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.
Tue, 14 May 2013 02:59:07 +1200 Author: Matt Riddell
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
Wed, 08 May 2013 04:05:31 +1200 Author: Matt Riddell
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!
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!
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.
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
Sat, 27 Apr 2013 03:43:09 +1200 Author: Matt Riddell
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
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.
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.
Asterisk 11.0.0-beta1 Now Available August 11, 2012 Average Vote: 10
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).
oFono 1.0 has been released November 10, 2011 Average Vote: 10
Steve Totaro has forwarded details of the latest release of a project called oFono.
Discount for Astricon 2012 August 31, 2012 Average Vote: 10
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.
Asterisk 10.0.0-rc1 Now Available November 11, 2011 Average Vote: 10
The Asterisk Development Team is pleased to announce the first release candidate of Asterisk 10.0.0.
AstLinux Custom Build Engine Now Available March 13, 2012 Average Vote: 10
Kristian Kielhofner has posted details of some work that has been completed in the AstLinux project to provide a custom build engine.
Hello World from the new guy June 21, 2012 Average Vote: 10
David Duffett has written his first post as Asterisk Community Director.
Asterisk 1.10 branch created July 14, 2011 Average Vote: 10
At 8:34am New Zealand time this morning a branch named 1.10 was created.
Asterisk Trunk moves from Berkley DB to SQLite 3. July 7, 2011 Average Vote: 10
While reading through the commit logs this morning I noticed that the planned change to SQLite 3 as the backend database has taken place.
Chapter on Asterisk Architecture May 27, 2011 Average Vote: 10
Russell Bryant has posted a note about a chapter on Asterisk Architecture.
New SIP channel driver project page November 23, 2012 Average Vote: 10
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.
Asterisk and Google August 26, 2011 Average Vote: 10
Malcolm Davenport has posted a request on behalf of Digium for someone to help maintain the Google channel driver when Google makes changes.
espeak module for Asterisk August 22, 2011 Average Vote: 10
Lefteris Zafiris has posted details of a new version of the app_espeak application for Asterisk - another speech synthesizer.
DAHDI-Linux and DAHDI-Tools 2.7.0-rc1 Now Available May 30, 2013 The Asterisk Development Team has announced the releases of: DAHDI-Linux-v2.7.0-rc1, DAHDI-Tools-v2.7.0-rc1 and dahdi-linux-complete-2.7.0-rc1+2.7.0-rc1
