------------------------
    A corner case that I don't know if it's implemented: If we receive and INVITE without SDP, we MUST add SDP offer to the reliable answer and the caller must add an SDP answer to the PRACK. This means that the session begins in the middle of the INVITE transaction.
    "Once an answer has been sent or received, the UA SHOULD establish the session based on the parameters of the offer and answer, even if the original INVITE itself has not been responded to." (section 5)

    * Security
    ----------
    "The PRACK request can be injected by attackers to force retransmissions of reliable provisional responses to cease. As these responses can convey important information, PRACK messages SHOULD be authenticated as any other request." (section 9)

    Todo
    ----
    * Add PRACK to the list of supported headers
    - done
    * Should we be able to REQUIRE prack? Based on what? _SIPREQUIREPRACK ?
    - not now
    * PRACK enabled globally and per device - user and peer
    - done
    * PRACK working when Asterisk is the UAS
    - done
    * PRACK working when Asterisk is the UAC
    - done
    * PRACK with authentication
    - not done
    * Requesting auth for PRACK
    - not done
    * PRACK in "sip show settings"
    - done
    * PRACK in "sip show peer xxx"
    - done
    * PRACK in "sip show channel xxx"
    - done

    ---------------------
    The patch is (C) Copyright by Edvina AB, Sollentuna, Sweden.
    Developed by Olle E. Johansson, oej at edvina dot net



    Asterisk 11 Feature Freeze Reminder
    Click to view a printable version Tue, 10 Jul 2012 15:34:22 +1200

    Matt Jordan has posted a reminder that the feature freeze for Asterisk 11 is nearly upon us:

    Hello everyone:

    As a friendly reminder, the feature freeze deadline of July 18th for
    Asterisk 11 is rapidly approaching. Please try to have all features that you'd
    like included for this release up on Review Board as soon as possible!

    Thanks,

    Matt

    --
    Matthew Jordan
    Digium, Inc. | Software Developer



    Asterisk 1.8.11-cert4, 1.8.13.1, 10.5.2, 10.5.2-digiumphones Now Available (Security Release)
    Click to view a printable version Sat, 07 Jul 2012 17:22:00 +1200

    The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.11 and Asterisk 1.8 and 10. The available security releases are released as versions 1.8.11-cert4, 1.8.13.1, 10.5.2, and 10.5.2-digiumphones.

    These releases are available for immediate download at
    http://downloads.asterisk.org/pub/telephony/asterisk/releases

    The release of Asterisk 1.8.11-cert4, 1.8.13.1, 10.5.2, and 10.5.2-digiumphones resolve the following two issues:

    * If Asterisk sends a re-invite and an endpoint responds to the re-invite with a provisional response but never sends a final response, then the SIP dialog structure is never freed and the RTP ports for the call are never released. If an attacker has the ability to place a call, they could create a denial of service by using all available RTP ports.

    * If a single voicemail account is manipulated by two parties simultaneously, a condition can occur where memory is freed twice causing a crash.

    These issues and their resolution are described in the security advisories.

    For more information about the details of these vulnerabilities, please read security advisories AST-2012-010 and AST-2012-011, which were released at the same time as this announcement.

    For a full list of changes in the current releases, please see the ChangeLogs:

    http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.11-cert4
    http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.13.1
    http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.5.2
    http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.5.2-digiumphones

    The security advisories are available at:

    * http://downloads.asterisk.org/pub/security/AST-2012-010.pdf
    * http://downloads.asterisk.org/pub/security/AST-2012-011.pdf

    Thank you for your continued support of Asterisk!



    WebSocket SIP Support
    Click to view a printable version Wed, 04 Jul 2012 09:23:58 +1200

    As has been discussed on twitter, there is a reviewboard entry for adding web sockets support for SIP to Asterisk!

    The reviewbaord entry

    A description of websockets



    app_swift beta release
    Click to view a printable version Fri, 22 Jun 2012 00:40:25 +1200

    Darren Sessions has posted details of the release of a new beta of app_swift:

    Hi folks,

    Just a note to let everyone know I've finally finished up the new BETA release of app_swift (now v3.0.1 b1).

    This release introduces some pretty major changes to app_swift such as:

    - The entire code-base has now been unified and the build system auto detects which Asterisk version you're using (yay! one branch!)

    - Auto-detection and support for both the Cepstral 5.0 and 6.0 engines

    - Support for Asterisk 1.4, 1.6, 1.8, 1.8 Certified, and 10

    - Asterisk 1.2 support has been dropped.


    I have only been able to do some basic testing with all these permutations of Asterisk and the Cepstral engines on a few of my machines here at the house and need some volunteers to help out and be guinea-pigs.

    Please email me directly with any feedback you might have.


    I've updated my github repo with the new app_swift code which can be downloaded using git.

    git clone git://github.com/dmsessions/app_swift.git


    Thanks,

    - D



    Hello World from the new guy
    Click to view a printable version Thu, 21 Jun 2012 11:24:44 +1200

    David Duffett has written his first post as Asterisk Community Director.

    Excerpt from his post:

    Hello Asterisk community!

    As you will have seen from Bryan John’s blog post some days ago, I am privileged to become the new Director for the worldwide Asterisk community.

    I want to thank Bryan for the work he has done while in the role of Community Director, and it is good to know that he is still involved in the project.

    Having been involved with Asterisk for some years, I have always felt an enormous sense of gratitude towards Mark Spencer (currently CTO at Digium), for creating Asterisk in the first place, and to the numerous people that have made contributions to the project – for giving us such an amazing communications tool kit.

    I see Asterisk as an IP communications engine which is as flexible as it is powerful. I see it as a communications application development environment, that allows the rapid construction of solutions to satisfy all kinds of requirements – and although it is in large part deployed as a business PBX, I also see Asterisk as a kind of flexible filler or glue that allows the connection of disparate systems that would either have not been possible, or would have been too costly, in the past.

    Coming from the computer telephony card industry, it was amazing to me to find that an application, such as IVR, that previously required a $10,000 telephony card, a number of (probably beardy) C programmers and a large chunk of time, could now be completed within a few minutes on a platform that was FREE.

    Read More...



    Asterisk 10.5.1 Now Available (Security Release)
    Click to view a printable version Fri, 15 Jun 2012 15:20:59 +1200

    The Asterisk Development Team has announced a security release for Asterisk 10.
    This security release is released as version 10.5.1.

    The release is available for immediate download at
    http://downloads.asterisk.org/pub/telephony/asterisk/releases

    The release of Asterisk 10.5.1 resolves the following issue:

    * A remotely exploitable crash vulnerability was found in the Skinny (SCCP) Channel driver. When an SCCP client sends an Off Hook message, followed by a Key Pad Button Message, a structure that was previously set to NULL is dereferenced. This allows remote authenticated connections the ability to cause a crash in the server, denying services to legitimate users.

    This issue and its resolution is described in the security advisory.

    For more information about the details of this vulnerability, please read security advisory AST-2012-009, which was released at the same time as this announcement.

    For a full list of changes in the current releases, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.5.1

    The security advisory is available at:

    * http://downloads.asterisk.org/pub/security/AST-2012-009.pdf

    Thank you for your continued support of Asterisk!



    Named ACLs - Use Cases (request for review/commentary)
    Click to view a printable version Thu, 14 Jun 2012 10:52:58 +1200

    Jonathan Rose has posted a request for review and ideas on a feature for Named ACLs based on a proposal by Olle Johansson and John Todd:

    Greetings asterisk-dev, I've been prototyping a feature that is a candidate for Asterisk 11 called Named ACLs based somewhat on an Astricon proposal by Olle E. Johansson and John Todd which could most simply be described as a system for defining ACL profiles to be used and shared among various consumers in Asterisk. Matt Jordan and I have been working on creating an Asterisk Wiki page to define the use cases for this system so that we could better establish the goals of this project and hopefully get some commentary to see if they sync up with the expectations of involved developers.

    https://wiki.asterisk.org/wiki/display/AST/Named+ACLs

    Goals for the initial implementation are described in the section Use Cases - Initial Implementation

    Goals for reaching feature completeness with Olle's original proposal are in Use Cases - Dynamic Named ACL Updating

    Please let us know if there is anything you feel should be included in this list of use cases that isn't already there or if any other changes need to be made.

    --
    Jonathan R. Rose
    Digium, Inc. | Software Engineer



    Asterisk 10.4.2 Now Available
    Click to view a printable version Thu, 31 May 2012 09:09:45 +1200

    The Asterisk Development Team has announced the release of Asterisk 10.4.2.
    This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

    The release of Asterisk 10.4.2 resolves several issues reported by the community and would have not been possible without your participation.
    Thank you!

    The following are the issues resolved in this release:

    * --- Resolve crash in subscribing for MWI notifications
    (Closes issue ASTERISK-19827. Reported by B. R)

    * --- Fix crash in ConfBridge when user announcement is played for more than 2 users
    (Closes issue ASTERISK-19899. Reported by Florian Gilcher)

    For a full list of changes in this release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.4.2

    Thank you for your continued support of Asterisk!



    Asterisk 1.8.12.2 Now Available
    Click to view a printable version Thu, 31 May 2012 09:05:04 +1200

    The Asterisk Development Team has announced the release of Asterisk 1.8.12.2.
    This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

    The release of Asterisk 1.8.12.2 resolves an issue reported by the= community and would have not been possible without your participation.
    Thank you!

    The following is the issue resolved in this release:

    * --- Resolve crash in subscribing for MWI notifications
    (Closes issue ASTERISK-19827. Reported by B. R)

    For a full list of changes in this release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.12.2

    Thank you for your continued support of Asterisk!



    Certified Asterisk 1.8.11-cert2; Asterisk 1.8.12.1, 10.4.1 Now Available (Security Release)
    Click to view a printable version Thu, 31 May 2012 09:01:15 +1200

    The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.11 and Asterisk 1.8 and 10. The available security releases are released as versions 1.8.11-cert2, 1.8.12.1, and 10.4.1.

    These releases are available for immediate download at
    http://downloads.asterisk.org/pub/telephony/asterisk/releases

    The release of Asterisk 1.8.11-cert2, 1.8.12.1, and 10.4.1 resolve the following two issues:

    * A remotely exploitable crash vulnerability exists in the IAX2 channel driver if an established call is placed on hold without a suggested music class. Asterisk will attempt to use an invalid pointer to the music on hold class name, potentially causing a crash.

    * A remotely exploitable crash vulnerability was found in the Skinny (SCCP) Channel driver. When an SCCP client closes its connection to the server, a pointer in a structure is set to NULL. If the client was not in the on-hook state at the time the connection was closed, this pointer is later dereferenced. This allows remote authenticated connections the ability to cause a crash in the server, denying services to legitimate users.

    These issues and their resolution are described in the security advisories.

    For more information about the details of these vulnerabilities, please read security advisories AST-2012-007 and AST-2012-008, which were released at the same time as this announcement.

    For a full list of changes in the current releases, please see the ChangeLogs:

    http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.11-cert2
    http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.12.1
    http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.4.1

    The security advisories are available at:

    http://downloads.asterisk.org/pub/security/AST-2012-007.pdf
    http://downloads.asterisk.org/pub/security/AST-2012-008.pdf

    Thank you for your continued support of Asterisk!



    New Community Support Manager at Digium - Rusty Newton
    Click to view a printable version Sat, 26 May 2012 12:11:06 +1200

    Kevin has posted a note about a new community support manager at Digium:

    We'd like you all to help us welcome Rusty Newton to Digium's Asterisk development and community support team! Rusty has been with Digium for over five years, starting in the Technical Support department and then moving to a sales position where he assisted customers with Asterisk and Switchvox solutions to their business needs. Prior to joining Digium he spent more than five years in the telecom industry, installing, configuring and maintaining PBXs. A couple of weeks ago he moved into a new role (for him and for Digium), Community Support Manager.

    In this role he'll be the primary person responsible for ensuring that Digium's community services are providing what the community members need, that the systems are operating properly, and that issues and questions are getting the attention they deserve. He'll be working closely with our Community Director as well, especially for events like AstriCon and others. He works directly with the software development team at Digium, which will allow him to focus almost exclusively on technical issues and discussions.

    We're quite excited that he has taken on this role and we expect that you will soon see the benefits of his activities across the community!



    Asterisk 11 Development: WebRTC/RTCWeb support
    Click to view a printable version Thu, 24 May 2012 10:28:39 +1200

    Kevin Fleming has posted details of a pretty major addition being worked on for Asterisk 11.

    Excerpt from the article:

    Early in 2012, the Asterisk development team at Digium got together to put together a list of projects we wanted to complete for the upcoming release of Asterisk 11. As you can imagine, there’s no shortage of feature requests out there… but we can’t do it all, and of course we still have to fix bugs, get maintenance releases out the door, and support the user community.

    After much discussion, we settled on the list of projects on this page; many of these have been on the wish lists produced from Asterisk Developer’s Conference sessions in past years. As of today, the team has made a great deal of progress on many of these projects, and you’ll see the benefits of that in Asterisk 11 when it is released later this year.

    However, there’s another exciting project we undertook as well, and we’ve kept this one a bit quiet until we had something to show off. For the past few months, Josh Colp (if you’ve been to an AstriCon recently, he’s the one you saw wandering the halls in a white fedora) has been working to add support for the nascent (but gaining traction) WebRTC effort. If you aren’t familiar with WebRTC (and its companion, RTCWeb), it’s an industry effort sponsored by many of the major browser manufacturers to integrate support for real-time communications (audio and video) directly into browsers, with no plugins or addons required. We think this is really going to be a game-changer for the VoIP community, as it will open the door to supporting custom, feature-rich applications on any device with a compatible browser, whether it is a laptop, tablet, smartphone, or anything else.

    Read More...



    Planned service outage for community services
    Click to view a printable version Thu, 24 May 2012 10:21:59 +1200

    On May 31, 2012 from approximately 9:00AM to 12:00PM (Central Daylight Time, GMT-5), the servers that Digium uses to provide many services to the Asterisk community will be relocated. This will mean that these services will be unavailable during most, if not all, of this time window. Once the move is complete, the services will be available again, with no user-visible changes.

    The services affected include:

    bamboo.asterisk.org
    code.asterisk.org
    downloads.digium.com
    downloads.asterisk.org
    git.asterisk.org
    issues.asterisk.org
    packages.asterisk.org
    reviewboard.asterisk.org
    svn.asterisk.org
    svnview.digium.com
    wiki.asterisk.org



    Asterisk 10.4.0 Now Available
    Click to view a printable version Fri, 04 May 2012 11:39:34 +1200

    The Asterisk Development Team has announced the release of Asterisk 10.4.0.
    This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

    The release of Asterisk 10.4.0 resolves several issues reported by the community and would have not been possible without your participation.
    Thank you!

    The following are the issues resolved in this release:

    * --- Prevent chanspy from binding to zombie channels
    (Closes issue ASTERISK-19493. Reported by lvl)

    * --- Fix Dial m and r options and forked calls generating warnings for voice frames.
    (Closes issue ASTERISK-16901. Reported by Chris Gentle)

    * --- Remove ISDN hold restriction for non-bridged calls.
    (Closes issue ASTERISK-19388. Reported by Birger Harzenetter)

    * --- Fix copying of CDR(accountcode) to local channels.
    (Closes issue ASTERISK-19384. Reported by jamicque)

    * --- Ensure Asterisk acknowledges ACKs to 4xx on Replaces errors
    (Closes issue ASTERISK-19303. Reported by Jon Tsiros)

    * --- Eliminate double close of file descriptor in manager.c
    (Closes issue ASTERISK-18453. Reported by Jaco Kroon)

    For a full list of changes in this release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.4.0

    Thank you for your continued support of Asterisk!



    Asterisk 1.8.12.0 Now Available
    Click to view a printable version Fri, 04 May 2012 11:37:33 +1200

    The Asterisk Development Team has announced the release of Asterisk 1.8.12.0.
    This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

    The release of Asterisk 1.8.12.0 resolves several issues reported by the community and would have not been possible without your participation.
    Thank you!

    The following are the issues resolved in this release:

    * --- Prevent chanspy from binding to zombie channels
    (Closes issue ASTERISK-19493. Reported by lvl)

    * --- Fix Dial m and r options and forked calls generating warnings for voice frames.
    (Closes issue ASTERISK-16901. Reported by Chris Gentle)

    * --- Remove ISDN hold restriction for non-bridged calls.
    (Closes issue ASTERISK-19388. Reported by Birger Harzenetter)

    * --- Fix copying of CDR(accountcode) to local channels.
    (Closes issue ASTERISK-19384. Reported by jamicque)

    * --- Ensure Asterisk acknowledges ACKs to 4xx on Replaces errors
    (Closes issue ASTERISK-19303. Reported by Jon Tsiros)

    * --- Eliminate double close of file descriptor in manager.c
    (Closes issue ASTERISK-18453. Reported by Jaco Kroon)

    For a full list of changes in this release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.12.0

    Thank you for your continued support of Asterisk!



    Certified Asterisk (SLA Supported GPL Code)
    Click to view a printable version Mon, 30 Apr 2012 11:19:36 +1200

    Malcolm Davenport has written a post about a new branching system for an SLA supported Open Source branching system for people who want super stable LTS releases of Asterisk.

    Excerpt from his post:

    Some users want the latest and greatest Asterisk. They’ve got a particular bug fix they’re keen about and they don’t want to mess around applying patches. If you’re this type of user, you’ll get the fix for your issues, and generally other people’s fixes, in the monthly point release.

    Some users really want it tested. They don’t care about anyone else’s bug fixes, except their own. In fact, they really don’t want anyone’s bug fixes but their own. That could break something. And when something breaks, Hulk gets angry. And when Hulk gets angry, well, you don’t want to make Hulk angry.

    Some users want additional functionality that can’t be part of the canonical release. Digium phones required all kinds of changes to Asterisk 1.8, and 10, and changes to support even more phone functionality will continue. That, of course, runs counter to the Asterisk release policy of not adding new features once a release branch is made.

    To address all of these different needs, we’ve cooked up a plan that, we hope, will satisfy most of the people, most of the time. For the first group of users (lots of bug fixes, frequent releases)? Stop reading. You’re all set; the world remains exactly the same.

    Read More...



    Asterisk 1.6.2.24, 1.8.11.1, 10.3.1 Now Available (Security Release)
    Click to view a printable version Tue, 24 Apr 2012 09:49:16 +1200

    The Asterisk Development Team has announced security releases for Asterisk 1.6.2, 1.8, and 10. The available security releases are released as versions 1.6.2.24, 1.8.11.1, and 10.3.1.

    These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases

    The release of Asterisk 1.6.2.24, 1.8.11.1, and 10.3.1 resolve the following two issues:

    * A permission escalation vulnerability in Asterisk Manager Interface. This would potentially allow remote authenticated users the ability to execute commands on the system shell with the privileges of the user running the Asterisk application.

    * A heap overflow vulnerability in the Skinny Channel driver. The keypad button message event failed to check the length of a fixed length buffer before appending a received digit to the end of that buffer. A remote authenticated user could send sufficient keypad button message events that the buffer would be overrun.

    In addition, the release of Asterisk 1.8.11.1 and 10.3.1 resolve the following issue:

    * A remote crash vulnerability in the SIP channel driver when processing UPDATE requests. If a SIP UPDATE request was received indicating a connected line update after a channel was terminated but before the final destruction of the associated SIP dialog, Asterisk would attempt a connected line update on a non-existing channel, causing a crash.

    These issues and their resolution are described in the security advisories.

    For more information about the details of these vulnerabilities, please read security advisories AST-2012-004, AST-2012-005, and AST-2012-006, which were released at the same time as this announcement.

    For a full list of changes in the current releases, please see the ChangeLogs:

    http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.24
    http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.11.1
    http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.3.1

    The security advisories are available at:

    http://downloads.asterisk.org/pub/security/AST-2012-004.pdf
    http://downloads.asterisk.org/pub/security/AST-2012-005.pdf
    http://downloads.asterisk.org/pub/security/AST-2012-006.pdf

    Thank you for your continued support of Asterisk!



    DAHDI-Linux 2.6.1, 2.5.1 and DAHDI-Tools 2.6.1, 2.5.1 Now Available
    Click to view a printable version Sun, 22 Apr 2012 14:02:48 +1200

    Shaun Ruffell has posted details of the latest release of DAHDI-Linux and DAHDI-Tools:

    The Asterisk Development Team has announced the releases of:
    DAHDI-Linux 2.6.1
    DAHDI-Linux 2.5.1
    DAHDI-Tools 2.6.1
    DAHDI-Tools 2.5.1
    DAHDI-Linux-Complete 2.6.1+2.6.1
    DAHDI-Linux-Complete 2.5.1+2.5.1

    These releases are available for immediate download at:
    http://downloads.asterisk.org/pub/telephony/dahdi-linux
    http://downloads.asterisk.org/pub/telephony/dahdi-tools
    http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete

    2.6.1 and 2.5.1 are bugfix releases of which the most noteable changes are:

    - Fix for Digium dual and quadspan cards in E1 mode when used with a hardware echocanceler that was introduced in 2.6.0.
    - Fix for intermittent failure to decode FSK caller ID on Digium voicebus analog cards introduced in 2.6.0.
    - Support for Linux kernel versions up to 3.4.

    Issues closed in these releases:
    DAHLIN-275: E1 spans have noise on some alternative channels when VPM is active
    DAHLIN-274: dahdi_dummy failes to compile
    DAHLIN-283: Disable Active State Power Management on PCIe links for DAHDI devices.
    DAHLIN-280: dahdi_dynamic_eth(ethmf,loc)
    DAHLIN-286: DAHDI driver wctdm24xxp does not compile with GCC 3.4.4
    DAHLIN-279: dahdi will not compile with CONFIG_DAHDI_ECHOCAN_PROCESS_TX
    DAHLIN-278: dahdi will not compile with CONFIG_DAHDI_NET
    DAHLIN-185: Dahdi dummy includes time.h, should be timer.h for low-res timer....
    DAHLIN-288: compilation error when CONFIG_DAHDI_WATCHDOG is defined
    And in the 2.5.1 release only:
    DAHLIN-272: No PCM on a TDM410 FXS module since r10167

    The DAHDI-Linux shortlog of changes since 2.6.0:

    Mike Sinkovsky (1):
    dahdi: Fix compilation when CONFIG_DAHDI_WATCHDOG is defined.

    Oron Peled (9):
    xpp: bugfix: fix bad refcount
    xpp: Don't deactivate XPDs on unregistration
    xpp: handle failures during dahdi_register_device()
    xpp: reset Astribank SPI busses
    xpp: FXS: better power-down to lower noise
    A parent-less device should not crash dahdi
    remove a duplicate dev_set_name()
    xpp: FXS: atomic vbat_h power handling
    xpp: FXS: added a 'lower_ringing_noise' parameter

    Shaun Ruffell (30):
    wctdm24xxp: FXS on-hook transmission timer incorrect.
    wct4xxp: VPM module creates noise on alternate channels on E1 spans.
    wctdm24xxp: Shorten RINGOFF debounce interval from 512ms to 128ms.
    xpp: Use 'bool' type for boolean module parameters on kernel versions >= 2.6.31.
    xpp: '%d' -> '%lu' when displaying module_refcount on kernel versions >= 3.3
    dahdi_dummy: Fix compilation since dahdi-linux 2.6.0.
    dahdi: Add dahdi_pci_disable_link_state for kernel < 2.6.25.
    wct4xxp: __t4_frame_in and __t4_framer_out slowdowns.
    wct4xxp: Add compile-time option to disable ASPM for PCIe devices.
    wcte12xp, wctdm24xxp: Add compile-time option to disable ASPM for PCIe devices.
    dahdi: Update dev_set_name / dev_name for RHEL 5.6+.
    dahdi_dynamic_eth: Move tx packet flushing to process context.
    dahdi_dynamic: Since dynamic devices are 'parentless' we must name them.
    dahdi_dynamic_eth: Prevent crash is packet arrives before span is fully configured.
    dahdi_dynamic_eth: Fix compilation on kernels < 2.6.22.
    wct4xxp: Disable all interrupts explicitly in interrupt handler.
    wct4xxp: Trivial formatting changes around request_irq.
    wctdm24xxp: Remove forward declaration of inline for GCC 3.4.4
    wctdm24xxp, wcte12xp: Allow VPMOCT032 firmware to be compiled into driver.
    dahdi_dynamic: Do not call into dahdi_dynamic without holding reference.
    dahdi_dynamic: Remove calls to __module_get().
    dahdi_dynamic: Close race on unload if red alarm timer was running when unloaded.
    dahdi_dynamic_eth: Make ztdeth_exit() symetrical with ztdeth_init() and fix race on unload.
    dahdi_dynamic_loc: Change and check the dyn->pvt pointer under lock.
    dahdi: Fix compilation when CONFIG_DAHDI_ECHOCAN_PROCESS_TX is defined.
    dahdi: Fix compilation when CONFIG_DAHDI_NET is defined.
    dahdi_dummy: Include timer.h instead of time.h
    wcb4xxp: Remove asm/system.h include.
    wcte12xp, wctdm24xxp, wct4xxp: Print warning about potential GPL violation w/HOTPLUG_FIRMWARE=no.
    xpp: Fix compilation when CONFIG_DAHDI_WATCHDOG is defined.

    Tzafrir Cohen (8):
    Build OSLEC EC if in the tree
    Astribank I firmwares rev. 7107
    USB_RECOV.hex: recovering from xpp hardware issues
    xpp: USB_FW rev 10401: minor 6FXS/2FXO caps issue
    xpp: firmwares to support E-Main 4
    xpp: firmwares: useless 0x1A at EOF
    FPGA_1161.201.hex rev 10532: fix reset of XR1000
    FPGA_1161.201.hex rev 10545: fix reset of XR1000

    The DAHDI-Linux diffstat from the 2.6.0 release:

     README                                        |   11 +-
     drivers/dahdi/Kbuild                          |   15 +-
     drivers/dahdi/dahdi-base.c                    |  106 +-
     drivers/dahdi/dahdi-sysfs.c                   |   15 +-
     drivers/dahdi/dahdi_dummy.c                   |   31 +-
     drivers/dahdi/dahdi_dynamic.c                 |   61 +-
     drivers/dahdi/dahdi_dynamic_eth.c             |   57 +-
     drivers/dahdi/dahdi_dynamic_loc.c             |    6 +-
     drivers/dahdi/firmware/Makefile               |    6 +
     drivers/dahdi/voicebus/Kbuild                 |   15 +-
     drivers/dahdi/voicebus/voicebus.h             |    9 +
     drivers/dahdi/voicebus/vpmoct.c               |   52 +-
     drivers/dahdi/wcb4xxp/base.c                  |    1 -
     drivers/dahdi/wct4xxp/Kbuild                  |    1 +
     drivers/dahdi/wct4xxp/base.c                  |   75 +-
     drivers/dahdi/wct4xxp/vpm450m.c               |    3 +-
     drivers/dahdi/wctdm24xxp/base.c               |   52 +-
     drivers/dahdi/wcte12xp/base.c                 |   12 +
     drivers/dahdi/xpp/card_fxs.c                  |   56 +-
     drivers/dahdi/xpp/card_global.c               |   27 +-
     drivers/dahdi/xpp/card_global.h               |    5 +-
     drivers/dahdi/xpp/firmwares/FPGA_1141.hex     | 1267 +-
     drivers/dahdi/xpp/firmwares/FPGA_1151.hex     | 1384 +-
     drivers/dahdi/xpp/firmwares/FPGA_1161.201.hex |20685 +++++++++++++++++++++++++
     drivers/dahdi/xpp/firmwares/FPGA_FXS.hex      | 1287 +-
     drivers/dahdi/xpp/firmwares/Makefile          |    6 +-
     drivers/dahdi/xpp/firmwares/USB_FW.201.hex    |  578 +
     drivers/dahdi/xpp/firmwares/USB_FW.hex        |  801 +-
     drivers/dahdi/xpp/firmwares/USB_RECOV.hex     |  421 +
     drivers/dahdi/xpp/xbus-core.c                 |   68 +-
     drivers/dahdi/xpp/xpd.h                       |    7 +
     drivers/dahdi/xpp/xpp_dahdi.c                 |    9 +-
     drivers/dahdi/xpp/xpp_dahdi.h                 |    1 +
     drivers/dahdi/xpp/xproto.c                    |   12 +-
     include/dahdi/kernel.h                        |   95 +-
     35 files changed, 24625 insertions(+), 2612 deletions(-)
    


    The DAHDI-Tools short log from 2.6.0 is:

    Tzafrir Cohen (13):
    xpp: clean 'Extrainfo' EEPROM field
    xpp: xpp_fxloader: improve output
    new build_tools/dahdi_sysfs_copy
    improve build_tools/dump_sys_state
    xpp: echo_loader.c: add AB id to messages
    xpp: bugfix: handle echo_loader errors
    xpp: Customizable license markers
    xpp: fix manpage of astribank_hexload
    xpp: Update product ID's in Dahdi::Hardware::USB
    xpp: oct612x: fix build warnings
    xpp: fix build warnings
    xpp: astribank_allow: useless debug on write
    xpp: astribank_hook: wait for udev to settle

    The DAHDI-Tools diffstat from 2.6.0:

     build_tools/dahdi_sysfs_copy                       |  142 ++++++++
     build_tools/dump_sys_state                         |  117 ++-----
     xpp/Makefile                                       |    2 +-
     xpp/astribank_allow.8                              |   10 +
     xpp/astribank_allow.c                              |  226 +------------
     xpp/astribank_hexload.8                            |   14 +-
     xpp/astribank_hook                                 |   27 ++
     xpp/astribank_license.c                            |  340 ++++++++++++++++++++
     xpp/astribank_license.h                            |   29 ++
     xpp/astribank_usb.h                                |   12 +
     xpp/echo_loader.c                                  |  168 ++++++----
     xpp/fpga_load.c                                    |   11 +-
     xpp/mpp.h                                          |    2 +-
     xpp/mpptalk.c                                      |   16 +-
     xpp/mpptalk_defs.h                                 |    2 +
     .../oct6100api/oct6100_api/oct6100_channel.c       |   75 -----
     .../oct6100api/oct6100_api/oct6100_chip_open.c     |   18 --
     .../oct6100api/oct6100_api/oct6100_conf_bridge.c   |   19 --
     .../oct6100api/oct6100_api/oct6100_interrupts.c    |   15 -
     .../oct6100api/oct6100_api/oct6100_tsst.c          |    3 -
     .../oct6100api/oct6100_miscellaneous_priv.h        |    2 -
     xpp/perl_modules/Dahdi/Hardware/USB.pm             |    1 +
     xpp/xpp_fxloader                                   |   32 +-
     xpp/xtalk/xtalk.c                                  |    5 +-
     24 files changed, 753 insertions(+), 535 deletions(-)
    


    For a full list of changes in these releases, please see the ChangeLogs at
    http://svn.asterisk.org/svn/dahdi/linux/tags/2.6.1/ChangeLog
    http://svn.asterisk.org/svn/dahdi/linux/tags/2.5.1/ChangeLog
    http://svn.asterisk.org/svn/dahdi/tools/tags/2.6.1/ChangeLog
    http://svn.asterisk.org/svn/dahdi/tools/tags/2.5.1/ChangeLog

    Issues found in this release can be reported in the DAHDI-Linux [1] and DAHDI-Tools [2] projects at https://issues.asterisk.org/jira

    Thank you for your continued support of Asterisk!

    --
    Shaun Ruffell
    Digium, Inc. | Linux Kernel Developer



    Release Announcement: Adhearsion 2.0 for Asterisk 1.8+
    Click to view a printable version Tue, 17 Apr 2012 10:32:51 +1200

    Ben Klang has posted details of the latest version of the Adhearsion framework:

    Today marks another milestone in the Adhearsion project: the release of Adhearsion 2.0.  There has been a fury of activity in the last few days as we have worked hard to update documentation and release a brand new look-and-feel for the Adhearsion website.  We hope you like it.

    So, with a small flourish and no small amount of relief, I'm pleased to announce the immediate availability of Adhearsion 2.0, the open source framework for the creation of voice applications.

    Here are some highlights of the changes relative to the latest Adhearsion 1.x:

    * Adhearsion now supports multiple telephony engines! In particular we support Asterisk (as always) as well newly added support for PRISM via the open-standard Rayo protocol

    CallControllers make telephone functionality more Ruby-esque, more testable and are scientifically shown to make you happier

    * A self-documenting configuration engine ("rake config:show")

    * A completely revamped plugin system makes adding and sharing Adhearsion functionality better than ever

    * Did I mention the new website design and documentation?

    * Way more stuff than I can reasonably list here.  You should check out the CHANGELOG and the Upgrade documentation.


    I would like to take a moment and recognize the team that made this happen.  The Adhearsion project has exploded in the last year, and many of the people who worked so hard to bring you Adhearsion 2 are actually new to the community within the last year!  A special thanks to Ben Langfeld who has driven much of this development effort and contributed fixes to many bugs and added new functionality in some of our dependency packages in the process of making this happen.  I also want to thank our sponsors, especially Tropo, for not only funding direct development, but helping to evangelize and organize.  Tropo has been a fantastic collaborator throughout Adhearsion's lifetime.

    Now, you might be thinking "all of the above sounds great, but how stable can it really be? Is it webscale?"  The answer is "very stable" and "yes", respectively.  But I don't want you to just take my word for it.  A few weeks back, I bet Ben Langfeld a double sawbuck (that is, an Andrew Jackson, a USD $20) that Adhearsion 2 wasn't ready to take a fully loaded server's worth of traffic.  And he muttered something about me not keeping the faith, and then took me up on that bet.  So now we're going to do it live.  In the next couple of weeks we are going to do a live broadcast of a load test, pushing Adhearsion to scale on both Asterisk and PRISM.  We are going to see just how "webscale" it is, and we're going to be streaming the event live on Ustream so you all can join in the fun.  The loser (hopefully me) will be well and truly prepared to take your jeers and fork over the cash.  Look for an announcement soon for where and when.  It's about as geeky fun as telephony gets.  I hope you'll come join us.

    In the meantime, go check out Adhearsion 2!

    On behalf of the Adhearsion 2 development team, thanks for being you.
    -- 
    Ben Klang



    New Community Developer: Michael Young
    Click to view a printable version Tue, 17 Apr 2012 10:19:47 +1200

    Matthew Jordan has posted details of a new community developer:

    Michael L. Young (elguero) has officially joined the community of Asterisk developers. Many of you who visit the Asterisk issue tracker may have interacted with Michael before. He has been instrumental in diagnosing, testing, and fixing many bugs, including the dreaded music on hold reload changing timing source bug in ASTERISK-17474. In addition to his bug stomping contributions, he has provided several new features to Asterisk, including the expansion of the security event framework to SIP. Most recently, he has added IPv6 support to the SIP portion of the security event framework.

    Please join me in welcoming Michael to the community developer team!

    Matthew Jordan
    Digium, Inc. | Software Developer



    Asterisk SIP Masterclass in Barcelona, Spain
    Click to view a printable version Fri, 13 Apr 2012 10:55:19 +1200

    Olle has posted details of the final Asterisk SIP Masterclass he is running:

    Friends,

    I've been running the Asterisk SIP Masterclass for many years now. It's time to run the last show - partly with new material. Compared with the very first Asterisk SIP Masterclass I would say that I've rewritten 90% of the material. That's what happens during the class. Students ask questions, you write a slide. The word changes, you write a slide. You realize you've been wrong, you delete or edit a slide. It's a moving target.

    The last one of these classes that I teach will be in Barcelona, Spain - June 11th to June 15th.

    * Why the last one?
    --------------------------
    Things change and you need to follow. During the last couple of years I've been running many, many in-house trainings and workshops covering both Asterisk, SIP in general and Kamailio. There seems to be more demand for customized trainings that boost a team and help them move forward. I will continue with these trainings, as well as try to come up with other trainings that will run just a few times - more lab oriented possibly.

    * What is this class?
    ---------------------------
    From the sales material at http://www.avanzada7.com:

    "This class is focusing on building a scalable SIP realtime network. With a combination of theory and practical labs, you will learn how to setup and configure Asterisk and Kamailio - the Open Source SIP server - in a scalable enterprise or service provider network. We will go through various kinds of setups and give you insight in the design of real SIP networks with Asterisk running in enterprise and service provider networks. The teacher Olle Johansson, has many years of experience as an Asterisk developer as well as a community member of Kamailio.org. By spending a week with Olle, you will get a lot of insight into current and future features, bugs and implementation details in a way that's hard to get otherwise.

    Olle is a consultant working with architecture and implementation of large scale communication platforms based on the SIP protocol. He has experience from service providers, universities, call center platforms as well as enterprise solutions. With experience of Unix and TCP/IP networking for over 20 years, he has a lot of insight and knowledge, which he is using as a teacher."

    The class is a five day high level class. You will meet not only myself, but also other students that work with these tools and protocols, learn from them and work together to solve issues in the labs. You need to have a basic knowledge of Linux (how to start/stop applications, edit text files and build applications) and Asterisk. This class is starting at a high level with Asterisk. If you rather use FreeSwitch but want to learn Kamailio that is no problem. You will just have to endure a few slides on Asterisk - but many of the issues apply to FreeSwitch as well as other PBXs too.

    The cost is 3.200 Euro ex VAT. Companies outside of EU do not pay VAT as well as companies in EU with a VAT registration number.

    If you have any questions or want to register, feel free to contact me directly.

    Looking forward to seeing you in Barcelona!

    Best regards,
    /Olle

    ---
    oej@edvina.net - http://edvina.net
    Open Unified Communication - building platforms with SIP and XMPP
    From PBX to large scale implementations for carriers. Contact us today!



    Recent FreePBX vulnerability attacks
    Click to view a printable version Wed, 11 Apr 2012 11:19:11 +1200

    Jared Geiger has posted details of an unfixed security vulnerability he is seeing exploited in FreePBX:

    We saw some activity related to this FreePBX unpatched vulnerability this past weekend on some hosted PBXes.

    http://seclists.org/fulldisclosure/2012/Mar/234

    Usually we see the typical SIP Vicious attacks, but this one is much more involved and dangerous.

    Excerpt from the disclosure:

    A remote command execution vulnerability and some XSS in current and earlier FreePBX versions due to missing input sanitization.

    FreePBX is a popular implementation (500,000 active phone systems) of Asterisk (telephony software) based around a web-based configuration interface and other tools. Some of these installations are on a public IP address.



    Speech recognition and Text to Speech using Google
    Click to view a printable version Wed, 11 Apr 2012 10:56:46 +1200

    I was reading through the Asterisk Users mailing list and came across a couple of really cool AGI scripts for doing speech recognition and converting speech to text.

    They are both by Lefteris Zafiris.

    The first one is described as:

    AGI script for the Asterisk open source PBX which allows you to use Google's voice synthesis engine to render text to speech.

    This script makes use of Google's translate text to speech service in order to render text to speech and play it back to the user. It supports a variety of different languages (See README for a complete list), local caching of the voice data and also supports 8kHz or 16kHz sample rates to provide the best possible sound quality along with the use of wideband codecs.

    Read More...

    The second one is for voice recognition:

    Speech recognition script for Asterisk that uses Google speech API.
    This AGI script makes use of Google's speech recognition engine in order to render speech to text and return it back to the dialplan as an asterisk channel variable. See README for a complete list of supported languages.

    Read More...

    He has actually written quite a few other scripts too:

    https://github.com/zaf/



    Asterisk 10.3.0 Now Available
    Click to view a printable version Fri, 30 Mar 2012 12:35:13 +1300

    The Asterisk Development Team has announced the release of Asterisk 10.3.0.

    This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

    The release of Asterisk 10.3.0 resolves several issues reported by the community and would have not been possible without your participation.

    Thank you!

    The following are the issues resolved in this release:

    • Fix potential buffer overrun and memory leak when executing "sip show peers"
      (Closes issue ASTERISK-19231. Reported by Thomas Arimont, Jamuel Starkey)

    • Fix ACK routing for non-2xx responses.
      (Closes issue ASTERISK-19389.)

    • Remove possible segfaults from res_odbc by adding locks around usage of odbc handle
      (Closes issue ASTERISK-19011. Reported by Walter Doekes)

    • Fix blind transfer parking issues if the dialed extension is not recognized as a parking extension.
      (Closes issue ASTERISK-19322. Reported by aragon)

    • Copy CDR variables when set during a bridge
      (Closes issue ASTERISK-16990.)

    • push 'outgoing' flag from sig_XXX up to chan_dahdi
      (Closes issue ASTERISK-19316. Reported by Jeremy Pepper)

    For a full list of changes in this release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.3.0

    Thank you for your continued support of Asterisk!



    Asterisk 1.8.11.0 Now Available
    Click to view a printable version Fri, 30 Mar 2012 12:31:38 +1300

    The Asterisk Development Team has announced the release of Asterisk 1.8.11.0.

    This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

    The release of Asterisk 1.8.11.0 resolves several issues reported by the community and would have not been possible without your participation.
    Thank you!

    The following are the issues resolved in this release:

    • Fix potential buffer overrun and memory leak when executing "sip show peers"
      (Closes issue ASTERISK-19231. Reported by Thomas Arimont, Jamuel Starkey)
    • Fix ACK routing for non-2xx responses.
      (Closes issue ASTERISK-19389.)
    • Remove possible segfaults from res_odbc by adding locks around usage of odbc handle
      (Closes issue ASTERISK-19011. Reported by Walter Doekes)
    • Fix blind transfer parking issues if the dialed extension is not recognized as a parking extension.
      (Closes issue ASTERISK-19322. Reported by aragon)
    • Copy CDR variables when set during a bridge
      (Closes issue ASTERISK-16990.)
    • push 'outgoing' flag from sig_XXX up to chan_dahdi
      (Closes issue ASTERISK-19316. Reported by Jeremy Pepper)



    For a full list of changes in this release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.11.0

    Thank you for your continued support of Asterisk!



    How to Configure the New Call Blocking Asterisk Collector
    Click to view a printable version Thu, 22 Mar 2012 14:59:52 +1300

    Eric has written a post on how to configure the call blocking Asterisk collector for their Humbug fraud blocking service.

    Excerpt:

    Humbug Telecom Labs has released the Silver Hawk version of our collector for Asterisk based systems

    This new collector enables companies running Asterisk version 1.4 (or later) to benefit from Humbug’s fraud blocking service. The blocking is currently available for the following alerts:

    For Asterisk version 1.4 and up:

    Business Hours
    Time Range
    For Asterisk versions 1.6 and up:

    Blacklist
    Community Blacklist
    Blacklist Country

    We will be expanding to cover all our alerts in the near future. For more details about the features in the latest release please see our blog about it.

    Read More...



    Asterisk 1.4.44, 1.6.2.23, 1.8.10.1, 10.2.1 Now Available (Security Releases)
    Click to view a printable version Mon, 19 Mar 2012 11:08:12 +1300

    The Asterisk Development Team has announced security releases for Asterisk 1.4, 1.6.2, 1.8, and 10. The available security releases are released as versions 1.4.44, 1.6.2.23, 1.8.10.1, and 10.2.1.

    These releases are available for immediate download at
    http://downloads.asterisk.org/pub/telephony/asterisk/releases

    The release of Asterisk 1.4.44 and 1.6.2.23 resolve an issue wherein app_milliwatt can potentially overrun a buffer on the stack, causing Asterisk to crash. This does not have the potential for remote code execution.

    The release of Asterisk 1.8.10.1 and 10.2.1 resolve two issues. First, they resolve the issue in app_milliwatt, wherein a buffer can potentially be overrun on the stack, but no remote code execution is possible. Second, they resolve an issue in HTTP AMI where digest authentication information can be used to overrun a buffer on the stack, allowing for code injection and execution.

    These issues and their resolution are described in the security advisory.

    For more information about the details of these vulnerabilities, please read the
    security advisories AST-2012-002 and AST-2012-003, which were released at the same
    time as this announcement.

    For a full list of changes in the current releases, please see the ChangeLogs:

    http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.4.44
    http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.23
    http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.10.1
    http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.2.1

    The security advisories are available at:

    http://downloads.asterisk.org/pub/security/AST-2012-002.pdf
    http://downloads.asterisk.org/pub/security/AST-2012-003.pdf

    Thank you for your continued support of Asterisk!



    AstLinux Custom Build Engine Now Available
    Click to view a printable version Tue, 13 Mar 2012 10:57:45 +1300

    Kristian Kielhofner has posted details of some work that has been completed in the AstLinux project to provide a custom build engine:

    Hello everyone,

    Apologies for the cross-post but I want to make sure this announcement reaches as many people as possible.

    Thanks to the efforts of Lonnie Abelbeck over the past week or so we now have a resource for dynamically building custom AstLinux images directly from a publicly accessible web interface!

    I can't begin to describe how cool this system is... It supports at least the following features:

    - Build AstLinux SVN
    - Build AstLinux 1.0.2
    - Build either AstLinux 1.0 or AstLinux SVN with either Asterisk 1.4 or 1.8
    - Include some drivers and not others
    - Upgrade your system directly from this build repository
    - Automatic notification via e-mail when your custom build completes
    - Images that match your selections are cached, making them available for download immediately
    - Plenty more to come (submit your requests please)!

    NOTE: To keep bots out of this service a username and password have been applied and it is as follows:

    username: admin
    password: astlinux

    The AstLinux Custom Build Engine is available immediately at the following URL:

    http://build.astlinux.org


    Thanks again Lonnie!

    --
    Kristian Kielhofner



    Asterisk 10.2.0 Now Available
    Click to view a printable version Tue, 06 Mar 2012 11:09:26 +1300

    The Asterisk Development Team has announced the release of Asterisk 10.2.0.

    This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

    The release of Asterisk 10.2.0 resolves several issues reported by the community and would have not been possible without your participation.

    Thank you!

    The following is a sample of the issues resolved in this release:

    * --- Prevent outbound SIP NOTIFY packets from displaying a port of 0 ---
    (Closes issue ASTERISK-19430. Reported by Schmooze Com)

    * --- Include iLBC source code for distribution with Asterisk ---
    (Closes issue ASTERISK-18943. Reported by Leif Madsen)

    * --- Fix callerid of originated calls ---
    (Closes issue ASTERISK-19385. Reported by or nix Fixed by Matt Riddell)

    * --- Fix outbound DTMF for inband mode of chan_ooh323 ---
    (Closes issue ASTERISK-19233. Reported, patched by Matt Behrens)

    * --- Create and initialize udptl only when dialog requests image media ---
    (Closes issue ASTERISK-16794. Reported by under, tested by Stefan Schmidt)

    * --- Don't prematurely stop SIP session timer ---
    (Closes issue ASTERISK-18996. Reported by Thomas Arimont)

    For a full list of changes in this release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.2.0

    Thank you for your continued support of Asterisk!



    Asterisk 1.8.10.0 Now Available
    Click to view a printable version Tue, 06 Mar 2012 11:07:12 +1300

    The Asterisk Development Team has announced the release of Asterisk 1.8.10.0.

    This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

    The release of Asterisk 1.8.10.0 resolves several issues reported by the community and would have not been possible without your participation.

    Thank you!

    The following is a sample of the issues resolved in this release:

    * --- Prevent outbound SIP NOTIFY packets from displaying a port of 0 ---
    (Closes issue ASTERISK-19430. Reported by Schmooze Com)

    * --- Include iLBC source code for distribution with Asterisk ---
    (Closes issue ASTERISK-18943. Reported by Leif Madsen)

    * --- Fix callerid of originated calls ---
    (Closes issue ASTERISK-19385. Reported by or nix - fixed by Matt Riddell)

    * --- Fix outbound DTMF for inband mode of chan_ooh323 ---
    (Closes issue ASTERISK-19233. Reported, patched by Matt Behrens)

    * --- Create and initialize udptl only when dialog requests image media ---
    (Closes issue ASTERISK-16794. Reported by under, tested by Stefan Schmidt)

    * --- Don't prematurely stop SIP session timer ---
    (Closes issue ASTERISK-18996. Reported by Thomas Arimont)

    For a full list of changes in this release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.10.0

    Thank you for your continued support of Asterisk!



    AstriCon 2012 Call for Speakers
    Click to view a printable version Tue, 06 Mar 2012 10:24:46 +1300

    Bryan Johns has posted a call for speakers for the 2012 Astricon in October:

    It’s that time of year again! AstriCon 2012 will be held October 23-25 at the downtown Sheraton Hotel in Atlanta, Georgia. As a Atlanta native (there aren’t that many of us) I am excited to have the Asterisk community join us in my home town for three days of Asterisk business, discussion and training.

    I am opening our call for speakers effective today. You can submit your talk using this form: http://www.astricon.net/2012/speaking.aspx

    Accepted speakers will be given an all conference pass in exchange for their contribution. I hope that you will join us in Atlanta for what is likely to be the largest and best AstriCon yet.



    AstriCon 2012 Call for Speakers
    Click to view a printable version Tue, 06 Mar 2012 10:24:46 +1300

    Bryan Johns has posted a call for speakers for the 2012 Astricon in October:

    It’s that time of year again! AstriCon 2012 will be held October 23-25 at the downtown Sheraton Hotel in Atlanta, Georgia. As a Atlanta native (there aren’t that many of us) I am excited to have the Asterisk community join us in my home town for three days of Asterisk business, discussion and training.

    I am opening our call for speakers effective today. You can submit your talk using this form: http://www.astricon.net/2012/speaking.aspx

    Accepted speakers will be given an all conference pass in exchange for their contribution. I hope that you will join us in Atlanta for what is likely to be the largest and best AstriCon yet.



    Video on New Digium Phones
    Click to view a printable version Mon, 05 Mar 2012 12:13:18 +1300

    Digium have created a video that explains a bit more about the new phones they have created:





    AstLinux 1.0.2 Release
    Click to view a printable version Wed, 29 Feb 2012 14:42:18 +1300

    The AstLinux team is happy to announce the release of version 1.0.2. This release features several security updates. All current users are encouraged to upgrade as soon as possible. Please see the documentation at http://doc.astlinux.org for upgrade details or the official release pages.

    Updates:
    Asterisk (1.8.9.2)
    DAHDI (2.5.0.2)
    Rhino(0.99.5b1)
    Wanpipe (3.5.24)
    The Sangoma BRI/Hybrid cards (A500 + B700) are now supported via DAHDI

    Security Fixes:
    PHP(5.3.10)
    OpenSSL(0.9.8t)

    New Features:
    A "Test SMTP Mail Relay" feature was added to verify msmtp configuration

    See the change log on either of these release pages for more details

    http://www.astlinux.org/release/102-asterisk-1892
    http://www.astlinux.org/release/102-asterisk-1443

    Enjoy,
    The AstLinux Team



    FreePBX Remote Exploit
    Click to view a printable version Fri, 17 Feb 2012 08:22:38 +1300

    There was a vulnerability in FreePBX which has been fixed but was posted to the BugTraq mailing list. Make sure your systems are up to date.

    FreePBX web interface remote vulnerability

    The admin username and password for the web interface is stored in plain text in this publicly accessible file:
    http://yourip/admin/modules/framework/bin/gen_amp_conf.php

    Which allows a hacker to access the web GUI and view the secrets(passwords) for each extension in plain test, as well as change the outbound routes.

    Further details on this exploit can be found here:
    http://linuxsecurityblog.com/2012/02/freepbx-vulnerable/

    This was claimed to have been fixed around December 2011, however the vulnerability never made it to bugtraq.
    http://www.freepbx.org/forum/freepbx/development/security-gen-amp-conf-php

    I have confirmed this has been fixed as of:
    FreePBX 2.10.0rc1.1 and distro release 1.88.210.57-1

    And it appears to have affected versions for at least 1 year prior to their fix.



    Asterisk Open Source - Now with more Long Term Support
    Click to view a printable version Wed, 15 Feb 2012 12:07:27 +1300

    Kevin Fleming has posted a blog entry on the changing of LTS releases for Asterisk. They will now happen every second year.

    Excerpt from his post:

    Starting with this year’s release of Asterisk 11 (scheduled for October, just in time for AstriCon), the Asterisk release policy will be changed in order to provide Long Term Support (LTS) releases on a more frequent basis.

    A little background: Asterisk 1.4, released in late 2006, was the first release labeled with the ‘LTS’ designation (although that didn’t happen until years later, when the Asterisk development team started using that designation). Asterisk 1.8, released in late 2010, was the second release labeled ‘LTS’. Asterisk 10 was released earlier this year, and is a standard (not LTS) release.

    Until recently, the plan has been to make LTS releases every three years, and make two standard releases in between those releases. However, since standard releases have a limited support lifetime, customers have requested releases that they can use for longer periods of time to be made on a more frequent basis.

    Read more...



    Asterisk 1.8.9.1 Now Available
    Click to view a printable version Wed, 08 Feb 2012 11:19:13 +1300

    The Asterisk Development Team has announced the release of Asterisk 1.8.9.1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

    The release of Asterisk 1.8.9.1 resolves several issues reported by the community and would have not been possible without your participation.
    Thank you!

    The following is a sample of the issues resolved in this release:

    * --- Fixes deadlocks occuring in chan_agent ---
    (Closes issue ASTERISK-19285. Reported by: Alex Villacis Lasso)

    * --- Ensure entering T.38 passthrough does not cause an infinite loop ---
    (Closes issue ASTERISK-18951. Reported-by: Kristijan Vrban)

    For a full list of changes in this release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.9.1

    Thank you for your continued support of Asterisk!



    Asterisk 10.1.1 Now Available
    Click to view a printable version Wed, 08 Feb 2012 11:14:51 +1300

    The Asterisk Development Team has announced the release of Asterisk 10.1.1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

    The release of Asterisk 10.1.1 resolves several issues reported by the community and would have not been possible without your participation.
    Thank you!

    The following is a sample of the issues resolved in this release:

    * --- Fixes deadlocks occuring in chan_agent ---
    (Closes issue ASTERISK-19285. Reported by: Alex Villacis Lasso)

    * --- Ensure entering T.38 passthrough does not cause an infinite loop ---
    (Closes issue ASTERISK-18951. Reported-by: Kristijan Vrban)

    For a full list of changes in this release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.1.1

    Thank you for your continued support of Asterisk!



    Digium Creates Phones for Asterisk
    Click to view a printable version Thu, 02 Feb 2012 12:03:19 +1300

    Steven Sokol has posted details of the release of Digium phones for Asterisk:

    It’s official! Today, we are pleased to provide a sneak preview of the NEW family of high-definition Digium IP phones, designed exclusively for Asterisk. The phones will officially launch and be available to ship in April 2012. As a part of the Asterisk Community, we want to provide you with an opportunity to learn more about the phones and see our special early release video announcement.

    What makes the new Digium IP phones special? These are the first phones designed to fully leverage the power of Asterisk, the world’s most widely adopted open source communications software, and Switchvox, Digium’s award-winning unified communications (UC) system. With Digium technology on both the server and the phone, you will benefit from the best possible performance, unprecedented integration and a uniquely customizable phone system – all at an extremely competitive price.

    "Digium’s new phones mark the launch of the next chapter in our history of innovation. These are the first phones designed specifically for Asterisk—with the tightest integration possible between the phones and Asterisk. The success of Asterisk began with the transfer of power from the hands of the proprietary phone vendors to the hands of end users and administrators of phone systems. And now we’ve done it again by bringing control to the desk phone."

    - Danny Windham, President and CEO of Digium

    Extending Digium’s history of innovation, these IP phones include an app engine with a simple yet powerful JavaScript API that lets you, or programmers in your organization, create custom apps that run on the phones. A suite of productivity applications has been created that work with both Asterisk and Switchvox.

    "The app engine is a game-changing feature that will allow developers to write their own apps that run on the phones. We have a community of more than 80,000 users and developers who create amazing things with Asterisk. I look forward to seeing the cool apps they will create with these innovative phones. As usual, we’re enabling developers to create solutions limited only by their imaginations."

    - Mark Spencer, Founder and Chief Technology Officer of Digium

    The Digium IP phones include the following models:

    D40—An entry-level HD IP phone with 2-line keys. This is Digium’s best value phone, designed for any employee in the company.

    D50—A mid-level HD IP phone with 4-line keys and 10 rapid dial/busy lamp field (BLF) keys with an easy to print paper label strip for the user’s most important contacts. This model is perfect for managers or users who need easy access to their key contacts and features directly from the desk phone.

    D70—An executive-level HD IP phone with 6-line keys and 10 rapid dial/busy lamp field (BLF) keys and real-time status information displayed on an additional LCD screen, allowing users to quickly navigate through up to 100 of their most important contacts. Designed for administrators or executives, the D70 offers top-of-the-line features.

    This is an exciting time in Digium’s history and we are glad to have you as a part of the Asterisk Community as we prepare to officially launch Digium phones!

    Once again, we invite you to share in our excitement and watch this special message from our founder, Mark Spencer, and our CEO, Danny Windham. We also hope you will take a moment to learn more about extending the power of Asterisk with Digium IP phones.

    Best regards,

    Steve Sokol
    Digium, Inc. | Asterisk Marketing Director



    Converting multiple exten lines to using same in Asterisk dialplan
    Click to view a printable version Tue, 24 Jan 2012 12:44:38 +1300

    Leif Madsen has written a post on his blog about converting Asterisk 1.4 dial plans to using the same feature in 1.8 dial plans.

    Excerpt from his post:

    Last week I wanted to start changing some 1.4 based Asterisk dialplan to a 1.8 based Asterisk system, and in that process wanted to convert lines like:

    exten => _NXXNXXXXXX,1,NoOp()
    exten => _NXXNXXXXXX,2,GotoIf($[...]?reject,1)
    exten => _NXXNXXXXXX,3,Dial(SIP/foo/${EXTEN})
    ...
    into using the same => prefix:

    exten => _NXXNXXXXXX,1,NoOp()
    same => n,GotoIf($[...]?reject,1)
    same => n,Dial(SIP/foo/${EXTEN})

    Read More...



    Asterisk 1.8.8.2 and 10.0.1 Now Available (Security Release)
    Click to view a printable version Tue, 24 Jan 2012 12:07:59 +1300

    The Asterisk Development Team has announced security releases for Asterisk 1.8 and 10. The available security releases are released as versions 1.8.8.2 and 10.0.1. Please note that the security vulnerability in Asterisk 1.8 and 10 does not exist for Asterisk versions 1.4 or 1.6.2.

    These releases are available for immediate download at
    http://downloads.asterisk.org/pub/telephony/asterisk/releases

    The release of Asterisk versions 1.8.8.2 and 10.0.1 resolves an issue wherein an attacker attempting to negotiate a secure video stream can crash Asterisk if video support has not been enabled and the res_srtp Asterisk module is loaded.

    The issue and its resolution is described in the security advisory.

    For more information about the details of these vulnerabilities, please read the security advisory AST-2012-001, which were released at the same time as this announcement.

    For a full list of changes in the current releases, please see the ChangeLogs:

    http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.8.2
    http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.0.1

    Security advisory AST-2012-001 is available at:

    http://downloads.asterisk.org/pub/security/AST-2012-001.pdf

    Thank you for your continued support of Asterisk!



    AstLinux 1.01 Released
    Click to view a printable version Mon, 16 Jan 2012 12:12:39 +1300

    The AstLinux Team would like to announce the release of 1.0.1. This version is available with either Asterisk 1.4.43 or Asterisk 1.8.8.3. A full changelog and upgrade (or new install) instructions are available on our website. Please follow the upgrade instructions carefully when upgrading from a release prior to 1.0.

    http://www.astlinux.org

    Please note that this release includes a change in the way PATA (ide) devices are handled by the kernel. Those devices are now handled by libata which references the drives as /dev/sdx instead of /dev/hdx.

    As always, please report any issues (and comments) to the AstLinux mailing list on Sourceforge. (link available at the above website).

    The AstLinux Team



    Asterisk 1.6.2.22 Now Available
    Click to view a printable version Tue, 20 Dec 2011 12:18:21 +1300

    The Asterisk Development Team has announced the release of Asterisk 1.6.2.22. This release is available for immediate download at
    http://downloads.asterisk.org/pub/telephony/asterisk/

    The release of Asterisk 1.6.2.22 corrects two flaws in sip.conf.sample related to AST-2011-013:

    * The sample file listed *two* values for the 'nat' option as being the default. Only 'yes' is the default.

    * The warning about having differing 'nat' settings confusingly referred to both peers and users.

    For a full list of changes in this release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.22

    Thank you for your continued support of Asterisk!



    Asterisk 1.8.8.0 Now Available
    Click to view a printable version Mon, 19 Dec 2011 13:38:02 +1300

    The Asterisk Development Team is pleased to announce the release of Asterisk 1.8.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

    The release of Asterisk 1.8.8.0 resolves several issues reported by the community and would have not been possible without your participation.

    Thank you!

    The following is a sample of the issues resolved in this release:

    * Updated SIP 484 handling; added Incomplete control frame When a SIP phone uses the dial application and receives a 484 Address Incomplete response, if overlapped dialing is enabled for SIP, then the 484 Address Incomplete is forwarded back to the SIP phone and the HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete application dialplan logic was automatically triggered; now, explicit dialplan usage of the application is required.
    (Closes ASTERISK-17288. Reported by: Mikael Carlsson Tested by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/1416/)

    * Prevent IAX2 from getting IPv6 addresses via DNS IAX2 does not support IPv6 and getting such addresses from DNS can cause error messages on the remote end involving bad IPv4 address casts in the presence of IPv6/IPv4 tunnels.
    (Closes issue ASTERISK-18090. Patched by Kinsey Moore)

    * Fix bad RTP media bridges in directmedia calls on peers separated by multiple Asterisk nodes.
    (Closes issue ASTERISK-18340. Reported by: Thomas Arimont. Closes issue ASTERISK-17725. Reported by: kwk. Tested by: twilson, jrose)

    * Fix crashes in ast_rtcp_write()
    (Closes issue ASTERISK-18570)
    Related issues that look like they are the same problem: (Issue ASTERISK-17560, ASTERISK-15406, ASTERISK-15257, ASTERISK-13334, ASTERISK-9977, ASTERISK-9716)
    Review: https://reviewboard.asterisk.org/r/1444/
    Patched by: Russell Bryant

    * Fix for incorrect voicemail duration in external notifications. This patch fixes an issue where the voicemail duration was being reported with a duration significantly less than the actual sound file duration.
    (Closes ASTERISK-16981. Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH Tested by: Matt Jordan)
    Review: https://reviewboard.asterisk.org/r/1443)

    * Prevent segfault if call arrives before Asterisk is fully booted.
    (Patched by alecdavis. https://reviewboard.asterisk.org/r/1407/)

    * Fix remote Crash Vulnerability in SIP channel driver (AST-2011-012)
    http://downloads.asterisk.org/pub/security/AST-2011-012.pdf

    * Fix locking order in app_queue.c which caused deadlocks
    (Closes issue ASTERISK-18101. Reported by Paul Rolfe, patched by Gregory Nietsky)
    (Closes issue ASTERISK-18487. Reported by Jason Legault, patched by Gregory Nietsky)

    * Fix regression in configure script for libpri capability checks
    (Closes issue ASTERISK-18687. Reported by norbert, patched by Richard Mudgett)

    * Prevent BLF subscriptions from causing deadlocks.
    (Closes issue ASTERISK-18663)
    Review: https://reviewboard.asterisk.org/r/1563/

    * Fix deadlock if peer is destroyed while sending MWI notice.
    (Closes issue ASTERISK-18747)
    Reported by: Gregory Hinton Nietsky

    * Fix issue with setting defaultenabled on categories that are already enabled by default.
    (Closes issue ASTERISK-18738)
    Reported by: Paul Belanger

    * Don't crash on INFO automon request with no channel AST-2011-014. When automon was enabled in features.conf, it was possible to crash Asterisk by sending an INFO request if no channel had been created yet.

    * Fixed crash from orphaned MWI subscriptions in chan_sip This patch resolves the issue where MWI subscriptions are orphaned by subsequent SIP SUBSCRIBE messages.

    * Default to nat=yes; warn when nat in general and peer differ AST-2011-013. It is possible to enumerate SIP usernames when the general and user/peer nat settings differ in whether to respond to the port a request is sent from or the port listed for responses in the Via header. In 1.4 and

    For a full list of changes in this release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.8.0

    Thank you for your continued support of Asterisk!



    Asterisk 10.0.0 Released
    Click to view a printable version Mon, 19 Dec 2011 10:50:17 +1300

    The Asterisk Development Team is proud to announce the release of Asterisk 10.0.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

    Asterisk 10 is the next major release series of Asterisk. It will be a Standard support release, similar to Asterisk 1.6.2. For more information about support time lines for Asterisk releases, see the Asterisk versions page:

    https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

    With the release of the Asterisk 10 branch, the preceding '1.' has been removed from the version number per the blog post available at

    http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/

    The release of Asterisk 10 would not have been possible without the support and contributions of the community.

    You can find an overview of the work involved with the 10.0.0 release in the summary:

    http://svn.asterisk.org/svn/asterisk/tags/10.0.0/asterisk-10.0.0-summary.txt

    A short list of available features includes:

    • T.38 gateway functionality has been added to res_fax.
    • Protocol independent out-of-call messaging support. Text messages not associated with an active call can now be routed through the Asterisk dialplan. SIP and XMPP are supported so far.
    • New highly optimized and customizable ConfBridge application capable of mixing audio at sample rates ranging from 8kHz-192kHz
    • Addition of video_mode option in confbridge.conf to provide basic video conferencing in the ConfBridge() dialplan application.
    • Support for defining hints has been added to pbx_lua.
    • Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB).
    • Much, much more!

    A full list of new features can be found in the CHANGES file.

    http://svn.asterisk.org/svn/asterisk/branches/10/CHANGES

    Also, when upgrading a system between major versions, it is imperative that you read and understand the contents of the UPGRADE.txt file, which is located at:

    http://svn.asterisk.org/svn/asterisk/branches/10/UPGRADE.txt

    Thank you for your continued support of Asterisk!



    AstLinux 1.0.0 release
    Click to view a printable version Mon, 19 Dec 2011 10:30:52 +1300

    The AstLinux Team is happy to announce the release of AstLinux 1.0.0. This release includes significant changes and improvements over past releases. Specific upgrade and new installation instructions are available at: http://www.astlinux.org

    Some of the highlights include:

    • Using eglibc instead of uClibc. This allows binary compatibility with add-ons that are provided as binary only (G.729 CODEC, Fax for Asterisk etc).
    • Newer Kernel which better supports newer hardware
    • Support for Jabber/Gtalk
    • Removed mISDN support (the zaphfc DAHDI driver is included for single port ISDN cards)

    A full changelog is available on the release pages. We provide versions with Asterisk 1.8 and 1.4.

    Because this is a major version change, there are some special considerations when upgrading. Please read the instructions very carefully to ensure no step is skipped.

    http://doc.astlinux.org/userdoc:upgrade-0.7

    Please report any issues with the release back to the AstLinux mailing list.

    Enjoy,

    The AstLinux Team



    Asterisk 10.0.0-rc2 Now Available
    Click to view a printable version Wed, 16 Nov 2011 14:35:24 +1300

    The Asterisk Development Team has announced the second release candidate of Asterisk 10.0.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

    The release of Asterisk 10.0.0-rc2 resolves several issues reported by the community and would have not been possible without your participation. Thank you!

    The following is a sample of the issues resolved in this release candidate:

    * Ensure that a null vmexten does not cause a segfault

    * Fix issue with ConfBridge participants hanging up during DTMF feature menu usage getting stuck in conference forever
    (closes issue ASTERISK-18829)
    Reported by: zvision

    * Fix app_macro.c MODULEINFO section termination
    (closes issue ASTERISK-18848)
    Reported by: Tony Mountifield

    For a full list of changes in this release candidate, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.0.0-rc2

    Thank you for your continued support of Asterisk!



    Asterisk 10.0.0-rc1 Now Available
    Click to view a printable version Fri, 11 Nov 2011 08:19:06 +1300

    The Asterisk Development Team is pleased to announce the first release candidate of Asterisk 10.0.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

    All Asterisk users are encouraged to participate in the Asterisk 10 testing process. Please report any issues found to the issue tracker, https://issues.asterisk.org/jira. It is also very useful to see successful test reports. Please post those to the asterisk-dev mailing list.

    All Asterisk users are invited to participate in the #asterisk-testing channel on IRC to work together in testing the many parts of Asterisk.

    Asterisk 10 is the next major release series of Asterisk. It will be a Standard support release, similar to Asterisk 1.6.2. For more information about support time lines for Asterisk releases, see the Asterisk versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

    A short list of features includes:

    • T.38 gateway functionality has been added to res_fax.
    • Protocol independent out-of-call messaging support. Text messages not associated with an active call can now be routed through the Asterisk dialplan. SIP and XMPP are supported so far.
    • New highly optimized and customizable ConfBridge application capable of mixing audio at sample rates ranging from 8kHz-192kHz
      (More information available at https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10 )
    • Addition of video_mode option in confbridge.conf to provide basic video conferencing in the ConfBridge() dialplan application.
    • Support for defining hints has been added to pbx_lua.
    • Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB).
    • Much, much more!

    A full list of new features can be found in the CHANGES file.

    http://svnview.digium.com/svn/asterisk/branches/10/CHANGES

    For a full list of changes in the current release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.0.0-rc1

    Thank you for your continued support of Asterisk!



    oFono 1.0 has been released
    Click to view a printable version Thu, 10 Nov 2011 12:11:26 +1300

    Steve Totaro has forwarded details of the latest release of a project called oFono:

    When oFono launched, I announced the project to other projects that it may compliment.

    oFono has hit the 1.0 Milestone and has some serious backing if you missed my post a year or so ago and never heard of it.

    Check it out...

    Thanks,
    Steve Totaro

    ---------- Forwarded message ----------
    From: Marcel Holtmann

    Hello everybody,

    I am pleased to announce that we have released oFono 1.0 this week. This marks a major step for oFono and we consider it fully feature complete for 2G and 3G telephony.

    oFono is released under GPL version 2 and is 100% open source. It includes support for the majority of data modem vendors and also full voice telephony support for Infineon (now IMC), ST-Ericsson, Nokia/ISI and also Calypso/Freerunner.

    All standard features including voice calls, supplementary services, text messaging, USSD, SIM Toolkit, network registration, multiple GPRS contexts and many more have been integrated with easy to use D-Bus APIs.

    With our 1.0 out of the door, we will continue to improve our CDMA support and also integrate LTE into oFono. So stay tuned for new features.

    The tarballs are not yet available due to the security breach on kernel.org, but will be uploaded as soon as service has been restored.

    Regards

    Marcel



    New Asterisk community developers
    Click to view a printable version Thu, 10 Nov 2011 11:49:03 +1300

    Two new community developers have been granted permission to begin committing code into the Asterisk repository.

    Walter Doekes (wdoekes) has been active in the community and on reviewboard for several months.

    David Woodhouse (dwmw2) has offered to maintain app_sms.

    Thanks, Walter and David!



    Astricon: Matt Jordan: Asterisk APIs
    Click to view a printable version Wed, 26 Oct 2011 06:25:36 +1300

    The final session before lunch is Matt Jordan talking about Asterisk APIs.

    The discussion is basically about some of the tools Asterisk provides to help you with development.

    This is about internal utilities, functions, objects etc.

    Why?:

    • Provides functionality not in C
    • Commonality across Asterisk
    • Avoid common errors
    • Add additional debugging
    • Develop faster, safer

    Some examples are string manipulations, string objects, ref counting, containers, events and channel bridging.

    String manipulations:

    ast_copy_string, art_strdup, ast_strndup, ast_strip, art_stip_quoted, art_trim_blanks, ast_true/ast_false, art_get_time_t/ast_get_timeval, ast_str_hash/ast_str_case_hash

    He then gave an example of a crash - strdup on a null string. Replace with ast_strdup and you're all good.

    String Objects
    Unknown length => lots of reallocs

    Alternatively use dynamic strings - auto growing, length, truncate etc, support vargs.

    Stringfields
    Pool of pre allocated strings
    Can grow
    Makes managing char */char[] in structs easier

    ast_str_create to create a string
    ast_str_set with string, initial pointer, variable args

    Reference Counting
    Objects that automatically track who is owning that object. When count reaches 0 the object is auto destroyed. Similar to objective C.
    Useful for information that goes stale or complex lifetimes (i.e channels).
    Things to watch out for: it's atomic, but global objects are not thread safe. Circular dependencies, if A depends on B's lifetimes and B depends on A's lifetime etc.

    He then provided an example of how to use them.

    Containers:
    There are two types: Vanilla (unsorted lists), Chocolate (hash tables), no strawberry :-)

    All support thread sage and non thread safe access.

    All support insert/delete/iterate

    When to use lists:

    Small number of things, difficult to hash, when you need to do the same thing to all items.

    When to use hash tables:

    Care about speed, large number of items, operate on one item at a time, objects have to be ref-counted

    He then provided some examples of usage.

    Asterisk Events
    Publish/subscribe API:

    • Voicemail MWI
    • Device state changes
    • Security events
    • Channel event logs

    Anyone can publish/subscribe (even external entities)

    Lots of room for expansion.

    Channel Bridging

    Standard API for connecting 2 to n channels - Various tech support, Video, Codec negotiation
    Feature examples:

    • Blind transfer
    • Attended 3 way transfer
    • Dial transfer
    • ConfBridge (Asterisk 10)





    Astricon: Shaun Ruffell: DAHDI
    Click to view a printable version Wed, 26 Oct 2011 06:00:22 +1300

    The next discussion is by Shaun Ruffell on the DAHDI drivers (Digium Asterisk Hardware Device Interface).

    He's discussing what DAHDI does, how to use it, the general architecture, some of the tools and future direction.

    DAHDI is basically a set of kernel modules for talking to hardware, mixing, conferencing and bridging. It also provides times, echo cans, generates tones (DTMF etc), provides gain control.

    It can also provide HDLC framing in software on raw channels.

    You need DAHDI for timing if you're pre 2.6.25 without timbered or prior to version 1.8 or meetme/page.

    For compiling you basically just need to make sure you have the kernel headers.

    With a version number of A.B.C.D you should be able to use any A.B.x.y version. Tools and DAHDI do not require same numbers.

    If possible run from DAHDI trunk. It changes quite slowly so it is really helpful if you can run trunk.

    dahdi_genconf is the traditional way to configure 99% of use cases. Use fxotune with analogue cards to tune the hybrid.

    An example would be:

    modprobe wct4xxp && dahdi_genconf_system && dahdi_cfg -vvf

    He then went though some slides showing how the modules interact. It's quite hard to describe in text so again for this part you'll need to download the slides after they've been uploaded.

    Coming up there will be an object called dahdi_device. This will allow devices to explain timing relationships.

    System start up sequence:

    power on
    udev
    dahdi init script
    loads modules
    possibly wait for udev to create /dev/dahdi files such as:

    • /dev/dahdi/ctl
    • /dev/dahdi/pseudo
    • /dev/dahdi/channel
    • /dev/dahdi/timer
    • /dev/dahdi/transcode
    • /dev/dahdi/{channel number}

    It will the start processing config. Timing/Coding preferences set, signaling modes for channels set up, echo cans are loaded and assigned, tone zones are loaded.

    Next it runs fxotune (if configured) followed by HPEC and Asterisk.

    Tools:

    dahdi_cfg, dahdi_tool, dahdi_scan, fxotune, dahdi_test, dahdi_monitor, dahdi_maint, patlooptest/patgen/pattest

    Future directions:

    • co evolve with the kernel (follows kernel coding conventions)
    • New and evolving hardware support
    • Support sysfs (will come in in 2.6 but won't be advertised till 2.7) - will speed up boot
    • Continue to optimize: moving stuff out of the interrupt handler, eliminate transcoding, clean up processing inside per millisecond interrupts)





    Astricon: Matt Jordan: Asterisk Developer Tools
    Click to view a printable version Wed, 26 Oct 2011 05:36:46 +1300

    The next topic is the developer tools.

    You'll need:

    Editor, Compiler, SVN.

    You'll likely eventually need:

    Debugging, testing, reviews, safely commit/merge code.

    For debugging you can use ast_log, art_verb, ast_debug, ast_backtrace.

    art_backtrace is a new addition.

    art_debug is for developers - not for sending to users.

    gdb:

    Command line, attach to existing process, run a process, do core dumps, UIs available (DDD, some IDEs)

    Asterisk also has things that are built in:

    Locks: core show locks (deadlocks, sometimes things look like deadlocks but aren't)

    Threads: core show threads (Orphaned threads, Snapshot)

    Others: Memory allocation tracing, object reference count tracing

    He then moved into a practical example. Creating a deadlock.

    He has purposely introduced a bug (a deadlock).

    First he starts with core show locks. Nothing shows up.

    He then originated a local channel.

    This caused a deadlock which he showed with core show locks.

    It shows all the locks. One thread is waiting for lock number 2. That thread is waiting on a lock that the other thread holds.

    Other third party tools that are useful:

    Wireshark
    Network sniffer

    Valgrind
    Memory/thread analysis tools
    Asterisk suppressions are in contrib/valgrind.supp
    Build options needed:
    MALLOC_DEBUG
    DONT_OPTIMIZE
    DEBUG_THREADS (optional)

    On the wiki there's a walkthrough on how to run Asterisk under valgrind.

    Next we moved on to unit testing.

    Why: it may be easier, additional level of verification etc.

    CLI commands to run/view results:

    test show registered
    test execute
    test generate results
    test show results

    Examples:

    asterisk/tests
    test_skel.c

    There is also retools, post-review etc.

    It's always ok to post a review.

    SVN: update, commit

    Repotools - Merge - allows you to merge to the correct branches etc.

    Automerge allows you to keep dev branches in sync. Can email you conflicts.

    More info is available on the Asterisk wiki



    Astricon: Jason Parker: Asterisk Module API
    Click to view a printable version Wed, 26 Oct 2011 04:55:53 +1300

    After a quick break we returned to the presentation with the next being by Jason.

    He will be showing us what is required to make a module for Asterisk and what they need to do.

    Module API essentials
    Command Line Interfaces
    Configuration Files
    Registering Applications

    For a basic module you need:

    load_module()

    Gets called by Asterisk when it gets loaded. To run code, load the module, initialize variables etc. It can either succeed or fail:

    AST_MODULE_LOAD_SUCCESS
    AST_MODULE_LOAD_DECLINE
    AST_MODULE_LOAD_FAILURE

    unload_module()

    Unload whatever you used. Free RAM. Returns 0 on success other on fail.

    reload_module()

    Happens on module reload. Updated a config file etc. Returns 0 on success other on fail.
    Module Information (one of these two):

    AST_MODULE_INFO(keystr, flags_to_set, desc, load_func, unload_func, reload_func, load_pri)

    A more complicated example.

    AST_MODULE_INFO_STANDARD(keystr, desc);

    This provides:

    keystr: license text (normally ASTERISK_GPL_KEY)
    desc: brief description of module

    He then went though four examples:

    res_nothing.c

    Does nothing.

    res_cliexample.c

    Register a CLI command

    res_configexample.c

    Reads config

    res_appexample.c

    Registers a dialplan app

    These examples will be available as PDF files after Astricon completes.

    Jason took us through a few short C files showing the usage of the above files.

    I'll try to get copies of these files and upload them here.

    Also have a look at app_skel in the Asterisk source.

    Update: the examples are below:

    res_appexample.c
    res_cliexample.c
    res_configexample.c
    res_nothing.c




    David Vossell: Asterisk Architecture
    Click to view a printable version Wed, 26 Oct 2011 04:39:25 +1300

    The next discussion was from David on the structure of Asterisk internals:

    He started by discussing Asterisk Core. His speech started with a quick discussion on the fact that the Asterisk Core is basically a tool.

    The core can't make calls or anything.

    You have Asterisk modules, most of which come under 5 categories:

    • Channel Drivers
    • Dialplan Apps
    • Dialplan Functions
    • Codec Translators
    • Resource Modules

    At a minimum you need two modules - a channel driver and a dialplan app for the end point to talk to.

    A channel is the most important piece of Asterisk. It forms a connection between Asterisk and an endpoint.

    It means that Asterisk doesn't need to know how the underlying stuff happens - it just asks the channel driver to read etc and the channel driver does it. This means Asterisk can treat all channels in the same way. It's basically a channel abstraction layer.

    After a channel is created it is sent to the dialplan.

    These you should be familiar with i.e. Queue, Dial, Background etc.

    The next topic was channel bridging. Media is abstracted by the art_frame. This is an abstraction layer than any channel tech can consume/create.

    Two types of bridging:

    • Generic Bridging - standard
    • Native Bridging - between the same channel techs

    The next section is on threads

    Asterisk is highly multithreaded.

    Network monitor threads read and write to the network (they queue frames to the PBX thread which push the frames to the dialplan app).

    Codec translation

    If two channels are speaking different codecs then translation modules need to transcode. A translation path is created between them.

    Every time a codec module is loaded Asterisk creates a translation path between all possible formats and optimizes it.

    I.E. g722->slin16->siren7

    The next topic is audiohooks.

    Some things need to consume media and possibly do something with it - i.e. mixmonitor. You register a callback to receive the frames and can change the data before it moves on (i.e. pitch shifting, volume adjustment etc).

    There is also the frame hook api. This allows you to receive anything - not just audio. I.E. T38 gateway, video, jitterbuffer dialplan application.

    Media formats

    Media formats used to be represented by a single bit in Asterisk. They never had much info associated with them. You never needed to know the sample size etc. This doesn't work for things like celt, video etc.

    For things like h264 you need to have things like variable bit rate, variable frame rate etc.

    Asterisk 10 and greater changed this to a format id and any custom attributes in the format attributes. Now you have things like resolution, frame rate, encoding_profile.



    AstriCon: Paul Belanger: Introduction to Asterisk Development
    Click to view a printable version Wed, 26 Oct 2011 04:07:17 +1300

    For the first session of the day I decided to follow the Introduction to Asterisk development.

    Leif started the day as an MC and introduced Paul Belanger who started off with the basics of Asterisk. How to get Asterisk, where the mailing lists are, what the irc groups are, how to use the bug tracker etc.

    By the way, if you're at AstriCon you should join the #astricon group on irc.freenode.net.

    Paul then moved on to reporting bugs - making sure people submit back traces, config files, SIP traces if it's related to SIP, and then getting it confirmed by someone else.

    Applying a patch from an existing issue is a good way to get started - does it apply, does it work, does it solve the problem it's supposed to be solving? If you know how to use diff and patch then this is an easy way to get started.

    The next topic he covered was some of the existing tools. The first being the review board. An easy way to see the progress of a patch from start through to completion.

    If you're doing development it's a really good idea to have a test for the test framework to go with it. It's more likely to be committed if the patch includes a test for the issue.

    Leif then took over to explain the release management policy.



    AstriDevCon final session
    Click to view a printable version Tue, 25 Oct 2011 11:11:44 +1300

    With a quarter of an hour break in the sun we're back to the table and ready to finish off the afternoon.

    We started with a discussion on music on hold and how to create a mixed music on hold source without actually mixing music with dialogue.

    Leif stated that you can do it with chanspy and theres an example in the Asterisk cookbook.

    We then thought we might prioritize the items we discussed in the brainstorm but it was decided that we would leave it to the wider community.

    There was a discussion about the deprecation of app_macro - although this can't be done until other apps that rely on it no longer need it.

    To finish off the afternoon we went outside for a group photo.



    AstriDevCon afternoon session - Asterisk 11
    Click to view a printable version Tue, 25 Oct 2011 08:04:44 +1300

    Having topped up on nutrients we are ready to begin the first session of the afternoon.

    The morning session basically covered where things were at last year and updated them based on progress etc.

    As the session progresses Leif Madsen has been adding content to the wiki page:

    https://wiki.asterisk.org/wiki/display/AST/AstriDevCon+2011

    We have some new faces in for the afternoon session. Namely:

    Brad Watkins: Compuware
    Abel Camarillo: Neuroservices Communications
    Nicolas Bouliane: Avencall
    Xavier Carcelle: Avencall
    David Deaton: Digium
    Mathieu De Bellefeuille: Nisse
    Alex Balashov: Evariste Systems
    Joel Schuweiler: Digium
    Shawn McCord: Cycore Systems
    Klaus Darilion: ipcom.at

    We started by bringing people up to date with what happened this morning.

    Alex Balashov brought up SIP path support with an offer to spend some time on it. Klaus Darilion had written the original patch and while it works it doesn't support realtime. Kevin Fleming said that as the backend of AstDB will be able to be stored in a realtime DB, the registration fields will be removed from realtime anyway.

    We then actually moved on to discuss the wiki. Do people use it etc? Basically yes. It needs to be updated more. It needs to be more clear how to become a contributing editor. It's been a good platform and allows creation of documentation straight from the source - meaning it keeps in sync.

    In the future the integration between Jira and the Wiki etc (such as linking to related bugs) will be added.

    The bug tracker supports voting but at this stage it's not being used - there is a danger of disappointing people if there are lots of votes but an issue isn't getting work on.

    Alex brought up the overwriting of rtp port number or rather the reusing of port numbers. The idea of doing more efficient RTP port allocations via maybe a linked list of oldest ports used etc. On a loaded system it can be easy to reuse an existing port.

    We then discussed config files. The sample config files have a lot of information in them - way too much for the average user. Maybe we should export the available options to the wiki etc. That way you could also have something that parsed the config files and tell you if it's correct.

    Return to last known state - if a reload fails return to a previously working state rather than failing. Also being able to just go back to a previous state after a restart rather than the current state. One other option is to use SVN etc to contain your config files.

    Another option would be to force Asterisk not to start if a module failed to load. This is mostly usefully in a cluster where there are other machines that can take over.

    There was a brief discussion of Async AGI and whether it would be supported in the future - mostly because Adhearsion was thinking of moving entirely to it.

    We've decided to have a break for a while till at least 4pm then do a wrap-up.



    AstriDevCon Test Framework
    Click to view a printable version Tue, 25 Oct 2011 06:32:39 +1300

    The next discussion focussed primarily on the test framework:

    Pretty much everything has moved to Python - another language could be added but would maybe be a better idea to abstract it and make it a web service or something.

    One of the things discussed was to add adhersion as a test. What kinds of tests could be used to confirm it's all working as expected.

    Adhearsion is already running internal tests but they are not actually run as part of the test framework.

    One of the problems at the moment is that if a test fails there is a delay fixing that test.

    It would likely need to be run as a secondary testing framework so it didn't break bamboo.

    Including Adhearsion support would be a good way to test AMI and AGI.

    We then moved on to checking for memory leaks in the test suite. Quite a bit of work has been done on this so far. This checks for things like whether there is anything remaining after a test is completed.

    Automated crash analysis is being worked on and will hopefully start to provide back traces in those situations.

    There is a SIP protocol tester that could be run but it takes around 50 minutes to run - there is an idea to run build tests once a day.

    Making tests more generic was discussed although it might not make sense - i.e. options to the dial command should produce the same results no matter what channel type was being used.

    Adding support for chan_dahdi would be pretty useful too. There's no reason a card would be needed to use chan_dahdi.

    More info here




    AstriDevCon second morning session
    Click to view a printable version Tue, 25 Oct 2011 05:53:16 +1300

    Having had a break to replenish caffeine levels we are back to it.

    The first discussion was on creating unique identifiers for errors so that they can be looked up in the wiki (basically so you can use it as a knowledge base).

    We next discussed Asterisk Manager filters on the fly so that you can add or remove filters in real time to a logged in session.

    Next was creating a spec for the Asterisk Manager. That way if anything changes in the Asterisk Manager the version number would be increased so that people could rely on a document for compliance. This would likely go hand in hand with the creation of a list of Asterisk Manager Events.

    We then discussed having the ability to have multiple TLS certificates so that for example if you're hosting multiple companies you could receive a cert and issue the correct cert based on the request that comes in without having to have an IP address per host.

    ODBC, Postgres and MySQL now reconnect to the database if the connection is dropped for realtime.

    We then moved on to talking about the current app_queue - it's unlikely to ever be replaced. There's now quite a bit of functionality for distributed queues etc, but not great info centrally. I.E. somewhere there needs to be an atomic decision maker. Lately irroot has been working on cleaning up queues. There's some work being done where if an agent is in two queues and they both have calls that only one queue will send the agent a call. He is also keen to do work on skills based routing for queues.

    Threadifying chan_sip - it would really require a rewrite - TCP did some work for this but nowhere near enough.

    Our next discussion is on the Test Framework.

    More info here




    How well does Asterisk perform with VMWARE
    Click to view a printable version Tue, 25 Oct 2011 05:46:05 +1300

    Greenfield Tech has posted an article on stress testing Asterisk under VMWare.

    Excerpt from the article:

    Everybody these days are big into cloud computing – be it due to cost constraints, hype requirements or simply because you don’t have anything else to do – cloud computing is here to stay and will be with us for the next 10 years at the least. About 2 years ago, GreenfieldTech was involved in the testing and adaptation of Asterisk into the Amazon EC2 cloud infrastructure – since then much has happened. Asterisk based EC2 AMI images had become a norm and you can find dozens if not hundreds of Asterisk installations on Amazon EC2. The one thing that people always ask: “How does a cloud perform? is it measurable?” – well, most of the EC2 measurements were based upon fairly simple applications, so the actual results varied – in addition, due to the nature of the Amazon Cloud, results varied from one cloud region to another, depending on your termination provider, inbound provider, inbound bandwidth, etc. In other words, a definite answer was somewhat hard to give.



    AstriDevCon 2011 and AstriCon 2011
    Click to view a printable version Tue, 25 Oct 2011 03:50:13 +1300

    Having arrived in Denver after a long flight from New Zealand the first of the days - AstriDevCon 2011.

    The day started with an introduction. The following were present:

    • Eric Klein: Humbug Telecom
    • Dunling Li: BTS
    • Chris May
    • Shaun Ruffell: Digium
    • Bryan Johns: Digium
    • Mark Murawski: Intellasoft
    • Kevin Fleming: Digium
    • Byron Clark: Jack Communications
    • Jamual Starkey: Headland Communications
    • Leif Madsen: LeifMadsen Enterprises, Inc / Digium
    • Matt Riddell: Asterisk Daily News
    • Nir Simonivich: Greenfield Technologies / Humbug Telecom
    • Paul Belanger: Digium
    • Claude Patry
    • Jason Parker: Digium
    • David Vossel: Digium
    • Matt Jordan: Digium
    • Brad Wakens: Compuware
    • Ben Klang: Mojo Lingo / Adhearsion

    Some cool stuff:

    Group variables being done - you can assign a channel to a group and specify variables associated: Review 464

    Pre dial stuff so you can for example do stuff with agents before the call actually goes out: Review 1229

    Ben Klang has written documentation for installing Asterisk 1.6.0.4 through to Asterisk 10 on Solaris.

    David Vossell removed the reliance on a bitfield for codecs. He integrated Silk, Celt and allowed for more information for things like video.

    He also did the confbridge code which provides high def audio and some cool video conferencing - tested and working.

    Security event framework got added. Need more info to be available. Initially just supported Manager login failures. Prior to Asterisk 10 SIP authentication details. More things will get added as companies make software that consume these events.

    Work is being done in Asterisk SCF to support matroska as a file container, but not yet into Asterisk.

    A patch will be added to provide an event that states which channel hung up a call.

    There was a lengthy discussion on changing verbosity levels on a per output basis - i.e. verbosity for console, log files etc. There will need to be further discussion on this topic - it has the potential to have quite a performance penalty.

    There's some work being done to move AstDB to allow it to use a realtime backend.

    There is a chan_rtmp driver on the bug tracker that needs quite a bit of work before it could go in and it's not currently being worked on much.

    Seems it's time for a break so I'll sign off for the moment and update again after.

    More info here



    Asterisk 1.8.7.1 Now Available (Security Release)
    Click to view a printable version Wed, 19 Oct 2011 11:25:04 +1300

    The Asterisk Development Team has announced a security release for Asterisk 1.8. The available security release is released as version 1.8.7.1.

    This release is available for immediate download at
    http://downloads.asterisk.org/pub/telephony/asterisk/releases

    The release of Asterisk 1.8.7.1 resolves an issue with SIP URI parsing which can lead to a remotely exploitable crash:

    Remote Crash Vulnerability in SIP channel driver (AST-2011-012)

    The issue and resolution is described in the AST-2011-012 security advisory.

    For more information about the details of this vulnerability, please read the security advisory AST-2011-012, which was released at the same time as this announcement.

    For a full list of changes in the current release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.7.1

    Security advisory AST-2011-012 is available at:

    http://downloads.asterisk.org/pub/security/AST-2011-012.pdf

    Thank you for your continued support of Asterisk!



    AppKonference 2.0 released
    Click to view a printable version Thu, 13 Oct 2011 10:45:39 +1300

    Paul Albrecht has posted details of the latest version of AppKonference:

    Hi,

    I have released an updated AppKonference today. This fix release includes the following changes:

    Speed up the conference thread by reusing mix buffers and removing unnecessary frame duplication.

    Remove video pass through feature.

    Change to c idioms for conditionals and use asterisk macros for memory allocations.

    You can download the latest source from source forge:

    sourceforge.net/projects/appkonference

    --
    Paul Albrecht



    New Asterisk community developer
    Click to view a printable version Thu, 06 Oct 2011 13:21:36 +1300

    Kevin Fleming has posted details of a new developer with commit privileges:

    Igor Goncharovsky has just been granted permission to begin committing code into the Asterisk repository; he's been working heavily on chan_unistim lately and has volunteered to take over maintenance of that module. His Subversion committer name is 'igorg'.

    Thanks, Igor!

    --
    Kevin P. Fleming
    Digium, Inc. | Director of Software Technologies



    AstriCon 2011 Developer Days - Topics Requested
    Click to view a printable version Tue, 27 Sep 2011 18:13:28 -0300

    Bryan M. Johns has posted a note about the upcoming AstriDevCon held in conjunction with Astricon:

    Ladies and Gentlemen, Boys and Girls, it's time once again for AstriDevCon. Held this year in conjunction with the AstriCon conference in Denver, AstriDevCon will take place on Monday October 24th and on Friday October 28th, from 8am until 5pm. The Monday session will be dedicated to discussion about Asterisk, and the Friday session will cover Asterisk SCF.

    Both sessions are open to all AstriCon attendees that are actively involved in the development of Asterisk and/or Asterisk SCF. If you're not already involved in the development of Asterisk SCF, but you're planning to become so, your attendance is welcome.

    If you can't attend in person, we'll be doing our best to make listen-only participation available via VoIP dial-in and we'll try to get a Freenode channel going.

    With this e-mail, we'd like to solicit topics for conversation during the sessions.

    We'll consolidate the ideas and responses into a page on the Asterisk Project wiki, which we'll be updating during the DevCon for those following along at home.

    We expect there will be conversation about the development completed in the past year, general discussion about the projects, and ideas/brainstorming/planning for development work to be done in the upcoming year.

    So, what do you want to talk about?

    Bryan M. Johns
    Digium | Community Director



    Asterisk 10.0.0-beta2 Now Available
    Click to view a printable version Tue, 27 Sep 2011 18:07:10 -0300

    The Asterisk Development Team is pleased to announce the second beta release of Asterisk 10.0.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

    With the release of the Asterisk 10 branch, the preceding '1.' has been removed from the version number per the blog post available at http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/

    All interested users of Asterisk are encouraged to participate in the Asterisk 10 testing process. Please report any issues found to the issue tracker, https://issues.asterisk.org/jira. It is also very useful to see successful test reports. Please post those to the asterisk-dev mailing list.

    All Asterisk users are invited to participate in the #asterisk-testing channel on IRC to work together in testing the many parts of Asterisk.

    Asterisk 10 is the next major release series of Asterisk. It will be a Standard support release, similar to Asterisk 1.6.2. For more information about support time lines for Asterisk releases, see the Asterisk versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

    A short list of features includes:

    * T.38 gateway functionality has been added to res_fax.

    * Protocol independent out-of-call messaging support. Text messages not associated with an active call can now be routed through the Asterisk dialplan. SIP and XMPP are supported so far.

    * New highly optimized and customizable ConfBridge application capable of mixing audio at sample rates ranging from 8kHz-192kHz

    * Addition of video_mode option in confbridge.conf to provide basic video
    conferencing in the ConfBridge() dialplan application.

    * Support for defining hints has been added to pbx_lua.

    * Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB).

    * Much, much more!

    A full list of new features can be found in the CHANGES file.

    http://svnview.digium.com/svn/asterisk/branches/10/CHANGES

    For a full list of changes in the current release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.0.0-beta2

    Thank you for your continued support of Asterisk!



    Asterisk 1.8.7.0 Now Available
    Click to view a printable version Sun, 25 Sep 2011 18:46:03 -0300

    The Asterisk Development Team announces the release of Asterisk 1.8.7.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

    The release of Asterisk 1.8.7.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you!

    Please note that a significant numbers of changes and fixes have gone into features.c in this release (call parking, built-in transfers, call pickup, etc.).

    NOTE:

    Recently, we were notified that the mechanism included in our Asterisk source code releases to download and build support for the iLBC codec had stopped working correctly; a little investigation revealed that this occurred because of some changes on the ilbcfreeware.org website. These changes occurred as a result of Google's acquisition of GIPS, who produced (and provided licenses for) the iLBC codec.

    If you are a user of Asterisk and iLBC together, and you've already executed a license agreement with GIPS, we believe you can continue using iLBC with Asterisk. If you are a user of Asterisk and iLBC together, but you had not executed a license agreement with GIPS, we encourage you to research the situation and consult with your own legal representatives to determine what actions you may want to take (or avoid taking).

    More information is available on the Asterisk blog:

    http://blogs.asterisk.org/2011/09/19/ilbc-support-in-asterisk-after-googles-acquisition-of-gips/

    The following is a sample of the issues resolved in this release:

    • Added the 'storesipcause' option to sip.conf to allow the user to disable the setting of HASH(SIP_CAUSE,) on the channel. Having chan_sip set HASH(SIP_CAUSE,) on the channel carries a significant performance penalty because of the usage of the MASTER_CHANNEL() dialplan function.

      We've decided to disable this feature by default in future 1.8 versions. This would be an unexpected behavior change for anyone depending on that SIP_CAUSE update in their dialplan. Please refer to the asterisk-dev mailing list more information:

      http://lists.digium.com/pipermail/asterisk-dev/2011-August/050626.html

    • Significant fixes and improvements to parking lots.
      (Closes issues ASTERISK-17183, ASTERISK-17870, ASTERISK-17430, ASTERISK-17452, ASTERISK-17452, ASTERISK-15792. Reported by: David Cabrejos, Remi Quezada, Philippe Lindheimer, David Woolley, Mat Murdock. Patched by: rmudgett)

    • Numerous issues have been reported for deadlocks that are caused by a blocking read in res_timing_timerfd on a file descriptor that will never be written to.

      A change to Asterisk adds some checks to make sure that the timerfd is both valid and armed before calling read(). Should fix: ASTERISK-18142, ASTERISK-18197, ASTERISK-18166 and possibly others.
      (In essence, this change should make res_timing_timerfd usable.)

    • Resolve segfault when publishing device states via XMPP and not connected.
      (Closes issue ASTERISK-18078. Reported, patched by: Michael L. Young. Tested by Jonathan Rose)

    • Refresh peer address if DNS unavailable at peer creation.
      (Closes issue ASTERISK-18000)

    • Fix the missing DAHDI channels when using the newer chan_dahdi.conf sections for channel configuration.
      (Closes issue ASTERISK-18496. Reported by Sean Darcy. Patched by Richard Mudgett)

    • Remove unnecessary libpri dependency checks in the configure script.
      (Closes issue ASTERISK-18535. Reported by Michael Keuter. Patched by Richard Mudgett)

    • Update get_ilbc_source.sh script to work again.
      (Closes issue ASTERISK-18412)

    For a full list of changes in this release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.7.0

    Thank you for your continued support of Asterisk!



    Chan SS7 2.1.0 released
    Click to view a printable version Thu, 22 Sep 2011 19:40:34 -0300

    Anders Baekgaard from Netfors has posted details of the release of the latest version of their chan_ss7:

    Netfors has released chan_ss7 v. 2.1.0.

    This release is compatible with Asterisk version 1.8.x as well as earlier versions.

    More information is available at http://www.netfors.com/chan_ss7

    Please provide comments tochan_ss7 at netters dot com.

    Best regards,

    --
    Anders Baekgaard
    Netfors



    iLBC support in Asterisk after Google acquires GIPS
    Click to view a printable version Tue, 20 Sep 2011 22:23:29 -0300

    The Asterisk Development Team has posted details about iLBC disappearing from Asterisk 1.4 and Asterisk 1.6.2:

    Recently, we were notified that the mechanism included in our Asterisk source code releases to download and build support for the iLBC codec had stopped working correctly; a little investigation revealed that this occurred because of some changes on the ilbcfreeware.org website. These changes occurred as result of Google's acquisition of GIPS, who produced (and provided licenses for) the iLBC codec.

    We've determined that the change necessary to fix Asterisk's iLBC build process is rather trivial, and so we're planning to make that change in Asterisk 1.8.7.0-rc2, and subsequently in 1.8.7.0. We are not planning on making new releases of Asterisk 1.4 and Asterisk 1.6.2, since they are in security-maintenance mode and this is not a security issue. Users who wish to make the same change on their own to their copies of those versions are of course welcome to do so.

    As part of the process of determining what had broken here, we also became aware that the ilbcfreeware.org website no longer offers the iLBC license agreement it used to offer; this agreement was required by the iLBC licensors (GIPS) in order for users to safely distribute and use iLBC (and this is why the Asterisk project does not include the iLBC source code directly with Asterisk). The removal of this license agreement also occurred as a result of the Google acquisition, but as of this moment no alternative has been made available for those who wish to use the iLBC source code published in RFC 3951 (which Asterisk uses).

    Google does have an alternative implementation of iLBC available as part of the WebRTC project, with a license that is compatible with Asterisk (and does not require written agreements from end users), but the codec_ilbc module in Asterisk cannot be built against the WebRTC implementation of iLBC. Until such time as we have an improved version of codec_ilbc, Asterisk users will have to continue using the RFC 3951 iLBC source code.

    Unfortunately, that leaves Asterisk users in a bit of a bind; if they had already signed and sent in the GIPS iLBC license agreement, we believe they can continue to safely use the existing iLBC implementation. New users, though, do not have the option of agreeing to a license agreement that would allow them to use the RFC 3951 iLBC source code, as there is no mechanism to do that currently available. We've contacted Google and they are aware of the dilemma, and have said that they will address it, but we don't have a timeframe for when an alternative license mechanism will be available.

    In summary, if you are a user of Asterisk and iLBC together, and you've already executed a license agreement with GIPS, we believe you can continue using iLBC with Asterisk. If you are a user of Asterisk and iLBC together, but you had not executed a license agreement with GIPS, we encourage you to research the situation and consult with your own legal representatives to determine what actions you may want to take (or avoid taking).

    -- Asterisk Development Team



    AstriCon 2011 Open Source Showcase
    Click to view a printable version Tue, 20 Sep 2011 22:20:33 -0300

    Bryan M. Johns has posted a note about free tables for Open Source products at Astricon:

    Digium is working hard to prepare for this year's AstriCon conference in Denver, Colorado. If you haven't seen the list of activities, speakers and keynotes please take a moment to stop by http://www.astricon.net for all the latest conference details.

    The purpose of this email is to solicit Open Source projects that are associated with Asterisk to participate in the Open Source Showcase at this year's event. This program provides you with a FREE table in our exhibition hall to promote your project.

    If you're interested in being a part of the Open Source Showcase, please email me directly at johns at digium dot com and I will gladly reserve a spot for you. Please keep in mind that tables are limited and will be allocated on a first-come, first-serve basis.

    We are looking forward to seeing you at AstriCon!

    Bryan M. Johns
    Digium, Inc. | Community Director



    Astricon 2011 Update
    Click to view a printable version Thu, 08 Sep 2011 19:50:13 -0300

    Bryan Johns has posted an update about Astricon. As you all know by now I will be speaking on the Wednesday morning. I hope to see you all there!

    Here's Bryan's post:

    This year’s Astricon event is less than 2 months away and it is shaping-up to be the best show yet! As our trip to Denver, Colorado approaches, I wanted to take a moment to bring everyone up to speed on the details of the show as they have been finalized. If you haven’t done so recently, please take a moment to stop by the Astricon site and take a look at the updated conference schedule and content announcements. Here’s a few important items for you to consider as you plan for Astricon.

    Tuesday is an important day

    For a number of years, Tuesday at Astricon has been referred to as a “pre-conference” day. This year, Tuesday is going to be chock full of quality content. Tuesday’s program will include developer workshops for both Asterisk and Asterisk SCF, an Innovator’s Forum content track, a VoIP security roundtable, Asterisk 123 and a handful of other great talks and announcements. Make sure your travel plans have you in Denver Monday night so that you can make the most of Tuesday’s planned activities.

    Two DevCon Days

    This year, we will be hosting DevCon days on both Monday and Friday. Monday will be allocated to the traditional AstriDevCon where community developers will convene to establish the features and development roadmap for Asterisk 11. On Friday we will hold the inaugural Asterisk SCF DevCon where we will be defining features and timelines for the next planned release of our new open source project. If you are a developer and want to participate in the future of either Asterisk or Asterisk SCF, please plan to attend one or the other or both of these sessions.

    Announcing the Innovator’s Forum

    This year we received a significant number of quality speaker proposals and we have struggled to fit in everything that we want to have included in the show. To expand our available slots for content, our innovation track is being moved to theater-style, half-day session called the Innovator’s Forum. This session will focus on companies and developers doing innovative things with Asterisk and will formatted as a “lightening talk” series. In this session we will also highlight Digium’s yet to be announced 2011 Innovation Award winner. The Innovator’s Forum is another reason why you’ll want to be at the show on Tuesday.

    Lots of special announcements

    Digium has a number of exciting announcements planned for this year’s Astricon. While I am not at liberty to divulge any of those here, you will want to be there for the big news. Don’t you want to be able to tell your friends and colleagues that you were there when it happened? Make sure you’re there on Tuesday to be a part of the excitement that will comprise this year’s Astricon.

    That’s all for now but keep your eye on this blog and on the Astricon site for all the latest updates on the show and the things to do while you are in Denver. I am excited to see everyone in October and we are looking forward to a great show. See you at Astricon!



    Asterisk 1.8.6.0 Now Available
    Click to view a printable version Wed, 31 Aug 2011 23:10:35 -0300

    The Asterisk Development Team announces the release of Asterisk 1.8.6.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

    The release of Asterisk 1.8.6.0 resolves several issues reported by the community and would have not been possible without your participation.

    Thank you!

    The following is a sample of the issues resolved in this release:

    * Fix an issue with Music on Hold classes losing files in playlist when realtime is used.
    (Closes issue ASTERISK-17875. Reported by David Cunningham. Patched by Igor Goncharovsky)

    * Resolve a potential crash in chan_sip when utilizing auth= and performing a 'sip reload' from the console.
    (Closes issue ASTERISK-17939. Reported by wdoekes. Patched by Richard Mudgett)

    * Address some improper sql statements in res_odbc that would cause an update to fail on realtime peers due to trying to set as "(NULL)" rather than an actual NULL.
    (Closes issue ASTERISK-17791. Reported by marcelloceschia. Patched by Tilghman Lesher)

    * Resolve issue where 403 Forbidden would always be sent maximum number of times regardless to receipt of ACK.
    (Patched by Richard Mudgett)

    * Resolve issue where if a call to MeetMe includes both the dynamic(D) and always request PIN(P) options, MeetMe will ask for the PIN two times: once for creating the conference and once for entering the conference.
    (Patched by Kinsey Moore)

    * Fix New Zealand indications profile based on http://www.telepermit.co.nz/TNA102.pdf
    (Closes issue ASTERISK-16263. Reported, Patched by richardf)

    * Segfault in shell_helper in func_shell.c
    (Closes issue ASTERISK-18109. Reported by Michael Myles, patched by Richard Mudgett)

    For a full list of changes in this release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.6.0

    Thank you for your continued support of Asterisk!



    Matt Riddell at Astricon this October
    Click to view a printable version Tue, 30 Aug 2011 17:03:17 -0300

    I will be speaking at Astricon this year in Denver, Colorado.

    While I've spoken at Astricon a couple of times in the past, this will be the first time I've spoken at Astricon in the states.

    It would be great to meet up with of many of you as possible, so head along and let me know if you're going.

    I will be speaking at 10:00am on the Wednesday (October 26th) about interfacing with the Asterisk Manager. How to do it, things to watch out for, using different languages etc.

    As we approach the date for Astricon I'll provide additional information and a run down of some of the things that will be happening. If you've been reading the Daily Asterisk News for a while, you may remember the Madrid Astricon where I reported to the news on a regular basis about what was going on. I'll do the same again this year.

    So, if you're exhibiting and you'd like me to come and have a look at your products and maybe post some photos and a writeup etc, just flick me a note.

    Anyway, Digium has just posted a note this morning about the conference:

    CRN proclaims, "Asterisk 10 may be one of the biggest IT developments of the year." Learn what Asterisk 10 can do for you. Join Asterisk developers, integrators and enthusiasts at Astricon, the Asterisk User Conference & Expo. This year’s annual event is in Denver, Colorado, October 25 – 27th.

    Highlights of the three-day conference include:

    Asterisk 10 Symposium
    Keynote addresses by two industry leaders – Marten Mickos, CEO of Eucalyptus Systems and Jack Waters, CTO of Level 3 Communications.
    Multiple Conference Tracks featuring Cloud, Enterprise/Large Scale, Innovation and Tutorial/Development
    The ability to network with a broad array of Asterisk enthusiasts.

    Register today and see for yourself why CRN thinks Asterisk 10 is big news this year!

    October 25-27, 2011
    Westin Westminster Resort
    Denver, CO - USA



    Scheduled Maintenance for Asterisk Project community services
    Click to view a printable version Mon, 29 Aug 2011 20:00:35 -0300

    On Thursday, September 1st, 2011, the Asterisk community services listed below will be undergoing maintenance (power distribution upgrades in the cabinet where the servers are located). The services will be shut down at approximately 11:30 PM CDT, and will return no later than 6:00 AM CDT on September 2nd. We apologize in advance for any inconvenience this may cause.

    The affected services are:

    downloads.digium.com
    downloads.asterisk.org
    bamboo.asterisk.org
    git.asterisk.org
    code.asterisk.org
    packages.asterisk.org
    svn.asterisk.org
    issues.asterisk.org
    reviewboard.asterisk.org
    svnview.asterisk.org
    wiki.asterisk.org



    Thunderbird extension using AMI to dial
    Click to view a printable version Thu, 25 Aug 2011 19:37:18 -0300

    Chris Hastie has posted a note about a click to dial extension he has created for Thunderbird.

    Hi

    I've just added direct support for AMI to a forthcoming version of TBDialOut, a Thunderbird extension for dialling direct from Thunderbird's address book. If anyone fancies testing it I'd be grateful for any feedback. If you feel like casting a critical eye over the code, or doing some translating, even better.

    AMI support is available in TBDialOut 1.7.0pre1, which can be found either at http://www.oak wood.co.uk/tbdialout/ or from the 'Development channel' at the bottom of the page at https://addons.mozilla.org/en-US/thunderbird/addon/tbdialout/

    Thanks for your help

    Chris



    Asterisk and Google
    Click to view a printable version Thu, 25 Aug 2011 19:33:30 -0300

    Malcolm Davenport has posted a request on behalf of Digium for someone to help maintain the Google channel driver when Google makes changes:

    Howdy,

    Are you a developer, skilled in the ways of Asterisk, who:
    A) is frustrated by the uncertainty of Google Talk and Asterisk working on any given day?
    or
    B) wants free stuff?
    or
    C) both of the above?

    Then keep reading!

    The Asterisk community needs your help. We need someone to take over maintenance of the Google channel driver in Asterisk (chan_gtalk). Someone who can fix it, when Google makes changes. Someone who can make it more resistant to breaking, when Google makes changes. Someone who can give it the love that it needs, when Google makes changes. And, if we're all lucky, someone who can consider sprucing it up so that it takes advantage of libjingle, by merging chan_gtalk and chan_jingle into one awesome channel driver.

    Are you that person?

    If so, head over to this wiki page:
    https://wiki.asterisk.org/wiki/display/AST/Help+Maintain+Google+Talk+and+Voice

    and add a comment detailing your Asterisk development background - additional experience in Jabber/Jingle/XMPP is a plus.

    The free stuff?

    If you're nominated to maintain the Google channel driver, Digium will provide you with a free pass (you're on your own for hotel, food, travel, etc.) to Astricon 2011. Digium will also toss in some swag: t-shirts, backpack, stickers, etc.

    More importantly, you'll probably earn the admiration of thousands. Literally.

    Who wants to help?


    --

    Malcolm Davenport
    Digium, Inc. | Senior Product Manager



    Asterisk additions to support Asterisk Test Suite for 1.8
    Click to view a printable version Mon, 22 Aug 2011 19:00:08 -0300

    Matt Jordan has posted a note about some changes to the test framework and posing a question as to whether it should be included in Asterisk 1.8:

    Hello -

    As a part of enhancing the Asterisk Test Suite, we've come up with a mechanism in Asterisk to notify tests through AMI of events that occur in an application. The need for this came from two observations:

    1. Some tests may report false errors due to their inherent timing driven nature. For example, currently, in an application that has a built in playback of an audio file, the test must use Wait(n) or some other sleep mechanism and attempt to wait until the voice prompt is finished before taking action. When a test has two Asterisk instances, running on a single core machine, the tight timing constraints imposed by this scenario can mean that the Asterisk instances will become out of sync with what the test developer expected, and the test can arbitrarily fail. This situation arises often in those applications that have built in menus whose prompts can change based on the state of the application, such as app_voicemail's VoiceMailMain application.

    2. Applications whose success is a hangup result can also be difficult to verify. While UserEvent can be used in a dialplan to check certain return paths (thus determining whether an application definitely failed), it becomes difficult to say whether or not an application succeeds. For example, Dial has multiple success paths where the caller is hung up on; however, the hang up could also be an error in the dialplan. The only verification mechanism is by inspection.

    A new AMI event, TestEvent, has been added that allows applications to send information to the test suite during test execution. Hence, a test can wait to be notified of an event completing before taking additional actions. The TestEvent, much like the UserEvent, allows arbitrary data to be sent to the receiving party. This allows a test developer to handle most conceivable situations that Asterisk applications will throw at them.

    This patch will be going into 10 and trunk. However, we would also like to apply this patch to the 1.8 branch, to assist us in better supporting and testing the Asterisk branches currently being maintained. Our concern with applying this to 1.8 is that the TestEvent will alter existing applications. We've taken steps to prevent this from having any impact by only enabling the TestEvent if the TEST_FRAMEWORK compiler flag (which can only be enabled in dev-mode) is turned on. By default, all statements related to the TestEvent are macro'd out if the TEST_FRAMEWORK compiler flag does not exist.

    Normally we do not apply new features to older branches; however, as this can greatly help us in testing and maintaining this branch, we'd like to make an exception. Please let us know if you feel there are any issues with this feature going into the 1.8 branch.

    Thanks,

    Matt Jordan

    --
    Matthew Jordan
    Digium, Inc. | Software Developer



    rana-dtmf-rtp-duration - adventures in DTMF/RTP
    Click to view a printable version Sun, 21 Aug 2011 19:52:02 -0300

    Olle has posted details of some work he is doing on DTMF:

    Just to update you on my work with DTMF:

    This all started when we noticed that the duration of DTMF on the incoming stream compared with what we sent was radically different. DTMF in RTP consist of a start packet, updates (continuation) and an end packet. The end packet contains the final duration.

    While working on this, I noticed DTMF delays in the feature code, something that I tried to solve and Terry added a silence generator to and committed, so that's no longer the case. If Asterisk listens to DTMF and the tone is not a feature code, we're no longer waiting for the END packet to start sending out.

    For longer tones, we receive DTMF on one RTP channel and send AST_CONTROL messages to the outbound channel, where we play them according to the RTP packet stream there. The updates where previously silently ignored, which caused issues. I have added an AST_CONTROL message with duration updates, so that we can play out properly as we receive incoming RTP packets.

    That fixed a lot of issues. We still run into issues, partly due to feature code and partly because of the RTP stack architecture, where the outbound playout is often delayed compared with the inbound DTMF. Yesterday I discovered that people may come up with the idea to start sending a new DTMF tone on the incoming stream, while we're still playing the outbound DTMF. This causes a lot of interesting issues, the result being that we jump to the new one in the middle of the old one without sending any END messages.

    For those cases, I will have to create a DTMF queue in the outbound channel, consisting of the END frames with a duration so I can start the playout immediately after the previous tone.

    Hopefully the end result of this work will be:

    - more correct duration going through the Asterisk PBX bridge on rtp2rtp calls
    - no dropped DTMF tones when they are played quickly

    The branch I'm working on is currently for 1.6.0, which is what's used in production at the customer site. It will be upgraded to 1.8 and trunk versions as soon as we have tested.

    Any feedback is of course, as always, welcome.

    Regards,
    /Olle



    espeak module for Asterisk
    Click to view a printable version Sun, 21 Aug 2011 19:41:59 -0300

    Lefteris Zafiris has posted details of a new version of the app_espeak application for Asterisk - another speech synthesiser:

    Version 2.0 of app_espeak just got released.

    eSpeak For Asterisk provides the "Espeak" dialplan application, which allows you to use the Espeak speech synthesizer with Asterisk.

    It supports the following languages:
    Afrikaans, Albanian, Armenian,Cantonese, Catalan, Croatian, Czech, Danish, Dutch, English, Esperanto, Estonian, Finnish, French, Georgian, German, Greek, Hindi, Hungarian, Icelandic, Indonesian, Italian, Kannada, Kurdish, Latvian, Lojban, Macedonian, Malayalam, Mandarin, Norwegian, Polish, Portuguese, Romanian, Russian, Serbian, Slovak, Spanish, Swahili, Swedish, Tamil, Turkish, Vietnamese, Welsh.

    It supports 8kHz and 16kHz sample rates to provide the best possible sound quality along with the use of wideband codecs. Works with asterisk 1.6 , 1.8 , 10.

    http://zaf.github.com/Asterisk-eSpeak/

    ----------------
    Lefteris Zafiris



    Flite module for asterisk
    Click to view a printable version Sun, 21 Aug 2011 19:28:19 -0300

    Lefteris Zafiris has posted details of the latest release of app_flite - an application to make use of the text to speech capabilities of FestivalLite:

    Version 2.0 of app_flite just got released.

    Flite For Asterisk provides the "Flite" dialplan application, which allows you to use the Flite TTS Engine with Asterisk. It supports 8kHz and 16kHz sample rates to provide the best possible sound quality along with the use of wideband codecs. It works with asterisk 1.6 , 1.8 , 10.

    http://zaf.github.com/Asterisk-Flite/

    ----------------
    Lefteris Zafiris



    Developer Information Update
    Click to view a printable version Mon, 15 Aug 2011 18:26:02 -0300

    Kevin has posted a note asking developers with commit privileges to create a personal space on the Asterisk wiki:

    Recently, we've become a bit lax in informing our community when new developers join the team (both on the Digium team and from the community itself). In an effort to improve the situation, I'm asking that *everyone* who has commit access to any part of the Asterisk or Asterisk SCF project repositories head over to https://wiki.asterisk.org. Once you are there, if you don't already have a 'personal space', create one (and choose the option to make it publicly visible, not private). If the option to create a personal space isn't available to you, find me on IRC or email and I'll make sure your account has the correct permissions.

    In the main page of your personal space, please write a paragraph or two about your experience, what you have done/are doing with Asterisk or Asterisk SCF, and anything else you think might be relevant or useful. Basically, introduce yourself to the community!

    Please include on this page the nicks/handles you use on various services: your committer username (for Subversion), your IRC nick, etc.

    Thanks in advance; hopefully this will help get the community of developers more familiar with each other. In the future, when someone is granted commit access to one of the projects, we'll ensure they fill out their personal space 'introduction' page first, and then send an announcement to this list that they've joined the team.

    --
    Kevin P. Fleming
    Digium, Inc. | Director of Software Technologies



    Call for testers for call parking feature
    Click to view a printable version Mon, 15 Aug 2011 18:10:11 -0300

    Richard Mudgett has posted a note asking people to test some fixes for call parking:

    Greetings.

    There are currently a bunch of issues reported against call parking. These issues are listed with a brief description in the patch I have up on review board to resolve them and other parking issues I found. A team branch from the v1.8 branch is also available to checkout until the patch is merged into the v1.8 and newer branches.

    The Asterisk Development Team would like some testers to check out the patch and to report on this thread your findings. We would like to compile some real world test scenarios for a wiki page to prevent call parking from getting this badly broken in the future.

    Thank you

    Richard



    DAHDI-Linux 2.5.0 and DAHDI-Tools 2.5.0 Released
    Click to view a printable version Mon, 08 Aug 2011 23:16:32 -0300

    The Asterisk Development Team is pleased to announce the release of DAHDI-Linux and DAHDI-Tools version 2.5.0

    DAHDI-Linux 2.5.0, DAHDI-Tools 2.5.0 and DAHDI-Linux-Complete 2.5.0+2.5.0 are available for immediate download at:

    http://downloads.asterisk.org/pub/telephony/dahdi-linux
    http://downloads.asterisk.org/pub/telephony/dahdi-tools
    http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete

    2.5.0 is a feature release and builds against kernels from 2.6.9 up to and including 3.0. Some of the more notable changes from the 2.4.0 release are:

    General DAHDI Changes:

    • Software and Hardware (if available) echo cancellation can be mixed on a line if desired. This also allows hardware echo cancellation to be disabled without reloading the drivers.
    • Channels can be configured to generate channel events when there are data under runs or over runs. Can be used when tracking down the source of data skips.
    • Debug flag can be set to show RBS bit changes in kernel log.
    • Much refactoring in in dahdi-base to enable tighter integration with udev/sysfs in upcoming releases.

    Updated Drivers:

    • wctdm24xxp, wcte12xp: Added support for VPMOCT032 hardware echo cancellation module.
    • wctdm24xxp: Fix regression with forward disconnect.
    • wctdm24xxp, wcte12xp: Decrease the amount of time spent in the interrupt handler.
    • wctdm24xxp: Added fastpickup module parameter to help with type-II caller id detection.
    • wcte12xp: Properly recognizes loopup / loopdown signals.
    • wct4xxp: Reduced kernel memory footprint and stack usage.
    • wct4xxp: Added module parameter to allow the rotary switch settings to be ignored at module load time.
    • wcb4xxp: Support for beroNet BN4S0e PCI Express card.
    • xpp: support for a hardware echo canceller. Requires "octasic" support in dahdi-tools (as of 2.5) and the firmware from http://updates.xorcom.com/astribank/hwec/
    • xpd_fxs - added parameter ring_trapez for a trapezoid (stronger) ring.

    Changes to DAHDI-tools:

    • xpp: low-level USB tools can now use a relative -D mmm/nnn instead of a full -D /dev/bus/usb/mmm/nnn .
    • xpp: Octasic support: oct612x code included to allow loading the Astribank hardware echo canceller module's firmware.
    • xpp: xpp_order doesn't work when more then 9 ABs are listed (xpp/perl_modules/Dahdi/Xpp.pm)

    For a full list of changes in these releases, please see the ChangeLogs at
    http://svn.asterisk.org/svn/dahdi/linux/tags/2.5.0/ChangeLog and
    http://svn.asterisk.org/svn/dahdi/tools/tags/2.5.0/ChangeLog

    Issues found in these release candidates can be reported in the DAHDI-Linux or DAHDI-tools project at https://issues.asterisk.org/jira

    Thank you for your continued support of Asterisk!



    Asterisk 10 Beta 1
    Click to view a printable version Sun, 24 Jul 2011 21:14:11 -0300

    Malcolm Davenport has posted some information about the new version of Asterisk.

    Excerpt from his post:

    Howdy,

    On the heels of Kevin Fleming’s announcement yesterday discussing the changes in the Asterisk versioning scheme, we’d like to formally announce that Asterisk 10, Beta 1 is now available for community testing. Asterisk 10, a Standard Support release, will be the next major release of Asterisk and follows the release of Asterisk 1.8 LTS, a Long Term Support release. For more information on the different releases of Asterisk, check out the Asterisk Versions page on the Wiki.

    Let’s talk about some of its new capabilities.

    A major focus of the Asterisk 10 development cycle was Asterisk’s support for media types. In versions of Asterisk 1.8 and prior, Asterisk supported a rather limited number of codecs due to some architectural limitations. Plumbing was ripped out, kitchens were remodeled, girders were swapped, and Asterisk 10 now has a media architecture that’s capable of handling both a nearly unlimited number of codecs as well as codecs with more complex parameters. What does this mean for users? First, it means that Asterisk now comes with some additional codecs, including the 32kHz variant of the Speex codec (previous versions of Asterisk only supported the 8kHz or 16kHz variants), Skype’s Superwideband SILK codec, and pass-through support for the 44.1kHz and 48kHz variants of the CELT format.

    Read the rest...



    Asterisk 10.0.0 Beta 1 Now Available
    Click to view a printable version Sun, 24 Jul 2011 20:32:04 -0300

    The Asterisk Development Team is pleased to announce the first beta release of Asterisk 10.0.0-beta1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

    With the release of the Asterisk 10 branch, the preceding '1.' has been removed from the version number per the blog post available at http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/

    All interested users of Asterisk are encouraged to participate in the Asterisk 10 testing process. Please report any issues found to the issue tracker, https://issues.asterisk.org/jira. It is also very useful to see successful test reports. Please post those to the asterisk-dev mailing list.

    All Asterisk users are invited to participate in the #asterisk-testing channel on IRC to work together in testing the many parts of Asterisk. Additionally users can make use of the RPM and DEB packages now being built for all Asterisk releases. More information available at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages

    Asterisk 10 is the next major release series of Asterisk. It will be a Standard support release, similar to Asterisk 1.6.2. For more information about support time lines for Asterisk releases, see the Asterisk versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

    A short list of included features includes:

    • T.38 gateway functionality has been added to res_fax.
    • Protocol independent out-of-call messaging support. Text messages not associated with an active call can now be routed through the Asterisk dialplan. SIP and XMPP are supported so far.
    • New highly optimized and customizable ConfBridge application capable of mixing audio at sample rates ranging from 8kHz-192kHz
    • Addition of video_mode option in confbridge.conf to provide basic video conferencing in the ConfBridge() dialplan application.
    • Support for defining hints has been added to pbx_lua.
    • Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB).
    • Much, much more!

    A full list of new features can be found in the CHANGES file.

    For a full list of changes in the current release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.0.0-beta1

    Thank you for your continued support of Asterisk!



    The Evolution of Asterisk (or: How We Arrived at Asterisk 10)
    Click to view a printable version Thu, 21 Jul 2011 14:48:44 -0300

    Kevin Fleming has posted an email explaining the change from Asterisk 1.10 to Asterisk 10 and explaining why there will be no Asterisk 2.x:

    We are fast approaching the seven-year anniversary of the release of Asterisk 1.0.0, which occurred at the first AstriCon in September, 2004. If you look back a little further, there were various "0.x" releases made as early as December of 1999... my, how time has flown!

    We’ve had quite a few ‘major’ releases of Asterisk since then, including 1.2, 1.4, and most recently, 1.8. Each of these releases has included significant changes, and sometimes architecture-improving changes. Each of them has also included substantial new functionality for Asterisk users. Along the way, we’ve been asked by many people in the community when we are going to start working on (or release) "Asterisk 2.0." Typically, we’ve responded by saying that will not happen until we can really justify such a significant change in the version number. Many open source projects have gone through similar progressions, and quite a number of them have in fact undergone complete (or nearly complete) rewrites resulting in new ‘major’ versions.

    The Asterisk project, however, has tried to avoid that level of disruption to its users. Instead we’ve focused on attempting to provide as much backwards compatibility between major releases as we could. As a result, each time we’ve released a new major version, the decision has been made that "No, this isn’t Asterisk 2.0," and we’ve continued with the version numbering scheme that Mark Spencer started all those years ago.

    Over the past few months though, as we’ve approached the first beta release of the next major version of Asterisk, we’ve been having a somewhat unexpected conversation: about just how different this release is going to be from the releases that most users in the community are using on their production Asterisk systems (primarily Asterisk 1.4, although there are still a lot of 1.2 users as well).

    In fact, even though it’s been an evolutionary process, not a revolutionary one, the next major Asterisk release really will be substantially different from Asterisk 1.4 in some very noticeable ways: wideband conferencing support, basic video conferencing support, support for a number of additional VoIP technologies, full-fledged FAX support, and many others.

    That has raised the question: Is this Asterisk 2.0? If not, will there ever be an Asterisk 2.0? After quite a lot of discussion, we came to the conclusion that this is not Asterisk 2.0, but that it’s also quite unlikely that there ever will be such a release; it wouldn’t be in the community’s best interests to release something that is fundamentally different (and not compatible) but still call it '‘Asterisk'.’ That then leaves the question we’ve been asked by many people: If there’s never going to be an Asterisk 2.0, why continue to call these releases "1.x"? What does the "1" mean, if it’s never going to change?

    The conclusion that we’ve reached, and that we hope you’ll agree with, is that Asterisk is always going to be Asterisk, and that you don’t need a "1." prefix on the version number to be able to identify it. So, starting with the next major release, we’re going to drop the "1." completely. The next major release, which was going to be Asterisk 1.10, will now be just "Asterisk 10" and subsequent major releases will be "Asterisk 11", "Asterisk 12", and so forth.

    We’ll continue with our plan to have both standard and long-term support releases of Asterisk, and we’ll update the Asterisk Project Wiki with this information as soon as the first Asterisk 10 beta goes out. In fact, this should occur very soon.

    As always, thanks to everyone for their continued support of Asterisk. That especially includes the developer community, the people that find and report issues, the people that help test patches and the people that devote their time to answering questions on IRC channels, the mailing lists and the forums. We hope to see everyone trying out the forthcoming beta, and we look forward to seeing you all at AstriCon 2011!

    --
    Kevin P. Fleming
    Digium, Inc. | Director of Software Technologies



    Russell Bryant: Taking On New Challenges
    Click to view a printable version Mon, 18 Jul 2011 20:15:05 -0300

    Russell has posted on his blog explaining about his leaving Digium:

    I began working on the Asterisk project in 2004. My work on Asterisk has led to an exciting career in open source software engineering. At the end of July 2011, I will be leaving Digium to take on some new challenges. Specifically, I will be joining the Cloud Infrastructure team at Red Hat as a Principal Software Engineer where I will be working on projects related to clustering, high availability, and systems management. Additionally, I will be moving back to Charleston, SC to be closer to my family.

    While I will no longer be working with Asterisk full time, I still plan to participate in the open source community. I am excited to watch both Asterisk and Asterisk SCF continue to evolve and grow. The engineering team at Digium, as well as the global Asterisk development community are as strong as they have ever been and will continue to accomplish big things.

    I have met many great people from all over the world in my time with Asterisk. Thank you all for making the past seven years so memorable.

    Best Regards,


    Russell Bryant



    Russell Bryant: Thank You for Seven Years of Excellence
    Click to view a printable version Mon, 18 Jul 2011 20:06:26 -0300

    Bryan Johns has posted a blog entry thanking Russell for seven years working with Digium and to wish him luck in his new job at Red Hat.

    I can't believe he's leaving! Russell has always been someone I associate with both Digium and Asterisk and his hard work and diplomacy will be sorely missed! It sometimes feels like the majority of the Digium staff have turned over since I started the Daily Asterisk News in 2004!

    I hope that you continue to be the calming influence in the Asterisk project!

    On to the blog post.

    Excerpt from the blog post:

    More than seven years ago, Russell Bryant joined the Asterisk development team as a student at Clemson University. Over his time with the Asterisk project, Russell has been instrumental in the creation, growth and management of the world’s most popular open source communications platform. However, seven years is a long time to work within the confines of a single project and a desire for new experiences coupled with a desire to be closer to family has brought Russell to the decision to leave Digium effective July 29th, 2011.

    Russell will join the Cloud Infrastructure Development Team at Red Hat. While Russell will no longer be an employee of Digium, he will continue to participate in the growth of Asterisk and Asterisk SCF as a member of our vibrant developer community. We look forward to building synergies with the Red Hat team through this new relationship and working to enhance the virtualized performance of Asterisk.

    Read More...



    AstLinux 0.7.9 Release
    Click to view a printable version Sun, 17 Jul 2011 19:39:44 -0300

    The AstLinux Team would like to announce the immediate availability of the 0.7.9 release. This release includes either Asterisk 1.4.42 or Asterisk 1.8.4.4. All current users are encouraged to upgrade to this release to take advantage of bug fixes and other updates to Asterisk.

    A full changelog is available at http://www.astlinux.org

    Current users can upgrade from the web interface or from the commandline.

    >From the CLI:

    (Asterisk 1.4)
    upgrade-run-image check http://mirror.astlinux.org/firmware
    should report astlinux-0.7.9
    upgrade-run-image upgrade http://mirror.astlinux.org/firmware

    (Asterisk 1.8)
    upgrade-run-image check http://mirror.astlinux.org/ast18-firmware
    should report astlinux-0.7.9
    upgrade-run-image upgrade http://mirror.astlinux.org/ast18-firmware

    --
    The AstLinux Team



    Asterisk 1.10 branch created
    Click to view a printable version Wed, 13 Jul 2011 17:53:28 -0300

    At 8:34am New Zealand time this morning a branch named 1.10 was created.

    The commit message reads:

    Create branch for Asterisk 1.10

    You can find the branch here:

    http://svn.digium.com/svn/asterisk/branches/1.10/

    Update:

    The UPGRADE.txt document contains:

    ===========================================================
    ===
    === Information for upgrading between Asterisk versions
    ===
    === These files document all the changes that MUST be taken
    === into account when upgrading between the Asterisk
    === versions listed below. These changes may require that
    === you modify your configuration files, dialplan or (in
    === some cases) source code if you have your own Asterisk
    === modules or patches. These files also include advance
    === notice of any functionality that has been marked as
    === 'deprecated' and may be removed in a future release,
    === along with the suggested replacement functionality.
    ===
    === UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
    === UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
    === UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6
    === UPGRADE-1.8.txt -- Upgrade info for 1.6 to 1.8
    ===
    ===========================================================

    From 1.8 to 1.10:

    cel_pgsql:
    - This module now expects an 'extra' column in the database for data added using the CELGenUserEvent() application.

    ConfBridge
    - ConfBridge's dialplan arguments have changed and are not backwards compatible.

    File Interpreters
    - The format interpreter formats/format_sln16.c for the file extension '.sln16' has been removed. The '.sln16' file interpreter now exists in the formats/format_sln.c module along with new support for sln12, sln24, sln32, sln44, sln48, sln96, and sln192 file extensions.

    HTTP:
    - A bindaddr must be specified in order for the HTTP server to run. Previous versions would default to 0.0.0.0 if no bindaddr was specified.

    Gtalk:
    - The default value for 'context' and 'parkinglots' in gtalk.conf has been changed to 'default', previously they were empty.

    chan_dahdi:
    - The mohinterpret=passthrough setting is deprecated in favor of moh_signaling=notify.

    pbx_lua:
    - Execution no longer continues after applications that do dialplan jumps (such as app.goto). Now when an application such as app.goto() is called, control is returned back to the pbx engine and the current extension function stops executing.
    - the autoservice now defaults to being on by default
    - autoservice_start() and autoservice_start() no longer return a value.

    Queue:
    - Mark QUEUE_MEMBER_PENALTY Deprecated it never worked for realtime members
    - QUEUE_MEMBER is now R/W supporting setting paused, ignorebusy and penalty.

    Asterisk Database:
    - The internal Asterisk database has been switched from Berkeley DB 1.86 to SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3 utility in the UTILS section of menuselect. If an existing astdb is found and no astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will convert an existing astdb to the SQLite3 version automatically at runtime.




    Benchmarking AGI performance in C, PHP, and Perl
    Click to view a printable version Tue, 12 Jul 2011 01:17:53 -0300

    Steve Edwards has run some benchmarks comparing various languages for writing AGI scripts in:

    Many times, I've made the statement that you can execute hundreds of AGIs written in C in the time it takes to load an interpreter and parse a script written in PHP or Perl.

    Recently, a Doubting Thomas asked me to substantiate my claim.

    I suspect nobody has made the effort to implement an AGI of any reasonable size and function in multiple languages.

    I'm guessing it may not really be all that important and the results would be too task specific to be relevant.

    I suspect once an AGI is executing, the choice of source language is unimportant. I'll go out on a limb and say executing:

    select prompt_path from foo where bar;
    stream file prompt_path "1234*"
    stream file you_entered ""
    say digits selected ""

    will execute in effectively the same time regardless of source language. Waiting for Asterisk to play a file or for your database to return a row is beyond the scope of your AGI.

    It's what you do between your AGI and database calls that will determine how much your choice of source language will impact the total execution time.

    Unless you're doing a lot of stuff in between these API calls, the only place you can make an impact is getting your code into memory and ready to execute.

    I 'wrote' 2 different AGIs in C, PHP, and Perl.

    The first AGI, 'null-agi' reads the AGI environment variables from STDIN and exits. To me, this is the bare minimum a program can do and call itself an AGI. Each AGI was less than 10 lines.

    The second AGI, 'neutered-agi' is an AGI of 'production length' (around 1,600 lines) and supporting access to a MySQL database. The AGI is of 'production length' but still exits after reading the AGI environment variables because we are measuring program startup time.

    For both AGIs, the C implementation used an AGI library I developed way too many years ago. The PHP implementation used PHPAGI. The Perl implementation used Asterisk::AGI.

    The C version of neutered-agi was based on a 'voicemail-like' AGI I wrote many years ago that stored the user credentials and messages in MySQL.

    The PHP version of neutered-agi was based on dialparties.agi (nicked from PIAF). dialparties.agi is only about 800 lines long, so I 'doubled it' by copying and pasting it into the same source file. While dialparties.agi does not use MySQL, MySQL is available in PHP without including additional header or class files.

    The Perl version of neutered-agi was based on agi-VDAD_ALL_outbound.agi (nicked from Vicidial).

    I ran the tests in 3 different environments:

    +-------------------+--------+--------+----------+
    | CPU               | RAM    | CentOS | Asterisk |
    |-------------------+--------+--------+----------|
    | Geode 500MHz      | 256 MB |    4.9 |   1.2.37 |
    | Atom D525 1.80GHz | 4 GB   |    5.6 |  1.8.4.1 |
    | Xeon 3.40GHz      | 2 GB   |    4.8 |   1.2.40 |
    +-------------------+--------+--------+----------+



    No swapping occurred during the tests.

    My dialplan executed each AGI 1,000 times to make the cumulative execution time more measurable. I wrote the dialplan in both 'inline dialplan' and an AEL 'for' loop. The initial execution times were the same so the test runs were made with the AEL version because it is more manageable. (5 lines versus 1,000 lines.)

    Here's the results for executing each AGI 1,000 times on each host in seconds:

    Geode:

    +----------+----------+--------------+
    | language | null-agi | neutered-agi |
    |----------+----------+--------------|
    | C        |        6 |            6 |
    | PHP      |      116 |          160 |
    | Perl     |       99 |          639 |
    +----------+----------+--------------+



    Atom:

    +----------+----------+--------------+
    | language | null-agi | neutered-agi |
    |----------+----------+--------------|
    | C        |        6 |            6 |
    | PHP      |       52 |           65 |
    | Perl     |       38 |          197 |
    +----------+----------+--------------+



    Xeon:

    +----------+----------+--------------+
    | language | null-agi | neutered-agi |
    |----------+----------+--------------|
    | C        |        2 |            2 |
    | PHP      |       40 |           47 |
    | Perl     |       10 |          107 |
    +----------+----------+--------------+



    Summary:

    Geode - Perl / C: 106
    Atom - Perl / C: 33
    Xeon - Perl / C: 54

    The C null-agi AGI was statically linked. I didn't have all the libraries needed to statically link neutered-agi on these boxes, but the dynamically linked versions of null-agi and neutered-agi took the same time to execute (16 seconds on the Geode) so I'm assuming statically linked versions of null-agi and neutered-agi would also take the same time to execute. It also helps support my original statement :)

    I guessing the Perl version of neutered-agi took a big hit from having to load the database code as well as the AGI framework (use DBI; use Asterisk::AGI;) while PHP only had to load the AGI framework (require_once "phpagi.php";).

    I'll let you decide if the methodology is meaningful to your environment. I don't consider myself to be a PHP or Perl expert, so if I've made some colossal blunder in my methodology, please let me know.

    I'm guessing you'd have to resurrect a Soekris net4801 from the way-back-machine to substantiate my orignal claim of 'hundreds.' I'll have to remind myself to say 'dozens' from now on.

    (I don't have a 'thing' against Soekris -- I just bought 2 off Ebay to play with.)

    --
    Thanks in advance,

    Steve Edwards



    Asterisk 1.8.5.0 Now Available
    Click to view a printable version Mon, 11 Jul 2011 18:50:27 -0300

    The Asterisk Development Team announces the release of Asterisk 1.8.5.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

    The release of Asterisk 1.8.5.0 resolves several issues reported by the community and would have not been possible without your participation.
    Thank you!

    The following is a sample of the issues resolved in this release:

    • Fix Deadlock with attended transfer of SIP call
      (Closes issue #18837. Reported, patched by alecdavis. Tested by Irontec, ZX81, cmaj)
    • Fixes thread blocking issue in the sip TCP/TLS implementation.
      (Closes issue #18497. Reported by vois. Patched by dvossel. Tested by vois, rossbeer, kowalma, Freddi_Fonet)
    • Be more tolerant of what URI we accept for call completion PUBLISH requests.
      (Closes issue #18946. Reported by GeorgeKonopacki. Patched by mmichelson)
    • Fix a nasty chanspy bug which was causing a channel leak every time a spied on channel made a call.
      (Closes issue #18742. Reported by jkister. Tested by jcovert, jrose)
    • This patch fixes a bug with MeetMe behavior where the 'P' option for always prompting for a pin is ignored for the first caller.
      (Closes issue #18070. Reported by mav3rick. Patched by bbryant)
    • Fix issue where Asterisk does not hangup a channel after endpoint hangs up. If the call that the dialplan started an AGI script for is hungup while the AGI script is in the middle of a command then the AGI script is not notified of the hangup.
      (Closes issue #17954, #18492. Reported by mn3250, devmod. Patched by rmudgett)
    • Resolve issue where leaving a voicemail, the MWI message is never sent. The same thing happens when checking a voicemail and marking it as read.
      (Closes issue ASTERISK-18002. Reported by Leif Madsen. Resolved by Richard Mudgett)
    • Resolve issue where wait for leader with Music On Hold allows crosstalk between participants. Parenthesis in the wrong position. Regression from issue #14365 when expanding conference flags to use 64 bits.
      (Closes issue #18418. Reported by MrHanMan. Patched by rmudgett)

    For a full list of changes in this release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.5.0

    Thank you for your continued support of Asterisk!



    IANA Registration for Enumservice IAX
    Click to view a printable version Mon, 11 Jul 2011 18:41:32 -0300

    Klaus Darilion posted a link on the facebook page for the Daily Asterisk News regarding the registration of IAX for e.164:

    Here's the introduction from the document:

    The E.164 to Uniform Resource Identifiers (URIs) [RFC3986] Dynamic Delegation Discovery System (DDDS) Application (ENUM) [RFC6116] transforms E.164 [E164] numbers into URIs using the Domain Name System (DNS) [RFC1035].

    IAX (Inter-Asterisk eXchange) [RFC5456] is an "all-in-one" protocol for handling multimedia in IP networks. It combines both control and media services in the same protocol.

    This document registers an Enumservice for the IAX [RFC5456] protocol according to the guidelines given in [RFC6117].

    Read More...



    Asterisk SCF Steering Committee: Call For Participation
    Click to view a printable version Sun, 10 Jul 2011 22:00:56 -0300

    Kevin Fleming has posted a note asking people to become involved in the Asterisk SCF Steering Committee:

    (Apologies in advance for the length of this message; please don't stop reading after the first couple of paragraphs )

    A little over a year ago, when the Asterisk SCF project was just getting started, we invited interested community members to help us form a steering committee. The committee's purpose was to help shape the direction of Asterisk SCF development, in order to ensure that it would serve the needs of community members.

    The committee was formed and began having somewhat regular meetings (via conference call); the meetings are documented (and recordings posted) here:

    https://wiki.asterisk.org/wiki/display/TOP/Steering+Committee

    In the beginning, the committee's task was fairly easy to define: we at Digium had already decided on some basic pieces of technology, and some design criteria, but we needed help figuring out exactly how we should assemble these pieces. In other words, we knew how to make things that people wanted to use, but we needed people to tell us *what* to make.

    Some of us at Digium then spent time interviewing various people and companies, in order to determine what would be most useful to them in an 'Asterisk SCF 1.0' release. We combined the results of those interviews into a candidate feature list, and asked the committee to help us determine whether that was the *right* list of features to build first. The group was in agreement, so development began in earnest.

    Now it's about seven months later, development has been continuing at a frenetic pace... but the committee has become disengaged. It's been difficult to get a reasonable number of members to participate in the (now monthly) conference calls, and there have been occasions where no community members have participated at all. It's understandable to have some members occasionally unable to join; everyone has regular jobs, travel sometimes interferes, etc., but we'd hoped to have at least two or three of the six community members be able to join each call.

    In just a couple of months, we intend to be making the first beta releases of Asterisk SCF available, and when that happens, it will be quite important that we have the community's involvement to ensure:

    • that we continue heading down the right path
    • that we're addressing the right problems and feature requests that arise once the beta testing process has started
    • that we're providing the right documentation, examples and other assistance to get interested users able to actually use Asterisk SCF in their networks

    So, I'm posting this call to see if we can get things moving in the right direction again; if you are interested in helping to shape the future of Asterisk SCF development, please speak up! It's not necessary to commit multiple hours per week to this effort, but it's important that those who volunteer to be a member of the steering committee be able to stay engaged with the project, participate on the conference calls, and be able to spend at least an hour reviewing documentation and other details a few days before the conference call in order to facilitate discussion and decision-making.

    For this to be a true Steering Committee, those of us doing the bulk of the Asterisk SCF development are willing to commit to documenting our research, describing that to the committee and working to reach a consensus on how we should proceed... but only if the committee itself is willing to commit to being a full participant in the process. Without that level of commitment, development grinds to a halt, or decisions have to be made without community interaction; neither of those are what we would wish to be the case.

    So here's the call to action: join the committee, and help us make Asterisk SCF into the best framework for building communications applications that you can envision!

    --
    Kevin P. Fleming
    Digium, Inc. | Director of Software Technologies



    Asterisk Trunk moves from Berkley DB to SQLite 3.
    Click to view a printable version Wed, 06 Jul 2011 18:16:15 -0300

    While reading through the commit logs this morning I noticed that the planned change to SQLite 3 as the backend database has taken place.

    Here's the commit message:

    There were some bugs in the very ancient version of Berkley DB that Asterisk used. Instead of spending the time tracking down the bugs in the Berekeley code we move to the much better documented SQLite 3.

    Conversion of the old astdb happens at runtime by running the included astdb2sqlite3 utility. The ast_db API with SQLite 3 backend should behave identically to the old Berkeley backend, but in the future we could offer a much more robust interface.

    We do not include the SQLite 3 library in the source tree, but instead rely upon the distribution-provided libraries. SQLite is so ubiquitous that this should not place undue burden on administrators.

    Here's the entry from the changes file:

    Asterisk Database
    -----------------
    * The internal Asterisk database has been switched from Berkeley DB 1.86 to SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3 utility in the UTILS section of menuselect. If an existing astdb is found and no astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will convert an existing astdb to the SQLite3 version automatically at runtime.

    The change was committed in revision 326589




    libpri 1.4.12 Now Available
    Click to view a printable version Wed, 06 Jul 2011 18:03:50 -0300

    The Asterisk Development Team announces the release of libpri version 1.4.12. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri/

    The following are some of the issues resolved in this release:

    * Add call transfer exchange of subaddresses support and fix PTMP call transfer signaling.

    * Invalid PTMP redirecting signaling as TE towards NT.

    * Add Q931_IE_TIME_DATE to CONNECT message when in network mode.
    (issue #18047 (JIRA PRI-114). Reported by: wuwu. Patched by rmudgett)

    * Swap of master/slave in pri_enslave() incorrect.
    (issue #18769 (JIRA PRI-120). Reported by: jcollie. Patched by jcollie)

    * Fix I-frame retransmission quirks.

    * Crash if NFAS swaps D channels on a call with an active timer.

    * DMS-100 not receiving caller name anymore.
    (issue #18822 (JIRA PRI-121). Reported by: cmorford. Patched by rmudgett)

    * B channel lost by incoming call in BRI NT PTMP mode.

    * Implement the mandatory T312 timer for NT PTMP broadcast SETUP calls.

    This release contains several new features, among them:

    1.) ETSI and Q.SIG Call Completion Supplementary Service (CCSS) support
    2.) ETSI Advice Of Charge (AOC) support
    3.) ETSI Explicit Call Transfer (ECT) support
    4.) ETSI Call Waiting support for ISDN phones
    5.) ETSI Malicious Call ID support
    6.) Add Display IE text handling options.

    For a full list of changes in this release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/libpri/releases/ChangeLog-1.4.12

    Thank you for your continued support of Asterisk!



    Asterisk 1.8.5-rc1 Now Available
    Click to view a printable version Thu, 30 Jun 2011 19:20:37 -0300

    The Asterisk Development Team has announced the first release candidate of Asterisk 1.8.5. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

    The release of Asterisk 1.8.5-rc1 resolves several issues reported by the community and would have not been possible without your participation.
    Thank you!

    The following is a sample of the issues resolved in this release candidate:

    * Fix Deadlock with attended transfer of SIP call
    (Closes issue #18837. Reported, patched by alecdavis. Tested by Irontec, ZX81, cmaj)

    * Fixes thread blocking issue in the sip TCP/TLS implementation.
    (Closes issue #18497. Reported by vois. Patched by dvossel. Tested by vois, rossbeer, kowalma, Freddi_Fonet)

    * Be more tolerant of what URI we accept for call completion PUBLISH requests.
    (Closes issue #18946. Reported by GeorgeKonopacki. Patched by mmichelson)

    * Fix a nasty chanspy bug which was causing a channel leak every time a spied on channel made a call.
    (Closes issue #18742. Reported by jkister. Tested by jcovert, jrose)

    * This patch fixes a bug with MeetMe behavior where the 'P' option for always prompting for a pin is ignored for the first caller.
    (Closes issue #18070. Reported by mav3rick. Patched by bbryant)

    * Fix issue where Asterisk does not hangup a channel after endpoint hangs up. If the call that the dialplan started an AGI script for is hungup while the AGI script is in the middle of a command then the AGI script is not notified of the hangup.
    (Closes issue #17954, #18492. Reported by mn3250, devmod. Patched by rmudgett)

    * Resolve issue where leaving a voicemail, the MWI message is never sent. The same thing happens when checking a voicemail and marking it as read.
    (Closes issue ASTERISK-18002. Reported by Leif Madsen. Resolved by Richard Mudgett)

    * Resolve issue where wait for leader with Music On Hold allows crosstalk between participants. Parenthesis in the wrong position. Regression from issue #14365 when expanding conference flags to use 64 bits.
    (Closes issue #18418. Reported by MrHanMan. Patched by rmudgett)

    * Fix timerfd locking issue.
    (Closes ASTERISK-17867, ASTERISK-17415. Patched by kobaz)

    For a full list of changes in this release candidate, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.5-rc1

    Thank you for your continued support of Asterisk!



    Asterisk 1.4.42 Now Available (Final Maintenance Release)
    Click to view a printable version Thu, 30 Jun 2011 19:16:04 -0300

    The Asterisk Development Team has announced the final maintenance release of Asterisk, version 1.4.42. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

    Please note that Asterisk 1.4.42 is the final maintenance release from the 1.4 branch. Support for security related issues will continue until April 21, 2012. For more information about support of the various Asterisk branches, see https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

    The release of Asterisk 1.4.42 resolves several issues reported by the community and would have not been possible without your participation.
    Thank you!

    The following is a sample of the issues resolved in this release:

    * Resolve regression with ring groups in the Dial() application
    (Closes issue ASTERISK-17874. Reported by mspuhler. Patched by elguero)
    * Resolve deadlock when using tab completion on the 'meetme kick' CLI command when an invalid (non-existent) conference room is specified.
    (Closes issue ASTERISK-17771. Reported, patched by zvision)
    * Resolve issue where voice frames could be dropped when checking for T.38 during early media.
    (Closes issue ASTERISK-17705. Reported, patched by oej)
    * Resolve issue where DYNAMIC_FEATURES would not activate after a recent DTMF fix.
    (Closes issue ASTERISK-17914. Reported by vrban. Patched by twilson)

    Additionally security announcements AST-2011-010, and AST-2011-011 have been resolved in this release.

    For a full list of changes in this release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.42

    Thank you for your continued support of Asterisk!



    Asterisk 1.4.41.2, 1.6.2.18.2, and 1.8.4.4 Now Available (Security Releases)
    Click to view a printable version Tue, 28 Jun 2011 18:57:28 -0300

    The Asterisk Development Team has announced the release of Asterisk versions 1.4.41.2, 1.6.2.18.2, and 1.8.4.4, which are security releases.

    These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases

    The release of Asterisk 1.4.41.2, 1.6.2.18.2, and 1.8.4.4 resolves the following issue:

    AST-2011-011: Asterisk may respond differently to SIP requests from an invalid SIP user than it does to a user configured on the system, even when the alwaysauthreject option is set in the configuration. This can leak information about what SIP users are valid on the Asterisk system.

    For more information about the details of this vulnerability, please read the security advisory AST-2011-011, which was released at the same time as this announcement.

    For a full list of changes in the current releases, please see the ChangeLog:

    ChangeLog-1.4.41.2
    ChangeLog-1.6.2.18.2
    ChangeLog-1.8.4.4

    Security advisory AST-2011-011 is available at:

    http://downloads.asterisk.org/pub/security/AST-2011-011.pdf

    Thank you for your continued support of Asterisk!



    Packaging and AsteriskSCF
    Click to view a printable version Sun, 26 Jun 2011 18:12:40 -0300

    Paul Belanger has been working on putting together a package for AsteriskSCF for Debian unstable:

    Greetings!

    I had some free time this weekend, so I decided to look and see what was required to package some of the dependencies use by AsteriskSCF.

    I decided to start with zeroc-ice, since this package already exists upstream in Debian unstable. Right now Debian support 3.4.1, talking to the maintainer of the package 3.4.2 should be release this week. So I spend a few hour on Friday night / Saturday morning, packaging 3.4.2.

    Now that zeroc-ice 3.4.2 was packaged, I then create a tarball from the Digium version of ice in git and started work to package it. Again, after a few hours of hacking I've successfully created zeroc-ice packages from the version in the Digium git repo. :)

    git://git.asterisk.org/asterisk-scf/release/ice

    There we're some differences in java/build.xml between the two versions, however I've patched the Digium version and reverted some changes to icegridadmin.

    ATM I've only created packages for Debian unstable, it should not be hard to port them to Ubuntu Oneiric. Ideally, when I have more time, I'd like to backport the package into Debian Squeeze / Wheezy and Ubuntu Lucid, Maverick and Natty.

    --
    Paul Belanger
    Digium, Inc. | Software Developer



    AstchannelsLive 3.0 wallboard software for Windows
    Click to view a printable version Thu, 23 Jun 2011 20:56:33 -0300

    Version 3.0 of the AstchannelsLive software has been released.

    The software is a free Windows application that allows you to monitor the status of agents and phones in Asterisk.

    The project has two websites:

    http://sourceforge.net/projects/astchannelslive/

    and

    http://www.astchannelslive.com/

    The software has been released under the GPL.

    There is a flash video of it in operation here:

    http://www.astchannelslive.com/vb/flash/astchannelslive.swf



    Ice 3.4.2 incorporated into Asterisk SCF repositories
    Click to view a printable version Thu, 23 Jun 2011 20:43:20 -0300

    Kevin P. Fleming has posted details on the inclusion of the latest version of Ice into Asterisk SCF:

    ZeroC recently released version 3.4.2 of Ice, and I've just finished integrating it into our 'ice' repository to be used for Asterisk SCF builds. We still have quite a number of changes to Ice that we'll continue working with ZeroC to get them merged, and all of them had to be rebased against Ice 3.4.2 to ensure that future code reviews would be as smooth as possible.

    As a result of these rebasing operations, the 'master' branch from the release repository cannot be simply 'pulled' into an existing clone of the repository. Instead, a forced update needs to be done; to do this, ensure that you have the 'master' branch checked out, and that you have no changes in your working directory, and then issue this command:

    # git pull origin +master:master

    This will tell git that you want your local master branch to be the same as the master branch from the origin repository, even though that means losing history you had in your master branch.

    After you've done this you can build and install Ice as the instructions on wiki.asterisk.org document, then update your slice-plugins repository (a normal update will work fine there) and build/install those too. There is no urgency to update your system to correct bugs; our repository already included the bug-fix patches that ZeroC had published for Ice 3.4.1.

    --
    Kevin P. Fleming
    Digium, Inc. | Director of Software Technologies



    Asterisk 1.4.41.1, 1.6.2.18.1, and 1.8.4.3 Now Available (Security Release)
    Click to view a printable version Thu, 23 Jun 2011 20:25:55 -0300

    The Asterisk Development Team has announced the release of Asterisk versions 1.4.41.1, 1.6.2.18.1, and 1.8.4.3, which are security releases.

    These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases

    The release of Asterisk 1.4.41.1, 1.6.2.18, and 1.8.4.3 resolves several issues as outlined below:

    * AST-2011-008: If a remote user sends a SIP packet containing a null, Asterisk assumes available data extends past the null to the end of the packet when the buffer is actually truncated when copied. This causes SIP header parsing to modify data past the end of the buffer altering unrelated memory structures. This vulnerability does not affect TCP/TLS connections.
    -- Resolved in 1.6.2.18.1 and 1.8.4.3

    * AST-2011-009: A remote user sending a SIP packet containing a Contact header with a missing left angle bracket (<) causes Asterisk to access a null pointer.
    -- Resolved in 1.8.4.3

    * AST-2011-010: A memory address was inadvertently transmitted over the network via IAX2 via an option control frame and the remote party would try to access it.
    -- Resolved in 1.4.41.1, 1.6.2.18.1, and 1.8.4.3


    The issues and resolutions are described in the AST-2011-008, AST-2011-009, and AST-2011-010 security advisories.

    For more information about the details of these vulnerabilities, please read the security advisories AST-2011-008, AST-2011-009, and AST-2011-010, which were released at the same time as this announcement.

    For a full list of changes in the current releases, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.4.41.1
    http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.18.1
    http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.4.3

    Security advisories AST-2011-008, AST-2011-009, and AST-2011-010 are available at:

    http://downloads.asterisk.org/pub/security/AST-2011-008.pdf
    http://downloads.asterisk.org/pub/security/AST-2011-009.pdf
    http://downloads.asterisk.org/pub/security/AST-2011-010.pdf

    Thank you for your continued support of Asterisk!



    Asterisk SCF: Major build system update
    Click to view a printable version Wed, 22 Jun 2011 19:33:40 -0300

    Kevin Fleming has posted an email about some pretty big changes to the Asterisk SCF build system:

    On Monday afternoon I merged a significant set of improvements to the Asterisk SCF build system, and along with that there were a number of changes in each of the component repositories to take advantage of the improvements. This message documents the new features (and caveats) of this improved build system:

    • The CTest module (from the CMake distribution) is now included as part of the build; the effect of this is that there is a BUILD_TESTING variable in CMake. It defaults to 'on', but can be turned off, and if that is done, none of the unit tests in the tree will be built, which can save quite a bit of time.
    • The prefix on all the build script functions and variables was changed from "asterisk_scf" to "astscf". This was mostly important for the variables, as it affects how CMake groups them in its GUI representation.
    • A number of functions in AsteriskSCF.cmake are there for use by the script itself; they shouldn't be called from CMakeLists.txt files in component repositories. These functions now have a double-underscore prefix to indicate that they are for internal use.
    • The build system defines a C++ preprocessor macro that is used to mark functions and classes as being 'exported' from their translation units; this is necessary on Windows. Previously this macro was called ASTERISK_SCF_ICEBOX_EXPORT, but it is now called ASTSCF_DLL_EXPORT, since it was not specific to IceBox services.
    • Many functions in AsteriskSCF.cmake now perform sanity checking when they are called to ensure that their prerequisites are met.
    • Components can now be built from sources in multiple directories, as long as astscf_component_init() and one of the astscf_component_build_() functions are called from the top-most directory that contains any part of the component. Subdirectories that are descended into via add_subdirectory() calls *between* the component initialization and build functions are free to add C++ sources, C++ headers and Slice files to the component.
    • Handling of Slice files (translation to C++ headers and sources, compilation and linking into components) is now done by defining 'collections', using astscf_slice_collection(). This function is extensively documented in AsteriskSCF.cmake, but the basics are that a 'collection' consists of a directory (or tree of directories) containing Slice files, possibly a directory or directories containing C++ headers generated from them, and possibly a library built from C++ sources generated from them. Global collections can be referenced by name from anywhere in the build system after they are defined; in addition, projects will have an automatic 'PROJECT' collection defined if they have a top-level 'slice' subdirectory. Installation of Slice files, headers and libraries is done by requesting that the collection be installed; all other details are taken care of by the build system itself. The only downside to this is that when a test or component has a Slice file that it uses entirely for internal purposes (i.e. it will not be installed on the target system), a collection must still be defined to be able to refer to it; this is called a 'local' collection.
    • Slice files are now translated and compiled on a per-component basis, even if multiple components in the same directory use the same Slice file(s).
    • If a Makefile-style build system is generated by CMake, the generated Makefiles will now provide a 'make uninstall' target.
    • When builds are performed using GCC, the build system will now ask the linker to ensure that libraries that are built do not have any unresolved symbols; this means that such problems will be caught at build/link time, instead of not being caught until run time. Such problems are typically caused by forgetting to tell the build system that a component makes use of an installed library.
    • Building on the Slice file collection support above, astscf_component_add_slices() will now automatically discover Slice files that the component needs as a result of using the one(s) specified in the function call, and if any of those also need to be translated/compiled for this component, they will be added to the component as if the developer had requested them directly. In addition, this will happen even for Slice files that have already been translated, compiled and linked into a library, if a Slice file from that collection is *explicitly* added to a component. In the near future, this will allow for component-specific translation of Slice files using the soon-to-be-released metadata overlay plugin.
    • astscf_build_library() can now accept a SHARED, STATIC or MODULE argument to indicate the type of library to be built; astscf_set_libtype() has been removed.
    • If a project has a top-level 'include' directory, it is no longer necessary to explicitly call include_directories() to have it added to the search path; astscf_project() will do that automatically.

    In addition to these build system changes, a number of component repositories had some noticeable changes:

    • The library produced by the ice-util-cpp repository is now called 'astscf-ice-util-cpp'.
    • The library produced by the util-cpp repository is now called 'astscf-util-cpp', and everything in this repository is built into that library (as opposed to being built into three or more libraries).
    • The library produced by the slice repository is now called 'astscf-api'.
    • The CMake build system is no longer brought into a gitall integrated build by the gitall-asterisk-scf.sh script; instead, a copy of the build system has been incorporated into the gitall repository itself, using the git 'subtree' merge mechanism. There is a page on wiki.asterisk.org describing how this was done and the steps to be taken to bring updates into the gitall repository.
    • In a number of project repositories, the directory containing Slice definitions for that project was called 'local-slice' or 'local_slice'; they are all now called 'slice', and their structure follows that of the main slice repository. This resulted in a number of changes to the namespaces that various interfaces reside in.
    • The Slice definitions in project repositories (which generally define replication classes and interfaces, and configuration interfaces) are now installed along with the main Asterisk SCF API Slice definitions.


    --
    Kevin P. Fleming
    Digium, Inc. | Director of Software Technologies



    Asterisk 1.4.42-rc2 Now Available
    Click to view a printable version Mon, 20 Jun 2011 19:46:52 -0300

    The Asterisk Development Team has announced the second release candidate of Asterisk 1.4.42. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

    Please note that Asterisk 1.4.42 will be the final maintenance release from the 1.4 branch. Support for security related issues will continue for one additional year. For more information about support of the various Asterisk branches, see https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

    The release of Asterisk 1.4.42-rc2 resolves several issues reported by the community and would have not been possible without your participation.
    Thank you!

    The following is a sample of the issues resolved in this release candidate:

    * Resolve regression with ring groups in the Dial() application
    (Closes issue ASTERISK-17874. Reported by mspuhler. Patched by elguero)

    * Resolve deadlock when using tab completion on the 'meetme kick' CLI command when an invalid (non-existent) conference room is specified.
    (Closes issue ASTERISK-17771. Reported, patched by zvision)

    * Resolve issue where voice frames could be dropped when checking for T.38 during early media.
    (Closes issue ASTERISK-17705. Reported, patched by oej)

    * Resolve issue where DYNAMIC_FEATURES would not activate after a recent DTMF fix.
    (Closes issue ASTERISK-17914. Reported by vrban. Patched by twilson)

    For a full list of changes in this release candidate, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.42-rc2

    Thank you for your continued support of Asterisk!



    Formatting cleanups during JIRA issue triage
    Click to view a printable version Tue, 14 Jun 2011 22:50:59 -0300

    Russell Bryant has posted a note for those helping triage issues:

    Just a quick tip for those helping triage issues on mantis: if you see an issue that has a config or code snippet in the description, you can update its formatting so that it's displayed in a fixed width font and in a grey box by using the {code} macro. For example:

    {code}
    [default]

    exten => s,1,NoOp(Just an example)
    {code}

    --
    Russell Bryant
    Digium, Inc. | Engineering Manager, Asterisk



    Packaging and AsteriskSCF
    Click to view a printable version Sun, 26 Jun 2011 18:12:40 -0300

    Paul Belanger has been working on putting together a package for AsteriskSCF dependencies for Debian unstable:

    Greetings!

    I had some free time this weekend, so I decided to look and see what was required to package some of the dependencies used by AsteriskSCF.

    I decided to start with zeroc-ice, since this package already exists upstream in Debian unstable. Right now Debian support 3.4.1, talking to the maintainer of the package 3.4.2 should be release this week. So I spend a few hour on Friday night / Saturday morning, packaging 3.4.2.

    Now that zeroc-ice 3.4.2 was packaged, I then create a tarball from the Digium version of ice in git and started work to package it. Again, after a few hours of hacking I\\\'ve successfully created zeroc-ice packages from the version in the Digium git repo. :)

    git://git.asterisk.org/asterisk-scf/release/ice

    There we\\\'re some differences in java/build.xml between the two versions, however I\\\'ve patched the Digium version and reverted some changes to icegridadmin.

    ATM I\\\'ve only created packages for Debian unstable, it should not be hard to port them to Ubuntu Oneiric. Ideally, when I have more time, I\\\'d like to backport the package into Debian Squeeze / Wheezy and Ubuntu Lucid, Maverick and Natty.

    --
    Paul Belanger
    Digium, Inc. | Software Developer



    Asterisk 1.8.4.2 Now Available (Security Release)
    Click to view a printable version Thu, 02 Jun 2011 19:08:19 -0300

    The Asterisk Development Team has announced the release of Asterisk version 1.8.4.2, which is a security release for Asterisk 1.8.

    This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases

    The release of Asterisk 1.8.4.2 resolves an issue with SIP URI parsing which can lead to a remotely exploitable crash:

    Remote Crash Vulnerability in SIP channel driver (AST-2011-007)

    The issue and resolution is described in the AST-2011-007 security advisory.

    For more information about the details of this vulnerability, please read the security advisory AST-2011-007, which was released at the same time as this announcement.

    For a full list of changes in the current release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.4.2

    Security advisory AST-2011-007 is available at:

    http://downloads.asterisk.org/pub/security/AST-2011-007.pdf

    Thank you for your continued support of Asterisk!



    Migration from Mantis to JIRA
    Click to view a printable version Wed, 01 Jun 2011 20:54:46 -0300

    Russell Bryant has posted details of the migration from Mantis to Jira:

    Greetings,

    A few weeks ago I posted a message about the upcoming migration from Mantis to JIRA for issues.asterisk.org. A lot of testing has been done and all known issues have been resolved. We have scheduled the migration for Sunday, June 5th. The issue tracker will be down most of the day as the migration takes place. Once the migration is complete, the issue tracker will be:

    https://issues.asterisk.org/jira/

    Mantis will still be available for some time, but will be read-only. If you have an account on Mantis, you will be able to log in to JIRA using the same username. All of your history will have been migrated. This account can also be used on wiki.asterisk.org.

    IMPORTANT NOTE: You will have to click the "forgot my password" link to reset your password before you can log in, though. It is not possible to migrate passwords from one to the other as they use a different hashing algorithm.

    For more information about how to use JIRA, see the JIRA user's guide:

    http://confluence.atlassian.com/display/JIRA042/JIRA+User%27s+Guide

    If you run into any problems after the migration has taken place, please report them in the "JIRA Help" project. If you would rather report something via email, email espiceland at digium dot com and me.

    Thanks,

    --
    Russell Bryant
    Digium, Inc. | Engineering Manager, Open Source Software



    Asterisk 1.10 - Upcoming Beta
    Click to view a printable version Tue, 31 May 2011 19:39:59 -0300

    Russell has posted an note about the fact that Asterisk 1.10 beta will be out around the end of June:

    Greetings,

    It has been a few months since the last big update on Asterisk 1.10.
    The Asterisk versions overview calls for the release of Asterisk 1.10 in October of this year. We need at least 3 months to go through the beta and RC process. That means that we need to release the first beta at the end of June.

    We should be as close to feature complete at the time of the first beta as possible. It's possible to slip a few more things in during beta, but we should minimize that as much as we can to help reduce the impact to the testing effort.

    We need to make a big push in June to get the things we want in 1.10 into Asterisk trunk. If you have features that you'd like to get in, please respond to this thread with a link to the issue on the issue tracker (and reviewboard if it's there too).

    Thanks,

    --
    Russell Bryant
    Digium, Inc. | Engineering Manager, Open Source Software



    Chapter on Asterisk Architecture
    Click to view a printable version Thu, 26 May 2011 18:48:56 -0300

    Russell Bryant has posted a note about a chapter on Asterisk Architecture:

    Greetings,

    The book "Architecture of Open Source Applications" is now available. It contains a chapter on Asterisk. Take a look if you're interested in an architectural overview of Asterisk from a developer's perspective. You can read it for free here:

    http://www.aosabook.org/

    I hope you find it useful.

    --
    Russell Bryant
    Digium, Inc. | Engineering Manager, Open Source Software



    Skype Issues Official Statement About The End Of Skype For Asterisk
    Click to view a printable version Thu, 26 May 2011 03:03:03 -0300

    Disruptive Telephony has posted an official note from Skype about the end of Skype for Asterisk. Thanks to Olle for pointing it out on Twitter.

    Excerpt from the article:

    Before writing my story yesterday about Skype killing off Skype For Asterisk, I had reached out to Skype's PR agency to see if there was any statement from Skype. There wasn't at the time, but today they sent over this statement from Jennifer Caukin, a spokeswoman for Skype:

    Skype made the decision to retire Skype for Asterisk several months ago, as we have prioritized our focus around implementing the IETF SIP standard in our Skype Connect solution. SIP enjoys the broadest support of any of the available signaling alternatives by business communications equipment vendors, including Digium. By supporting SIP in favor of alternatives, we maximize our resources and continue to reinforce our commitment to delivering Skype on key platforms where we can meet the broadest customer demand.

    Being a huge advocate of open standards, I of course applaud Skype's commitment to supporting SIP. However, as I noted two years ago in my detailed review of what was then "Skype For SIP" (and is now "Skype Connect") the fundamental difference between Skype For Asterisk and Skype's SIP offering is this:

    Skype For Asterisk is/was two-way - you can make outbound calls TO Skype users.

    You can't do that with Skype Connect. You can receive calls from Skype users. You can receive calls to PSTN numbers that come in across the Skype network. You can make outbound calls to PSTN numbers via the Skype network. But you can't make outbound calls to Skype users.

    Read the rest...



    The long slow death of Skype for Asterisk
    Click to view a printable version Wed, 25 May 2011 20:11:32 -0300

    Tim Panton has posted more information on the death of Skype for Asterisk. Well worth a read.

    Excerpt from his article:

    It has been a while coming, but now at least the news is here. Skype have declined to renew their agreement with Digium to produce Skype for Asterisk.

    I’ll miss it. (I’ve had a lot of fun with it – see earlier blog posts)

    SFA (as it is sometimes known) is/was unique in that it allows a 3rd party developer to control a skype experience without being installed on the client’s desktop. It runs server side, in the rich environment of the Asterisk developer ecosystem.

    When I wanted to bridge in chat from a different IM system, SFA was there, chat messages turned up as Asterisk events and I could manage them with my normal tool set. All I had to do was buy a license, no qualification, no pre-approval of the application, no certification of the hardware, just a creditcard, a click-through license and away you go.

    SFA did presence properly too, your Skype status was available to asterisk, so calls could be routed to skype if you were online and to your cellphone if you weren’t (Skype can already do that you say – true enough – but with Asterisk I could whitelist callerid numbers and times of day, so that only the few would get to my cellphone out of office hours.)

    Read the rest...



    Product Notification: Skype for Asterisk - end of sale - July 26, 2011
    Click to view a printable version Tue, 24 May 2011 21:27:13 -0300

    Product notification:

    Skype for Asterisk will not be available for sale or activation after July 26, 2011.

    Skype for Asterisk was developed by Digium in cooperation with Skype. It includes proprietary software from Skype that allows Asterisk to join the Skype network as a native client. Skype has decided not to renew the agreement that permits us to package this proprietary software. Therefore Skype for Asterisk sales and activations will cease on July 26, 2011.

    This change should not affect any existing users of Skype for Asterisk. Representatives of Skype have assured us that they will continue to support and maintain the Skype for Asterisk software for a period of two years thereafter, as specified in the agreement with Digium. We expect that users of Skype for Asterisk will be able to continue using their Asterisk systems on the Skype network until at least July 26, 2013. Skype may extend this at their discretion.

    Skype for Asterisk remains for sale and activation until July 26, 2011. Please complete any purchases and activations before that date.

    Thank you for your business.

    Digium Product Management



    Asterisk 1.8.4.1 Now Available
    Click to view a printable version Tue, 24 May 2011 19:27:13 -0300

    The Asterisk Development Team has announced the release of Asterisk 1.8.4.1.
    This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

    The release of Asterisk 1.8.4.1 resolves several issues reported by the community. Without your help this release would not have been possible.
    Thank you!

    Below is a list of issues resolved in this release:

    * Fix our compliance with RFC 3261 section 18.2.2. (aka Cisco phone fix)
    (Closes issue #18951. Reported by jmls. Patched by wdoekes)

    * Resolve a change in IPv6 header parsing due to the Cisco phone fix issue.
    This issue was found and reported by the Asterisk test suite.
    (Closes issue #18951. Patched by mnicholson)

    * Resolve potential crash when using SIP TLS support.
    (Closes issue #19192. Reported by stknob. Patched by Chainsaw. Tested by vois, Chainsaw)

    * Improve reliability when using SIP TLS.
    (Closes issue #19182. Reported by st. Patched by mnicholson)

    For a full list of changes in this release candidate, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.4.1

    Thank you for your continued support of Asterisk!



    AstLinux 0.7.8 Release
    Click to view a printable version Sun, 22 May 2011 23:36:12 -0300

    The AstLinux Team would like to announce the immediate availability of the 0.7.8 release. This release includes either Asterisk 1.4.41 or Asterisk 1.8.4. All current users are encouraged to upgrade to this release to take advantage of bug fixes and other updates to Asterisk.

    Please note that there is a bug in Asterisk 1.8.4 that will prevent Cisco 79xx phones from registering.

    A full changelog is available at http://www.astlinux.org

    Current users can upgrade from the web interface or from the commandline.

    From the CLI:

    Asterisk 1.4
    upgrade-run-image check http://mirror.astlinux.org/firmware
    should report astlinux-0.7.8
    upgrade-run-image upgrade http://mirror.astlinux.org/firmware

    Asterisk 1.8
    upgrade-run-image check http://mirror.astlinux.org/ast18-firmware
    should report astlinux-0.7.8
    upgrade-run-image upgrade http://mirror.astlinux.org/ast18-firmware

    --
    The AstLinux Team
    http://www.astlinux.org



    Asterisk 1.4.42-rc1 Now Available
    Click to view a printable version Thu, 19 May 2011 00:27:58 -0300

    The Asterisk Development Team has announced the first release candidate of Asterisk 1.4.42. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

    Please note that Asterisk 1.4.42 will be the final maintenance release from the 1.4 branch. Support for security related issues will continue for one additional year. For more information about support of the various Asterisk branches, see
    https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

    The release of Asterisk 1.4.42-rc1 resolves several issues reported by the community and would have not been possible without your participation.
    Thank you!

    The following is a sample of the issues resolved in this release candidate:

    * Fix ordering of voicemails if additional voicemails are left while deleting other voicemails.
    (Closes issues #19032, #18582, #18692, #18998. Patched by Tilghman Lesher, Alec Davis, elguero. Reported, tested by: alecdavis, tootai, afosorio)

    * Fix issues with ISDN calls changing B channels during call negotiations. The handling of the PROCEEDING message was not using the correct call structure if the B channel was changed.
    (Closes issue #18231, #18313, #18488. Reported by destiny6628, jpokorny. Patched by rmudgett)

    * Fix detection of OpenSSL 1.0
    (Closes issue #19093. Reported, patched by tzafrir)

    * Be more flexible with unknown chunks in wav files.
    (Closes issue #18306. Reported, patched by jhirsch)

    * If sox fails when processing a voicemail, don't delete the original file.
    (Closes issue #18111. Reported by sysreq. Patched by seanbright)

    For a full list of changes in this release candidate, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.42-rc1

    Thank you for your continued support of Asterisk!



    Asterisk GUI 2.1.0-rc1 Now Available
    Click to view a printable version Thu, 12 May 2011 20:09:57 -0300

    The Asterisk Development Team has announced the first release candidate of Asterisk-GUI 2.1.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk-gui

    The release of Asterisk-GUI 2.1.0-rc1 resolves several issues reported by the community. Without your help this release would not have been possible.
    Thank you!

    Please help test this release candidate against Asterisk 1.8, and report any issues to https://issues.asterisk.org/ in the Asterisk-GUI project. Thanks!

    Below is a sample of the issues resolved in this release:

    New Features:

    • Compatible with Asterisk 1.6.x and 1.8
    • New option to record MeetMe conferences
    • 14459 - AddQueueMember support

    Improvements:

    • Better compatibility with webkit browsers
    • Real Asterisk version comparison with greater than/less than functionality
    • Cache Asterisk version calculations for improved browser responsiveness
    • default to alwaysauthreject = yes
    • New options in configuration of vpmadt032.

    Bug Fixes:

    • Fix display of sample bulk user input lines
    • Fix use of reserved Javascript keyword "char" in context parsing method.
    • ASTGUI-383 - Modify file-caching behavior to cache contexts separately.
    • ASTGUI-354 - Fix bug in extensions.conf app argument parsing function where string was added to return array only if it was empty.
    • Fix bug where attempting to empty a context which is already empty causes an error and prohibits the callback function from being executed.
    • Handle errors pertaining to unknown status from manager events more gracefully for 1.8 compatibility.
    • Show agents with technology "DAHDI" correctly.
    • 18646 - Fix issue where BLF does not work for users configured in the GUI
    • 17316 - Fix issue where conference interface always empty
    • 18080 - Fix issue where subscribecontext was not validated correctly

    Thank you for your continued support of Asterisk!



    Splices - how to manage multiple media sessions
    Click to view a printable version Tue, 10 May 2011 19:31:12 -0300

    Olle has posted an entry to his blog discussing true multimedia during a communication.

    Excerpt from his post:

    Imagine working at your desk, getting a phone call from your friend Randy. You answer on your IP phone. Being Randy, he suddenly wants to play a new jingle he created while being in the mood the day before. The phone speaker is not a good device for a cool guitar riff - is it? On the same desk you have your PC with a softphone that supports HD voice and really cool loudspeakers. Why not transfer the audio to the PC while still using the phone’s microphone? After that, you want to play a video for Randy. Now you want to add media from your laptop to the call - while the call is still managed by the IP phone.

    Read More...



    Need help defining a stackexchange for telephony
    Click to view a printable version Tue, 10 May 2011 18:50:39 -0300

    Simon P. Ditner has posted a note about trying to set up a system similar to StackOverflow but for telephony:

    For those of that are fans of stackoverflow.com, and stackexchange.com, there's an effort to define a telephony stackexchange site. It's still in the definition phase. What it needs to move forwards is more votes on on/off topic questions, and perhaps some better questions to vote for or against.

    If you're interested in helping out, or following the progress, visit:
    http://area51.stackexchange.com/proposals/12932/telephony/

    Cheers,
    spd



    Asterisk 1.8.4 Now Available
    Click to view a printable version Tue, 10 May 2011 18:41:03 -0300

    The Asterisk Development Team has announced the release of Asterisk 1.8.4. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

    The release of Asterisk 1.8.4 resolves several issues reported by the community.
    Without your help this release would not have been possible. Thank you!

    Below is a sample of the issues resolved in this release:

    * Use SSLv23_client_method instead of old SSLv2 only.
    (Closes issue #19095, #19138. Reported, patched by tzafrir. Tested by russell and chazzam.

    * Resolve crash in ast_mutex_init()
    (Patched by twilson)

    * Resolution of several DTMF based attended transfer issues.
    (Closes issue #17999, #17096, #18395, #17273. Reported by iskatel, gelo, shihchuan, grecco. Patched by rmudgett)

    NOTE: Be sure to read the ChangeLog for more information about these changes.

    * Resolve deadlocks related to device states in chan_sip
    (Closes issue #18310. Reported, patched by one47. Patched by jpeeler)

    * Resolve an issue with the Asterisk manager interface leaking memory when disabled.
    (Reported internally by kmorgan. Patched by russellb)

    * Support greetingsfolder as documented in voicemail.conf.sample.
    (Closes issue #17870. Reported by edhorton. Patched by seanbright)

    * Fix channel redirect out of MeetMe() and other issues with channel softhangup
    (Closes issue #18585. Reported by oej. Tested by oej, wedhorn, russellb. Patched by russellb)

    * Fix voicemail sequencing for file based storage.
    (Closes issue #18498, #18486. Reported by JJCinAZ, bluefox. Patched by jpeeler)

    * Set hangup cause in local_hangup so the proper return code of 486 instead of 503 when using Local channels when the far sides returns a busy. Also affects CCSS in Asterisk 1.8+.
    (Patched by twilson)

    * Fix issues with verbose messages not being output to the console.
    (Closes issue #18580. Reported by pabelanger. Patched by qwell)

    * Fix Deadlock with attended transfer of SIP call
    (Closes issue #18837. Reported, patched by alecdavis. Tested by alecdavid, Irontec, ZX81, cmaj)

    Includes changes per AST-2011-005 and AST-2011-006
    For a full list of changes in this release candidate, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.4

    Information about the security releases are available at:

    http://downloads.asterisk.org/pub/security/AST-2011-005.pdf
    http://downloads.asterisk.org/pub/security/AST-2011-006.pdf

    Thank you for your continued support of Asterisk!



    AstriCon 2011 goes to Denver
    Click to view a printable version Tue, 10 May 2011 18:26:59 -0300

    Lisa King has posted a blog entry on the upcoming Astricon event which will be held in October in Denver.

    Excerpt from her post:

    It has been a great privilege to watch AstriCon “grow up.” And I am really excited to announce this year’s event location: October 25-27, 2011 we will descend upon the city of DENVER, Colorado! The event is set for The Westin Westminster. Located just outside of Denver with a backdrop of the scenic Rocky Mountains, The Westin Westminster combines the best of both worlds – the comforts and accessibility of the city and the feel and relaxation of the countryside.

    The meeting areas and exhibit space are conveniently located on one level and within easy view of one other. This should make it much easier to find all of the various tracks and meeting rooms within the conference area, and still be close enough to check out all the great exhibits. The hotel has very comfortable sleeping rooms, which will allow all of us a good night’s rest after the activities of the day. Not one to call it a day too early? No problem! Once the scheduled activities end, there are plenty of restaurants and entertainment venues directly across the street for you to enjoy. For those of you who made last year’s event, this location has a very similar feel and layout to the Renaissance in Phoenix.

    Read More...



    Mantis to JIRA migration testing
    Click to view a printable version Mon, 09 May 2011 20:20:21 -0300

    Russell Bryant has posted a note explaining the move from Mantis to Jira for issue tracking for Asterisk:

    Greetings,

    The Asterisk issue tracker, issues.asterisk.org, has been running the Mantis issue tracker since 2003. We have begun migrating issues.asterisk.org to run a new issue tracker, JIRA. It integrates well with other Atlassian tools we are already using, including Bamboo and Confluence.

    Please take some time to test this instance of JIRA and see if things work how you would expect them to:

    https://issues.asterisk.org/jira/

    If you have an account on Mantis, you can log in using the same username. All of your history will have been migrated. You will have to click the "forgot my password" link to reset your password before you can log in, though.

    The data that is imported into this test instance is a couple of months old, but is representative of what the migration would look like. Please keep in mind that any changes made on this test instance will be lost.

    If you find any problems, please report them on Mantis in the "Mantis" project. If you have any questions or comments, please post to the asterisk-dev list.

    Thanks,

    --
    Russell Bryant
    Digium, Inc. | Engineering Manager, Open Source Software



    Trixbox CE - undocumented web admin
    Click to view a printable version Wed, 04 May 2011 19:38:08 -0300

    This one is a little old (March 2011) and I only just saw it. Basically there was an undocumented web admin account.

    Excerpt from the release:

    Trixbox CE, an Asterisk and FreePBX based system ships with undocumented web admin.

    The admin web interface can be accessed by user wwwadmin which grants full access to PBX configuration including all trunks/extensions credentials and config changes.

    Read More...



    Digium Update - Wednesday May 3rd - Back in Business
    Click to view a printable version Wed, 04 May 2011 19:12:23 -0300

    Danny has posted what will likely be the last update regarding the loss of power in Alabama:

    As of last evening, power was restored to the Digium building! By noon today we should have the facility and all systems transferred back to commercial power and once again resume operations at full capacity. Our sincere thanks go out to all those who offered and provided support to Digium over the past week. Additionally, I’d like to thank the Digium team for the incredible efforts on the part of so many people to keep the company going for the past 6 days in the presence of so many obstacles. While things will quickly get back to business as usual for our company – life as usual for many in our community will be a long time coming. Our hearts go out to all of those impacted by this disaster.

    Danny



    The Importance of Looking Ahead
    Click to view a printable version Wed, 04 May 2011 00:07:34 -0300

    Bryan Johns has posted a blog entry on the retiring of Asterisk versions 1.4 and 1.6. Quite a bit has been discussed about this in the mailing lists via a myriad of different opinions. I personally am using 1.2, 1.4 and 1.8 across various sites without problems.

    Excerpt from his post:

    Last week marked the retirement of the 1.4 and 1.6 branches of Asterisk to security maintenance only. This means that the 1.4 and 1.6 versions of the Asterisk open source telephony framework will only receive necessary security patches going forward and all feature enhancements, bug-fixes and other new work will be applied to the 1.8 LTS (Long Term Support) version and the future delivery of the upcoming 1.10 (Standard Support) version, anticipated later this year. Version 1.8 of Asterisk, along with the older 1.4 version, marked as a Long Term Support, bug fixes are made four years past initial version release. Standard Support releases, such as 1.6.x and 1.10, receive bug fixes for one year past initial version release. This does not mean that you cannot continue to use 1.4 or 1.6. It simply means that Digium will be orienting it’s focus to the support of 1.8 LTS and to the development of 1.10.

    I joined Digium late last year following many years of implementing Asterisk solutions (beginning at version 1.0) in corporate America and I understand the reasons why some choose to stay with older versions like 1.4. They are perceived to be more stable and reliable and they are more familiar to the user because they have been available for a longer period of time. Still, in order to spur the evolution of Asterisk as technology the Asterisk community has to limit the extent to which we look backward and focus more of our available energy on the current and future versions of the software.

    Read More...



    Bug triage delayed due to storms in Alabama
    Click to view a printable version Tue, 03 May 2011 23:56:16 -0300

    Leif Madsen has posted an update on bug triaging:

    After the tornado's that hit Alabama last week, Digium HQ is still offline except for critical services. As the building is running on generator power currently, there will be some delay in triaging issues for the next couple of days.

    Please be patient while things get back to normal, and apologies in advance for the long lead time on triaging your issues.

    Updates about the situation at Digium HQ can be found on the Digium blog at http://blogs.digium.com/

    While issues may remain in the New status until they can be triaged, commenting and work on existing issues can continue, and is of course encouraged.

    Thanks!
    Leif Madsen -- Bug Marshal



    Digium Update - Monday May 2nd
    Click to view a printable version Mon, 02 May 2011 18:19:06 -0300

    Danny has posted another update on the progress on getting power restored to Digium:

    Efforts continue today across North Alabama as the region struggles to recover from last week’s storms. The communications infrastructure is greatly improved and power has now been restored to a number of areas, but unfortunately not to Digium’s headquarters. As a result, Digium is once again operating on generator power in a limited fashion. Most Digium services including digium.com, asterisk.org, and asteriskexchange.com are operational. A small number of Digium services are temporarily unavailable including the master subversion server for Asterisk. Support staff are in place assisting customers with technical issues and the sales and operations staff are now able to process and ship orders without delay.

    Our thoughts and prayers remain with our friends and colleagues who have suffered loss during this difficult time. Updates will be posted here as the situation evolves

    Sourced from blogs.digium.com



    Digium Update - Friday April 29
    Click to view a printable version Fri, 29 Apr 2011 09:40:39 -0300

    Danny has posted an update on the situation over at Digium:

    Following Wednesday’s violent weather, Digium continues to operate with a skeleton staff. As of this morning, power to the region has not been restored, and local carriers and ISPs are experiencing intermittent service outages. Consequently, access to Digium personnel and services remains unreliable. However, support and operations personnel are in place to assist customers with any critical technical issues and priority product needs. If initial attempts to contact us prove unsuccessful – please be persistent. Updates will be posted here as the situation evolves.



    origsvn.digium.com down
    Click to view a printable version Thu, 28 Apr 2011 18:34:47 -0300

    Russell has posted a note to say that the OrigSVN server is down and commits will likely have to wait till next week:

    Greetings,

    origsvn.digium.com, the master subversion server, is down and may not be back until early next week. This outage does not affect svn access to the public mirror, svn.digium.com/svn.asterisk.org.

    Alabama was hit with some pretty severe storms yesterday and much of the northern part of the state will be without power for 4 to 5 days. This server is not on the company's generator. Commits will have to hold off until next week. Feel free to make your git jokes here. :-)

    --
    Russell Bryant
    Digium, Inc. | Engineering Manager, Open Source Software



    Digium Operations Impacted by Tornado Outbreak
    Click to view a printable version Thu, 28 Apr 2011 18:25:42 -0300

    Danny Windham has posted a note about Alabama being hit by the worst tornado outbreak in 30 years.

    All of our thoughts are with those affected both directly and indirectly. Digium offices are closed at the moment although all services continue to run.

    Excerpt from his post:

    Dear customers, friends and colleagues of Digium,

    The region of the US where Digium is headquartered (North Alabama) was hit hard yesterday by the worst tornado outbreak in 30 years. Thankfully, we are unaware of any injuries to Digium employees. However, several employees, and many of our friends, neighbors, and colleagues have sustained significant personal property loss or damage. Power is out across the entire region (over 1M people) with over 80 transmission lines that supply the region having been destroyed. Estimates are that over 200 people were killed, 40 of them in north Alabama, 8 in our local community. A state of emergency has been declared and a dusk to dawn curfew has been enacted.

    Read More...



    Astricon 2011 Call for Speakers
    Click to view a printable version Wed, 27 Apr 2011 23:28:11 -0300

    Bran Johns has posted a call for speakers for the next Astricon.

    Excerpt from his post:

    It’s that time of year again! Digium is excited to announce Astricon 2011 to be held October 25-27 this year in Denver, Colorado. Now that we know where and when this year’s conference will be held, we are putting out a Call for Speakers to the community of Asterisk users, developers and partners.

    Read More...



    Asterisk SCF Performs at SIPit 28
    Click to view a printable version Wed, 27 Apr 2011 23:18:23 -0300

    Bryan Johns has posted a blog entry about the recent SIPit 28 that was hosted by Digium.

    Excerpt from his post:

    Digium was honored to host the SIP Forum’s SIPit 28 interoperability testing event April 11-15 in Huntsville, Alabama. At this event, Asterisk SCF was put through a series of tests to gauge its compliance with SIP standards and its performance interoperating with other vendors’ SIP implementations. The Asterisk SCF development team at Digium was excited to see that the platform, while still early in its development, performed well. A number of issues with Asterisk SCF were addressed and repaired during the event and the product continues its march toward a 1.0 release by the end of 2011.

    Read More...



    Request for Reviews for Asterisk: The Definitive Guide
    Click to view a printable version Tue, 26 Apr 2011 20:01:46 -0300

    Leif Madsen has written a post on his blog asking people to read the new Asterisk book, and if you like it post a review to Amazon.

    Contents of his post:

    Recently Jim Van Meggelen, Russell Bryant and myself released the 3rd edition of the Asterisk book published by O’Reilly Media titled ‘Asterisk: The Definitive Guide‘.

    We have released this book under a Creative Commons license in the spirit of Open Source software. It is available in its entirety at http://ofps.oreilly.com/titles/9780596517342/ and for purchase through Amazon

    If you have a chance to review the book, and feel it is a useful and well written resource, we would appreciate your favorable review on Amazon.com. One of the biggest factors in sales is favorable reviews, and with better sales comes the ability to dedicate more time to writing books like this one (which are typically written in our “free time”).

    Any comments, suggestions, or constructive criticisms are always welcome.

    Thanks!
    Leif Madsen.



    Asterisk 1.4.40.2, 1.4.41 and 1.6.2.18 now available.
    Click to view a printable version Tue, 26 Apr 2011 19:45:32 -0300

    The Asterisk Development Team has announced the release of new versions of Asterisk.

    Asterisk 1.4.40.2

    The Asterisk Development Team has announced the release of Asterisk 1.4.40.2.
    This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

    After the initial release of AST-2011-006, a regression was found and then resolved. This release contains the correct change.

    * Reverted part of r314607, as it can introduce a regression. Specifically, the security check for the "system" privilege was removed. If a user had the "call" privilege but not the "system" privilege, they would lose the ability to execute the system app and dialplan functions that run commands in a shell. This branch never used the "system" privilege for that purpose and did not need to be patched.
    (Related to AST-2011-006)

    For a full list of changes in this release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.40.2

    Thank you for your continued support of Asterisk!

    Asterisk 1.4.41

    The Asterisk Development Team has announced the release of Asterisk 1.4.41. This
    release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

    The release of Asterisk 1.4.41 resolves several issues reported by the community and would have not been possible without your participation.
    Thank you!

    The following is a sample of the issues resolved in this release:

    * Only offer codecs both sides support for directmedia.
    (Closes issue #17403. Reported, patched by one47)

    * Resolution of several DTMF based attended transfer issues.
    (Closes issue #17999, #17096, #18395, #17273. Reported by iskatel, gelo, shihchuan, grecco. Patched by rmudgett)
    NOTE: Be sure to read the ChangeLog for more information about these changes.

    * Fix channel redirect out of MeetMe() and other issues with channel softhangup
    (Closes issue #18585. Reported by oej. Tested by oej, wedhorn, russellb. Patched by russellb)

    * Fix voicemail sequencing for file based storage.
    (Closes issue #18498, #18486. Reported by JJCinAZ, bluefox. Patched by jpeeler)

    * Guard against retransmitting BYEs indefinitely during attended transfers with chan_sip.
    (Review: https://reviewboard.asterisk.org/r/1077/)


    In addition to the changes listed above, commits to resolve security issues AST-2011-005 and AST-2011-006 have been merged into this release. More information about AST-2011-005 and AST-2011-006 can be found at:

    http://downloads.asterisk.org/pub/security/AST-2011-005.pdf
    http://downloads.asterisk.org/pub/security/AST-2011-006.pdf

    After the initial release of AST-2011-006, a regression was found and then resolved. This release contains the correct change.

    For a full list of changes in this release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.41

    Thank you for your continued support of Asterisk!

    Asterisk 1.6.2.18

    The Asterisk Development Team has announced the release of Asterisk 1.6.2.18.
    This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

    The release of Asterisk 1.6.2.18 resolves several issues reported by the
    community and would have not been possible without your participation.
    Thank you!

    The following is a sample of the issues resolved in this release:

    * Only offer codecs both sides support for directmedia.
    (Closes issue #17403. Reported, patched by one47)

    * Resolution of several DTMF based attended transfer issues.
    (Closes issue #17999, #17096, #18395, #17273. Reported by iskatel, gelo, shihchuan, grecco. Patched by rmudgett)
    NOTE: Be sure to read the ChangeLog for more information about these changes.

    * Resolve deadlocks related to device states in chan_sip
    (Closes issue #18310. Reported, patched by one47. Patched by jpeeler)

    * Fix channel redirect out of MeetMe() and other issues with channel softhangup
    (Closes issue #18585. Reported by oej. Tested by oej, wedhorn, russellb. Patched by russellb)

    * Fix voicemail sequencing for file based storage.
    (Closes issue #18498, #18486. Reported by JJCinAZ, bluefox. Patched by jpeeler)

    * Guard against retransmitting BYEs indefinitely during attended transfers with chan_sip.
    (Review: https://reviewboard.asterisk.org/r/1077/)

    In addition to the changes listed above, commits to resolve security issues AST-2011-005 and AST-2011-006 have been merged into this release. More information about AST-2011-005 and AST-2011-006 can be found at:

    http://downloads.asterisk.org/pub/security/AST-2011-005.pdf
    http://downloads.asterisk.org/pub/security/AST-2011-006.pdf

    For a full list of changes in this release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.18

    Thank you for your continued support of Asterisk!



    The new ConfBridge application is now in Asterisk Trunk
    Click to view a printable version Mon, 25 Apr 2011 20:07:05 -0300

    David Vossel has posted details of the inclusion of the new ConfBridge application which has been merged into trunk - it has support for codecs all the way up to 192khz!

    His post:

    Howdy,

    I am proud to announce that after a good bit of development, community feedback, testing, and code review, the brand new ConfBridge application has been officially merged into Asterisk Trunk!!! http://svnview.digium.com/svn/asterisk?view=revision&revision=314598

    If you are already familiar with ConfBridge from Asterisk 1.6.X and 1.8, forget everything you know. This is a completely revamped, highly optimized, and feature rich conferencing application capable of mixing sample rates from 8khz all the way up to 192khz! Exciting right?! So Go! use it, test it, and report back! Tell us what you like, what you don't like, if you want a feature that doesn't yet exist, and report bugs. This conferencing application has huge potential and we need community feedback. Asterisk 1.10 isn't that far away, and once it is branched adding new functionality to this application may not be possible, so start using it now!

    To help get you started, Malcolm Davenport has written some fantastic documentation on the asterisk.org wiki. It can be found below.
    https://wiki.asterisk.org/wiki/display/AST/ConfBridge+1.10

    New conference join and leave sounds have been created for this application, but will not be available officially until the next sounds release. If you can not wait until then you can find them attached to this issue, https://issues.asterisk.org/view.php?id=19165.

    If you start using the ConfBridge application and find that you are interested in writing a new feature for it, feel free to use me as a resource by email or IRC. I'm happy to review your code and anything else I can to do make this application successful.

    Thanks!

    --
    David Vossel
    Digium, Inc. | Software Developer, Open Source Software



    Final call for changes to Asterisk 1.4 and 1.6.2
    Click to view a printable version Mon, 25 Apr 2011 19:52:58 -0300

    Greetings,

    Today (Note: 22nd of April - sorry missed this one) is scheduled to be the last day of full maintenance for the Asterisk 1.4.X and 1.6.2.X release series. Here is what will happen next:

    Release candidates for 1.4.41 and 1.6.2.18 are currently pending and will be released very soon. After that, we plan to make one more release for 1.4.X and 1.6.2.X that contains normal bug fixes (1.4.42 and 1.6.2.19). These releases will include any fixes that have been made since the release candidates for 1.4.41 and 1.6.2.18 were made.

    We would like to allow for a 1 week grace period to get final changes in. If you have any patches that have not yet been committed to 1.4 or 1.6.2, please commit them very soon. If you have patches that are written but need review, please note them in this thread and we will get them reviewed as soon as we can.

    Open bug reports against 1.4 and 1.6.2 will not immediately be closed. We will evaluate them for whether they affect supported versions (1.8) before closing anything. In some cases, we may need assistance from the reporter to help determine if an issue is still a problem in Asterisk 1.8.

    These changes will allow the development team to focus more on addressing issues that are reported against Asterisk 1.8. Now let's work together to move Asterisk forward!

    Thanks,

    --
    Russell Bryant
    Digium, Inc. | Engineering Manager, Open Source Software



    Asterisk 1.4.40.1, 1.6.1.25, 1.6.2.17.3, and 1.8.3.3 Now Available (Security Releases)
    Click to view a printable version Mon, 25 Apr 2011 19:30:14 -0300

    The Asterisk Development Team has announced security releases for Asterisk branches 1.4, 1.6.1, 1.6.2, and 1.8. The available security releases are released as versions 1.4.40.1, 1.6.1.25, 1.6.2.17.3, and 1.8.3.3.

    These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases

    The releases of Asterisk 1.4.40.1, 1.6.1.25, 1.6.2.17.3, and 1.8.3.3 resolve two issues:

    * File Descriptor Resource Exhaustion (AST-2011-005)
    * Asterisk Manager User Shell Access (AST-2011-006)

    The issues and resolutions are described in the AST-2011-005 and AST-2011-006 security advisories.

    For more information about the details of these vulnerabilities, please read the security advisories AST-2011-005 and AST-2011-006, which were released at the same time as this announcement.

    For a full list of changes in the current releases, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.4.40.1
    http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.1.25
    http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.17.3
    http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.3.3

    Security advisory AST-2011-005 and AST-2011-006 are available at:

    http://downloads.asterisk.org/pub/security/AST-2011-005.pdf
    http://downloads.asterisk.org/pub/security/AST-2011-006.pdf

    Thank you for your continued support of Asterisk!



    AppKonference 1.7
    Click to view a printable version Mon, 18 Apr 2011 20:45:00 -0300

    Paul Albrecht has posted details of the latest version of AppKonference. It will now work with Asterisk 1.8.

    Hi,

    I have released an updated AppKonference today. This fix release includes the following changes:

    Fix to compile module with Asterisk 1.8. Note: To compile the module you must set the new Makefile variable "ASTERISK_SRC_DIR" to point to your Asterisk source.

    Changed the frame mixing algorithm to reduce the number of calls to buffer mixing routine.

    Changed the output frame queuing routine, eliminating the sequential search through out frames for each conference member.

    Use SIMD instructions for frame mixing. Note: this has only been tested on Asterisk 1.4. Later Asterisk branches seem to leave the slinear frames on an inappropriate byte boundary so some asterisk hacks are necessary for Asterisk 1.6/1.8. If you want to try this feature uncomment the "VECTORS" variable in the Makefile.


    You can download the latest source from source forge:

    sourceforge.net/projects/appkonference

    --
    Paul Albrecht



    SIPit 28 completed
    Click to view a printable version Sun, 17 Apr 2011 20:36:42 -0300

    Olle has posted an entry to his blog about the completion of SIPit 28 that was hosted by Digium.

    An excerpt from his post:

    SIPit 28 was hosted by Digium in Huntsville, Alabama, USA the week of April 11-15, 2010. There were 54 attendees from 19 companies visiting from 10 countries, using 40 distinct implementations in the interoperability tests.

    SIPit, organized by the SIP Forum, is one of the foundations that make SIP work across vendors and implementations. Twice each year, developers from all around the world meet and test, discuss, learn and fix issues both in implementations and standards. During SIPit events, many bugs in the RFCs - or just missing explanations - has been found. Under the leadership of Robert Sparks, SIPit has become the primary event for all SIP developers. Edvina proudly organized SIPit #26 in Stockholm in May 2010.

    Read More...



    Asterisk Tech Tips: Asterisk Cookbook
    Click to view a printable version Wed, 13 Apr 2011 20:17:15 -0300

    Steven Sokol has posted details of the next Asterisk Tech Tips event taking place next Thursday, the 21st of April.

    Excerpt from his post:

    I’m pleased to announce that Russell Bryant and Leif Madsen, two giants of the Asterisk community, will host the next Asterisk Tech Tips event, scheduled for Thursday, April 21. The dynamic duo co-authored Asterisk Cookbook, a new how-to filled with a laundry list of Asterisk recipes. Russell and Leif will introduce the book and each will present their favorite recipe – live and in person!

    Read More...



    Online training for Asterisk offers new dialplan video for free
    Click to view a printable version Wed, 13 Apr 2011 20:13:29 -0300

    Digium has released another of the Asterisk Essentials videos for free although the full pack will retail for between $299 and $349 depending on when you purchase it.

    Excerpt from their release:

    Two weeks ago, Digium announced Asterisk Essentials, our first online training course. It consists of more than 6 hours of video, where we introduce Asterisk and Digium, walk through the background and basics of Asterisk, and demonstrate how to install and configure Asterisk and many of its features. We named the course Asterisk “Essentials” because it’s focused on rapidly covering the most critical concepts an administrator needs to understand to work with Asterisk. The whole course is full of working examples you can try on your own, too.

    At launch, we gave away one of the introductory videos that talks about the course layout and goals, as well as a technical video on IVR Basics in Asterisk. (Be sure to review Allison Smith’s “IVR Commandments” series of blog posts after watching this video!).

    Today, we’re pleased to offer another free video from Asterisk Essentials covering Dialplan Variables. These free training videos should give you a feel for the style and quality of the course.

    Asterisk Essentials retails for $349, but is available through July 1 for just $299. Visit the Asterisk Essentials overview to learn more about the course, see system requirements and to view the free videos.

    Read More...



    CEL Logging to MySQL - Please Test
    Click to view a printable version Mon, 11 Apr 2011 20:38:17 -0300

    Jonathan Penny has been working on a MySQL module for the new Call Event Logging system in Asterisk and now needs it to be tested:

    I’ve recently finished an add-on module for CEL logging to MySQL, and it needs to be tested.

    The feature is being tracked at https://issues.asterisk.org/view.php?id=19058

    And the patch is available at https://issues.asterisk.org/file_download.php?file_id=29110&type=bug

    Thank You,

    -Jonathan Penny



    DAHDI-Linux 2.4.1.2 Released
    Click to view a printable version Mon, 11 Apr 2011 20:26:44 -0300

    The Asterisk Development Team announces the release of DAHDI-Linux 2.4.1.2.

    DAHDI-Linux 2.4.1.2 and DAHDI-Linux-Complete 2.4.1.2+2.4.1 are available for immediate download at:

    http://downloads.asterisk.org/pub/telephony/dahdi-linux
    http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete

    2.4.1.2 is a maintenance release that resolves a conflict with RHEL 5.6. RHEL 5.6 backported the definition of dev_name from kernel 2.6.26. DAHDI also had this definition backported. The result was that DAHDI would fail to compile.
    The issue was originally reported in https://issues.asterisk.org/view.php?id=18992

    Issues found in these releases can be reported in the DAHDI-linux project at https://issues.asterisk.org

    Thank you for your continued support of Asterisk!



    SIPit 28 to be hosted by Digium
    Click to view a printable version Mon, 04 Apr 2011 20:08:26 -0300

    Kevin has posted an entry on the Asterisk website about the upcoming SIP interoperability testing event.

    Excerpt from his post:

    From April 11 through April 15, 2011, the 28th bi-annual SIPit testing event, organized by the SIP Forum, will be held in Huntsville, Alabama at the Jackson Conference Center. Digium is hosting the event, with additional sponsorship from Polycom, Inc.

    As a member of the worldwide Voice over IP community for more than ten years, we’ve participated in multiple SIPit events, working with vendors from around the world to ensure that Asterisk interoperates well with a variety of SIP endpoints and servers. At SIPit 28, our testing teams will be testing both Asterisk 1.8 (and the future Asterisk 1.10) and the in-development Asterisk Scalable Communications Framework (SCF).

    Read more...



    DAHDI-Linux 2.4.1.1 Released
    Click to view a printable version Mon, 04 Apr 2011 18:42:45 -0300

    The Asterisk Development Team announces the release of DAHDI-Linux 2.4.1.1.

    DAHDI-Linux 2.4.1.1 and DAHDI-Linux-Complete 2.4.1.1+2.4.1 are available for immediate download at:

    http://downloads.asterisk.org/pub/telephony/dahdi-linux
    http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete

    2.4.1.1 is a maintenance release to fix a regression in DAHDI-Linux 2.4.1 from DAHDI-Linux 2.4.0 which prevented the LEDS on the TDM410 and AEX410 panel from lighting when the wctdm24xxp module loads (#18939).

    Issues found in these releases can be reported in the DAHDI-linux project at https://issues.asterisk.org

    Thank you for your continued support of Asterisk!



    Asterisk Dev Team Releases Binary Packages
    Click to view a printable version Tue, 29 Mar 2011 18:43:07 -0400

    The Asterisk Developer Team has announced the availability of binary packages.

    Excerpt from the announcement:

    Compile no more my friends, for binary packages are here! Yesterday (March 28, 2011) the Asterisk development team opened the doors on a new package repository for users of Debian and Ubuntu, two of the most popular Linux distributions. Those repos complement the existing RedHat Enterprise Linux / CentOS Linux repositories that Digium has maintained for the past several years. As of now you can quickly and easily install and maintain Asterisk using apt (Ubuntu/Debian) or yum (RHEL/CentOS) package management utilities.

    Read More...



    Asterisk Cookbook Is Done
    Click to view a printable version Sun, 27 Mar 2011 21:10:24 -0400

    Leif Madsen has posted details on the completion of the Asterisk Cookbook.

    Just a quick update that the Asterisk Cookbook is complete! We’ll be releasing it under a Creative Commons license (just like Asterisk: The Definitive Guide) again. It’ll be a 24 recipe e-book right now, with it growing over time and eventually becoming a full print edition once the size warrants it. Of course if you purchase now you help spur on additional incentive to continue making it larger.

    Read More...



    GenerationD launches HAAST Lite
    Click to view a printable version Sun, 27 Mar 2011 21:00:03 -0400

    Asterisk software developer GenerationD systems has launched a Lite version of their HAAST Asterisk clustering software, serving the needs of smaller companies.

    High Availability ASTerisk (HAAST) is a software product which creates a high availability / clustered pair out of an existing pair of Asterisk servers. HAAST can detect a range of failures on one Asterisk server and automatically transfer control to another Asterisk Server, resulting in a telephony environment with minimal down time.

    HAAST is a 100% software solution, with switchover in as little as 10 seconds. Built-in intelligent network control allows for a single IP address shared between multiple servers, so clients/phones automatically connect to the active Asterisk server . Built-in replication of configuration between servers also reduces maintenance and support activities.

    HAAST is an easy to use solution, with shell (command line), telnet, and web interfaces, suitable for beginners and experts alike. HAAST is ideal for demanding telephony environments like call centers and mid-to-large size businesses, as well as for small-businesses looking for high PBX uptime using low-cost off the shelf components.

    Further information is available at www.generationd.com



    Echo Cancellation Hardware for Xorcom Asterisk Solutions
    Click to view a printable version Sun, 27 Mar 2011 20:54:52 -0400

    Xorcom, a privately-held manufacturer of business telephony interfaces and appliances, announces that it is now shipping a hardware echo canceller module for its award-winning Astribank™ telephony interfaces and Asterisk-based IP-PBX appliances. The module provides a high level of echo cancellation and voice enhancement while reducing load on the CPU as compared to OSLEC, the software-based echo cancellation solution provided in the standard Asterisk framework.

    New Echo Cancellation Module Increases Number of Simultaneous Calls Supported

    Until now, Xorcom recommended that their customers use the OSLEC (Open Source Line Echo Canceller) provided as part of the Asterisk operating system. Central Processing Units (CPUs) are so powerful these days that normally the OSLEC has no impact on the IP-PBX operation. However, if the telephony system is used in high density call environments, and/or where additional applications are involved, such as call centers that need call recording functionality, offloading the echo cancellation processing to a separate hardware module makes sense.

    “By reducing the load on the server’s CPU, the number of simultaneous calls that are supported on the telephony system is increased significantly,” states Izzy Gal, VP Innovation at Xorcom. “We’ve set the echo tail size to 128 taps and our published load tests show that we achieve the same number of simultaneous calls as when we disable the software echo cancellation completely.”

    About Xorcom

    Founded in 2004 and privately-held, Xorcom designs and manufactures business telephony solutions that support both traditional and VoIP communication. Xorcom solutions are easy to install and maintain, and are cost-effective since there are no per user license fees. Various utilities provide backup and restore, immediate recovery, and a full redundancy solution for the entire PBX to maximize uptime. Xorcom’s bundled solution for hospitality features the only MICROS-Fidelio certified Property Management System (PMS) interface for Asterisk. Xorcom sells its products via a worldwide distribution channel and OEM partners. For more information, visit: www.xorcom.com.



    Asterisk 1.8 Packages for Debian and Ubuntu
    Click to view a printable version Wed, 23 Mar 2011 18:36:49 -0400

    Digium has posted a note about the fact they will now be providing Debian and Ubuntu packages in addition to the current CentOS and RHEL packages:

    Greetings,

    Digium has been providing rpm packages for the latest versions of
    Asterisk that are compatible with RHEL 5 and CentOS 5 for quite some
    time now. We are pleased to announce that we will now be providing deb
    packages for both Debian and Ubuntu. As of now, we have Asterisk 1.8
    packages available for the following distribution versions:

    * Debian 6.0 (squeezy)
    * Ubuntu 10.04 (lucid)
    * Ubuntu 10.10 (maverick)
    * Ubuntu 11.04 (natty)

    This effort is not intended to replace packaging of Asterisk in the
    official Debian or Ubuntu repositories. Our repositories are for
    providing access to major versions of Asterisk that are newer than what
    is included. We are exploring ways to work as closely as possible with
    the Debian and Ubuntu package maintainers to ensure that we do not
    duplicate efforts and that we provide the best possible result for users
    of Asterisk.

    For information on how to set up your system to use our repositories,
    please refer to the following page on the Asterisk wiki:

    https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages

    If you have any problems related to the repositories or the packages themselves, please report them in the "AsteriskNOW and Packages" project on the Asterisk issue tracker, http://issues.asterisk.org/.

    Thanks,

    -- Asterisk Development Team --



    Asterisk Tech Tips - Thursdays At Noon
    Click to view a printable version Mon, 21 Mar 2011 18:50:07 -0400

    Steve Sokol has posted details of a new initiative over at Digium to provide a tech tips series - first one this Thursday.

    Excerpt from his post:

    Get your geek on and join us this Thursday (March 24, 12:00 PM CDT) for Asterisk Tech Tips, a new webinar series that gets down and dirty with Asterisk. Each Tech Tips episode starts out with an in-depth tutorial covering one of the many awesome features of Asterisk. After the tutorial we open up the floor for general Q&A. You’re welcome to ask about the topic of the week or any other Asterisk-related questions you might have. We’ll have a number of Asterisk experts available to answer questions.

    Read more...



    Asterisk 1.6.1.23, 1.6.1.17.1 and 1.8.3.1 Now Available (Security Releases)
    Click to view a printable version Wed, 16 Mar 2011 19:46:42 -0400

    The Asterisk Development Team has announced security releases for Asterisk branches 1.6.1, 1.6.2, and 1.8. The available security releases are released as versions 1.6.1.23, 1.6.2.17.1, and 1.8.3.1.

    These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases

    The releases of Asterisk 1.6.1.23, 1.6.2.17.1, and 1.8.3.1 resolve two issues:

    * Resource exhaustion in Asterisk Manager Interface (AST-2011-003)
    * Remote crash vulnerability in TCP/TLS server (AST-2011-004)

    The issues and resolutions are described in the AST-2011-003 and AST-2011-004 security advisories.

    For more information about the details of these vulnerabilities, please read the security advisories AST-2011-003 and AST-2011-004, which were released at the same time as this announcement.

    For a full list of changes in the current releases, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.1.23
    http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.17.1
    http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.3.1

    Security advisory AST-2011-003 and AST-2011-004 are available at:

    http://downloads.asterisk.org/pub/security/AST-2011-003.pdf
    http://downloads.asterisk.org/pub/security/AST-2011-004.pdf

    Thank you for your continued support of Asterisk!



    Peering into the future of Asterisk
    Click to view a printable version Wed, 16 Mar 2011 18:15:23 -0400

    Malcolm Davenport has posted an article about the future of Asterisk.

    Here's an excerpt from his post:

    Having completed the release of Asterisk 1.8 this past October, we’ve since turned our attention to the next release of Asterisk, tentatively numbered 1.10. We expect, subject to change at any time, to have Asterisk 1.10 ready to go in time for Astricon, this fall, in the October-ish timeframe. This means that we’ll have betas and release candidates beginning in the Summer. Unlike Asterisk 1.8, Asterisk 1.10 will *not* be a long-term-support release, which means it’ll only be supported for 1 year from its release; see https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions for more information. But…it’ll be full of new features, features of which you’ll probably want to take advantage.

    Read More...



    Asterisk Cookbook Ready For Review
    Click to view a printable version Wed, 16 Mar 2011 16:56:05 -0400

    Leif Madsen has posted a link on his blog to the Open Feedback Publishing System for the Asterisk Cookbook.

    Just a quick note that we’ve finished the Asterisk Cookbook (electronic edition) and it is now ready for review! We’re hoping to have all comments submitted within the next week.

    We’re using OFPS (Open Feedback Publishing System) again. You can read the book and help review here: http://ofps.oreilly.com/titles/9781449303822/



    First Look at Asterisk SCF Developer APIs
    Click to view a printable version Thu, 10 Mar 2011 18:09:32 -0400

    Kevin Fleming has posted a link to a look at the new Asterisk SCF APIs:

    I recently gave a presentation at FOSDEM 2011 about Asterisk SCF Developer APIs… the presentation, sample code and some details are available over on the Asterisk wiki (here). Enjoy!

    Excerpt from the wiki page:

    Recently I had the pleasure of attending my first FOSDEM, FOSDEM 2011. It was an intense weekend of open source presentations and conversations, and was quite enjoyable (although exhausting). Of course, it doesn't hurt that FOSDEM is held in Brussels, which has some of the best food I've ever had in Europe! (No, they aren't kidding about how good the frites are... you really do need to try them)

    At this event I participated in the Open Source Telephony Dev Room, where I gave a presentation about the developer APIs we are building in Asterisk SCF. While it's still pretty early in the project's life, we've got a pretty good idea of how we want them to work, because we've had years of application/system developers telling us what they really need from a communications platform. Just before FOSDEM, the Asterisk SCF development team agreed on what the very first Extension Point that we'd build was going to be: SIP authentication. In Asterisk SCF, an Extension Point is a place where a developer can 'hook' their own code to modify/influence the behavior of another component, without having to look at that component's source code (or modify it, as is usually the case). In most cases, they will be places where a component has to make a decision about how to proceed while handling a request, but the decision necessarily involves some sort of policy to be defined. In most applications, the policy is usually defined by a set of (possibly quite complex) configuration options, which a system administrator can set (hopefully) to tell the system exactly how they wish it to behave.

    Read More...



    IP-PBX from Xorcom Awarded Best of Show at ITEXPO East 2011
    Click to view a printable version Thu, 10 Mar 2011 15:58:28 -0400

    Xorcom today announced that its new XE Series IP-PBX for VoIP and traditional business communication solutions received a “Best of Show Award” in the On-site Product Launch category at TMC’s recent ITEXPO East 2011 in Miami. Like the existing XR2000 and XR3000 series, these new PBX handle both traditional (Public Switched Telephone Network) and VoIP calls. The new series, however, features microprocessor-based temperature control with redundant fans, redundant hard disk drives, internal backup and recovery support, and an LCD (Liquid Crystal Display) touch panel for easy access to system parameters.

    The ITEXPO Best of Show Awards recognize innovative products and solutions that were featured in the exhibit hall during the event. This designation serves to highlight the technological achievement and creative product features that set these winners apart from the rest of the crowd. TMC’s editorial staff evaluated the pre-show nominations, and met with vendors to inspect and review the displayed products on the first day of the Expo.

    “Feedback from our channel drove us to create the new series,” states Eran Gal, CEO and co-founder of Xorcom. “On the one hand, our current systems’ modularity created an almost unlimited number of configurations, which is attractive. However, this type of flexibility can also be confusing. We realized we could divide our customers into two groups: those with standard telephony requirements, and those whose call volumes and additional applications mandated a more powerful phone system, or that have stricter policies about networking and reliability. The first group is served by our existing XR series, while the latter will benefit from the more robust XE series.”

    “ITEXPO in Miami was one of the best events we’ve ever produced, and this year’s Best-of-Show winners faced an even greater challenge differentiating themselves from a large and highly competitive field,” said Rich Tehrani, CEO and conference chairman for TMC. “We’re proud to honor companies like Xorcom with a Best-of-Show Award, and we thank them for sharing their innovative solutions with our attendees.”

    A full list of the winners is posted on TMC’s Web site at www.tmcnet.com. The next ITEXPO event is scheduled to take place in Austin, TX, September 13-15, 2011. Visit Xorcom there in booth 519.

    About Xorcom
    Founded in 2004 and privately-held, Xorcom designs and manufactures business telephony solutions that support both traditional and VoIP communication. Xorcom solutions are easy to install and maintain, and are cost-effective since there are no per user license fees. Various utilities provide backup and restore, immediate recovery, and a full redundancy solution for the entire PBX to maximize uptime. Xorcom’s bundled solution for hospitality features the only MICROS-Fidelio certified Property Management System (PMS) interface for Asterisk. Xorcom sells its products via a worldwide distribution channel and OEM partners. For more information, visit: http://www.xorcom.com



    Olle, SIP and IPV6
    Click to view a printable version Wed, 09 Mar 2011 17:03:50 -0400

    Olle has posted an article on his progress towards IPv6 with SIP.

    Excerpt from the article:

    During the last couple of months, I’ve been trying to understand how to migrate SIP to a world with dual stacks or only IPv6 stacks. In order to get momentum, I’ve started to create a repository of information on Edvina’s SIPv6 site. I’ve also started a new Twitter flow and a Facebook page. Please follow the project there.

    Dan York of Voxeo encouraged this work even further by writing an article on the Voxeo blog called “Will You Join In Olle’s Crusade for VoIP and IPv6?” Great support - thanks, Dan!

    Read More...



    Debugging the Asterisk Dialplan with Verbose
    Click to view a printable version Mon, 07 Mar 2011 18:37:32 -0400

    Russell has posted an article on using verbose to debug the Asterisk dialplan.

    An excerpt from the article:

    Leif Madsen and I are working on a new book, the Asterisk Cookbook. One of the recipes that I am working on this morning is a method of adding debug statements into the Asterisk dialplan. I came up with a GoSub() routine that can log messages based on log level settings that are global, per-device, or per-channel. Here’s a preview. I hope you find it useful!
    Channel logging GoSub() routine.

    * ARG1 – Log level.
    * ARG2 – The log message.

    Channel logging using this routine will be sent to the Asterisk console at verbose level 0, meaning that they will show up when you want them to regardless of the current “core set verbose” setting. This routine uses a different method, values in AstDB, to control what messages show up.

    Read More...



    AstLinux 0.7.7 released
    Click to view a printable version Sun, 06 Mar 2011 20:09:50 -0400

    The AstLinux Team would like to announce the immediate availability of the 0.7.7 release:

    This release includes either Asterisk 1.4.40 or Asterisk 1.8.3. All current users are encouraged to upgrade to this release to take advantage of bug fixes and other updates to Asterisk.

    PPTP was added as a possible VPN option.

    A full changelog is available at http://www.astlinux.org


    Current users can upgrade from the web interface or from the commandline.

    From the CLI:

    (Asterisk 1.4)
    upgrade-run-image check http://mirror.astlinux.org/firmware

    should report astlinux-0.7.7

    upgrade-run-image upgrade http://mirror.astlinux.org/firmware

    (Asterisk 1.8)
    upgrade-run-image check http://mirror.astlinux.org/ast18-firmware

    should report astlinux-0.7.7

    upgrade-run-image check http://mirror.astlinux.org/ast18-firmware

    --
    The AstLinux Team



    DAHDI-Linux 2.4.1 and DAHDI-Tools 2.4.1 Released
    Click to view a printable version Fri, 04 Mar 2011 17:55:20 -0400

    The Asterisk Development Team is pleased to announce the release of DAHDI-Linux and DAHDI-Tools version 2.4.1.

    DAHDI-Linux 2.4.1, DAHDI-Tools 2.4.1, and DAHDI-Linux-Complete 2.4.1+2.4.1 are available for immediate download at:
    http://downloads.asterisk.org/pub/telephony/dahdi-linux
    http://downloads.asterisk.org/pub/telephony/dahdi-tools
    http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete

    2.4.1 is a maintenance release of the DAHDI drivers and tools packages. Some of the more notable changes are:

    * Support for compilation against kernel versions from 2.6.9 up to and including 2.6.38-rc6.

    * wct4xxp: PCI-express cards go through an extended reset at start by default.

    * wcte12xp, wctdm24xxp: Disable read-line multiple PCI command, which increases compatibility in some systems.

    * xpp: Fixes init error for PRI devices with < 4 ports.

    * tonezone: Add Macao, China to tone zone data.

    * dahdi_genconf: Don't generate configurations that use channel 16 on E1 CAS.

    For a full list of changes in these releases, please see the ChangeLogs at http://svn.asterisk.org/svn/dahdi/linux/tags/2.4.1/ChangeLog and http://svn.asterisk.org/svn/dahdi/tools/tags/2.4.1/ChangeLog

    Issues found in these release candidates can be reported in the DAHDI-linux or DAHDI-tools project at https://issues.asterisk.org

    Thank you for your continued support of Asterisk!



    A2Billing 1.9 (Cuprum) released
    Click to view a printable version Tue, 01 Mar 2011 17:01:42 -0400

    Areski has posted details of the latest release of A2Billing:

    As A2Billing enters its 7th year of providing its Open Source Switch and Billing system, version 1.9 has been released which includes a number of exciting new features.

    One of the most important of the new features of A2Billing is A-Leg charging, that is to say billing per minute on the ingress leg of the call for DID resale and delivery.

    The A-Leg features allow for a number of new products with flexible pricing that can easily be configured and marketed to your customers. For instance:

    • Geographic telephone numbers billed per minute in addition to a monthly rental.
    • Non-Geographic numbers Billed per minute, with a monthly rental, such as toll-free services.
    • Revenue Share numbers, with the ability to credit a customer's account when those numbers are called.
    • On-net calling.
    • Voice conferencing service billed per user per minute.

    Needless to say, there have been a number of bug fixes and security enhancements so we would urge you to upgrade. We have moved development onto Github, so you can follow our work at https://github.com/Star2Billing/a2billing

    Upgrades on the systems previously installed by us are quick and easy, so if you want to upgrade, then please contact us for pricing. For those wanting a commercial install with training and, optionally, ongoing support, may wish to consider our Managed Install Bundle as described at http://www.star2billing.com/consultancy/managed-install/

    Yours,
    Areski & Joe, and the rest of The Star2Billing Team



    Using exec to set externaddr in sip.conf
    Click to view a printable version Mon, 28 Feb 2011 17:24:35 -0400

    Leif Madsen has put together a script to use cURL via PHP to set externaddr in sip.conf.

    Excerpt from the article:

    Today I was working on a system, and knowing that the system is going to get moved, and that often one of the things forgotten is to update the externaddr= option in sip.conf (when Asterisk is sitting behind NAT), I decided to put together a little script that returns the external IP address of the system. Using this script along with an #exec in the sip.conf file will make it so the address gets updated when the system is moved to the new physical location. I used the php5-curl package on Ubuntu.

    Read More...



    Asterisk 1.4.40, 1.6.2.17 and 1.8.3 released.
    Click to view a printable version Mon, 28 Feb 2011 16:25:21 -0400

    The Asterisk Development Team has announced the release of Asterisk 1.4.40, 1.6.2.17 and 1.8.3.

    Asterisk 1.4.40

    The Asterisk Development Team has announced the release of Asterisk 1.4.40. This release is available for immediate download at

    http://downloads.asterisk.org/pub/telephony/asterisk/

    The release of Asterisk 1.4.40 resolves several issues reported by the community and would have not been possible without your participation.
    Thank you!

    The following is a sample of the issues resolved in this release:

    * Correct issue where res_config_odbc could populate fields with invalid data.
    (Closes issue #18251, #18279. Reported by bcnit, zerohalo. Tested by trev, jthurman, elguero, zerohalo. Patched by tilghman)

    * Resolve issue where re-transmissions of SUBSCRIBE could break presence.
    (Closes issue #18075. Reported by mdu113. Patched by twilson)

    * Resolve issue in res_odbc where it may crash when a query fails.
    (Closes issue #18243. Reported, patched by ks3)

    * Fix CPU spike when pressing DTMF after agent login.
    (Closes issue #18130. Reported by rgj. Patched by jpeeler)

    * Fix cross-compiling issue.
    (Closes issue #18301. Reported, patched by abelbeck)

    * This version of Asterisk includes the new Compiler Flags option BETTER_BACKTRACES which uses libbfd to search for better symbol information within both the Asterisk binary, as well as loaded modules, to assist when using inline backtraces to track down problems.
    (Patched by tilghman)

    * Resolve several issues with DTMF based attended transfers.
    (Closes issues #17999, #17096, #18395, #17273. Reported by iskatel, gelo, shihchaun, grecco. Patched by rmudgett).
    NOTE: Be sure to read the ChangeLog for more information about these changes.

    * Fix regression that changed behavior of queues when ringing a queue member.
    (Closes issue #18747, #18733. Reported by vrban. Patched by qwell.)

    Additionally, this release has the changes related to security bulletin AST-2011-002 which can be found at

    http://downloads.asterisk.org/pub/security/AST-2011-002.pdf

    For a full list of changes in this release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.40

    Thank you for your continued support of Asterisk!

    Asterisk 1.6.2.17

    The Asterisk Development Team has announced the release of Asterisk 1.6.2.17. This release is available for immediate download at
    http://downloads.asterisk.org/pub/telephony/asterisk/

    The release of Asterisk 1.6.2.17 resolves several issues reported by the community and would have not been possible without your participation.
    Thank you!

    The following is a sample of the issues resolved in this release:

    * Resolve duplicated data in the AstDB when using DIALGROUP()
    (Closes issue #18091. Reported by bunny. Patched by tilghman)

    * Correct issue where res_config_odbc could populate fields with invalid data.
    (Closes issue #18251, #18279. Reported by bcnit, zerohalo. Tested by trev, jthurman, elguero, zerohalo. Patched by tilghman)

    * When using cdr_pgsql the billsec field was not populated correctly on unanswered calls.
    (Closes issue #18406. Reported by joscas. Patched by tilghman)

    * Resolve issue where re-transmissions of SUBSCRIBE could break presence.
    (Closes issue #18075. Reported by mdu113. Patched by twilson)

    * Fix regression causing forwarding voicemails to not work with file storage.
    (Closes issue #18358. Reported by cabal95. Patched by jpeeler)

    * This version of Asterisk includes the new Compiler Flags option BETTER_BACKTRACES which uses libbfd to search for better symbol information within both the Asterisk binary, as well as loaded modules, to assist when using inline backtraces to track down problems.
    (Patched by tilghman)

    * Resolve several issues with DTMF based attended transfers.
    (Closes issues #17999, #17096, #18395, #17273. Reported by iskatel, gelo, shihchaun, grecco. Patched by rmudgett).
    NOTE: Be sure to read the ChangeLog for more information about these changes.

    * Resolve issue where no Music On Hold may be triggered when using res_timing_dahdi.
    (Closes issues #18262. Reported by francesco_r. Patched by cjacobson. Tested by francesco_r, rfrantik, one47)

    * Fix regression that changed behavior of queues when ringing a queue member.
    (Closes issue #18747, #18733. Reported by vrban. Patched by qwell.)

    Additionally, this release has the changes related to security bulletin AST-2011-002 which can be found at

    http://downloads.asterisk.org/pub/security/AST-2011-002.pdf

    For a full list of changes in this release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.17

    Thank you for your continued support of Asterisk!

    Asterisk 1.8.3

    The Asterisk Development Team has announced the release of Asterisk 1.8.3. This release is available for immediate download at
    http://downloads.asterisk.org/pub/telephony/asterisk/

    The release of Asterisk 1.8.3 resolves several issues reported by the community and would have not been possible without your participation. Thank you!

    The following is a sample of the issues resolved in this release:

    * Resolve duplicated data in the AstDB when using DIALGROUP()
    (Closes issue #18091. Reported by bunny. Patched by tilghman)

    * Ensure the ipaddr field in realtime is large enough to handle IPv6 addresses.
    (Closes issue #18464. Reported, patched by IgorG)

    * Reworking parsing of mwi => lines to resolve a segfault. Also add a set of unit tests for the function that does the parsing.
    (Closes issue #18350. Reported by gbour. Patched by Marquis)

    * When using cdr_pgsql the billsec field was not populated correctly on unanswered calls.
    (Closes issue #18406. Reported by joscas. Patched by tilghman)

    * Resolve memory leak in iCalendar and Exchange calendaring modules.
    (Closes issue #18521. Reported, patched by pitel. Tested by cervajs)

    * This version of Asterisk includes the new Compiler Flags option BETTER_BACKTRACES which uses libbfd to search for better symbol information within both the Asterisk binary, as well as loaded modules, to assist when using inline backtraces to track down problems.
    (Patched by tilghman)

    * Resolve issue where no Music On Hold may be triggered when using res_timing_dahdi.
    (Closes issues #18262. Reported by francesco_r. Patched by cjacobson. Tested by francesco_r, rfrantik, one47)

    * Resolve a memory leak when the Asterisk Manager Interface is disabled.
    (Reported internally by kmorgan. Patched by russellb)

    * Reimplemented fax session reservation to reverse the ABI breakage introduced in r297486.
    (Reported internally. Patched by mnicholson)

    * Fix regression that changed behavior of queues when ringing a queue member.
    (Closes issue #18747, #18733. Reported by vrban. Patched by qwell.)

    * Resolve deadlock involving REFER.
    (Closes issue #18403. Reported, tested by jthurman. Patched by jpeeler.)

    Additionally, this release has the changes related to security bulletin AST-2011-002 which can be found at
    http://downloads.asterisk.org/pub/security/AST-2011-002.pdf

    For a full list of changes in this release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.3

    Thank you for your continued support of Asterisk!



    Introducing the new ConfBridge
    Click to view a printable version Wed, 23 Feb 2011 21:02:40 -0400

    David Vossel has posted details of the new ConfBridge in Asterisk:

    Howdy,

    Over the next couple of weeks I will be working on a new feature set for ConfBridge. In order to accomplish some of the features I have planned, a new configuration method for ConfBridge will be necessary. This new configuration method is what I am bringing to you all today. Before I get started on implementation I want to tell you all what I have planned and open the floor for feedback.

    At the moment ConfBridge's configuration is centralized entirely in the dialplan. The new configuration method will be a combination of a new confbridge.conf file and a new ConfBridge application syntax that takes advantage of the options declared in that config file. This will result in a ConfBridge application that is much more powerful than the current application, but not backwards compatible with the ConfBridge syntax used in 1.8. The lack of backwards compatibility with the previous syntax is unfortunate but necessary to achieve the new functionality we have planned.

    --- ConfBridge Configuration Concepts

    Bridge Profiles - Contains options such as mixing interval, internal sample rate, and other bridge specific options.

    User Profiles - Contains options such as conference pins, marked user, noise gate filters, volume control, and other user specific options.

    Menus - Custom mappings of DTMF sequences to conference actions. This allows for different menus to be assigned to each individual user depending on their permissions.

    --- ConfBridge Dialpan Syntax

    The ConfBridge dialplan application now uses these three configuration concepts, bridge profiles, user profiles, and menus as arguments. The new syntax will look like this ConfBridge([confno][,bridge profile[,user profile[,menu]]]). The only required argument to ConfBridge is the confno. In the case of when no user or bridge profiles are provided, the options declared in the [general] section of confbridge.conf will be used.

    --- Sample Configuration exercising these concepts.

    ; The general section is used to build the default bridge and user profile. Both bridge
    ; and user options can be declared here under this section. When ConfBridge is called without
    ; any bridge or user profile arguments, the options in this section are exercised.

    [general]
    mix_interval=20
    mix_sample_rate=dynamic ;adjust to best quality sample rate for the channels joined.
    denoise=yes ; denoise every channel that joins the bridge

    ; Custom Bridge profiles can be made. These options take precedence over the general section when
    ; this profile is used.
    [custom_bridge]
    type=bridge
    mix_interval=40
    mix_sample_rate=32khz ;fixed 32khz internal sample rate for mixing.
    wait_for_moderator=yes
    end_conf_moderator=yes
    join_sound=custom-join
    exit_sound=custom-exit
    auto_record=filename.wav

    ; Custom user profile can be made as well. All users who join
    ; with this profile will be muted by default.
    [silent_users]
    type=user
    mute=yes

    ; Custom Menus are created by mapping DTMF sequences to ConfBridge
    ; actions. This is a custom menu for the silent users so they can
    ; adjust their listening volume.
    [silent_menu]
    type=menu
    '*'=playback_continue(silent-conf-menu)
    '*1'=increase_rx_volume
    '*2'=decrease_rx_volume

    ; Any number of user profiles can be created and applied to a
    ; single bridge. Channels joining the bridge with this
    ; user profile will be asked for a pin and be marked as
    ; a moderator.
    [moderator_users]
    type=user
    pin=1234
    moderator=yes

    ; This menu is not that useful and is just for example purposes.
    ; It shows how more complex menu structures can be created.
    [moderator_menu]
    type=menu
    '*'=playback_continue(conf-menu) ;play a prompt while continuing to gather DTMF sequence
    '*1'=toggle_mute
    '*2'=toggle_deaf
    '*3'=toggle_deaf, toggle_mute ;notice multiple actions can be associated with a single DTMF sequence
    '*4'=dialplan_exec(context,priority)
    '*5'=playback_continue(conf-menu2) ;play another prompt while collecting more of the DTMF sequence
    '*51'=increase_tx_volume
    '*52'=decrease_tx_volume
    '*53'=increase_rx_volume
    '*54'=decrease_rx_volume

    --- Using the sample configuration in the dialplan with the ConfBridge app.

    ; users who can only listen to the conference call this extension
    exten => muted_listener,1,Answer()
    exten => muted_listener,n,ConfBridge(1111, custom_bridge, silent_user, silent_menu)

    ;Moderators call this extension but have to put in a pin because they are a moderator_user.
    exten => moderator,1,Answer()
    exten => moderator,n,Conf(1111, custom_bridge, moderator_user, moderator_menu)
    -----------------------------------------------

    I look forward to any feedback you have to offer on this new configuration method!

    David Vossel
    Digium, Inc. | Software Developer, Open Source Software
    445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
    Check us out at: www.digium.com & www.asterisk.org
    The_Boy_Wonder in #asterisk-dev



    Adhearsion 1.0.1 Released
    Click to view a printable version Wed, 23 Feb 2011 18:39:31 -0400

    Ben Klang has posted details of the latest version of Adhearsion:

    The Adhearsion team announces the release of Adhearsion version 1.0.1. Adhearsion is an open source Ruby-language framework for creating telephony applications. This update primarily addresses compatibility with newer versions of other software but also adds native support for Bundler to newly created Adhearsion applications.

    Here are some highlights from the changelog:

    * Handling of new Asterisk 1.6/1.8 events
    * Improved control of Asterisk Queues
    * Two new dialplan methods have been added: say_chars and say_phonetic
    * Ruby 1.9 is now an officially supported platform
    * Fix compatibility with Rails 3
    * Bundler now included by default for new Adhearsion applications

    Not bad for a dot release! You can read the full CHANGELOG here.

    As always I'd like to thank the Adhearsion community for their contributions to this release. Special thanks to contributors Ben Langfeld, Robert Jackson and Matthew Clark.

    To install Adhearsion just type gem install adhearsion at your nearest command prompt. For help getting started, checkout our Wiki and Getting Started pages. As always, you can find us on irc.freenode.net #adhearsion or our Google Groups mailing list. Contributors welcome! Check out the sources on Adhearsion's Github.

    /BAK/
    --
    Ben Klang



    Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 Now Available
    Click to view a printable version Tue, 22 Feb 2011 17:12:23 -0400

    The Asterisk Development Team has announced security releases for Asterisk branches 1.4, 1.6.1, 1.6.2, and 1.8. The available security releases are released as versions 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4.

    These releases are available for immediate download at

    http://downloads.asterisk.org/pub/telephony/asterisk/releases

    The releases of Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 resolve an issue that when decoding UDPTL packets, multiple stack and heap based arrays can be made to overflow by specially crafted packets. Systems configured for T.38 pass through or termination are vulnerable. The issue and resolution are described in the AST-2011-002 security advisory.

    For more information about the details of this vulnerability, please read the security advisory AST-2011-002, which was released at the same time as this announcement.

    For a full list of changes in the current release, please see the ChangeLog:

    ChangeLog-1.4.39.2
    ChangeLog-1.6.1.22
    ChangeLog-1.6.2.16.2
    ChangeLog-1.8.2.4

    Security advisory AST-2011-002 is available at:

    http://downloads.asterisk.org/pub/security/AST-2011-002.pdf

    Thank you for your continued support of Asterisk!



    AST-2011-002: Multiple array overflow and crash vulnerabilities in UDPTL code
    Click to view a printable version Mon, 21 Feb 2011 17:47:52 -0400

    Asterisk Project Security Advisory - AST-2011-002

Product

Asterisk

Summary

Multiple array overflow and crash vulnerabilities in UDPTL code

Nature of Advisory

Exploitable Stack and Heap Array Overflows

Susceptibility

Remote Unauthenticated Sessions

Severity

Critical

Exploits Known

No

Reported On

January 27, 2011

Reported By

Matthew Nicholson

Posted On

February 21, 2011

Last Updated On

February 21, 2011

Advisory Contact

Matthew Nicholson <mnicholson@digium.com>

CVE Name




Description

When decoding UDPTL packets, multiple stack and heap based arrays can be made to overflow by specially crafted packets. Systems doing T.38 pass through or termination are vulnerable.


Resolution

The UDPTL decoding routines have been modified to respect the limits of exploitable arrays.


In asterisk versions not containing the fix for this issue, disabling T.38 support will prevent this vulnerability from being exploited. T.38 support can be disabled in chan_sip by setting the t38pt_udptl option to ?no? (it is off by default).


t38pt_udptl = no


The chan_ooh323 module should also be disabled by adding the following line in modles.conf.


noload => chan_ooh323


Affected Versions

Product

Release Series


Asterisk Open Source

1.4.x

All versions

Asterisk Open Source

1.6.x

All versions

Asterisk Business Edition

C.x.x

All versions

AsteriskNOW

1.5

All versions

s800i (Asterisk Appliance)

1.2.x

All versions


Corrected In

Product

Release

Asterisk Open Source

1.4.39.2, 1.6.1.22, 1.6.2.16.2, 1.8.2.4

Asterisk Business Edition

C.3.6.3


Patches

URL

Branch

http://downloads.asterisk.org/pub/security/AST-2011-002-1.4.diff

1.4

http://downloads.asterisk.org/pub/security/AST-2011-002-1.6.1.diff

1.6.1

http://downloads.asterisk.org/pub/security/AST-2011-002-1.6.2.diff

1.6.2

http://downloads.asterisk.org/pub/security/AST-2011-002-1.8.diff

1.8



Links



Asterisk Project Security Advisories are posted at http://www.asterisk.org/security

This document may be superseded by later versions; if so, the latest version will be posted at http://downloads.digium.com/pub/security/AST-2011-002.pdf and http://downloads.digium.com/pub/security/AST-2011-002.html


Revision History

Date

Editor

Revisions Made

02/21/11

Matthew Nicholson

Initial Release


Asterisk Project Security Advisory - AST-2011-002
Copyright © 2011 Digium, Inc. All Rights Reserved.
Permission is hereby granted to distribute and publish this advisory in its original, unaltered form.





AstLinux 0.7.6 Released
Click to view a printable version
Sun, 20 Feb 2011 17:13:11 -0400

The AstLinux Team is happy to announce the latest release (0.7.6).

There are several security updates as well as feature enhancements/improvements. All current users are encouraged to update.

A full changelog is available at: http://www.astlinux.org

Both Asterisk 1.4.39.1 and Asterisk 1.8.2.3 are supported on separate firmware images.

Current users can upgrade from the web interface or from the commandline.

From the CLI:

Asterisk 1.4

upgrade-run-image check http://mirror.astlinux.org/firmware

should report astlinux-0.7.6

upgrade-run-image upgrade http://mirror.astlinux.org/firmware

Asterisk 1.8

upgrade-run-image check http://mirror.astlinux.org/ast18-firmware

should report astlinux-0.7.6

upgrade-run-image check http://mirror.astlinux.org/ast18-firmware

--
The AstLinux Team



First HD Conference In Asterisk
Click to view a printable version Sun, 20 Feb 2011 17:07:27 -0400

Leif Madsen has posted details of a successful public HD conference using Asterisk:

Today Russell Bryant and I had the first public HD conference using Asterisk! There had been other testing done by Malcolm Davenport and David Vossel (who is the developer working on this integration) internally, but this was the first public HD enabled conference using Asterisk (as far as I’m aware).

It worked really well! People were able to join in, and those who were calling in via G722 were able to hear each other in wideband, while the other people in the conference who were using narrowband codecs like GSM and ulaw didn’t hear any difference in the audio from the participants. It’s very cool that just because someone joins in a narrowband codec that it doesn’t bring the quality of the audio down for all participants. Very nice!

This is a feature that is slated to be in Asterisk 1.10, and it’s comforting to know just how well this works already. You can check out the team branch where David Vossel is working at http://svn.asterisk.org/svn/asterisk/team/dvossel/fixtheworld_phase2

Enjoy!



Chan SS7 2.0.0 released
Click to view a printable version Thu, 17 Feb 2011 19:44:49 -0400

Anders Baekgaard from Netfors has posted details of the latest release of their SS7 Channel for Asterisk.

Netfors has released chan_ss7 v. 2.0.0.

This release has a major new feature: it has premilinary support for ANSI MTP and ANSI ISUP. More information is available at http://www.netfors.com/chan_ss7

Please provide comments to chan_ss7 at netfors dot com.

Best regards,
--
Anders Baekgaard
Netfors

Also, Abdul Basit provided the following details about the release:

Great news.
Just for the list, i pasted the specifications here for quick overview:

Specfication for version 2.0.0
- Compatible with Asterisk 1.2.x, 1.4.x and 1.6.x.
- MTP2 (Q.703) implementation
- MTP3 (Q.704) implementation (subset).
- ISUP (Q.76x) implementation (mostly complete).
- Supports Dahdi/Zaptel compatible digital interfaces, e.g. Redfone, Sangoma, Digium, Openvox
- Facilities for MTP2 packet protocol analysis using e.g. wireshark/ethereal
- Supports high call volumes
- Supports multiple linksets with different OPCs/DPCs
- Supports linksets with multiple links.
- Supports load sharing and MTP changeover.
- Supports multiple hosts (cluster) configuration with load sharing and failover.
- Flexible Dial command syntax for SS7 to allow routing to different linksets.


New in version 2.0.0
- Preliminary ANSI support.

--
Regards,

Abdul Basit



Asterisk 1.10 Update
Click to view a printable version Wed, 16 Feb 2011 18:23:18 -0400

Russell Bryant has posted some information about Asterisk 1.10:

Greetings,

Shortly after the release of Asterisk 1.8, we had a developer meeting and discussed some of the projects that people would like to see in Asterisk 1.10. We discussed the schedule there a bit, as well. Now that Asterisk 1.8 has settled down and we are well into the development cycle for Asterisk 1.10, it is a good time to revisit the plans for the next release.

At Digium, the biggest thing we have been working on for 1.10 so far is replacing the media infrastructure in Asterisk. Most of the critical and invasive plumbing work is done and has been merged into trunk. Next we're looking at building up some features on top of that, such as adding more codecs, enhancing ConfBridge() to support additional sampling rates (HD conferencing), adding features that exist in MeetMe() but not ConfBridge(), and enhancing codec negotiation.

Of course, many others have been working on new developments as well. I would encourage you to respond if you'd like to provide an update on some new things that you're working on.

We would like to release Asterisk 1.10 roughly a year after Asterisk 1.8. This will be a standard release, not LTS. To have the release out in the October time frame, we need to branch off 1.10 (feature freeze) at the end of June. At that point we will begin the beta and RC process. If you're working on new development projects that you would like to get into Asterisk 1.10, please keep this timeline in mind.

As always, comments and questions are welcome.

Thanks,

--
Russell Bryant
Digium, Inc. | Engineering Manager, Open Source Software
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA



AstLinux: Beta images for 0.7.6
Click to view a printable version Mon, 24 Jan 2011 18:28:48 -0400

Darrick Hartman has posted details of the latest beta images for AstLinux 0.7.6 - the embedded Asterisk distribution:

We're getting close to having 0.7.6 ready to go. There were some fairly significant updates which we would like to see some additional people help with testing.

The major changes are:

Dahdi: Update to 2.4.0
Asterisk: Update to 1.4.39.1; 1.8.2.1
Wanpipe: Update to 3.5.18 which now has several working configuration utilities (not all, but there are several that are now working).

There are a few other less critical updates, but we really could use some broader exposure on the Wanpipe testing. I've tested it on an A200 compatible setup with fxo and fxs modules.

If you want to help test, Asterisk 1.4 firmware is available:

upgrade-run-image check http://mirror.astlinux.org/beta-firmware (should say astlinux-0.7-4729)
upgrade-run-image upgrade http://mirror.astlinux.org/beta-firmware

Asterisk 1.8 firmware is available:

upgrade-run-image check http://mirror.astlinux.org/beta-18firmware (should say astlinux-0.7-4729)
upgrade-run-image upgrade http://mirror.astlinux.org/beta-18firmware

Please report back any results (positive or negative) along with the type of hardware used.

Thanks,

Darrick
--
Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com



Article number 2500
Click to view a printable version Mon, 24 Jan 2011 18:22:12 -0400

This article marks number 2500. We have been producing articles on Asterisk and related products now since 2004.

Don't forget you can send us articles or notes you'd like posted by clicking on the Submit Article link at the top of the Daily Asterisk News.

If you have any suggestions for how you'd like the news changed or any ideas for articles, just let me know by emailing me: matt@venturevoip.com

Hope you've enjoyed the last 2500 articles and that you enjoy the next 2500 articles :-)



Crystal Provisioning
Click to view a printable version Sun, 23 Jan 2011 21:01:04 -0400

Crystal Provisioning - The way to: deploy, configure and manage any sip phones.
Save Time and Money
Plug in the phone and the Crystal Provisioning will auto set the phone with your company profile.
Crystal Provisioning features:

  • Crystal Provisioning auto scans the network for new phones.
  • Crystal Provisioning displays the phones details:
    • Phone name and model and fabricator
    • Phone Mac address
    • Phone firmware version
    • Subscriber name.
    • Phone IP address

  • Configure all your phones from one single interface.
  • Manage multiply company’s (different phone configuration for every company/department)
  • Manage Single Phone book and up load the list to any phone model or fabricator.
  • Mange List of your company PBX extensions and their register status
  • Crystal Provisioning will alert concerning IP address conflicts
  • Remote phone reboot
  • Direct phone admin interface from Crystal Provisioning





People Tech CRM
Click to view a printable version Sun, 23 Jan 2011 20:17:21 -0400

PeopleTech have sent details of their CRM package:

PeopleTech Systems CRM is a Customer Relationship Management software package developed on Java platform. The product focuses mainly on Sales, Marketing, and Service (help desk) sectors, and has been encouraging partners to use its framework to customize it to meet many different demands and also complete stable solution for all your telemarketing requirements.

PeopleTech Systems CRM family includes Dialer, Inbound Routing, and Advanced CRM, Inbound Outbound Campaign Management, IVR, BI Reporting, Voice and SMS Broadcasting.

Primary Benefits are

Our IVR system employs the very latest in IVR technology and Open Standards Telephony Servers, Advanced Speech Recognition (ASR), Interactive Voice Response (IVR), Text-to-Speech (TTS) and connectivity to databases

Connecting IVRS with existing web, legacy database and customer service systems already in place and integration API’s are available for web service and Sockets

Using standards to integrate IVR Software with the rest of your infrastructure provides a number of opportunities for businesses to improve customer service and reduce costs
A spring framework Java based architecture of this application makes it stable , scalable and robust and it can be integrated with Asterisk , Free switch or any SIP based IPBX, MS SQL for Dash board and BI reporting system.

PeopleTech systems offer this software at affordable pricing along with Computer Telephony hardware and Software building blocks.

About PeopleTech systems Private Limited.

PeopleTech systems are a leading provider of Software and infrastructure solutions that enable Enterprises and telecommunication companies to deliver Voice, Data and multimedia services to their Customers. People tech has implemented turnkey telecom, IVR and contact center projects in India, Bangladesh, Malaysia, Singapore, USA, UK, and Middle East

Contact Email: sales@peopletech.co.in
Contact phone: +91-44-26260180
http://www.peopletech.co.in



Asterisk 1.8.2.2 Now Available (Security Release)
Click to view a printable version Thu, 20 Jan 2011 21:35:00 -0400

The Asterisk Development Team has announced a release for the security issue described in AST-2011-001.

Due to a failed merge, Asterisk 1.8.2.1 which should have included the security fix did not. Asterisk 1.8.2.2 contains the the changes which should have been included in Asterisk 1.8.2.1.

This releases is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases

The releases of Asterisk 1.4.38.1, 1.4.39.1, 1.6.1.21, 1.6.2.15.1, 1.6.2.16.2, 1.8.1.2, and 1.8.2.2 resolve an issue when forming an outgoing SIP request while in pedantic mode, which can cause a stack buffer to be made to overflow if supplied with carefully crafted caller ID information. The issue and resolution are described in the AST-2011-001 security advisory.

For more information about the details of this vulnerability, please read the security advisory AST-2011-001, which was released at the same time as this announcement.

For a full list of changes in the current release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.2.2

Security advisory AST-2011-001 is available at:

http://downloads.asterisk.org/pub/security/AST-2011-001.pdf

Thank you for your continued support of Asterisk!



Safi Communications Suite 1.5.0 Beta
Click to view a printable version Wed, 19 Jan 2011 18:53:35 -0400

Safi Systems have announced a new deployment of their communications suite (for graphically controlling Asterisk call flow).

Excerpt from their blog post:

We at Safi Systems are happy to announce the 1.5.0 Beta release of SafiWorkshop and SafiServer, now collectively referred to as the Safi Communications Suite (or SCS for short). It’s been a long time coming and if you’ve been waiting for us to finally put together another release, we appreciate your patience. We’ve been busy for the past several months, both with SCS product development as well as some SCS-related contracting opportunities that have come our way.

If you tried out the SafiServer/SafiWorkshop 1.3.0 Alpha; you know we experimented with an Apache Servicemix integration with the hopes of incorporating many of the powerful SOA-related features offered by that platform into the SafiServer. Unfortunately, integrating the two systems proved problematic and, in the end was deemed not worth the extra effort and resultant code bloat. Partially as a result of that effort, however, the SafiServer has been reworked and is now leaner, more modular, and more stable than ever before. Many of the features promised by the Servicemix integration such as email, file I/O, SNMP, SMS messaging, scheduling, and others will be implemented natively and will be appearing in the form of Safi ActionPaks in the near future.

Read More...



AST-2011-001: Stack buffer overflow in SIP channel driver
Click to view a printable version Wed, 19 Jan 2011 17:16:22 -0400

Asterisk Project Security Advisory - AST-2011-001

Product: Asterisk
Summary: Stack buffer overflow in SIP channel driver
Nature of Advisory: Exploitable Stack Buffer Overflow
Susceptibility: Remote Authenticated Sessions
Severity: Moderate
Exploits Known: No
Reported On: January 11, 2011
Reported By: Matthew Nicholson
Posted On: January 18, 2011
Last Updated On: January 18, 2011
Advisory Contact: Matthew Nicholson
CVE Name:

Description: When forming an outgoing SIP request while in pedantic mode, a stack buffer can be made to overflow if supplied with carefully crafted caller ID information. This vulnerability also affects the URIENCODE dialplan function and in some versions of asterisk, the AGI dialplan application as well. The ast_uri_encode function does not properly respect the size of its output buffer and can write past the end of it when encoding URIs.

Resolution: The size of the output buffer passed to the ast_uri_encode function is now properly respected.

In asterisk versions not containing the fix for this issue, limiting strings originating from remote sources that will be URI encoded to a length of 40 characters will protect against this vulnerability.

exten => s,1,Set(CALLERID(num)=${CALLERID(num):0:40})
exten => s,n,Set(CALLERID(name)=${CALLERID(name):0:40})
exten => s,n,Dial(SIP/channel)

The CALLERID(num) and CALLERID(name) channel values, and any strings passed to the URIENCODE dialplan function should be limited in this manner.

Affected Versions:

Product Release Series
Asterisk Open Source 1.2.x All versions
Asterisk Open Source 1.4.x All versions
Asterisk Open Source 1.6.x All versions
Asterisk Open Source 1.8.x All versions
Asterisk Business Edition C.x.x All versions
AsteriskNOW 1.5 All versions
s800i (Asterisk Appliance) 1.2.x All versions

Corrected In:

Product Release
Asterisk Open Source 1.4.38.1, 1.4.39.1, 1.6.1.21, 1.6.2.15.1,
1.6.2.16.1, 1.8.1.2, 1.8.2.1
Asterisk Business Edition C.3.6.2

Patches:
URL Branch
http://downloads.asterisk.org/pub/security/AST-2011-001-1.4.diff 1.4
http://downloads.asterisk.org/pub/security/AST-2011-001-1.6.1.diff 1.6.1
http://downloads.asterisk.org/pub/security/AST-2011-001-1.6.2.diff 1.6.2
http://downloads.asterisk.org/pub/security/AST-2011-001-1.8.diff 1.8

Asterisk Project Security Advisories are posted at
http://www.asterisk.org/security

This document may be superseded by later versions; if so, the latest version will be posted at
http://downloads.digium.com/pub/security/AST-2011-001.pdf and
http://downloads.digium.com/pub/security/AST-2011-001.html

Revision History
Date Editor Revisions Made
2011-01-18 Matthew Nicholson Initial Release

Asterisk Project Security Advisory - AST-2011-001
Copyright (c) 2011 Digium, Inc. All Rights Reserved.
Permission is hereby granted to distribute and publish this advisory in its original, unaltered form.



Weekly Asterisk Developer Conference Call
Click to view a printable version Wed, 19 Jan 2011 16:28:49 -0400

Bryan M. Johns has posted details about a new weekly Asterisk developer call:

Digium is happy to announce a new, weekly Asterisk developer conference call!

This call will begin on Thursday, January 27 2011 and will be held every Thursday thereafter.

Information, agendas and ICS downloads can be found on the Asterisk wiki here:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Weekly+Developer+Conference+Call

We look forward to your participation!

Bryan M. Johns
Digium, Inc. | Community Director
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
direct: +1 256-428-6007
Check us out at : www.asterisk.org or www.digium.com



OrderlyQ - The Answer to Call Center Queuing
Click to view a printable version Tue, 18 Jan 2011 17:35:15 -0400

OrderlyQ have sent through a note about their Call Center Queue Solution:

OrderlyQ is a unique add-on for Asterisk that is designed to ensure that your call center is as efficient as possible. If there are no agents available to answer a caller then the caller is given an estimated wait time, advised to call back after that time and a place is reserved in the queue for them so that they are answered when they call back.

OrderlyQ Features

Easy to install without changing existing set up.
Deploys automatically as required.
Free no-obligation trial.
Optimises call centers.

OrderlyQ Advantages

Cuts call abandonment - answer up to 41% more calls.
Improves caller satisfaction - complaints about queuing can be cut to less than 1 in 1000 calls.
Cost - much cheaper than 'agent callback' systems.

Email sales@orderlyq.com, or browse www.orderlyq.com



Asterisk Voicemail for the future - building blocks instead of monolithic design
Click to view a printable version Sun, 16 Jan 2011 17:33:28 -0400

Olle has written a post about some ideas for a new voicemail structure for Asterisk in 1.10.

Excerpt from his post:

Building services in Asterisk is about combining a lot of small services into larger building blocks. The Asterisk platform provides you with a large set of functions, like calling out, answering, recording, playing prompts and listening for DTMF. In combination with simple logic control statements, much like any other programming language, you can use these to build complex services. Asterisk very much follows the design principles behind the UNIX operating system: small building blocks combined with a script language may build very complex larger services.

Read more...



Asterisk 1.4.39, 1.6.2.16, 1.8.2 Now Available
Click to view a printable version Sun, 16 Jan 2011 16:56:33 -0400

The Asterisk Development Team has announced the release of Asterisk 1.4.39, 1.6.2.16 and 1.8.2.

I've separated out the release notes:

Asterisk 1.4.39

The Asterisk Development Team has announced the release of Asterisk 1.4.39. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.4.39 resolves several issues reported by the community and would have not been possible without your participation. Thank you!

The following is a sample of the issues resolved in this release:

  • Resolve issue where channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call.
    (Closes issue #18171. Reported by: SantaFox)
    (Closes issue #18185. Reported by: kwemheuer)
    (Closes issue #18211. Reported by: zahir_koradia)
    (Closes issue #18230. Reported by: vmarrone)
    (Closes issue #18299. Reported by: mbrevda)
    (Closes issue #18322. Reported by: nerbos)
  • Fix bugs in saying numbers using the Swedish language syntax
    (Closes issue #18355. Reported, patched by oej)
  • Fix not stopping MOH when transfered local channel queue member is answered. The problem here is only present when local channels are used with the MOH passthru option as well as no optimization (/nm).
    Patched by jpeeler.
  • Improve handling of REGISTER requests with multiple contact headers.
    Patched by jpeeler.
  • app_followme: Don't create a Local channel if the target extension does not exist.
    (Closes issue #18126. Reported, patched by junky)
  • Revert code that changed SSRC for DTMF.
    (Closes issue #17404, #18189, #18352. Reported by sdolloff, marcbou. rsw686. Tested by cmbaker82)
  • Resolve issue where REGISTER request with a Call-ID matching an existing transaction is received it was possible that the REGISTER request would overwrite the initreq of the private structure.
    (Closes issue #18051. Reported by eeman. Patched, tested by twilson)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.39

Thank you for your continued support of Asterisk!

Asterisk 1.6.2.16

The Asterisk Development Team has announced the release of Asterisk 1.6.2.16.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.6.2.16 resolves several issues reported by the community and would have not been possible without your participation. Thank you!

The following is a sample of the issues resolved in this release:

  • Fix cache of device state changes for multiple servers.
    (Closes issue #18284, #18280. Reported, tested by klaus3000. Patched, tested by russellb)
  • Resolve issue where channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call.
    (Closes issue #18171. Reported by: SantaFox)
    (Closes issue #18185. Reported by: kwemheuer)
    (Closes issue #18211. Reported by: zahir_koradia)
    (Closes issue #18230. Reported by: vmarrone)
    (Closes issue #18299. Reported by: mbrevda)
    (Closes issue #18322. Reported by: nerbos)
  • Linux and *BSD disagree on the elements within the ucred structure. Detect which one is in use on the system.
    (Closes issue #18384. Reported, patched, tested by bjm, tilghman)
  • app_followme: Don't create a Local channel if the target extension does not exist.
    (Closes issue #18126. Reported, patched by junky)
  • Revert code that changed SSRC for DTMF.
    (Closes issue #17404, #18189, #18352. Reported by sdolloff, marcbou. rsw686. Tested by cmbaker82)
  • Resolve issue where REGISTER request with a Call-ID matching an existing transaction is received it was possible that the REGISTER request would overwrite the initreq of the private structure.
    (Closes issue #18051. Reported by eeman. Patched, tested by twilson)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.16

Thank you for your continued support of Asterisk!

Asterisk 1.8.2

The Asterisk Development Team has announced the release of Asterisk 1.8.2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.2 resolves several issues reported by the community and would have not been possible without your participation. Thank you!

The following is a sample of the issues resolved in this release:

  • 'sip notify clear-mwi' needs terminating CRLF.
    (Closes issue #18275. Reported, patched by klaus3000)
  • Patch for deadlock from ordering issue between channel/queue locks in app_queue (set_queue_variables).
    (Closes issue #18031. Reported by rain. Patched by bbryant)
  • Fix cache of device state changes for multiple servers.
    (Closes issue #18284, #18280. Reported, tested by klaus3000. Patched, tested by russellb)
  • Resolve issue where channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call.
    (Closes issue #18171. Reported by: SantaFox)
    (Closes issue #18185. Reported by: kwemheuer)
    (Closes issue #18211. Reported by: zahir_koradia)
    (Closes issue #18230. Reported by: vmarrone)
    (Closes issue #18299. Reported by: mbrevda)
    (Closes issue #18322. Reported by: nerbos)
  • Fix reloading of peer when a user is requested. Prevent peer reloading from causing multiple MWI subscriptions to be created when using realtime.
    (Closes issue #18342. Reported, patched by nivek.)
  • Fix XMPP PubSub-based distributed device state. Initialize pubsubflags to 0 so res_jabber doesn't think there is already an XMPP connection sending device state. Also clean up CLI commands a bit.
    (Closes issue #18272. Reported by klaus3000. Patched by Marquis42)
  • Don't crash after Set(CDR(userfield)=...) in ast_bridge_call. Instead of setting peer->cdr = NULL, set it to not post.
    (Closes issue #18415. Reported by macbrody. Patched, tested by jsolares)
  • Fixes issue with outbound google voice calls not working. Thanks to az1234 and nevermind_quack for their input in helping debug the issue.
    (Closes issue #18412. Reported by nevermind_quack. Patched by dvossel)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.2

Thank you for your continued support of Asterisk!



Asterisk: The Definitive Guide, First Draft Complete
Click to view a printable version Thu, 13 Jan 2011 18:20:24 -0400

Leif Madsen has posted a link about the first draft of the new Asterisk book being completed.

Excerpt from his post:

Today we marked the finishing of Asterisk: The Definitive Guide (3rd Edition) by sending the first draft off to the O’Reilly production team. We finished nearly on schedule (within a few days, which is remarkable considering we added 150 more pages than originally intentioned) and have our favourite copy editor Rachel Head (formerly Wheeler) who did the first edition of Asterisk: The Future of Telephony who we emphatically enjoyed working with. We’re looking forward to it again.

Read more...



ZORG new C++ and Java ZRTP implementation public release
Click to view a printable version Wed, 12 Jan 2011 18:12:03 -0400

Andrea Cristofanini from PrivateWave has sent us across a press release about a new Open Source encryption implementation:

Milan, January 12th 2011 - PrivateWave Italia S.p.A, an Italian company engaged in developing technologies for privacy protection and information security in voice telecommunications, is pleased to announce the release of ZORG, a new Open Source ZRTP protocol implementation available for download from http://www.zrtp.org.

ZRTP provides end-to-end key exchange with Elliptic Curve Diffie-Hellmann 384bit and AES-256 SRTP encryption.

ZORG has been originally developed and implemented in PrivateWave's PrivateGSM voice encryption products available for the following platforms: Blackberry, Nokia and iOS (iPhone).

Zorg C++ has been integrated with PJSIP open source VoIP SDK and it's provided as integration patch against PJSIP 1.8.5. It has been tested on iPhone, Symbian, Windows, Linux and Mac OS X.

Zorg Java has been integrated within a custom version of MJSIP open source SDK on Blackberry platform and it includes memory usage optimizations required to reduce at minimum garbage collector activity.

Both platforms have separated and modular cryptographic back-ends so that the cryptographic algorithms implementation could be easily swapped with other ones.

ZORG is licensed under GNU AGPL and source code is available on github at https://github.com/privatewave/ZORG .

We are releasing it under open source and in coherence with our approach to security as we really hope that it can be useful for the open source ecosystem to create new voice encryption systems in support of freedom of speech.

More than 20 pjsip-based open source VoIP encryption software (several written in Java) could directly benefit from ZORG release.

We would be happy to receive proposal of cooperation, new integration, new cryptographic back-ends, bug scouting and whatever useful to improve and let ZRTP affirm as voice encryption standard.

Zorg is available from http://www.zrtp.org.



Open Source Telephony at FOSDEM 2011
Click to view a printable version Mon, 10 Jan 2011 18:11:38 -0400

Russell Bryant has posted details of the talks on telephony which will take place at FOSDEM 2011 on the 6th of February:

FOSDEM (Free and Open Source software Developers European Meeting) is an amazing conference held each year in Brussels, Belgium. I have been lucky enough to attend in 2009 and 2010. Both times I was very impressed with the number of attendees and the quality of the talks.

For FOSDEM 2011, I am proud to take part in the conference by organizing a day of talks on open source telephony. The talks will take place on Sunday, February 6th. Without any further delay, here are the talks that are scheduled for the open source telephony dev room at FOSDEM 2011.

  • “Introduction to Asterisk Developement”, Russell Bryant
  • “Digital PSTN Connectivity with Asterisk”, Jakub Klausa
  • “Mobicents 2.0, The Open Source Java Communication Platform”, Jean Deruelle
  • “Scaling location services in large SIP networks with Kamailio”, Henning Westerholt, Marius Zbihlei
  • “Unifying SIP and Web worlds with Lua (Kamailio)”, Daniel-Constantin Mierla
  • “XiVO IPBX OpenHardware”, Xavier Carcelle
  • “Unified Communications - Future (Yate and YateClient)”, Diana Cionoiu
  • “Asterisk SCF (Scalable Communications Framework)”, Kevin P. Fleming
  • “Developing rich VoIP applications with SIPSIMPLE SDK”, Saúl Ibarra Corretgé
  • “SIP Communicator: Building a Multi-Protocol Multi-OS Communications Client”, Emil Ivov

I hope to see you there!



Asterisk: The Definitive Guide -- Call For Reviewers
Click to view a printable version Mon, 10 Jan 2011 18:00:59 -0400

The next Asterisk book is ready for technical review - help out if you can!

Hey all!

We're getting VERY close to having the first draft of the next Asterisk book, Asterisk: The Definitive Guide ready to be sent off to production. We're very close to meeting our target dates, but our review timeline is very tight. Only about 2 weeks!

Each morning we're continuing to work on the book, taking in your comments, reviewing chapters, testing dialplan and installation steps, and all that good stuff.

However, we've been looking at this book since May 2010 and our eyes are starting to get glazed :) We'd love for the community to have a look at the book and offer some constructive criticism.

It's far too late to take requests for things to cover. What we have is what we're going to get in for this edition. After we finish this book though we plan on continuing to update it, so there will be a chance to take suggestions again soon.

For now, head on over to http://ofps.oreilly.com and check out the book (updated this morning). There are a couple of bugs in the OFPS software which are causing comments to not be available after chapter 8, but we're hoping to have those resolved by Friday. However, we do have this fancy mailing list that we can use.

Russell, Jim and myself will be monitoring this list for comments, and we'll try and get all of them satisfied before publication. If there is a particular area we're covering that you're an expert in, we'd love to have you focus on that chapter. You can email me back directly for more information on what we might be looking for in that type of situation.

We do have editors to help with grammar and spelling, but pointing anything out is certainly useful. The best use of your time though is testing the dialplan snippets, the installation instructions for both Ubuntu and CentOS (we're covering two Linux distributions this time around, which increases the testing load significantly), and making sure anything we're explaining is concise, relates to what we're talking about, and makes sense. The goal is to build an Asterisk system from scratch, so following through our dialplan via the chapters to make sure it all continues to build on itself would be ideal.

Additionally, if you see any sections which say, "see chapter XXX for more information" that are not links, please let us know, as those are meant to be placeholders until the chapters existed and we could link back to them. Now that all chapters are created, we should be linking to the appropriate locations. If you're reading a section and notice a good spot to reference another part of the book (for example, lets say we're talking about database functionality in one of the other chapters, and there is an appropriate spot in the Database Integration chapter to link to), then let us know!

Thanks for your interest! Books should be shipping sometime between March and April. Pre-orders are available now at http://oreilly.com/catalog/9780596517342. And yes, we'll be releasing under a Creative Commons license like the last two books, so you'll have access to the book at any time online.

This book has been pretty much written from the ground up, and is well over 600 pages of content. It's been a lot of work, but we hope you like it!

Thanks!
Russell, Jim and Leif.



AppKonference 1.6
Click to view a printable version Thu, 16 Dec 2010 19:52:10 -0400

Paul Albrecht has posted details of the latest release of AppKonference:

Hi,

I have released an updated AppKonference today. This fix release includes the following changes:

* Deprecated input smoothing and output packing because packet size is fixed at 20 milliseconds so these functions are unnecessary.

* Added a compiler flag, CACHE_CONTROL_BLOCKS, for conference and member block reuse. To disable control block reuse edit the Makefile or clear it on the command line: make CACHE_CONTROL_BLOCKS=0.

* Changed the conference event privilege to user to reduce the number of events an application has to monitor to support a conference.

You can download the latest source from source forge:

sourceforge.net/projects/appkonference

--
Paul Albrecht



Asterisk 1.8.1.1 Now Available
Click to view a printable version Thu, 16 Dec 2010 16:17:37 -0400

The Asterisk Development Team has announced the release of Asterisk 1.8.1.1.

This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.1.1 resolves two issues reported by the community since the release of Asterisk 1.8.1.

* Don't crash after Set(CDR(userfield)=...) in ast_bridge_call. Instead of setting peer->cdr = NULL, set it to not post.
(Closes issue #18415. Reported by macbrody. Patched, tested by jsolares)

* Fixes issue with outbound google voice calls not working. Thanks to az1234 and nevermind_quack for their input in helping debug the issue.
(Closes issue #18412. Reported by nevermind_quack. Patched by dvossel)

For a full list of changes in this release candidate, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.1.1

Thank you for your continued support of Asterisk!



Asterisk 1.4.38, 1.6.2.15, 1.8.1 Now Available
Click to view a printable version Wed, 08 Dec 2010 16:24:45 -0400

The Asterisk Development Team has announced the release of Asterisk 1.4.38, 1.6.2.15 and 1.8.1

These releases are available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

Asterisk 1.4.38

The release of Asterisk 1.4.38 resolves several issues reported by the community and would have not been possible without your participation. Thank you!

The following is a sample of the issues resolved in this release:

* Add ability for Asterisk to try both the encoded and unencoded subscription URI for a match in hints.
(Closes issue #17785. Reported, tested by ramonpeek. Patched by tilghman)

* Set the caller id on CDRs when it is set on the parent channel.
(Closes issue #17569. Reported, patched by tbelder)

* Ensure user portion of SIP URI matches dialplan when using encoded characters
(Closes issue #17892. Reported by wdoekes. Patched by jpeeler)

* Fix a crash in res_jabber by ensuring that we don't alter memory after it's freed.
(Closes issue #17387. Reported, tested by jmls. Patched by tilghman)

* Fix problem with qualify option packets for realtime peers never stopping. The option packets not only never stopped, but if a realtime peer was not in the peer list multiple options dialogs could accumulate over time.
(Closes issue #16382. Reported by lftsy. Tested by zerohalo. Patched by jpeeler)

* Multiple fixes related to Local channels.

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.38

Thank you for your continued support of Asterisk!

Asterisk 1.6.2.15

The release of Asterisk 1.6.2.15 resolves several issues reported by the community and would have not been possible without your participation. Thank you!

The following is a sample of the issues resolved in this release:

* When using chan_skinny, don't crash when parking a non-bridged call.
(Closes issue #17680. Reported, tested by jmhunter. Patched, tested by DEA)

* Add ability for Asterisk to try both the encoded and unencoded subscription URI for a match in hints.
(Closes issue #17785. Reported, tested by ramonpeek. Patched by tilghman)

* Set the caller id on CDRs when it is set on the parent channel.
(Closes issue #17569. Reported, patched by tbelder)

* Ensure user portion of SIP URI matches dialplan when using encoded characters
(Closes issue #17892. Reported by wdoekes. Patched by jpeeler)

* Resolve issue where Party A in an analog 3-way call would continue to hear ringback after party C answers.
(Patched by rmudgett)

* Fix problem with qualify option packets for realtime peers never stopping. The option packets not only never stopped, but if a realtime peer was not in the peer list multiple options dialogs could accumulate over time.
(Closes issue #16382. Reported by lftsy. Tested by zerohalo. Patched by jpeeler)

* Multiple fixes related to Local channels.

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.15

Thank you for your continued support of Asterisk!

Asterisk 1.8.1

The release of Asterisk 1.8.1 resolves several issues reported by the community and would have not been possible without your participation. Thank you!

The following is a sample of the issues resolved in this release:

* Fix issue when using directmedia. Asterisk needs to limit the codecs offered to just the ones that both sides recognize, otherwise they may end up sending audio that the other side doesn't understand.
(Closes issue #17403. Reported, patched by one47. Tested by one47, falves11)

* Resolve issue where Party A in an analog 3-way call would continue to hear ringback after party C answers.
(Patched by rmudgett)

* Fix playback failure when using IAX with the timerfd module.
(Closes issue #18110. Reported, tested by tpanton. Patched by jpeeler)

* Fix problem with qualify option packets for realtime peers never stopping. The option packets not only never stopped, but if a realtime peer was not in the peer list multiple options dialogs could accumulate over time.
(Closes issue #16382. Reported by lftsy. Tested by zerohalo. Patched by jpeeler)

* Fix issue where it is possible to crash Asterisk by feeding the curl engine invalid data.
(Closes issue #18161. Reported by wdoekes. Patched by tilghman)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.1

Thank you for your continued support of Asterisk!



Asterisk Media Architecture project
Click to view a printable version Wed, 08 Dec 2010 16:03:51 -0400

David Vossel has written a post about rewriting the way Asterisk deals with media:

Howdy,

I'm working on a long term project that involves completely reworking how media is handled in Asterisk. I have spent the last month researching solutions for the problems in Asterisk's current media architecture and creating a high level design document outlining my approach to fix them.

Here is a link to the design proposal on the new Asterisk wiki.
https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal

This document covers quite a bit of topics. The best place to start reading to decide if this is something you are interested in or not is the "Project Requirements" section, but I'm going through out some requirement keywords for you below to help out.

  • SILK
  • H.264
  • Improved translation path generation to work with all media types
  • Video transcoding
  • Multiple Steams with translation paths
  • No format bit field limits
  • Renegotiation media after call setup
  • Negotiating media formats containing attributes (Request video with the parameters you actually want)

I need your feedback. If any of this interests you then read the document, or at least the areas that concern you, and post your comments. Please keep this discussion on the -dev list.

Thanks!

David Vossel
Digium, Inc. | Batman Developer, Open Source Software



Pinana updates - distributed device states over SIP
Click to view a printable version Thu, 02 Dec 2010 20:31:43 -0400

Olle has posted details of his work on the Pinana branch to allow blinking lights to work over a distributed architechture:

Hi!

Brad Watkins (Marquis42) and I have been working on project pinana for a while. A lot of time was spent on architecture, something we locked down at Astricon and went from architecture work to coding. We now have the follwowing state:

- Asterisk can subscribe to remote device states by adding hints
- Asterisk can PUBLISH device state updates to a presence server

This way, Asterisk can subscribe to the state of an extension in another Asterisk server. If you have two servers that are load balanced somehow, you can have a hint that points to the local SIP device and to the state of a remote device in the other server. Regardless of where the call turns up, your state will change and force a change of the extension state and ... the lamp will start blinking!

The PUBLISH part still needs some polishing, but the idea here is to use a Kamailio SIP server to aggregate the device states and handle the distribution. All Asterisk servers PUBLISH to Kamailio and subscribe from kamailio. Each Asterisk server now can send ONE publish and Kamailio will distribute the update to the rest of the group. If you add or remove a server, you don't have to change the configuration of the rest of the servers.

We're at a state where Asterisk sends PUBLISH requests and Kamailio accepts them. It's a good start. There are issues with state changes during a PUBLISH transaction that we're working on handling properly now.

The code is written for 1.4 and will be ported to 1.8 and for inclusion in trunk. We based the subscribe part on the code for remote MWI subscriptions in trunk, and the PUBLISH part is using the EPA framework for trunk. We've found issues with both these architectures that we will propose fixes for in 1.8 and trunk.

If you want to test, the code is in the svn branch at http://svn.digium.com/svn/asterisk/team/group/pinana-publish-1.4

Note that we do work with this branch, so at some points it might not be in compilation state or even crash badly. If so, find us and we'll help you find some stable version to test.

This work is funded by Telekompetanse AS in Oslo, Norway for a project in Akkershus Fylke. Much of the architecture is based on earlier discussions with Sitelink in the UK. The Kamailio module for dialog-info presence that we used was developed by Klaus Darillion, another well-known Asterisk and Kamailio developer.

My personal feeling is that this will add a lot of functionality to Asterisk in large scale installations. It's something I wanted to do for many years. Feels great to see the bits and pieces coming together and actually work as planned!

Regards,
/Olle



Support text messages outside of a call
Click to view a printable version Thu, 02 Dec 2010 20:25:19 -0400

Russell has posted a reviewboard entry with the aim of supporting text messages outside of the context of a call - this will be quite a significant advance for Asterisk!

Here's the description from the reviewboard entry:

This branch contains a proposal for adding protocol independent support for processing text messages into and out of the dialplan, outside of a call. The file doc/asterisk-messaging.txt contains more details on the proposal. The introduction of the document is quoted here:

"Asterisk has some limited support today for messaging. The support that exists primarily includes passing text messages in the context of a call. The SIP and IAX2 protocols have support for this, but that's it.

There are a couple of other messaging protocols that are supported: Skype and XMPP (Jabber). The support of these is very minimal and not very integrated into the architecture of Asterisk since these messages are not in the context of a phone call. They provide a combination of dialplan and manager interface interfaces that are specific to each protocol. There just is no current architectural concept of dealing with text messages.

The purpose of this proposal is to introduce text messaging into the architecture of Asterisk. For messaging support to exist in the true spirit of Asterisk architecture, the design needs to achieve the following two goals:

a) Protocol Independence
b) Scriptable message routing

The rest of this document goes through some details about how these goals will be achieved in a way that is both architecturally compatible with Asterisk as well as practical to implement."

----------

In addition to the documented proposal, I have made some good progress on implementation. Here is what is done so far:

  • core modifications to allow sending incoming messages through the dialplan
  • core modifications to allow outbound messages from the dialplan
  • modifications to res_jabber to allow inbound and outbound messages in the new architecture
  • The beginning of changes to chan_sip, with support for inbound and outbound MESSAGE outside of a call





ATCOM release GSM Asterisk card
Click to view a printable version Tue, 30 Nov 2010 16:13:45 -0400

ATCOM has sent out a note to inform people about their new GSM card which uses DAHDI:

ATCOM officially releases its new PCI based telephony cards with GSM channels - AX4G which supports up to 4 GSM channels and fully compatible with open source Asterisk systems.

With this product, Asterisk users can perfectly integrate their fixed IP telephony system with the mobile nework.

The applications could be GSM connectivity for a PBX, Mobile PBX, GSM VoIP gateway, SMS gateway and GSM callback services.

For more information, please visit:

http://www.atcom.cn/AX4G.html



EasyITSP released to public
Click to view a printable version Mon, 29 Nov 2010 20:32:33 -0400

Matthew from lemens-ts.com has posted details of an Open Source ITSP project which works with Asterisk:

EasyITSP, an open source ITSP/Hosted PBX solution that sits on top of Digium's open source Asterisk PBX, has been released to the public. EasyITSP allows people with little experience to easily manage and run an ITSP and/or Hosted PBX services with its small footprint and its easy to use admin and customer portals. EasyITSP is quickly gaining ground in its market with constant development of new features. To learn more and view a demo, visit http://lemens-ts.com/easyitsp



AstLinux 0.7.4 Release now available
Click to view a printable version Sun, 28 Nov 2010 21:53:07 -0400

The AstLinux Team is happy to announce the release of AstLinux 0.7.4. This is a dual release which allows you to chose between Asterisk 1.4.36 or 1.8.0.

There are several security updates and other improvements. All current AstLinux users should upgrade as soon as feasible.

One of the more significant additions includes preliminary IPv6 support.

The two releases can be viewed here.

http://www.astlinux.org/release/074-asterisk-1436
http://www.astlinux.org/release/074-asterisk-180

A full changelog is available on those pages.

Current users can upgrade either from the web interface or via the command line.

upgrade-run-image check http://mirror.astlinux.org/firmware
(http://mirror.astlinux.org/ast18-firmware for Asterisk 1.8 firmware)
The version should be reported as 0.7.4

upgrade-run-image upgrade http://mirror.astlinux.org/firmware

The AstLinux Team



AstLinux Beta images available
Click to view a printable version Wed, 24 Nov 2010 17:36:53 -0400

Darrick Hartman has posted details of the availability of beta images for the 0.7 branch:

I forgot to send out the email, but beta images from the 0.7 branch are available to upgrade an existing install.

They are in the 'beta-firmware' repository.

upgrade-run-image check http://mirror.astlinux.org/beta-firmware
upgrade-run-image upgrade http://mirror.astlinux.org/beta-firmware

These are the Asterisk 1.4 images. I still need to build and upload the Asterisk 1.8 images from the same base.

Darrick



Asterisk 1.6 and Kamailio 3.1 realtime integration tutorial
Click to view a printable version Wed, 24 Nov 2010 16:56:40 -0400

Daniel-Constantin Mierla has posted an upgrade to his tutorial on integrating Asterisk and Kamailio:

Hello,

I got the time to upgrade my tutorial about Asterisk and Kamailio realtime integration to latest stable release of Kamailio, version 3.1.0 (out on Oct 6, 2010).

You can find the document at:

http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb

Besides making it work for v3.1.x, the Kamailio config file has some new features included:
* IP authentication - can be enabled via define WITH_IPAUTH
* TLS support - can be enabled via define WITH_TLS
- TLS to UDP translation and vice-versa is done automatically by Kamailio in case you configure Asterisk on UDP
* detection of DoS attacks - can be enabled via define WITH_ANTIFLOOD
- banning automatically traffic from attacker IP addresses for a specific time interval
* restructuring of configuration file for better modularity and highlighting of functionalities such as registrar, location server, within-dialog request routing

Hope it is useful for some people within this community.

Next step, naturally, is to upgrade the tutorial for latest Asterisk, 1.8.0, just need some time to get familiar with it.

Cheers,
Daniel

--

Daniel-Constantin Mierla
Kamailio (OpenSER) Advanced Trainings
Nov 22-25, 2010, Berlin, Germany
Jan 24-26, 2011, Irvine, CA, USA
http://www.asipto.com



libpri 1.4.11.5 Now Available
Click to view a printable version Tue, 23 Nov 2010 01:03:47 -0400

The Asterisk Development Team has announced the release of libpri 1.4.11.5.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri/

The release of libpri 1.4.11.5 resolves several issues reported by the community and would not have been possible without your participation.
Thank you!

The following are some of the issues resolved in this release:

* Prevent a CONNECT message from sending a CONNECT ACKNOWLEDGE in the wrong state.
(issue #17360. Reported by: shawkris. Patched by rmudgett)

* Made Q.921 delay events to Q.931 if the event could immediately generate response frames.
(closes issue #17360. Reported by: shawkris. Patched by rmudgett)

* BRI PTMP: Active channels not cleared when the interface goes down.
(closes issue #17865. Reported by: wimpy. Patched by rmudgett)

* Segfault in pri_schedule_del() - ctrl value is invalid.
(closes issue #17522)
(closes issue #18032. Reported by: schmoozecom. Patched by rmudgett)

* Crash when receiving an unknown/unsupported message type.
(closes issue #17968. Reported by: gelo. Patched by rmudgett)

* B410P gets incoming call packets on ISDN but Asterisk doesn't see the call.
(closes issue #18232. Reported by: lelio. Patched by rmudgett)

For a full list of changes in the current release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/libpri/ChangeLog-1.4.11.5

Thank you for your continued support of Asterisk!



AstriCon 2010 Keynote: The Future of Open Source Communications
Click to view a printable version Thu, 11 Nov 2010 19:11:37 -0400

Steven Sokol has posted a link to the video of the keynote speech by Kevin Fleming on Asterisk SCF.





Asterisk 1.6.2.14 Released
Click to view a printable version Thu, 11 Nov 2010 17:13:38 -0400

The Asterisk Development Team has announced the release of Asterisk 1.6.2.14. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.6.2.14 resolves several issues reported by the community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

* Fix issue where session timers would be advertised as supported even when session-timers=refuse was set in sip.conf. Also fix interoperability problems with session timer behavior in Asterisk.
(Closes issue #17005. Reported by alexcarey. Patched by dvossel)

* Parse all "Accept" headers for SIP SUBSCRIBE requests.
(Closes issue #17758. Reported by ibc. Patched by dvossel)

* Fix issue where queue stats would be reset on reload.
(Closes issue #17535. Reported by raarts. Patched by tilghman)

* Fix issue where MoH files were no longer rescanned on during a reload.
(Closes issue #16744. Reported by pj. Patched by Qwell)

* Fix issue with dialplan pattern matching where the specificity for pattern ranges and pattern characters was inconsistent.
(Closes issue #16903. Reported, patched by Nick_Lewis)

For a full list of changes in the current release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.14

Thank you for your continued support of Asterisk!



Asterisk 1.4.37 Released
Click to view a printable version Thu, 11 Nov 2010 16:52:06 -0400

The Asterisk Development Team has announced the release of Asterisk 1.4.37. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.4.37 resolves several issues reported by the community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

* Fix issue with decoding ^-escaped characters in realtime (res_pgsql)
(Closes issue #17790. Reported denzs. Patched by Qwell)

* Don't send a devstate change on poke_noanswer if the state did not change.
(Closes issue #17741. Reported, patched by schmidts)

* Transmit silence when reading DTMF in ast_readstring. Otherwise you could get issues with DTMF timeouts causing hangups.
(Closes issue #17370. Reported, patched by makoto)

* Fix to SIP extension state update (deadlock issues)
(Closes issue #17888. Reported by zerohalo. Patched by dvossel)

* Fix issue with MoH where it doesn't recover cleanly when it can't play a file and would just stop, instead of continuing to find the next playable file in the MoH class.
(Closes issue #17807. Reported by kshumard. Patched by bbryant)


For a full list of changes in the current release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.37

Thank you for your continued support of Asterisk!



Asterisk and Asterisk SCF Project Wiki
Click to view a printable version Wed, 03 Nov 2010 17:41:53 -0400

For those of you who may have missed the announcements made last week at AstriCon 2010, the Asterisk and Asterisk SCF projects now have a Wiki site available at:

https://wiki.asterisk.org

This site contains a great deal of Asterisk documentation, development plans and other content, with more to come. It also contains Asterisk SCF documentation, development history and much more.

The wiki site will allow you to keep track of what is posted/changed there in various ways, including RSS feeds, direct subscription to pages you are interested, and even subscription to entire project spaces. However, if you want to watch an entire project space, it will be more efficient for the wiki site if you instead subscribe to one of the new mailing lists we've setup at lists.digium.com.

The asterisk-wiki-changes list will get notified for all content changes and comments posted in the Asterisk project space; the asterisk-scf-wiki-changes list will get the same sort of notifications for the Asterisk SCF project space.



Asterisk community services powered by Atlassian tools
Click to view a printable version Tue, 02 Nov 2010 16:17:51 -0400

Kevin Fleming has posted a note to thank Atlassian for the free donation of their software tools to the Asterisk and Asterisk SCF projects:

Some of you have already noticed we've chosen a number of Atlassian tools to provide services to the Asterisk and Asterisk SCF communities (Confluence, Crowd, Crucible, Fisheye and Bamboo). Of course, we're not alone in this since many other open source projects have chosen these tools as well, but I'd just like to state again how happy we are that Atlassian is willing to license these tools at no cost to open source projects. The tools are really powerful, easy to use and flexible, and there's no doubt that our community will be better off for having chosen them.

Thanks again Atlassian!

--

Kevin P. Fleming
Digium, Inc. | Director of Software Technologies



New Asterisk Documentation wiki
Click to view a printable version Mon, 01 Nov 2010 17:59:58 -0400

The doc directory plus other documentation for Asterisk is moving to a new Asterisk wiki - separate from the voip-info wiki. Russell is asking is people have any comments:

Greetings,

As you have probably seen by now, there is a new project wiki available:

http://wiki.asterisk.org

This wiki was originally created to host documentation for the Asterisk SCF project. We then realized that it would be very valuable to host all of our Asterisk documentation there, as well. At this point, everything from the doc/ directory (from Asterisk 1.8) has been imported into the wiki. In addition, we have created a tool that syncs the XML based documentation from the Asterisk source with the wiki (applications, functions, AMI actions, AGI commands).

Now that the wiki has been made available, I would like to strip most of the documentation from the doc/ directory in favor of the wiki. It does not make sense to have the documentation in more than one place. The wiki supports exporting the documentation as a PDF, so we can still include the docs in a tarball.

Before I go and start ripping files out of the source tree, I wanted to post this to give people an opportunity to ask questions or provide feedback.

Thanks,

--

Russell Bryant
Digium, Inc. | Engineering Manager, Open Source Software



AstriDevCon 2010 Recap
Click to view a printable version Mon, 01 Nov 2010 17:53:06 -0400

Russell Bryant has posted a recap of AstriDevCon that occurred at Astricon.

Greetings,

We held a developer meeting on Friday, October 29th, immediately following AstriCon. Thank you very much to everyone that participated! I wanted to send out some notes on what we discussed for those that could not attend. The following page shows the topics that were discussed. Feel free to post any comments or questions to this list.

https://wiki.asterisk.org/wiki/display/AST/AstriDevCon+2010

Thanks,

--

Russell Bryant
Digium, Inc. | Engineering Manager, Open Source Software



Asterisk SCF
Click to view a printable version Sun, 31 Oct 2010 18:16:28 -0400

The Asterisk SCF project was announced at Astricon. It will be the world's first high performance, distributed, scalable, fault-tolerant, open source communications framework.

You can read the full write up on it over on the new Asterisk wiki.

An excerpt from that site:

The Asterisk SCF Mission Statement
Asterisk Scalable Communications Framework (SCF) is a highly-scalable, distributed, extensible open-source communications platform and application suite.

Asterisk SCF Goals

  • Scalability
  • High Availability and Fault Tolerance
  • Extensibility
  • Performance

Asterisk SCF Is
Asterisk SCF is a collection of fine-grained software components that focus on very specific functional tasks using a message-oriented and distributed architecture. Asterisk SCF provides a flexible deployment model, taking advantage of modern multi-core processors and operating across servers in multiple locations. Asterisk SCF provides a set of building blocks for communications applications, interfaces and services.

Asterisk SCF Achieves
Asterisk SCF achieves interoperability between a wide variety of of extensible voice, video, and data communication technologies using an extensible set of Media, Channel, Routing, Bridging and Discovery Services.

Asterisk SCF Works
Asterisk SCF works by creating fined-grained components that interoperate via a publicly-defined messaging API, where the opportunity exists at many points for new components to interject themselves into the system, either as replacements for other components, intermediaries between components, or observers of the system interactions.

Asterisk SCF defines a wide array of publishable events, as well as hook mechanisms for altering the operation of individual components. Configurable Discovery Mechanisms in Asterisk SCF are designed to ensure that new technologies and components can be rapidly integrated.

Scalability in Asterisk SCF is achieved by supporting load-balancing among the various system components. With each major function implemented in its own software module, users can make, during deployment, resource allocation decisions based on specific needs.



AstLinux: Beta Release with Asterisk 1.8.0
Click to view a printable version Wed, 27 Oct 2010 05:53:14 -0300

Darrick Hartman has posted details of the latest beta releases of AstLinux with the released version of Asterisk 1.8.0:

After several additional changes, we finally have a fairly complete release that contains Asterisk 1.8.0 (final). As with most Asterisk releases, I would be hesitant to use this on a production system until it's been tested a bit more. If you have the ability to test, please do so.

More information about the release is available here:

http://www.astlinux.org/release/07-4578-asterisk-180

A member of the community has stepped up and has been working on a new gui installer for use from Windows. It needs to be tested more, but is something that should be available for our next official release. This work is based on NSIS from my understanding.

Please report back your experiences with this beta release. Unless there are major issues reported, we will do a dual release with both Asterisk 1.8 and Asterisk 1.4 available as separate images in the next week or so.

Darrick



QueueMetrics 1.6.2 released
Click to view a printable version Mon, 25 Oct 2010 21:36:24 -0300

Lenz has posted details of the latest release of QueueMetrics:

This release of QueueMetrics fixes a number of issues that were present in previous releases and improves graphs in the main section, especially multi-line and SLA graphs.
We also offer native, high-performance support for 64-bit host systems.

We added Twitter and Facebook as easy means of contact - if you have an account on either, we suggest to add us so you can receive news and updates in real-time.

This release works with the same activation keys you used on your current version of QueueMetrics.

As always, we value your feedback on how to improve QueueMetrics to make it a better fit to your needs.

Enjoy the update,

Lorenzo Emilitri
Managing Director, Loway

---

Important changes

  • Improved graphs: report graphs have been improved so that they are easier to read - two-value distributions used to be hard to read, because it was hard to tell which line meant which dataset. This is now easier to read because we added legends.
  • New multi-period logic for SLA graphs: it is now possible to have SLA timing broken down in two parts, e.g. every 5 seconds up to 30 seconds and every 10 seconds up to 100 seconds. This works for all SLA graphs.
  • Many common issues fixed
  • Direct support for 64 bit systems: the new RPM installers will install an optimized JVM when running on 64-bit systems, that are now often used as hosts for QM.

Major changes

These are the major upgrades over the previous versions:

#1162 - New graph: Inclusive Answered SLA [UN18]
#1163 - New multi-period logic for SLA timing
#1172 - QM now shows the supervisor's name and not logins
#1181 - Improved distribution graphs. Now with data legends.
#1185 - Broken SLA graphs
#1178 - 64-bit JVM

Minor changes and bugs fixed

#1180 - Taken-lost call graphs were not visible in 1.6.x
#1177 - The INFO IVR field was not reported correctly in multi-stint mode
#1152 - Graphs would not be correct for non-latin languages
#1144 - Error when creating some SQL queries
#1145 - Error for expired session on call detail popup
#1146 - Error when selecting locale
#1160 - Report list would crash if no report was available
#1170 - Rare error exporting to XML
#1175 - Turned off logging for "Unknown Verb"
#1182 - An extension is required for hotdesking agents

FAQ:
#1139 - New FAQ on using "Sliding Window" mode and on why having logged-on agents is a good idea
#1173 - Wrong links on the FAQs

---


Stay in touch quickly with Facebook and Twitter

You can easily stay up-to-date with what's new in QueueMetrics by adding us to your Twitter or Facebook account:

- on Twitter, we are at http://twitter.com/queuemetrics

- on Facebook, we are at http://www.facebook.com/group.php?gid=265016726083

We will post last minute and development news on both sites.

---

New offices

Loway will be moving to a new, larger office space by the end of next month.

Our contact information will change and - most important - out main telephone number will be different. We will post on the website as soon as we have the correct contact information.

---

QueueMetrics translations

QueueMetrics is now available in the following languages:

  • German
  • English
  • Spanish / Spain
  • Spanish / Latin America
  • Italian
  • French
  • Portuguese
  • Polish
  • Russian
  • Dutch
  • Hungarian
  • Slovak
  • Japanese

Thanks to our translators for making it possible!

If you would like to help translate QueueMetrics to a new language, please contact us.

New support portal

In order to improve the reliability and speed of our support, we have just rolled out a new support portal for our clients. This way it is easy for you to keep track of the statuses of your support requests, and no more e-mail is lost due to over-zealous anti-spam filters.

The new support portal is located at support.loway.ch and you can get your account and open up a ticket just by sending an email to support@loway.ch. If you would like to try it now, do not hesitate to send a test email to get your access credentials.

How to upgrade from a previous version

If you installed originally using yum, you should simply type:

yum update queuemetrics

and then point your browser to http://myserver:8080/queuemetrics/dbtest to update the database.

You should then copy the web.xml and configuration.properties from the old release to the new one.

Do not forget to make a backup copy of the database before running the update (just in case...)

---

Loway
via Ligornetto 17/a
6854 San Pietro
Switzerland

tel. +41 91 630-9886
fax. +39 0332 3568135

http://www.loway.ch
http://queuemetrics.com

All texts and data are copyright by Loway and may be re-used only with indication of their source.



Encrypted mobile to landline phone calls with Asterisk
Click to view a printable version Mon, 25 Oct 2010 04:46:10 -0300

Andrea Cristofanini from PrivateWave has posted a hugely detailed howto explaining the process required to set up encrypted mobile to landline phone calls with Asterisk 1.8, Snom 300 and PrivateGSM Enterprise:

How-to-asterisk.1.8_SRTPTLS_snom300_pgsm.pdf

Here's a snippet from the introduction:

Introduction
GSM has been cracked and and now mobile phone calls can be intercepted with cheap hardware and Open Source tools.
VoIP can be intercepted with simple windows point and click VoIP interception tool.
VoIP encryption protocols exists to protect from eavesdropping:

  • SRTP does end-to-site voice encryption
  • SDES does key exchange for SRTP
  • SIP/TLS encrypt signaling channel over which SDES keys are exchanged


Till now Asterisk, the most used Open Source telephony engine, did not support it but Asterisk 1.8 finally supports SRTP voice encryption with SDES key exchange!
This howto explains the building of an integrated mobile to landline phone calling platform with:

  • Asterisk 1.8 RC2
  • PrivateGSM Enterprise for mobile secure calling (iPhone, Blackberry and Nokia S60)
  • Snom 300 for landline secure calling

The procedures described below will guide you through the following steps:
1. Install Asterisk
2. Configure Asterisk
3. Configure Snom 300
4. Configure PrivateGSM Enterprise
5. Make secure calls
Please note that the security provided is end-to-site, that means that no one except the PBX system administrator can
eavesdrop in on the call.
If you need end-to-end security (typical government need) you need to use ZRTP that does end-to-end encryption.



Mobile phone anti-tapping solution
Click to view a printable version Sun, 24 Oct 2010 20:07:36 -0300

PrivateWave, the secure voice communications specialist, today launches an anti-tapping solution, designed to ensure the safe transfer of information between mobile phones and protect businesses from the loss of confidential information - a bigger priority than ever, with industrial espionage estimated to cost international businesses over 126 billion pounds every year.

Using its PrivateGSM software, PrivateWave has developed the Enterprise VoIP Security Suite (EVSS) which is designed to secure phone conversations between landline VoIP phones and mobile devices, protecting business communications. The software is compatible with Nokia, iPhone and Blackberry mobile devices and will soon be fully compatible with devices based on Android.

The development of the EVSS is the result of the encryption of mobile communications and the development of secure VoIP solutions. This integrated system guarantees full protection of voice communications from intrusions, securing all PBX conversations from landline phones to mobile devices and vice versa, resulting in maximum privacy of all employee conversations.

"With traditional channels of identity theft now closing, individuals are increasingly targeting unprotected voice conversations to obtain confidential information as it is almost always uncharted territory for businesses when considering security options," says Carlo Marchini, CEO, PrivateWave. "Recent wiretapping or 'phone hacking' stories reported in the media reveal how easy it has become for individuals to create low-cost illegal phone tapping systems and how this type of interception is a growing threat", he adds. "It has therefore become increasingly important, particularly from a financial and business point of view, for companies to adopt a robust anti-tapping solution that will protect these private conversations".

Created from the recent merger by incorporation of zerozero39 into the former Khamsa Italia, PrivateWave's plans for the UK market are part of a broader company strategy to strengthen its presence in foreign markets and in key target sectors including business enterprise, public sector, not for profit organisations, and business professionals.

Marchini concludes: "Our Enterprise VoIP Security Suite is the first solution in the world able to secure company mobile-PBX and fixed phone conversations which are often at risk of illegal tapping. Bringing our solution to the UK market is an exciting opportunity for us and we are confident it will be a success".

About PrivateWave
PrivateWave was founded in 2005 with the aim of bring innovation to the secure communications market. Since the beginning, the company chose to cooperate with universities leveraging on the incubation by Acceleratore d'Impresa of the Milan Politecnico and CPStartup of the Switzerland University. Once completed the development of the technology, in late October 2009 the company became an entirely Italian private company and then - in mid October 2010 - changed its name in PrivateWave Italia SpA, following the merger with zerozero39 srl. PrivateWave Italia expects to close 2010 with a revenue of about 1 million Eur and a growth target of 2.5 million Eur in 2011. For further information, please visit www.privatewave.com



Asterisk 1.8.0 Now Available
Click to view a printable version Thu, 21 Oct 2010 17:43:32 -0300

The Asterisk Development Team is proud to announce the release of Asterisk 1.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

Asterisk 1.8 is the next major release series of Asterisk. It will be a Long Term Support (LTS) release, similar to Asterisk 1.4. For more information about support time lines for Asterisk releases, see the Asterisk versions page.

http://www.asterisk.org/asterisk-versions

The release of Asterisk 1.8.0 would not have been possible without the support and contributions of the community. Since Asterisk 1.6.2, we've had over 500 reporters, more than 300 testers and greater than 200 developers contributed to this release.

You can find a summary of the work involved with the 1.8.0 release in the summary:

http://svn.asterisk.org/svn/asterisk/tags/1.8.0/asterisk-1.8.0-summary.txt

A short list of available features includes:

  • Secure RTP
  • IPv6 Support in the SIP channel driver
  • Connected Party Identification Support
  • Calendaring Integration
  • A new call logging system, Channel Event Logging (CEL)
  • Distributed Device State using Jabber/XMPP PubSub
  • Call Completion Supplementary Services support
  • Advice of Charge support
  • Much, much more!

A full list of new features can be found in the CHANGES file.

http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=markup

For a full list of changes in the current release candidate, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0

Thank you for your continued support of Asterisk!



Asterisk: The Definitive Guide book available for pre order and public review
Click to view a printable version Wed, 20 Oct 2010 19:40:35 -0300

The 3rd edition of the Asterisk book is on its way and available for public review and pre-order!

Today the 3rd edition of the O'Reilly published Asterisk book took big leap forward, by becoming available for pre-order, and public review!

You can find information about the book and pre-order it here:
http://oreilly.com/catalog/9780596517342

Additionally, you can sign up and help us make it the best book on Asterisk ever by helping to review it! The review site is brand new and we may run into some bugs and such, but this is a big step in the right direction. Check it out here:

http://ofps.oreilly.com

Feedback and comments encouraged and welcomed! Note that the book is still being written, so you may see some section blank.

Thanks!
Leif Madsen, Russell Bryant, and Jim Van Meggelen



Xorcom wins best of show at it expo west 2010
Click to view a printable version Wed, 20 Oct 2010 17:44:04 -0300

Xorcom, a privately-held manufacturer of business telephony interfaces and appliances based on Asterisk open source software, announces today that its Complete Concierge solution for hospitality, comprised of the Xorcom Asterisk®-based PBX and MICROS Fidelio-certified PMS (Property Management System) interface, received the "Best of Show" award in the "Open Source" category at TMC's ITEXPO West 2010 held earlier this month in Los Angeles, California.

Integrated PBX and PMS Improves Staff Efficiency and Guest Satisfaction

Complete Concierge by Xorcom is a cost-effective, brand-new alternative to traditional hotel and motel telephony systems. It leverages existing hotel infrastructure, such as cabling and analog phone sets, and has its own built-in call accounting software package with programmable rate tables. For properties that are already using a Property Management System, the Complete Concierge integrates seamlessly with the leading PMS packages, such as Micros Fidelio Opera (who certified the Complete Concierge interface as interoperable with their PMS in July 2009), Protel, Amadeus, Optima, Brilliant and Newhotel. On the other hand, for smaller properties that cannot justify the investment in a PMS package, Complete Concierge provides its own built-in PMS communications functionality as well. It also supports SIP phones.

"Xorcom's Complete Concierge, based on the Asterisk open source telephony platform, is the most cost-effective hotel telephone system on the market," claims Eran Gal, Xorcom CEO. "Hotels can now upgrade their telephone system to reap the benefits of open source telephony, without sacrificing PMS integration or hospitality-specific features."

New Standards Set by TMC at ITEXPO
"ITEXPO in LA was the largest and most successful event we've ever produced, and this year's Best-of-Show winners faced an even greater challenge differentiating themselves from a large and highly competitive field," said Rich Tehrani, CEO and group editor-in-chief of TMC. "We're proud to honor companies like Xorcom with a Best-of-Show Award, and we thank them for sharing their innovative solutions with our attendees."

About the Best of Show Awards
The ITEXPO Best of Show Awards recognize innovative products and solutions that were featured in the Exhibit Hall during the event. This designation serves to highlight the technological achievement and creative product features that set these winners apart from the rest of the crowd. TMC's editorial staff evaluated the pre-show nominations, and met with vendors to inspect and review the displayed products on the first day of the Expo. A full list of the winners is posted on TMC's Web site at www.tmcnet.com.

Availability
Complete Concierge is being demonstrated in the Xorcom booth #316 at next week's AstriCon conference and exhibition, being held in National Harbor, Maryland from Oct. 26-28. Complete Concierge is available for sale now from Xorcom\'s worldwide reseller channel. A price quote for a configuration matching specific requirements, and the location of a nearby reseller, can be obtained by completing this form.

About Xorcom
Founded in 2004, Xorcom is a privately-held IP-PBX manufacturer. Xorcom harnesses the power of Asterisk® Open Source IP-PBX the most rapidly growing telephony platform in the world to design and produce leading-edge hardware telephony solutions for commercial installations. Today, Xorcom offers the widest and most flexible range of solutions and hardware platforms in the Asterisk market. System integrators, telecom equipment manufacturers, and customer premise telephony and VoIP providers use Xorcom products to provide added value to their end users. Xorcom sells its products via a worldwide distribution channel and OEM partners. For more information, visit: www.xorcom.com.



AstLinux: Beta images with Asterisk 1.8
Click to view a printable version Wed, 20 Oct 2010 17:38:35 -0300

Darrick Hartman has posted details of some beta AstLinux images with Asterisk 1.8 on them:

Beta images are available for testing with Asterisk 1.8.0-rc3

The release page is here:

http://www.astlinux.org/release/07-4556-asterisk-180-rc3

You probably want to start with a new set of config files for Asterisk as the changes are significant.

http://svnview.digium.com/svn/asterisk/branches/1.8/UPGRADE.txt?view=markup

Please provide feedback on these images or the ISO if you test.

Darrick
--
Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com



Pre-Astricon Greeting from Allison Smith
Click to view a printable version Mon, 18 Oct 2010 21:52:39 -0300

Allison (the voice of Asterisk) has posted a blog entry about the upcoming Astricon conference.

An excerpt from the article:

I have been asked to put my Asterisk Blog Series “The 15 Commandments of IVR” on hold, briefly, in order to blog about the most highly- anticipated yearly event in telephony– AstriCon — which is rapidly approaching. All eyes are on National Harbor, MD — an exciting, cosmopolitan location for the convention, and quite a departure from our usual idyllic desert setting of recent years.

For those prospective attendees who need a little extra convincing of *any* kind as to why one should attend AstriCon, I’ve boiled down some essential points which reinforce the truism of why AstriCon is a “must-attend” for anyone who is *anyone* in the Asterisk ecosystem.

Read More...



Asterisk 1.8.0 Release Candidate 5 Now Available
Click to view a printable version Mon, 18 Oct 2010 20:57:47 -0300

The Asterisk Development Team has announced the fifth release candidate of Asterisk 1.8.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

With all currently 1.8.0 blocker issues closed, Asterisk 1.8.0-rc5 is currently scheduled to become the full release of Asterisk 1.8.0.

All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any issues found to the issue tracker, https://issues.asterisk.org/. It is also very useful to see successful test reports. Please post those to the asterisk-dev mailing list.

Asterisk 1.8 is the next major release series of Asterisk. It will be a Long Term Support (LTS) release, similar to Asterisk 1.4. For more information about support time lines for Asterisk releases, see the Asterisk versions page.

http://www.asterisk.org/asterisk-versions

The release of Asterisk 1.8.0-rc5 was triggered by some last minute platform compatibility IPv6 changes. In addition, the availability of the English sound prompts with Australian accents has been added.

A full list of new features can be found in the CHANGES file.

http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=markup

For a full list of changes in the current release candidate, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc5

Thank you for your continued support of Asterisk!



Asterisk 1.8.0 Release Candidate 4 Now Available
Click to view a printable version Mon, 18 Oct 2010 18:18:11 -0300

The Asterisk Development Team has announced the fourth release candidate of Asterisk 1.8.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

With all currently 1.8.0 blocker issues closed, Asterisk 1.8.0-rc4 is currently scheduled to become the full release of Asterisk 1.8.0.

All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any issues found to the issue tracker, https://issues.asterisk.org/. It is also very useful to see successful test reports. Please post those to the asterisk-dev mailing list.

Asterisk 1.8 is the next major release series of Asterisk. It will be a Long Term Support (LTS) release, similar to Asterisk 1.4. For more information about support time lines for Asterisk releases, see the Asterisk versions page.

http://www.asterisk.org/asterisk-versions

With the availability of the Asterisk 1.8.0 release candidates, the binary add-on modules for Asterisk produced by Digium have been updated with new versions that are compatible with Asterisk 1.8. The availability of these modules will assist with the testing of Asterisk 1.8.0 in a wider variety of situations.

This release candidate contains fixes since the last release candidate as reported by the community. A sampling of the changes in this release candidate include:

* Additional fixups in chan_gtalk that allow outbound calls to both Google Talk and Google Voice recipients. Adds new chan_gtalk enhancements externip and stunaddr.
(Closes issue #13971. Patched by dvossel)

* Resolve manager crash issue.
(Closes issue #17994. Reported by vrban. Patchd by dvossel)

* Documentation updates for sample configuration files.
(Closes issues #18107, #18101. Reported, patched by lathama, lmadsen)

* Resolve issue where faxdetect would only detect the first fax call in chan_dahdi.
(Closes issue #18116. Reported by seandarcy. Patched by rmudgett)

* Resolve issue where a channel that is setup and torn down *very* quickly may not have the right call disposition or ${DIALSTATUS}.
(Closes issue #16946. Reported by davidw. Review
https://reviewboard.asterisk.org/r/740/)

* Set TCLASS field of IPv6 header when SIP QoS options are set.
(Closes issue #18099. Reported by jamesnet. Patched by dvossel)

* Resolve issue where Asterisk could crash on shutdown when using SRTP.
(Closes issue #18085. Reported by st. Patched by twilson)

* Fix issue where peers host port would be lost on a SIP reload.
(Closes issue #18135. Reported, tested by lmadsen. Patched by dvossel)

A short list of available features includes:

* Secure RTP
* IPv6 Support in the SIP channel driver
* Connected Party Identification Support
* Calendaring Integration
* A new call logging system, Channel Event Logging (CEL)
* Distributed Device State using Jabber/XMPP PubSub
* Call Completion Supplementary Services support
* Advice of Charge support
* Much, much more!

A full list of new features can be found in the CHANGES file.

http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=markup

For a full list of changes in the current release candidate, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc4

Thank you for your continued support of Asterisk!



AstLinux: Beta images and ISO
Click to view a printable version Sun, 17 Oct 2010 19:56:24 -0300

Darrick Hartman has posted a note about new beta images and an ISO for AstLinux:

I just finished uploading some beta images and an ISO from SVN 4552 from the 0.7 branch. This image includes several updates and new features. (specifically IPv6 support and IMAP voicemail storage). These beta images are still based on Asterisk 1.4.36 and are ONLY AVAILABLE FOR geni586 and net5501 (ISO only on geni586)

This is a BETA test. DO NOT use on production equipment unless you understand the risks and back up your configurations prior to doing so. I'm looking for feedback on the 'install' process both when booted from the Live CD and when attempting to install from a Windows PC. That process has changed significantly and actually works now (or at least it should work).

To install from the ISO boot the iso, then type 'install' at the boot prompt. The web page that loads in Windows should be pretty straight-forward.

Please CAREFULLY read and understand the following before testing:

1) Asterisk

If you are using an existing config, you WILL need to add a line to the asterisk/modules.conf file or asterisk WILL crash. This is due to adding a new module which allows for voicemail storage in IMAP. For new installations, this will be done automatically (we've added a noload line to the default modules.conf file found in /stat/etc/asterisk.

noload => app_voicemail_imap.so

If you want to try the imap storage, you would then need to load this, and noload app_voicemail.so instead. Only one of the two can be loaded or you will have issues.

We will be including this option in future AstLinux releases. It's still not decided if we'll include it in Asterisk 1.4 releases or only in AstLinux releases that contain Asterisk 1.8 (when that is released).

2) Firewall:

The fireall configuration MUST be backed up if you want/need to revert at any time in the future. Those files are /mnt/kd/arno-iptables-firewall/* and /mnt/kd/rc.conf.d/gui.firewall.conf The version of Arno's firewall included in this build includes some changes which will be handled by the upgrade-arno-firewall script and web interface, but cannot be reverted from the web interface.

If you've read all that and want to continue, you can upgrade your existing test box by using the 'beta-firmware' repository.

upgrade-run-image check http://mirror.astlinux.org/beta-firmware

It should report astlinux-0.7-4552 (or newer)

If it does, upgrade with:

upgrade-run-image upgrade http://mirror.astlinux.org/beta-firmware

Again, this will ONLY work on net5501 and geni586 systems. I have not built these beta images for other platforms yet. These are the two most commonly used platforms based on web stats.

The ISO is available here:

http://mirror.astlinux.org/downloads/iso/geni586/astlinux-0.7-4552.iso

Please test the 'install' function from the boot prompt or the bat file install from a Windows machine. Be VERY careful to select the right target device in Windows.

Looking forward to some feedback on these latest changes.

Darrick
--

Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com



AstriCon Update
Click to view a printable version Thu, 14 Oct 2010 18:35:56 -0300

John Todd has posted an article about the upcoming Astricon conference - less than two weeks away!

Excerpt from the article:

AstriCon is less than two weeks away! If you haven’t booked your flight to Washington DC, now’s your chance! The main hotel (the Gaylord) is pretty booked, but that’s OK – there are still rooms a few hundred feet away at some of the hotels around the complex (Aloft, Wyndham, Hampton Inn, Residence Inn) and there are more hotels within a short drive/cab of the venue.

Speakers at AstriCon

We’ve got some great last-minute speakers to announce – I’m pleased to say that Ruben Sousa will be giving a talk on one of the largest open-source Asterisk installations in the world (100,000 users, 184 servers) which is arrayed across the University system in Portugal. We have a really solid line-up this year of talks focused on security and scalability from Kevin Lynn, Sandro Gauci, the Great Olle Johansson, and more! Many of the most active community developers, integrators, and speakers will be on hand, along with some very interesting announcements from Digium including the yearly roadmap and status update from the Digium engineering group – don’t miss out on hearing what’s new and what’s coming up! With the huge number of features that have been added to 1.8, it’s possible that you’ll learn from someone at the show how the newest release of Asterisk can benefit your organization in a way you never expected.

Read More...



Twin Cities Business: Call Control
Click to view a printable version Thu, 14 Oct 2010 18:29:27 -0300

Digium has posted a link to a press release about Asterisk use in the call centre.


Excerpt from the article:

When Pete Hainey started his call center business in 1993, few call center agents had computer monitors.

Today, fewer and fewer have phones.

Hainey is president and founder of Customer Elation, a Bloomington company that manages call center operations for retail, health care, and other organizations. Like most call centers these days, its agents receive calls through their computers—possible with voice over Internet protocol (VOIP), which allows telephone calls to travel over the same connections that carry e-mail and Internet data.

During the past five years, VOIP has overtaken traditional telephony infrastructure as the new standard in call centers, bringing with it a flurry of new features for call routing and reporting. VOIP has drastically reduced long-distance fees, enabled easy integration of home agents and remote facilities, and significantly lowered the cost of hardware.

“All of these are really huge changes in what was a fairly predictable, staid industry 10 years ago,” says Steve Sokol, marketing director for Asterisk, a free software program that runs on Linux and allows call centers to tap into VOIP, sponsored by Alabama-based Digium.

Read More...



AstriCon 2010 Conference Sessions Spotlight
Click to view a printable version Wed, 13 Oct 2010 18:10:01 -0300

Digium has sent out a mailer about some of the conference sessions at the upcoming Astricon (26th-28th of October 2010).

A great deal of the value you get from your investment in AstriCon is the first-class education you receive in the conference sessions.

At AstriCon 2010 in D.C., your full conference pass gives you access to all four individual tracks, plus pre-conference workshops on Tuesday, October 26.

The program this year is focused on the 'now' of Asterisk, and the all-important 'What's next?' Here's a look at a few key sessions you won't want to miss while you are there.

  • Just how vulnerable is your phone system?
    Presented by: Sandro Gauci
    Sandro is the author of the well-known "SIPVicious" password penetration test kit, which is used by both white hat tiger teams as well as others as one of the most popular brute-force attack systems on SIP platforms. He'll talk about methods by which SIP platforms are attacked on the Internet, what vulnerabilities administrators should look for, and how mitigate these issues on Asterisk. Don't miss this talk by one of the most well-known names in the SIP security community!
  • Orbitz & Asterisk: How We Did It
    Presented by Jim Kerr
    Orbitz implemented Asterisk in their infrastructure to handle customer calls. Learn how this huge travel management agency integrated Asterisk into their legacy system, how they deployed across many call centers, and how they achieved PCI security compliance with their installation.
  • Asterisk Development Update
    Presented by Russell Bryant
    Russell is the lead for Open-Source Asterisk development at Digium. Hear about what's been happening in the last year with Asterisk development, and what features and functions will be showing up in the near future. This is a standing-room only session each year; don't miss the chance to hear this informal and information- packed session with the core Asterisk developers.
  • Connecting Systems Together
    Presented by Jim Van Meggelen
    Jim is one of the authors of the famous "Asterisk: The Future of Telephony" book. His latest area of focus involves connecting disparate Asterisk systems together, which for any medium or large-sized organization is always a challenge. How to manage endpoints? What routing method should be used to connect offices? How about connecting Asterisk to global routing networks of other organizations? Get your island of phones connected after listening to Jim's talk.
  • Ask the Experts
    Presented by a panel of Digium Experts
    This is the AstriCon equivalent of "Stump The Band". We'll have a panel of Digium Sales Engineers, Support staff, and Engineers who will give some short examples of the toughest problems they've encountered and solved, and then the audience will be invited to throw out their most difficult problems to see if there can be some lightning insights into the issue. Some they'll solve, some they won't, but it'll be enlightening either way!
  • IPv6 in Asterisk 1.8
    Presented by Simon Perreault
    IPv6 is coming to Asterisk! The Internet is running low on IPv4 address space, and many large providers in Asia, Europe, and North America are moving towards IPv6 as a solution. Asterisk will support IPv6 in 1.8, and Simon will talk about the changes, configuration requirements, and the fundamentals of IPv6 required to get Asterisk running. Then, as part of a double-long session, Simon will walk through configuration of an entire platform (operating system and Asterisk and softclient) to get calls working on a purely IPv6 network. This is a vital talk to see if your network is moving to IPv6!


Register now and take advantage of the unmatched education and networking opportunities at AstriCon 2010 in our new East Coast location.

AstriCon is the one place every year where the full community of Asterisk developers, end-users, and hardware manufacturers meet to discuss progress, and to learn from each other. By not being there, you not only miss the opportunity to improve your own knowledge, but to improve your business as well.
See you in D.C. in three weeks.
The AstriCon Team



Asterisk 1.8.0 Release Candidate 3 Now Available
Click to view a printable version Sun, 10 Oct 2010 18:32:08 -0300

The Asterisk Development Team has announced the third release candidate of Asterisk 1.8.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any issues found to the issue tracker, https://issues.asterisk.org/. It is also very useful to see successful test reports. Please post those to the asterisk-dev mailing list.

Asterisk 1.8 is the next major release series of Asterisk. It will be a Long Term Support (LTS) release, similar to Asterisk 1.4. For more information about support time lines for Asterisk releases, see the Asterisk versions page.

http://www.asterisk.org/asterisk-versions

With the availability of the Asterisk 1.8.0 release candidates, the binary add-on modules for Asterisk produced by Digium have been updated with new versions that are compatible with Asterisk 1.8. The availability of these modules will assist with the testing of Asterisk 1.8.0 in a wider variety of situations.

This release candidate contains fixes since the release candidate as reported by the community. A sampling of the changes in this release candidate include:

* Still build chan_sip even if res_crypto cannot be built (use, but not depend)
(Reported by a user on the mailing list. Patched by tilghman)

* Get notifications for call files only when a file is closed, not when created
(Closes issue #17924. Reported by mkeuter. Patched by abeldeck)

* Fixes to chan_gtalk to allow outbound DTMF support to work correctly. Gtalk expects the DTMF to arrive on the RTP stream and not via jingle DTMF signalling.
(Patched by dvossel. Tested by malcolmd)

* Fixes to allow chan_gtalk to communicate with the Gmail web client.
(Patched by phsultan and dvossel)

* Fix to GET DATA to allow audio to be streamed via an AGI.
(Closes issue #18001. Reported by jamicque. Patched by tilghman)

* Resolve dnsmgr memory corruption in chan_iax2.
(Closes issue #17902. Reported by afried. Patched by russell, dvossel)

A short list of available features includes:

* Secure RTP
* IPv6 Support in the SIP channel driver
* Connected Party Identification Support
* Calendaring Integration
* A new call logging system, Channel Event Logging (CEL)
* Distributed Device State using Jabber/XMPP PubSub
* Call Completion Supplementary Services support
* Advice of Charge support
* Much, much more!

A full list of new features can be found in the CHANGES file.

http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=checkout

For a full list of changes in the current release candidate, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc3

Thank you for your continued support of Asterisk!



Integrics releases Enswitch 3.3
Click to view a printable version Sun, 10 Oct 2010 18:15:42 -0300

Integrics are pleased to announce the release of Enswitch 3.3, a complete integrated solution for commercial telephony services such as:

  • Full featured multi-tenant hosted PBX.
  • ITSP (Internet Telephony Service Provider).
  • SIP trunking.
  • VoIP for WiFi providers.
  • Toll-free and number translation services.
  • Calling cards.

Enswitch is in production today with carriers worldwide on systems from hundreds of users on single machines to over 150,000 users on redundant/failover clusters. A list of new features can be found at:

http://integrics.com/products/enswitch/guides/latest/en/presales/new/

More links, including a full list of features and a working demo of the web interface, are at:

http://integrics.com/products/enswitch/



Star2Billing News
Click to view a printable version Thu, 07 Oct 2010 16:22:28 -0300

Areski has posted details of lots of news articles, including a new version of A2Billing, a new version of CDR Stats, a desk at Astricon, a speech at VoIP2Day and a flight to The Village Telco Project.

Here are some excerpts from the news stories:

A2Billing 1.8.1 has been released. It includes the usual round of bug fixes, as well as some new call-back features sponsored by our customers, one being the way we answer the call to make A2Billing more compatible with Mobile Phone Call-Back Dialler software, and another feature called IVR call-back, where the customer is prompted for the number to call first, then the system calls them back and connects them.

--

Some months ago, we released version 1 of CDR-Stats, to replace the 8 year old Asterisk–Stat. Development of CDR-Stats has continued apace, and we are now pleased to release version 1.1.0

--

Star2Billing have been given an Open Source desk by Digium at Astricon in Washington this year, and both Joe and Areski will be attending.

--

Star2Billing have been invited to speak on their popular IP-Centrex and Multi-Tenant PBX solution at the VoIP2Day conference in Madrid, and will be giving an overview of how this product is assembled out of Open Source components including A2Billing.

--

Areski is flying down to South Africa this week to see the official launch of the Village Telco Project which is sponsored by the Shuttleworth Foundation.

--

You can read the full articles over here.



OpenSIPS Bootcamp training and certification
Click to view a printable version Thu, 07 Oct 2010 16:16:55 -0300

We've had an article posted:

OpenSIPS bootcamp training and certification in Edison, NJ USA, from Nov 15-19.

Learn how to build a SIP server using OpenSIPS

The OpenSIPS 1.6 Bootcamp is a full 5 day (40 hours) intensive training providing in depth coverage of OpenSIPS Installation, Configuration and Administration. All the knowledge that is transferred to you will be strongly backed-up by practice sessions where you will get hands-on experience in handling OpenSIPS SIP Server. The training is structured to offer 50% - 50% between theoretical and practical sessions. Asterisk users can use OpenSIPS for SIP load balancing or use OpenSIPS for large installations integrated with Asterisk as a media server or SIP gateway.

Certification

Optionally, you can take the OCP (OpenSIPS Certified Professional) in the end of the training to proof your knowledge and become an OpenSIPS guru.

More information at http://www.opensips.org/Training or by email bootcamp_at_opensips.org



Tropo Now Speaks Asterisk Gateway Interface (AGI)
Click to view a printable version Sun, 03 Oct 2010 19:42:55 -0300

Jason Goeke has posted a link to an article about Tropo support for Asterisk via AGI.

Excerpt from the article:

The Asterisk community is a vibrant one, one that we actively support through our sponsorship and advocacy of Adhearsion. We have decided to take it a step further and created a Tropo Scripting application that turns Tropo into a giant Asterisk application platform in the cloud. You can now run just about any Asterisk AGI application on Tropo.

Tropo AGItate was started on the Nerd Bird (good to have in-flight WiFi) from San Jose to Austin, on my way to LoneStar Ruby Conference. Jim Freeze – the organizer of LSRC – had recently been to AdhearsionConf in San Francisco; I wanted to be able to show something extra special during my talk there. On that one flight, I was able to get the basics working and show Tropo emitting AGI during my talk, just like that. (For those non-Asterisk folks out there, AGI is an API that lets external applications connect in to Asterisk and fully control it).

Read more...:



Scheduling automated calls between two participants with res_calendar
Click to view a printable version Wed, 29 Sep 2010 20:56:42 -0300

Leif has posted an awesome snippet from the next edition of the Asterisk book.

Excerpt from his blog post:

Here is a little dialplan snippet I wrote this morning for the next edition of the Asterisk book. While I’m not going to delve into all the aspects of setting up res_calendar like we do in the book, I thought for those of you who might already have this working might enjoy it.

Read More...



Asterisk IMAP and Gmail
Click to view a printable version Tue, 28 Sep 2010 19:56:20 -0300

Leif Madsen has posted an article on integrating Asterisk Voicemail with Gmail.

Excerpt from his blog post:

Today I was working on the next edition of the Asterisk book and wanted to see if I could get Asterisk IMAP voicemail support to work with Gmail. I had tried doing this a few times in the past without success, but since I had spent some time documenting and testing against Dovecot last week for another client and gotten everything working, I figured I had a good base to start trying to connect to the Gmail IMAP servers.

Read More...



AstLinux 0.7.3 released
Click to view a printable version Tue, 28 Sep 2010 19:22:37 -0300

Darrick Hartman has posted details of the release of AstLinux version 0.7.3:

The AstLinux Team is happy to announce the release of AstLinux 0.7.3. This update contains mostly bug fixes and security updates. All current users of AstLinux are encouraged to update to this release.

Updating can be performed from the web interface or from the command line using a few simple commands.

For the Changelog and other instructions, please visit:

http://www.astlinux.org/release/073

Enjoy,

Darrick



First Micros Fidelio Certified Asterisk Interface Enables Open Source PBX to Penetrate Hospitality Market
Click to view a printable version Tue, 21 Sep 2010 08:03:11 -0300

Xorcom, a privately-held manufacturer of business telephony interfaces and appliances based on Asterisk open source software announces Complete Concierge, a new product comprised of an Asterisk-based PBX with an integrated property management system (PMS) interface. The Complete Concierge allows businesses in the hospitality sector (hotels, motels, resorts, spas, etc.) to reap the benefits of open source telephony while utilizing the leading Property Management System (PMS) software in one bundled solution.

Integrated PBX and PMS Improves Staff Efficiency and Guest Satisfaction
Complete Concierge by Xorcom is a cost-effective, brand-new alternative to traditional hotel and motel telephony systems. It leverages existing hotel infrastructure, such as cabling and analog phone sets, and has its own built-in call accounting software package with programmable rate tables. For properties that are already using a Property Management System, the Complete Concierge integrates seamlessly with the leading PMS packages, such as Micros Fidelio Opera, Protel, Amadeus, Optima, Brilliant and Newhotel. On the other hand, for smaller properties that cannot justify the investment in a PMS package, Complete Concierge provides its own built-in PMS communications functionality as well. It also supports SIP phones.

Comprehensive Communications for Hotel Guests and Staff
Complete Concierge is a complete, all-powerful system that enables guests to access their own private communications from various sources. Plus, Complete Concierge supports flexible staff-to-guest, and staff-to-staff communication procedures to help maintain an efficient, effective service.

Hospitality Communications Features


  • Integrated wake-up

  • Integrated operator panel

  • Front desk check-in and check-out

  • Guest&#146;s name applied to room upon check-in

  • Vacant status applied to room upon check-out

  • Opening of phone lines at guest check-in

  • Closing of phone lines at guest check-out

  • Opening of guest voicemail box upon check-in

  • Purging of voice messages upon check-out

  • Built-in room status system

  • Built-in mini-bar management



General Communications Features

  • Unlimited number of extensions

  • Extension groups

  • Auto attendants

  • Voicemail boxes

  • Voicemail-to-email

  • Fax-to-email

  • Fring for iPhone and Droid integration

  • Call monitoring

  • Call recording

  • Integrated paging over IP telephone speakers

  • Ring groups

  • Custom messages

  • Music-on-hold



  • Management Features
    Complete Concierge comes with secure Web-based administration and a rapid recovery system for reliable system back-up and restore. It supports up to 1,500 analog ports and seamlessly switches between PSTN and VoIP.

    System Requirements


    • Xorcom IP-PBX &#150; model to match property requirements, such as number and type of telephony interfaces
    • Windows PC (XP and up, Windows server not required)



    Availability
    Complete Concierge is available now from Xorcom\'s worldwide reseller channel. To receive a price quote for the configuration that matches your requirements, and locate the nearest Xorcom partner, go to this Web page.

    Contact Info
    Ruth Bridger, VP Marketing, Xorcom Ltd.
    Tel: +972 (0)4 995-1972
    e-mail: ruth.bridger@xorcom.com

    About Xorcom
    Founded in 2004, Xorcom is a privately-held IP-PBX manufacturer. Xorcom harnesses the power of Asterisk® Open Source IP-PBX &#150; the most rapidly growing telephony platform in the world &#150; to design and produce leading-edge hardware telephony solutions for commercial installations. Today, Xorcom offers the widest and most flexible range of solutions and hardware platforms in the Asterisk market. For more information, visit: www.xorcom.com.

    Asterisk is a registered trademark of Digium, Inc. Complete Concierge is a trademark of Xorcom, Ltd. All other trademarks are the property of their respective owners.



    Submit your own stories
    Click to view a printable version Mon, 20 Sep 2010 21:45:29 -0300

    We have made a pretty big update to the Daily Asterisk News. You can now submit your own stories! There is now a link at the top of the Daily Asterisk News. Just click the link, fill out the form and we will check over your story and submit it.

    This is all under trial at the moment, so bear with us while we test it out and see what type of articles are submitted.

    They all need to be related to Asterisk, and I'll be reviewing each article, so keep the HTML down and make the articles relevant.



    Asterisk 1.6.2.12 and 1.6.2.13 Now Available
    Click to view a printable version Wed, 15 Sep 2010 21:28:49 -0300

    The Asterisk Development Team has announced the release of Asterisk 1.6.2.12 and 1.6.2.13.

    These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

    The release of Asterisk 1.6.2.12 resolves several issues reported by the community and would have not been possible without your participation.
    Thank you!

    The following is a sample of the issues resolved in this release:

    * Fix issue where DNID does not get cleared on a new call when using immediate=yes with ISDN signaling.
    (Closes issue #17568. Reported by wuwu. Patched by rmudgett)
    * Several updates to res_config_ldap.
    (Closes issue #13573. Reported by navkumar. Patched by navkumar, bencer.
    Tested by suretec)
    * Prevent loss of Caller ID information set on local channel after masquerade.
    (Closes issue #17138. Reported by kobaz, patched by jpeeler)
    * Fix SIP peers memory leak.
    (Closes issue #17774. Reported, patched by kkm)
    * Add Danish support to say.conf.sample
    (Closes issue #17836. Reported, patched by RoadKill)
    * Ensure SSRC is changed when media source is changed to resolve audio delay.
    (Closes issue #17404. Reported, tested by sdolloff. Patched by jpeeler)
    * Only do magic pickup when notifycid is enabled.
    A new way of doing BLF pickup was introduced into 1.6.2. This feature adds a call-id value into the XML of a SIP_NOTIFY message sent to alert a subscriber that a device is ringing. This option should only be enabled when the new 'notifycid' option is set, but this was not the case. Instead the call-id value was included for every RINGING Notify message, which caused a regression for people who used other methods for call pickup.
    (Closes issue #17633. Reported, patched by urosh. Patched by dvossel. Tested by: dvossel, urosh, okrief, alecdavis)

    For a full list of changes in the current release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.12

    Thank you for your continued support of Asterisk!

    Update: 1.6.2.13 released as well:

    This release resolves an issue where the .version and ChangeLog files were not updated for 1.6.2.12. Asterisk 1.6.2.13 has no additional changes from 1.6.2.12 other than the .version, ChangeLog and summary files.

    For a full list of changes in the current release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.13

    Thank you for your continued support of Asterisk!



    Asterisk 1.4.36 Now Available
    Click to view a printable version Wed, 15 Sep 2010 21:27:10 -0300

    The Asterisk Development Team has announced the release of Asterisk 1.4.36. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

    The release of Asterisk 1.4.36 resolves several issues reported by the community and would have not been possible without your participation.
    Thank you!

    The following is a sample of the issues resolved in this release candidate:

    * Fix issue where DNID does not get cleared on a new call when using immediate=yes with ISDN signaling.
    (Closes issue #17568. Reported by wuwu. Patched by rmudgett)

    * Fix issue where SIP promiscuous redirect could fail to dial the redirect (app_queue).

    * Fixes issue with translator frame not getting freed. This issue prevented G.729 licenses from being freed up.
    (Closes issue #17630. Reported by manvirr. Patched by dvossel)

    * Ensure SSRC is changed when media source is changed to resolve audio delay.
    (Closes issue #17404. Reported, tested by sdolloff. Patched by jpeeler)

    * Q931 - Sending PROGRESS after sending ALERTING is a protocol error.
    (Closes issue #17874. Reported, patched by nic_bellamy)

    For a full list of changes in the current release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.36

    Thank you for your continued support of Asterisk!



    Asterisk 1.8.0-beta5 Now Available
    Click to view a printable version Mon, 13 Sep 2010 20:09:35 -0300

    The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta5.

    This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

    All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any issues found to the issue tracker, http://issues.asterisk.org/. It is also very useful to see successful test reports. Please post those to the asterisk-dev mailing list.

    Asterisk 1.8 is the next major release series of Asterisk. It will be a Long Term Support (LTS) release, similar to Asterisk 1.4. For more information about support time lines for Asterisk releases, see the Asterisk versions page.

    http://www.asterisk.org/asterisk-versions

    This release contains fixes since the last beta release as reported by the community. A sampling of the changes in this release include:

    * Fix issue where TOS is no longer set on RTP packets.
    (Closes issue #17890. Reported, patched by elguero)

    * Change pedantic default value in chan_sip from 'no' to 'yes'

    * Asterisk now dynamically builds the "Supported" header depending on what is enabled/disabled in sip.conf. Session timers used to always be advertised as being supported even when they were disabled in the configuration.
    (Related to issue #17005. Patched by dvossel)

    * Convert MOH to use generic timers.
    (Closes issue #17726. Reported by lmadsen. Patched by tilghman)

    * Fix SRTP for changing SSRC and multiple a=crypto SDP lines. Adding code to Asterisk that changed the SSRC during bridges and masquerades broke SRTP functionality. Also broken was handling the situation where an incoming INVITE had more than one crypto offer.
    (Closes issue #17563. Reported by Alexcr. Patched by twilson)

    Asterisk 1.8 contains many new features over previous releases of Asterisk.
    A short list of included features includes:

    * Secure RTP
    * IPv6 Support in the SIP Channel
    * Connected Party Identification Support
    * Calendaring Integration
    * A new call logging system, Channel Event Logging (CEL)
    * Distributed Device State using Jabber/XMPP PubSub
    * Call Completion Supplementary Services support
    * Advice of Charge support
    * Much, much more!

    A full list of new features can be found in the CHANGES file.

    http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=checkout

    For a full list of changes in the current release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-beta5

    Thank you for your continued support of Asterisk!



    AstriDevCon signups
    Click to view a printable version Thu, 09 Sep 2010 18:11:52 -0300

    John Todd has posted a note about the AstriDevCon occurring during the AstriCon conference:

    [shaking the trees again for developers who are interested in meeting face-to-face in October - sign up on the form linked below, please! We need to figure out how many tables, what size room, etc. - it seems that quite a few of you who I know are coming haven't yet filled in the form. Please just let us know you'll be attending, even if you've talked with Russell or Kevin or myself directly... we have short memories. :-) ]


    This AstriCon will see the re-establishment of the AstriDevCon - the mini-conference within a conference! This is a full-day, discussion-oriented meeting where core developers can have direct and uninterrupted access to each other in a more direct way than usual. These sessions often result in massively productive results, either as new features are contemplated and developed, or as old problems are finally vanquished through the application of many eyes and hands.

    AstriDevCon will be on Friday, October 29. It's expected to be the whole day in one of the rooms from AstriCon - from 9:00ish until... when it ends that evening. It's expected that there will be some activity the prior evening as well, as AstriCon ends in the afternoon but many of you may already be in town and gravity ends up pulling people towards the room...

    Seats are limited. Preference will be given to developers who are active participants in the issue tracker, patch submission, and asterisk-dev mailing lists and IRC channels. There is no charge for attending AstriDevCon, though registration is required.

    IMPORTANT: If you know someone who is an Asterisk developer, but doesn't frequent the asterisk-dev list, please forward this to them. There are a surprising number of people who make significant contributions but who are possibly out of the range of this message, and we want to ensure everyone who qualifies knows about this opportunity.

    Register here:
    http://bit.ly/astridevcon2010

    The DevCon is fairly unstructured, but it is typically the case that there are discussions on:

    - security issues
    - priority of work issues
    - general brainstorming on new features
    - difficult bugs in the issuetracker
    - protocol (SIP, IAX2, etc.) questions and decisions
    - integration of third-party applications

    JT

    ---
    John Todd
    Digium, Inc. | Asterisk Open Source Community Director



    Asterisk 1.8.0-beta5 Now Available
    Click to view a printable version Wed, 08 Sep 2010 20:59:36 -0300

    The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta5.

    This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

    All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any issues found to the issue tracker, http://issues.asterisk.org/. It is also very useful to see successful test reports. Please post those to the asterisk-dev mailing list.

    Asterisk 1.8 is the next major release series of Asterisk. It will be a Long Term Support (LTS) release, similar to Asterisk 1.4. For more information about support time lines for Asterisk releases, see the Asterisk versions page.

    http://www.asterisk.org/asterisk-versions

    This release contains fixes since the last beta release as reported by the community. A sampling of the changes in this release include:

    * Fix issue where TOS is no longer set on RTP packets.
    (Closes issue #17890. Reported, patched by elguero)

    * Change pedantic default value in chan_sip from 'no' to 'yes'

    * Asterisk now dynamically builds the "Supported" header depending on what is enabled/disabled in sip.conf. Session timers used to always be advertised as being supported even when they were disabled in the configuration.
    (Related to issue #17005. Patched by dvossel)

    * Convert MOH to use generic timers.
    (Closes issue #17726. Reported by lmadsen. Patched by tilghman)

    * Fix SRTP for changing SSRC and multiple a=crypto SDP lines. Adding code to Asterisk that changed the SSRC during bridges and masquerades broke SRTP functionality. Also broken was handling the situation where an incoming INVITE had more than one crypto offer.
    (Closes issue #17563. Reported by Alexcr. Patched by twilson)

    Asterisk 1.8 contains many new features over previous releases of Asterisk.
    A short list of included features includes:

    • Secure RTP
    • IPv6 Support in the SIP Channel
    • Connected Party Identification Support
    • Calendaring Integration
    • A new call logging system, Channel Event Logging (CEL)
    • Distributed Device State using Jabber/XMPP PubSub
    • Call Completion Supplementary Services support
    • Advice of Charge support
    • Much, much more!

    A full list of new features can be found in the CHANGES file.

    http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=checkout

    For a full list of changes in the current release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-beta5

    Thank you for your continued support of Asterisk!



    libpri 1.4.11.4 Now Available
    Click to view a printable version Thu, 02 Sep 2010 21:23:20 -0300

    The Asterisk Development Team has announced the release of libpri 1.4.11.4.

    This release is available for immediate download at
    http://downloads.asterisk.org/pub/telephony/libpri/

    The release of libpri 1.4.11.4 resolves several issues reported by the community and would have not been possible without your participation.

    Thank you!

    The following are some of the issues resolved in this release:

    * Fix issue where calling name is not successfully processed on inbound QSIG PRI calls from Mitel PBX.
    (Closes issue #17619. Reported by: jims8650. Patched by rmudgett)

    * Added missing code specified by Q.921 (Figure B.8 Page 85) when receive RNR in "Timer Recovery" state.
    (Closes issue #16791. Reported by: alecdavis. Patched by alecdavis)

    * Fixed issue where incoming calls specifying the channel using a slot map could not negotiate a B channel correctly.

    * Add support to receive ECMA-164 2nd edition OID name ROSE messages.

    * Fixed issue where ISDN BRI PTMP TE does not recover from line faults.
    (Closes issue #17570. Reported by: jcovert. Patched by rmudgett)

    * Q.921 improvements from comparing Q.921 SDL diagrams with implementation.

    * Q.921/Q.931 message debug output improvements.

    For a full list of changes in the current release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/libpri/ChangeLog-1.4.11.4

    Thank you for your continued support of Asterisk!



    New CDR Stats Package
    Click to view a printable version Tue, 31 Aug 2010 23:12:18 -0300

    This one has been a long time coming. A new CDR stats package from Areski to replace the 7 year old stalwart for viewing Asterisk call detail records.

    While writing the previous article about Trixbox/FreePBX I came across a link to Areski's latest creation and thought I'd share it with you all!

    cdr stats screen shotHere's a description from the CDR-Stats web page:

    "CDR-Stats is a CDR viewer for Asterisk Call Data Records. (The Freeswitch version is under development). It allows you to interrogate your CDR to provide reports and statistics via a simple to use, yet powerful, web interface."

    As we've come to expect with Areski's work (I'm kinda assuming it's his work here through the myriad of linkages), the layout is beautiful, clean and to the point.

    Here's a description of the product (released under AGPL3) and Star2Billing:

    The motivation behind those who create open source software is their belief in quality and the hope that the users that find their creation useful, and that those users will donate their time and money to make it even better.

    This does not just apply to those with programming skills, for instance even those with no programming knowledge can make immense contributions by reporting bugs, writing documentation, and helping other new users to understand CDR-Stats.

    Another way that you can contribute to the the development of CDR-Stats is to purchase consultancy from us, as the revenue from this goes back into improving our products. This may not be the case with third party consultants.

    Star2Billing S.L. was formed in 2009 to provide commercial support, installation and training to telecommunications companies that wished to deploy A2Billing.

    A2Billing is both a Telecoms Switch and a Billing Platform, first launched in 2005, for those wishing to provide telecom services such as VoIP, calling cards and callback, as well as a variety of other products. Please see http://www.star2billing.com for a list of some of the products that Star2Billing can provide.

    CDR-Stats is the latest piece of software development from the Star2Billing stable, and is a complete ground up re-write of the famous Asterisk-Stat, which has been around for over 7 years as a CDR viewer and analyser for Asterisk, and has been included in FreePBX to view and analyse CDR.



    GUI changes from Trixbox, FreePBX, 2600hz, BlueBox
    Click to view a printable version Tue, 31 Aug 2010 22:39:16 -0300

    Ok, bear with me on this one. If you understand all the ramifications, FreePBX has split to a new project called BlueBox contained within the 2600hz project. This obviously has implications for Trixbox that uses FreePBX to provide quite a bit of functionality.

    So let's start at the start :)

    This news article involves a few products. We'll start with TrixBox and breakdown some of the things that go into making it. The situation is a bit like an onion. A whole lot of layers that we can peel back.

    trixboxSo, Trixbox is: http://fonality.com/trixbox/. Trixbox used to be Asterisk @ Home (obviously it ended up being bigger than something for just home users).

    Trixbox (I know, it is supposed to have a lower case T) contains various Open Source applications all bundled up into an ISO distribution (basically that means that it can be distributed as a DVD or CD and installed from there).

    Some of the products which Trixbox contain are:

    Areski Asterisk CDR Stats
    Flash Operator Panel
    FreePBX
    And many many more.

    freepbxOk, so the last one on that list (other than many many more) is FreePBX.

    Here's a description from their website:

    FreePBX is an easy to use GUI (graphical user interface) that controls and manages Asterisk, the world's most popular open source telephony engine software. FreePBX has been developed and hardened by thousands of volunteers over tens of thousands man hours. FreePBX has been downloaded over 5,000,000 times and estimates over 500,000 active phone systems. If you don't know about FreePBX, you are probably paying too much for your phone system.

    So, FreePBX is basically the GUI which is used to configure Asterisk within Trixbox (I know I'm simplifying a bit).

    A few weeks back, a news story was posted about the fact that version 3 of FreePBX was being spun off into its own project:

    http://www.freepbx.org/news/2010-08-03/v3-spun-off-to-give-it-full-independence

    An excerpt:

    "It became clear that the best thing that could be done for both projects was to spin v3 off into its own identitty thus allowing both FreePBX and the new rewrite to flourish and serve the community in the best possible way. The new project, still run by Darren, is named 2600hz Project and will be the new home to allow v3 to flourish, while FreePBX (v2) continues to evolve and serve the large installed base that it enjoys today."

    Which brings us along to this newly formed 2600hz project.

    An excerpt from their website:

    2600hz is home to a collection of open-source telephony software that enables the use of the FreeSWITCH, Asterisk and YATE switching libraries. Initially built around the blue.box project, we aim to provide a collection of software to power your GUI, your cloud-based telephony switch and/or your monitoring and maintenance tool set.

    We're nearly there!

    So, the core of the 2600hz project is the blue.box project: http://www.voipkb.com/wiki/index.php/About_BlueBox

    Description from that page:

    BlueBox, formerly known as FreePBX v3, is an open-source project run as part of the 2600hz project. It is a rename of the FreePBX v3 project. The project is still run by all the original developers who were active on the FreePBX v3 project, but is now sponsored and operated solely by the 2600hz Foundation.

    The project name and home was changed after issues arose in regards to legacy leadership, community confusion about ownership/direction and varied requirements that stemmed from the original FreePBX v2 project.

    The project remains under heavy development with additional features and services being added weekly.

    So! Where to from here.

    FreePBX is apparently going to continue on its own path. Trixbox will likely still contain FreePBX (or will it move to blue.box) and in all of this I've found a new release of Areski's CDR stats - will post a news story straight after this one :)

    Would love to know your thoughts!



    RazorQuotePBP Asterisk Payment Module
    Click to view a printable version Tue, 31 Aug 2010 00:30:52 -0300

    RazorQuote has sent us a press release about the launch of RazorQuotePBP, a native Asterisk module that allows any Asterisk connected device to accept credit card payments.

    Here's the press release for RazorQuote

    Boca Raton, FL. – August 30, 2010 – RazorQuote, the discount merchant services company, today launched RazorQuotePBP™, a native Asterisk module that allows any Asterisk® connected device to accept credit card payments. Using RazorQuotePBP, the Asterisk development community can now create professional, e-commerce, pay-by-phone applications using Asterisk®.

    This initial release of RazorQuotePBP includes the ability to accept payments, issue credits, and void charges. It features support for multiple merchant accounts, AVS, CVV, “verify input” mode and a full web based administrative and reporting interface. Future versions will include voice recognition, recurring payment, check processing, secure card information storage and support for alternative payment systems like PayPal® and Google Checkout®. RazorQuotePBP supports both the 1.4 and 1.6 versions of Asterisk and is available in both 32bit and 64bit versions.

    “We are committed to bringing a full featured, PCI compliant, payment capability to the Asterisk® platform. It is our hope that the RazorQuotePBP module will unleash a flood of mobile ecommerce applications based on the Asterisk platform. After all, the only standard interface on all mobile phones is the phone application itself. We believe that Asterisk running our RazorQuotePBP module is uniquely positioned to take advantage of this attractive opportunity”, said Brian Young, CEO of RazorQuote.

    RazorQuotePBP will initially only be available to businesses with U.S. bank accounts and requires a RazorQuote merchant account. For more details or to check out a live demonstration, go to www.razorquote.com/razorquotepbp.html



    CloudVox: Install an open source Asterisk phone app and get 250 dollars
    Click to view a printable version Mon, 30 Aug 2010 01:10:57 -0300

    CloudVox is running a competition for people to receive 250 dollars for writing up some documentation for Open Source applications on CloudVox - first in first served.

    Excerpt from their article:

    Install an open source Asterisk phone app, get $250

    We’re trying to make powerful phone and SMS apps easier to use, which means documenting how to deploy and use them. It’s a chance to contribute to open source apps and make $250 for your effort.

    HOW IT WORKS

    • Setup 1 of the 7 open source Asterisk apps below to run against Ifbyphone’s cloud-scale Asterisk hosting service, Cloudvox.
    • Document what you did to get it running, like as a blog post and screenshots. Include enough detail that a technical person could follow your instructions to install it themselves, and proof that it works (best proof: a phone number, video/screencast, or recorded call).
    • Collect $250

    We hope that the combination of powerful free apps, straightforward docs, and easy pay-as-you-go Asterisk hosting will encourage others to deploy these apps and write new ones.

    THE APPS

    • Web-MeetMe (PHP). Schedules and manages conference calls.
    • BigBlueButton (Asterisk-Java), a presentation, video, and audio collaboration system (and Google Summer of Code project) tailored for higher education (Asterisk-Java).
    • MonAst (Python), an Ajax-based app to monitor and hangup calls, queues, and conferences.
    • AsterCRM (PHP), an open source contact center for Asterisk, with screen pops, click-to-call, Web-controlled monitoring, post-call surveys, Google Maps integration, and more. Part of AsterCC.
    • Queue-Tip (Ruby/Adhearsion), which analyzes IVR call queues and runs agent-specific reports.
    • Flash Operator Panel (FOP) (PHP). Via a Web browser, manage Asterisk calls and conferences.
    • Asterisk WEB/PHP Event Monitor, which stores Asterisk Manager Interface (AMI) events to MySQL then displays them in realtime.

    Read More...



    AstriCon approaches
    Click to view a printable version Wed, 25 Aug 2010 04:05:25 -0300

    John Todd has posted a note about the upcoming AstriCon conference in Washington, DC, and the innovation awards:

    Just a reminder: AstriCon is coming up in October in Washington, DC (http://www.astricon.net/) and we're looking forward to seeing you there!

    We're getting to the deadline for Innovation Awards for this year. What's an Innovation Award? The Innovation Award is designed to recognize developers, customers and partners for outstanding achievements that are improving business processes, overcoming technology challenges and enhancing the company's bottom line. Digium picks five different categories in which certain projects or companies have excelled in the last year creating amazing things with Asterisk. The awards are presented at AstriCon.

    If you think you're doing something great with Asterisk, send it in! It's a great opportunity to be recognized as a leader in Asterisk development, implementation, and innovation. August 1 is the deadline.

    More details here - http://www.digium.com/en/company/awards/innovation.php

    Send your Innovation Award proposal to Julie Webb (jwebb at digium dot com) for inclusion.

    AstriCon in general:

    I'll take this opportunity to ask everyone again to get your reservations in for AstriCon this year! We're looking forward to a really good show, in a city slightly less oven-like than the past three years. The conference has a fantastic line-up of speakers and as always, offers the opportunity to talk with people in an informal setting about their real-world experiences with Asterisk, VoIP, different hardware, methods of implementation, and make all sorts of connections that you just can't get without meeting face-to-face. Washington DC is convenient from Europe, with direct flights to IAD (Dulles), DCA (Reagan International), and BWI (Baltimore Washington) airports from most major European and South American cities.

    JT

    ---
    John Todd
    Digium, Inc. | Asterisk Open Source Community Director



    Announcing Adhearsion 0.8.5
    Click to view a printable version Wed, 25 Aug 2010 03:53:31 -0300

    Ben Klang has posted a note about the latest release of Adhearsion - a framework for developing Asterisk based solutions using Ruby:

    We in the Adhearsion community are happy to announce the release of version 0.8.5 of our framework. Adhearsion is a featureful framework for developing Asterisk-based applications using the Ruby programming language. This latest release adds exciting new support for XMPP within Adhearsion applications. Additionally we have focused on fixing outstanding issues, improving documentation and including contributions from many new community members.

    There are several easy ways to get Adhearsion:

    • “gem install adhearsion” in your favorite terminal window
    • Download the tarball (http://github.com/adhearsion/adhearsion/tarball/0.8.5)
    • Clone our Git repository (http://github.com/adhearsion/adhearsion.git)

    New in this version of Adhearsion:

    • Support for connecting to XMPP, querying roster status, sending and receiving messages and other XMPP features. A sample component is included to get you started quickly.
    • Allow routing calls into specific Adhearsion dialplan contexts using the AGI URI. Example: agi://localhost/my_stuff will send the call into the “my_stuff” context.
    • Allow using static MeetMe conference room definitions
    • Extend logging objects into ActiveRecord and Blather
    • Dozens of smaller bugfixes and enhancements; see the CHANGELOG

    Now that the Adhearsion API has been stable for nearly two years, we will be looking for a 1.0 release to be made in the next couple of months based on the current software. The 0.8.5 version may be considered a 1.0 Release Candidate. Our community is making a concerted effort to improve documentation, enhance examples, create shareable components and fix any remaining bugs in time for the big 1.0.

    To learn more about the Adhearsion project, visit our website at http://adhearsion.com. Our active community may be found on our mailing list or in IRC (irc.freenode.net #adhearsion).

    A special shout out to the following people who helped bring you Adhearsion 0.8.5:

    • Ben Langfeld (XMPP support, documentation, code review)
    • Michel Vaillancourt (Testing, feedback, ideas)
    • Jason Goecke and Tropo (Continued support of Adhearsion and AdhearsionConf, project mentoring)
    • Chris Matthieu (Creation of AHNHub.com)
    • Jay Phillips (Work on AHNHub.com, ideas and discussion at AdhearsionConf)

    And a thanks to all of the people who reported bugs on our bugtracker, sent us pull requests on Github, answered questions on the mailing list or in IRC, and generally made our community the awesome group that it is. Thanks!

    /BAK/
    --
    Ben Klang
    Alkaloid Networks LLC
    http://projects.alkaloid.net



    Asterisk 1.8.0-beta4 Now Available
    Click to view a printable version Tue, 24 Aug 2010 18:00:48 -0300

    The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta4.

    This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

    All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any issues found to the issue tracker, http://issues.asterisk.org/. It is also very useful to see successful test reports. Please post those to the asterisk-dev mailing list.

    Asterisk 1.8 is the next major release series of Asterisk. It will be a Long Term Support (LTS) release, similar to Asterisk 1.4. For more information about support time lines for Asterisk releases, see the Asterisk versions page.

    http://www.asterisk.org/asterisk-versions

    This release contains fixes since the last beta release as reported by the community. A sampling of the changes in this release include:

    * Fix parsing of IPv6 address literals in outboundproxy
    (Closes issue #17757. Reported by oej. Patched by sperreault)

    * Change the default value for alwaysauthreject in sip.conf to "yes".
    (Closes issue #17756. Reported by oej)

    * Remove current STUN support from chan_sip.c. This change removes the current broken/useless STUN support from chan_sip.
    (Closes issue #17622. Reported by philipp2. Review: https://reviewboard.asterisk.org/r/855/)

    * PRI CCSS may use a stale dial string for the recall dial string. If an outgoing call negotiates a different B channel than initially requested, the saved original dial string was not transferred to the new B channel. CCSS uses that dial string to generate the recall dial string.
    (Patched by rmudgett)

    * Split _all_ arguments before parsing them. This fixes multicast RTP paging using linksys mode.
    (Patched by russellb)

    * Expand cel_custom.conf.sample. Include the usage of CSV_QUOTE() to ensure data has valid CSV formatting. Also list the special CEL variables that are available for use in the mapping. There are also several other CEL fixes in this release.
    (Patched by russellb)


    Asterisk 1.8 contains many new features over previous releases of Asterisk.
    A short list of included features includes:

    • Secure RTP
    • IPv6 Support in the SIP Channel
    • Connected Party Identification Support
    • Calendaring Integration
    • A new call logging system, Channel Event Logging (CEL)
    • Distributed Device State using Jabber/XMPP PubSub
    • Call Completion Supplementary Services support
    • Advice of Charge support
    • Much, much more!

    A full list of new features can be found in the CHANGES file.

    http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=checkout

    For a full list of changes in the current release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-beta4

    Thank you for your continued support of Asterisk!



    AstriDevCon: October 29th, Washington DC
    Click to view a printable version Sun, 22 Aug 2010 22:48:29 -0300

    John Todd has posted a note about the AstriDevCon conference which occurs within the Astricon conference.

    This AstriCon will see the re-establishment of the AstriDevCon - the mini-conference within a conference! This is a full-day, discussion-oriented meeting where core developers can have direct and uninterrupted access to each other in a more direct way than usual. These sessions often result in massively productive results, either as new features are contemplated and developed, or as old problems are finally vanquished through the application of many eyes and hands.

    AstriDevCon will be on Friday, October 29. It's expected to be the whole day in one of the rooms from AstriCon - from 9:00ish until... when it ends that evening. It's expected that there will be some activity the prior evening as well, as AstriCon ends in the afternoon but many of you may already be in town and gravity ends up pulling people towards the room...

    Seats are limited. Preference will be given to developers who are active participants in the issue tracker, patch submission, and asterisk-dev mailing lists and IRC channels. There is no charge for attending AstriDevCon, though registration is required.

    IMPORTANT: If you know someone who is an Asterisk developer, but doesn't frequent the asterisk-dev list, please forward this to them. There are a surprising number of people who make significant contributions but who are possibly out of the range of this message, and we want to ensure everyone who qualifies knows about this opportunity.

    Register here:
    http://bit.ly/astridevcon2010

    The DevCon is fairly unstructured, but it is typically the case that there are discussions on:

    • security issues
    • priority of work issues
    • general brainstorming on new features
    • difficult bugs in the issuetracker
    • protocol (SIP, IAX2, etc.) questions and decisions
    • integration of third-party applications

    Questions? Reply back to this email and I'll try to answer as quickly as possible.

    JT

    ---
    John Todd
    Digium, Inc. | Asterisk Open Source Community Director



    The XV Commandments of IVR
    Click to view a printable version Mon, 16 Aug 2010 21:24:15 -0300

    An update on the 15 tips for creating effective IVR systems by Allison Smith - the Voice of Asterisk.

    Excerpt from the article:

    We hope that you have been enjoying the IVR Clinic from our guest blogger, Allison Smith. The fifteen-part-series of blogs is designed to assist in the writing of smooth-flowing, efficiently-working IVR systems. As the voice of Asterisk, Allison Smith, has a wealth of experiences recording IVR prompts and has unique insight into the pitfalls that lead to awkward-running and ineffective IVR systems. Allison’s blog will help facilitate the writing of clean, easy-to-navigate prompts for your business phone system.

    Read More



    Daily Asterisk News goes social
    Click to view a printable version Mon, 16 Aug 2010 04:37:35 -0300

    Hi all, I have added a like button for news stories and the ability to like the Facebook Daily Asterisk News page.

    There are a couple of benefits with this - first off it makes the Facebook page a real entity. Secondly it allows you to post stories there which I can then publish to the Daily Asterisk News.

    As always I'm fully open to any suggestions as to how this might be made better :)



    Allison Smith Hilarity
    Click to view a printable version Wed, 11 Aug 2010 00:33:34 -0300

    Randy has posted a note with a link to some funny recordings made by Allison Smith - the voice of Asterisk:

    Greetings and salutations Asterisk community,

    I've been contacted by a man who has generously posted some prompts he commissioned from Allison Smith. If you haven't heard Allison in humor mode, you owe it to yourself to hear this. Joey Lindstrom has decided to place these in the public domain and he's asking Digium to include them in the Asterisk prompts collection. Because he's encouraging anyone who want to use them to do so, feel free to download the collection:

    http://vuc.li/FunnyAllison

    Kudos to Joey for this, his way, he says, to give something back to the people who make Asterisk so special.

    That's YOU!

    /r

    ps: Allison is at her top form too. Make sure you follow Allison on Twitter, she's @voicegal



    Asterisk 1.8.0-beta3 Now Available
    Click to view a printable version Wed, 11 Aug 2010 00:03:58 -0300

    The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta3.

    This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

    All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any issues found to the issue tracker, http://issues.asterisk.org/. It is also very useful to see successful test reports. Please post those to the asterisk-dev mailing list.

    Asterisk 1.8 is the next major release series of Asterisk. It will be a Long Term Support (LTS) release, similar to Asterisk 1.4. For more information about support time lines for Asterisk releases, see the Asterisk versions page.

    http://www.asterisk.org/asterisk-versions

    This release contains fixes since the last beta release as reported by the community. A sampling of the changes in this release include:

    * Fix a regression where HTTP would always be enabled regardless of setting.
    (Closes issue #17708. Reported, patched by pabelanger)

    * ACL errors displayed on screen when using dynamic_exclude_static in sip.conf
    (Closes issue #17717. Reported by Dennis DeDonatis. Patched by mmichelson)

    * Support "channels" in addition to "channel" in chan_dahdi.conf.
    (https://reviewboard.asterisk.org/r/804)

    * Fix parsing error in sip_sipredirect(). The code was written in a way that did a bad job of parsing the port out of a URI. Specifically, it would do badly when dealing with an IPv6 address.
    (Closes issue #17661. Reported by oej. Patched by mmichelson)

    * Fix inband DTMF detection on outgoing ISDN calls.
    (Patched by russellb and rmudgett)

    * Fixes issue with translator frame not getting freed. This issue prevented g729 licenses from being freed up.
    (Closes issue #17630. Reported by manvirr. Patched by dvossel)

    * Fixed IPv6-related SIP parsing bugs and updated documention.
    (Reported by oej. Patched by sperreault)

    * Add new, self-contained feature FIELDNUM(). Returns a 1-based index into a list of a specified item. Matches up with FIELDQTY() and CUT().
    (Closes #17713. Reported, patched by gareth. Tested by tilghman)

    Asterisk 1.8 contains many new features over previous releases of Asterisk.
    A short list of included features includes:

    * Secure RTP
    * IPv6 Support
    * Connected Party Identification Support
    * Calendaring Integration
    * A new call logging system, Channel Event Logging (CEL)
    * Distributed Device State using Jabber/XMPP PubSub
    * Call Completion Supplementary Services support
    * Advice of Charge support
    * Much, much more!

    A full list of new features can be found in the CHANGES file.

    http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=checkout

    For a full list of changes in the current release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-beta3

    Thank you for your continued support of Asterisk!



    Asterisk 1.6.2.11 Now Available
    Click to view a printable version Wed, 11 Aug 2010 00:01:23 -0300

    The Asterisk Development Team has announced the release of Asterisk 1.6.2.11.

    This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

    The release of Asterisk 1.6.2.11 resolves several issues reported by the community and would have not been possible without your participation. Thank you!

    The following are a few of the issues resolved by community developers:

    * Send DialPlanComplete as a response, not as a separate event. Otherwise, it goes to all manager sessions and may exclude the current session, if the Events mask excludes it.
    (Closes issue #17504. Reported, patched by rrb3942)

    * Allow the "useragent" value to be restored into memory from the realtime backend. This value is purely informational. It does not alter configuration at all.
    (Closes issue #16029. Reported, patched by Guggemand)

    * Fix rt(c)p set debug ip taking wrong argument Also clean up some coding errors.
    (Closes issue #17469. Reported, patched by wdoekes)

    * Ensure channel placed in meetme in ringing state is properly hung up. An outgoing channel placed in meetme while still ringing which was then hung up would not exit meetme and the channel was not properly destroyed.
    (Closes issue #15871. Reported, patched by Ivan)

    * Correct how 100, 200, 300, etc. is said. Also add the crazy British numbers.
    (Closes issue #16102. Reported, patched by Delvar)

    * cdr_pgsql does not detect when a table is found. This change adds an ERROR message to let you know when a failure exists to get the columns from the pgsql database, which typically means that the table does not exist.
    (Closes issue #17478. Reported, patched by kobaz)

    * Avoid crashing when installing a duplicate translation path with a lower cost.
    (Closes issue #17092. Reported, patched by moy)

    * Add missing handling for ringing state for use with queue empty options.
    (Closes issue #17471. Reported, patched by jazzy)

    * Fix reporting estimated queue hold time. Just say the number of seconds (after minutes) rather than doing some incorrect calculation with respect to minutes.
    (Closes issue #17498. Reported, patched by corruptor)

    For a full list of changes in the current release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.11

    Thank you for your continued support of Asterisk!



    Asterisk 1.4.35 Now Available
    Click to view a printable version Tue, 10 Aug 2010 23:52:08 -0300

    The Asterisk Development Team has announced the release of Asterisk 1.4.35.

    This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

    The release of Asterisk 1.4.35 resolves several issues reported by the community and would have not been possible without your participation.
    Thank you!

    The following are a few of the issues resolved by community developers:

    * Ensure channel placed in meetme in ringing state is properly hung up.
    (Closes issue #15871. Reported, patched by Ivan)

    * If all members are paused, the wrong status is indicated.
    (Closes issue #17576. Reported, patched by ramonpeek)

    * Fix logging message for stale nonce.
    (Closes issue #17582. Reported, patched by kenner)

    * Interpret device state AST_DEVICE_UNKNOWN as extension state AST_EXTENSION_NOT_INUSE.
    (Closes issue #16035. Reported by francesco_r. Patched by viniciusfontes)

    * Resolve T.38 negotiation regression.
    (Closes issue #16705. Reported by mpiazzatnetbug. Patched by ebroad)

    For a full list of changes in the current release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.35

    Thank you for your continued support of Asterisk!



    Asterisk and Kamailio (openser) realtime integration
    Click to view a printable version Wed, 04 Aug 2010 23:13:48 -0300

    Daniel-Constantin Mierla posted a writeup on combining Asterisk and Kamailio:

    Hello,

    I put together a new tutorial about asterisk realtime integration with kamailio (openser). This time the database used is the one of asterisk, also call routing logic is controlled by asterisk, here is the link:

    http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb

    Practically is an easier way to scale starting from existing asterisk installations.

    The other (old) version I wrote for long time, using kamailio database and asterisk just for media services, is available at:
    http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+1.4+With+Kamailio+1.5.x

    Hope is useful for some of you!
    Daniel

    --
    Daniel-Constantin Mierla
    Kamailio (OpenSER) Advanced Training
    Miami, Fl, USA - June 21-23, 2010
    http://www.asipto.com/index.php/kamailio-advanced-training/



    The Everything Asterisk Video Collection
    Click to view a printable version Wed, 04 Aug 2010 22:34:02 -0300

    Steven Sokol has posted a blog entry on Asterisk Video Resources:

    Asterisk.org Video Resources PageAs part of our ongoing effort to take over the world with Asterisk, I’ve put together a new “Asterisk Video Resources” page on the Asterisk.org site. It contains about 40 videos. Some are tutorials, others are background material. Some are a bit out of date (dare I say, retro?) but are still worth a look. Many of them are community contributed — yet another validation of the open source way of doing things. Over the next few weeks you should see more content pouring in and, with any luck, a bit more organization.

    If you happen to have any Asterisk videos that you would like to share (including any hilarious candid moments from AstriCon), please send them to me. (ssokol [at] asterisk [dot] org). Better yet, show up to AstriCon and shoot some new fun!

    Cheers,

    -S



    Leif Madsen: Speaking at AstriCon 2010
    Click to view a printable version Wed, 04 Aug 2010 19:51:38 -0300

    Leif has posted a note to his blog about the fact he will be speaking at Astricon this year:

    Hey everyone!

    Sorry for the lack of updates recently. We’ve been hard at work on the next version of the Asterisk book, which is shaping up to be the best book ever written on Asterisk. The scope and detail we’re working towards is ambitious, but will be very much worth it. Of course the book will be focused on the upcoming Asterisk 1.8 releases (which was recently branched and betas are now available for testing!).

    My talk this year at AstriCon will be “5 Things You Didn’t Know Asterisk Could Do” which is really an update on various new features in Asterisk 1.8 which you may or may not be aware of. Where possible (time permitting) I’ll be showing configuration setups for the options, or at least give you the overview of the features, what it can and can not do, along with links to some documentation. If I get really ambitious I might create blog posts covering all the features that I can then point back to after the talk!

    So far, it looks like I’ll be talking at 1:45pm on October 27th, 2010 (which appears to be immediately after lunch… I’ll have to be peppy to wake people up from their food comas!). The Agenda-at-a-Glance has been updated, and you can see me on the 27th: http://www.astricon.net/agendaAtaGlance.aspx

    AstriCon is always a great time with some very intelligent people, and I always learn something new each year. I’ve been to several different conferences, and this is probably one of the best balanced conferences in terms of sales vs. technical. If you’re into technical talks, but enjoy the polish that a conference with sales booths provides, you’re really going to enjoy AstriCon. I’ve been participating since 2004 and have never missed a conference. Can’t say that about any of the other conferences I’ve attended before.



    SIP STUN support testing
    Click to view a printable version Sun, 01 Aug 2010 23:10:43 -0300

    David Vossel has posted a note asking for help with testing of some stun changes:

    Howdy!

    I know very little about STUN, but it was obvious to me after reading this post, http://forums.asterisk.org/viewtopic.php?t=74252, that the issue of chan_sip using the same UDP socket for SIP traffic and STUN traffic wasn't good. I wrote a patch that I hope with make our existing STUN support more reliable by handling STUN queries on a dedicated socket. This patch can be found attached to this issue, https://issues.asterisk.org/view.php?id=17622.

    The patch can also be checked out in the svn branch below.

    svn co http://svn.digium.com/svn/asterisk/team/dvossel/sip_stun_support_improved

    If you have encountered any reliability issues using STUN support in chan_sip in the past, please this this patch/branch and post your results.

    Thanks!

    David Vossel
    Digium, Inc. | Software Developer, Open Source Software



    Automated Testing Update
    Click to view a printable version Thu, 29 Jul 2010 22:35:00 -0300

    Russell Bryant has posted details of a new mailing list for automated testing of Asterisk and some information on the progress that has been made. There is no way to say how important I think this work is. It really makes a huge difference to Asterisk and the ability to use it in an enterprise environment. Really great work!

    Here's his mail:

    Greetings,

    A while back, I posted a message about an effort to improve automated testing in the Asterisk project. I wanted to give an update on how that project has progressed for those that have not been following along very closely.

    We started using Bamboo as a continuous integration tool, which you can find running at http://bamboo.asterisk.org/. Note that some of the pass/fail statistics on there are a bit skewed, as the Bamboo server was just rebuilt and things were failing as everything was put back together.

    A lot of really good automated test cases have been developed, and more are constantly being added. There are currently 85 test cases that run against Asterisk trunk after every change to the code. While some tests are small in scope, many of them cover significant call scenarios, such as various methods of doing transfers and call parking.

    I apologize for the previous flood of Bamboo emails to the -dev list. :-) I now have a new mailing list created for those that would like to subscribe to those messages.

    http://lists.digium.com/mailman/listinfo/test-results

    Additionally, one of the latest updates to our Bamboo setup is automated testing code coverage analysis. It will tell us exactly what code ran as a result of our automated test cases. It provides a good metric to start using to help identify areas of Asterisk that are in need of more test cases. You can find the code coverage reports for the latest builds of Asterisk trunk and 1.8 on Linux in the artifacts tab when viewing the details of a build.

    http://bamboo.asterisk.org/browse/AST-TRUNK/latest
    /images/asterisk.gif
    I'm proud of the progress we have made so far and am excited to continue aggressive development of automated test cases for Asterisk. The tests we have are already catching problems on a regular basis. The resulting quality improvements make the job of the development team easier, as well as result in a better experience for end users.

    If you're looking for a way to contribute to Asterisk and you are more comfortable writing scripts instead of C code, then the external test suite is a great way to get involved and help out.

    Thank you all for your continued support of Asterisk!

    Best Regards,

    --
    Russell Bryant
    Digium, Inc. | Engineering Manager, Open Source Software

    Update: Mark Michelson has posted the following:

    <shameless plug>

    If you're itching to learn more about the Asterisk test suite and how to get involved in the testing effort, I will be giving a presentation at Astricon in October that will give all the details!

    </shameless plug>

    If you want to help with the testing effort but are not sure how to start, check out Leif Madsen's blog post about setting up the testsuite here:
    http://blogs.asterisk.org/2010/04/29/installing-the-asterisk-test-suite/. Now that 1.8 has entered the beta stage, this is the ideal time to be adding new tests. We want the beta to be all about ironing out new bugs instead of fixing regressions. If we have the tests in place, we can be more sure that we won't be introducing regressions and 1.8 can be both the most feature-rich and stable version of Asterisk yet.

    I'm PUMPED!
    Mark Michelson



    Asterisk 1.8.0-beta2 Now Available
    Click to view a printable version Tue, 27 Jul 2010 16:12:14 -0300

    The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta2.

    This release is available for immediate download at
    http://downloads.asterisk.org/pub/telephony/asterisk/

    All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any issues found to the issue tracker, http://issues.asterisk.org/. It is also very useful to see successful test reports. Please post those to the asterisk-dev mailing list.

    Asterisk 1.8 is the next major release series of Asterisk. It will be a Long Term Support (LTS) release, similar to Asterisk 1.4. For more information about support time lines for Asterisk releases, see the Asterisk versions page.

    http://www.asterisk.org/asterisk-versions

    This release contains fixes since the last beta release as reported by the community. Some of the changes include:

    • Remove duplicate -c flag when using $(INSTALL)
      (Closes issue #17695. Reported, patched by pabelanger)

    • Don't re-register CDR module on reload.
      (Closes issue #17304. Reported, tested by jnemeth. Patched by tilghman)

    • Don't assume qlog is open.
      (Closes issue #17704. Reported, tested by vrban. Patched by pabelanger)

    • Expand the correct value within AST_OPTION_ONLY.
      (Closes issue #17703. Reported by stuarth. Patched by seanbright)

    • Allow for systems without locale support to be usable.
      (Closes issue #17697. Reported, patched by pprindeville. Tested by mmichelson)

    • Fixes for sounds/Makefile to install on systems using older GNU make.
      (Closes issue #17716. Reported by farisraouf. Patched by tilghman, qwell, seanbright)

    • Update logger.conf.sample to include documentation about new 'fax' logger level.
      (Closes issue #17715. Reported, tested by vrban. Patched by pabelanger)



    Asterisk 1.8 contains many new features over previous releases of Asterisk.
    A short list of included features includes:

    • Secure RTP
    • IPv6 Support
    • Connected Party Identification Support
    • Calendaring Integration
    • A new call logging system, Channel Event Logging (CEL)
    • Distributed Device State using Jabber/XMPP PubSub
    • Call Completion Supplementary Services support
    • Advice of Charge support
    • Much, much more!

    A full list of new features can be found in the CHANGES file.

    http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=checkout

    For a full list of changes in the current release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-beta2

    Thank you for your continued support of Asterisk!



    HumBug - Pre BETA Launch Registration
    Click to view a printable version Tue, 27 Jul 2010 03:59:49 -0300

    Nir Simionovich has posted details of the beta of the new call analytics service:

    Hi All,

    I'd like to introduce you to something that we've been working on the past few months now.

    Over the course of the past 6 months, we've been working on developing an analytical service for Asterisk based PBX systems. Primarily, currently supporting FreePBX based PBX systems, however, can cater to other variants as well.

    So, what is it really about? it's all about your call data and getting an insight to information beyond that of merely CDR records. The humbug project aims at providing a call analytics service, similar to Google Analytics, without you needing to change anything in your dialplan or configuration. Simply insert a small PERL based agent to your Asterisk PBX system, and we'll collect the information directly from the manager port. The agent is fully open sourced, so you can look into it, make sure we don't do anything malicious and simply join in.

    As time progresses and more data is accumulated, we'll be launching a call fraud analysis services to accompany the analytical service. The analytical services are totally FREE for you to use. Just go to the humbug website at http://www.humbuglabs.org and register. We'll contact you directly for the agent download. We welcome any critics, advice, code modifications and ideas. If' you've encountered fraud issues with your PBX, we'd love to hear about it, and analyze your case, so we may add it to our fraud signature database.

    Asterisk PBX makers who wish to integrate humbug services into their offering are welcome to do so - just let us know, so we'll know who does what and where :-)

    Kind Regards,
    Nir Simionovich
    The Humbug Project



    Branch Merging Changes
    Click to view a printable version Sun, 25 Jul 2010 22:42:20 -0300

    Russell Bryant has posted details of some changes to the way developers need to commit code to Asterisk because of the newly released 1.8 branch:

    Greetings,

    The Asterisk 1.8 branch has been created. (Yay!)

    As a related change to this, there have been some changes to the process of merging changes between branches. The merge order is now:

    1.4 --> 1.6.2 --> 1.8 --> trunk

    So, the 1.6.2 branch has the following properties:

    branch-1.4-merged
    branch-1.4-blocked

    The 1.8 branch has:

    branch-1.6.2-merged
    branch-1.6.2-blocked

    ... etc.

    Other than the order of the branches in the merge process (and the corresponding property names), the usage of the svnmerge tool for merging between branches remains the same.

    Thanks,

    --
    Russell Bryant
    Digium, Inc. | Engineering Manager, Open Source Software



    Asterisk 1.8.0-beta1 is Now Available
    Click to view a printable version Sun, 25 Jul 2010 22:27:54 -0300

    The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta1. This release marks the beginning of the testing process for the eventual release of Asterisk 1.8.0.

    This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

    All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any issues found to the issue tracker, http://issues.asterisk.org/. It is also very useful to hear successful test reports. Please post those to the asterisk-dev mailing list.

    Asterisk 1.8 is the next major release series of Asterisk. It will be a Long Term Support (LTS) release, similar to Asterisk 1.4. For more information about support time lines for Asterisk releases, see the Asterisk versions page.

    http://www.asterisk.org/asterisk-versions

    Asterisk 1.8 contains many new features over previous releases of Asterisk.

    A short list of included features includes:

    • Secure RTP
    • IPv6 Support
    • Connected Party Identification Support
    • Calendaring Integration
    • A new call logging system, Channel Event Logging (CEL)
    • Distributed Device State using Jabber/XMPP PubSub
    • Call Completion Supplementary Services support
    • Advice of Charge support
    • Much, much more!

    A full list of new features can be found in the CHANGES file.

    For a full list of changes in the current release, please see the ChangeLog

    Thank you for your continued support of Asterisk!



    Asterisk 1.6.2.10 Now Available
    Click to view a printable version Sun, 25 Jul 2010 21:31:34 -0300

    The Asterisk Development Team has announced the release of Asterisk 1.6.2.10.
    This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

    The release of Asterisk 1.6.2.10 resolves several issues reported by the community and would have not been possible without your participation.
    Thank you!

    The following are a few of the issues resolved by community developers:

    * Allow users to specify a port for DUNDI peers.
    (Closes issue #17056. Reported, patched by klaus3000)

    * Decrease the module ref count in sip_hangup when SIP_DEFER_BYE_ON_TRANSFER is set.
    (Closes issue #16815. Reported, patched by rain)

    * If there is realtime configuration, it does not get re-read on reload unless the config file also changes.
    (Closes issue #16982. Reported, patched by dmitri)

    * Send AgentComplete manager event for attended transfers.
    (Closes issue #16819. Reported, patched by elbriga)

    * Correct manager variable 'EventList' case.
    (Closes issue #17520. Reported, patched by kobaz)

    In addition, changes to res_timing_pthread that should make it more stable have also been implemented.

    For a full list of changes in the current release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.10

    Thank you for your continued support of Asterisk!



    Asterisk 1.4.34 Now Available
    Click to view a printable version Sun, 25 Jul 2010 21:15:54 -0300

    The Asterisk Development Team has announced the release of Asterisk 1.4.34.

    This release is available for immediate download at
    http://downloads.asterisk.org/pub/telephony/asterisk/

    The release of Asterisk 1.4.34 resolves several issues reported by the community and would have not been possible without your participation.
    Thank you!

    The following are a few of the issues resolved by community developers:

    * Allow users to specify a port for DUNDi peers.
    (Closes issue #17056. Reported, patched by klaus3000)

    * Decrease the module ref count in sip_hangup when SIP_DEFER_BYE_ON_TRANSFER is set.
    (Closes issue #16815. Reported, patched by rain)

    * First caller into a dynamic conference new enters the pin once.
    (Closes issue #15878. Reported, patched by pabelanger)

    * Send AgentComplete manager events in the event of blind and attended transfers.
    (Closes issue #16819. Reported, patched by elbriga)

    For a full list of changes in the current release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.34

    Thank you for your continued support of Asterisk!



    AppleRaisin - AstDB over realtime
    Click to view a printable version Thu, 22 Jul 2010 19:57:01 -0300

    Olle has posted a note about his awesome AppleRaisin branch which provides the ability to store AstDB in realtime. This would make for a much simpler failover and clustering situation:

    https://issues.asterisk.org/view.php?id=17681

    I opened a bug for Appleraisin today. I think this would be a good addition to 1.8, but the code needs a guardian angel with some time to take it through reviewboard and maybe add some final missing functions for the CLI database commands to be fully supported.

    This is a feature for everyone running FreePBX or any other GUI that uses the astdb heavily and need some way of doing failover without having to replicate disks with drbd or similar tools.

    The 1.4 version of the code has been run in production for quite a long time successfully.

    Cheers,
    /O



    QueueMetrics 1.6.1 released
    Click to view a printable version Thu, 22 Jul 2010 00:30:54 -0300

    Lenz has posted a note to inform us that QueueMetrics version 1.6.1 has been released. This release offers a large number of bug fixes, misc improvements and new developements including hotdesking:

    QueueMetrics version 1.6.1 has been released today. This release offers a large number of bug fixes, misc improvements and new developements. Some are worth pointing out:

    • Hotdesking - a lot of our customers complain that after switching to ADDMEMBER-style logins, it is not possible to keep agents separate from their extensions. Hotdesking solves this in an easy and efficient manner, and makes it all transparent if you use the Agent's page.
    • To make implementing Hotdesking easier, we release the new User Manual with a section explaining all the details, and we also release version 4.0 of the TrixBox/FreePBX tutorial using hotdesking.
    • Automatic language negotiation - QueueMetrics will automatically speak your preferred language.
    • License expiration notice - you will get a visual notification before your license actually expires, thus freeing you to remember when it is due.
    • We have a new "Advanced configuration" manual that explains a number of topics - it is a "work in progress" but worth checking out.
    • If you update using RPMs, QueueMetriocs will use Java 6 instead of Java 5. This improves system performce and gives access to a plethora of live monitoring tools.
    • Logging has been reduced to avoid excessive I/O load on busy systems

    Major changes:

    • #857 - New SubqueueMode that propagates all subqueue events to the parent queue
    • #1029 - Automatic language negotiation at session start
    • #1001 - SIP channel hotdesking
    • #1061 - Hotdesking - Quick database sync
    • #1062 - Hotdesking - Changes to the Agent's page
    • #1117 - QM will show a message before its license is due to expire.

  • Minor updates:
    • #1118 - Removed excessive logging
    • #1125 - Keeping track of completion codes on muti-stint calls
    • #1112 - Exporting UTF-8 characters over XML-RPC
    • #1012 - Improved number parsing resiliency in the Configuration Wizard
    • #1105 - Too many reports were shown in the 00-All page
    • #753 - No more conflicts with Fedora RPMs
    • #1038 - Multiple content-type on page bug.
    • #1078 - Traffic Analysis and Agent Performance reports were not visible
    • #1031 - Time offset was not computed correctly when running in cluster mode
    • #1036 - Export of any report over Robot mode
    • #1039 - Hotdesking - wrong reporting code passed to Asterisk dialplan
    • #550 - Manual call closure on qlog_opencall
    • #571 - Loader scripts now LSB-compliant
    • #1011 - Fixed dependency not present in Java 1.4
    • #1037 - Fixed method not present in Java 1.4
    • #1040 - Fixed potential issue with algebraic expressions
    • #1067 - AGAW issue made it sometimes not runnable in QM 1.6.0.1
    • #1086 - Date format was not correct for cz_CZ
    • #1088 - Missing unused decoder file
    • #1123 - Encoding issue on QA form
    • #1004 - Short code visible with the report editor
    • #1030 - The CALLMONITOR key was incorrectly used instead of the MON_ADIO one.
    • #981 - Some &nbsp; on the Agent Session Editor page (ported on main trunk)
    • #982 - The session total duration is not correctly calculated (ported on main trunk)
    • #983 - Cannot adjust the time for some sessions (ported on main trunk)
    • #1110 - Supervisors should be Analysts (ported on main trunk)
    • #1031 - Problem with time-zone offset in Cluster mode.
    • #1033 - Excessive logging on catalina.out

    You can download the latest version immediately from the downloads page.As an alternative, if you run RHEL or CentOS (or any distribution derived from them, like e.g. TrixBox, Nerd Vittles and AsteriskNOW), you can install it automatically using yum - see the installation page.

    If you would like to write a language pack for your native language, it's very easy and it only takes a couple of hours' work. See the Translating QueueMetrics document from the Manuals page.

    As always, we value your feedback on how to improve QueueMetrics to make it a better fit to your needs.





    Asterisk 1.8 Branch Creation
    Click to view a printable version Wed, 21 Jul 2010 21:25:51 -0300

    Russell Bryant has posted a note to inform us of the creation of the 1.8 branch of Asterisk.

    Greetings,

    As you may have seen in a comment in another message, it is the intention to have the 1.8 branch created by the end of the week (likely on Friday). IPv6 support was the last major pending project, and the final related patches are up on reviewboard. We need to get the testing process going on 1.8 as soon as possible so we can move forward toward the official release.

    Ideally, all features that will be in 1.8 should be in trunk by the time the branch is created. However, while we're still early in the beta process, I think it is reasonable to consider making some exceptions for some last minute additions on a case by case basis.

    If you have anything pending, please try to get it in as soon as possible. If you have something that can't make it by Friday, but you think could be done in the next couple of weeks, then that's still a possibility. Let's talk about it.

    Thanks,

    --
    Russell Bryant
    Digium, Inc. | Engineering Manager, Open Source Software



    app_swift v2.0 released
    Click to view a printable version Wed, 21 Jul 2010 01:07:12 -0300

    Like a few of these news stories that I will be posting over the next couple of days this is a little old - hope it is not something you have already seen. This one is for a new version of the app_swift text-to-speech module for Asterisk 1.2, 1.4, and 1.6:

    Anyway, here's the announcement from Darren Sessions:

    Hi all,

    Thought I'd mention that the new version of the app_swift text-to-speech module for Asterisk 1.2, 1.4, and 1.6 has been released at it's new home on the Asterisk Forge.

    http://forge.asterisk.org/gf/project/app_swift/

    For those that are unaware, app_swift provides a direct interface with the Cepstral text-to-speech engine so instead of having to call the Cepstral engine and write then read an audio file (i.e. disk I/O), you can call the library directly and "stream" the audio straight to the Asterisk channel. Additionally, the app_swift module supports DTMF detection with a max digits and timeout value as well (similar to the AGI get data functionality).

    The new version of app_swift has been built and tested on the latest releases of Asterisk for each of their respective code-bases (1.2.40, 1.4.32, and 1.6.2.8) using the Cepstral 5.x libraries.

    Any questions or feedback, please let me know.

    Thanks,

    - Darren



    Back to life
    Click to view a printable version Wed, 21 Jul 2010 00:51:22 -0300

    Hey all - I am back online after some pretty big projects which have taken all my time. Will be updating the Asterisk news over the next few days.

    If you've posted comments etc, they'll start appearing over the next couple of days.

    If you've got an article you'd like to see posted, mail me at matt at venturevoip dot com.



    Libpri 1.4.11.2 Now Available
    Click to view a printable version Tue, 08 Jun 2010 23:26:56 -0300

    The Asterisk Development Team has announced the release of version 1.4.11.2 of libpri. This release is available for immediate download at

    http://downloads.asterisk.org/pub/telephony/libpri/

    This release fixes situation where Q.SIG calling name in FACILITY message was not reported to the upper layer:

    * pri_facility.c: Q.SIG calling name in FACILITY message not reported to the upper layer. Q.SIG can send the CallingName, CalledName, and ConnectedName in stand alone FACILITY messages. If the CallingName was not sent in the SETUP message, the caller id name was not reported to the upper layer.
    (Closes issue #17458. Reported, tested by: jsmith. Patched by rmudgett)

    For a full list of changes in the current release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/libpri/releases/ChangeLog-1.4.11.2

    Thank you for your continued support of Asterisk!



    12 mantis issues Ready for Review
    Click to view a printable version Tue, 01 Jun 2010 23:12:40 -0300

    Paul Belanger has posted a note about some issues in the Asterisk issue tracker which are marked as ready for review and asks if people could have a look at them:

    List,

    I'd like to get some feedback about the following issue. Some are newly marked 'Ready for Review', other have been labeled 'Ready for Review' for some time.

    If you have time, please take a minute to review them, most are trivial patches.

    ---
    [patch] Use pkg-config to find gmime libraries

    [patch] chan_iax2 sends command RINGING in answer state

    [patch] Add new AGI command: PARK

    [patch] incorrect playback when using say_date_with_format_es on one o'clock (Spanish)

    [patch] sip peers loaded with realtime doesn't load useragent

    [patch] Makefile: remove ASTBINDIR variable

    ast_gethostbyname doesn't set h_length if argument is an IP Address

    [patch] Asterisk ignores changes to realtime queue member table after initial startup

    [patch] Paused members in queue with higher weight

    [patch] AGI returns bogus "510 Invalid or unknown command"

    [patch] MusicOnHold produces a crash

    [patch] Stuck channel using FEATD_MF if caller hangs up at the right time

    --
    Paul Belanger
    Polybeacon | Consultant
    Jabber: paul dot belanger at polybeacon dot com | IRC: pabelanger (Freenode)
    blog.polybeacon.com



    Asterisk 1.6.2.8 Now Available
    Click to view a printable version Tue, 01 Jun 2010 23:00:54 -0300

    The Asterisk Development Team has announced the release of Asterisk 1.6.2.8.

    This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

    The release of Asterisk 1.6.2.8 resolves several issues reported by the community, and would have not been possible without your participation.

    Thank you!

    The following are a few of the issues resolved by community developers:

    * Enable auto complete for CLI command 'logger set level'.
    (Closes issue #17152. Reported, patched by pabelanger)

    * Make the mixmonitor thread process audio frames faster.
    (Closes issue #17078. Reported, tested by geoff2010. Patched by dhubbard)

    * Add missing 'useragent' field to sip-friends.sql file.
    (Closes issue #17171. Reported, patched by thehar)

    * Add example dialplan for dialing ISN numbers (http://www.freenum.org)
    (Closes issue #17058. Reported, patched by pprindeville)

    * Fix issue with double "sip:" in header field.
    (Closes issue #15847. Reported, patched by ebroad)

    * Add ability to generate ASCII documentation from the TeX files by running 'make asterisk.txt'.
    (Closes issue #17220. Reported by lmadsen. Tested, patched by pabelanger)

    * When StopMonitor() is called, ensure that it will not be restarted by a channel event.
    (Closes issue #16590. Reported, patched by kkm)

    * Small error in the T.140 RTP port verbose log.
    (Closes issue #16998. Reported, patched by frawd. Tested by russell)

    For a full list of changes in the current release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.8

    Thank you for your continued support of Asterisk!



    Asterisk 1.4.32 Now Available
    Click to view a printable version Tue, 01 Jun 2010 22:55:11 -0300

    The Asterisk Development Team has announced the release of Asterisk 1.4.32. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

    The release of Asterisk 1.4.32 resolves several issues reported by the community, and would have not been possible without your participation.

    Thank you!

    The following are a few of the issues resolved by community developers:

    * Make the mixmonitor thread process audio frames faster.
    (Closes issue #17078. Reported, tested by: geoff2010. Patched by dhubbard)

    * When StopMonitor is called, ensure that it will not be restarted by a channel event.
    (Closes issue #16590. Reported, patched by: kkm)

    * Fix up hidecallerid feature in chan_dahdi.
    (Closes issue #17143, #7321. Reported, patched by djenson99)

    * Resolve deadlocks in chan_local.
    (Closes issue #17185. Reported, tested by schmoozecom, GameGamer43)

    * Ensure channel state is not incorrectly set in the case of a very early answer.
    (Closes issue #17067. Reported, patched by tzafrir)

    * Registration fix for SIP realtime.
    (Closes issue #17266. Reported, patched by Nick_Lewis. Tested by sberney)

    For a full list of changes in the current release, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.32

    Thank you for your continued support of Asterisk!



    Libpri 1.4.11 Released
    Click to view a printable version Mon, 31 May 2010 02:18:39 -0300

    The Asterisk Development Team has announced the release of version 1.4.11 of libpri. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri/

    This release contains many fixes and new features, among them being:

    1.) Support for NT-PTMP BRI links, including support for multiple TEIs and connecting of BRI phones.

    2.) Support for allowing persistent Q.921 drops on both NT and TE PTMP links, as well as automatically requesting that Q.921 data links reactivate when needed by Q.931.

    3.) T309 is enabled by default.

    4.) Problems with Keypad Facility Digits were addressed.

    5.) A number of additional service related features were added: Connected Line Information, HOLD/RELEASE support, Call Deflection/Call Rerouting, as well as partial subaddress support. They are supported in the Q.SIG and EuroISDN switch types, and most currently require using the trunk version of Asterisk.

    6.) Many potential and realized Q.921 related problems, particularly during retransmissions and other scenarios involving medium to high packet loss.

    For a full list of changes in the current release candidates, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/libpri/releases/ChangeLog-1.4.11

    Thank you for your continued support of Asterisk!



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