Olle has posted a blog entry on SIPit and Asterisk:

SIPit is the main interoperability event for all things SIP. It’s organized by the SIP Forum and creates good feedback to the IETF. Asterisk has been participating in SIPit during many years and in many variants - videocaps, Marc Blanchet’s IPv6 branch and the standard Digium releases. All these tests has lead to a large amount of improvements for Asterisk and have helped us to build a network with other developers in the business, a network which helps when we have bugs that involve interoperability with these devices or servers. SIPit has proven important for the success of Asterisk, and thus it is also important for everyone in the Asterisk community.

Now, when we are working on the next long-term release (1.8) we really need to test again and make sure that we interoperate properly. New stuff, like Terry’s SRTP branch, my RTCP work and the call completion and caller ID update work needs serious testing. We need feedback to be able to fix the issues with the TCP and TLS support. (more…)

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Randulo has posted details of a 24 hour conference call being held on the 26th of March - no excuse to not be there no matter what time zone you are in.

I've always missed it because of the time and/or my workload - this one I'll definitely have to make it to!

Anyway, here's his mail:

24 hours of VUC and a chance for those of you in Asia and the Southern Hemisphere to join us at least once live at a decent time!

On the third birthday of the VUC (formerly Asterisk Users Conference) we will be on the air for 24 consecutive hours beginning at 3AM EDT Friday on on through the next 24 hours. The record for the longest VUC is currently 8 hours.

We will be talking about Asterisk and VoIP but also about things that affect our community, including net neutrality, port blocking, censorship and how online communities are using voice over the net (VoIP as we'd call it) to improve lives.

http://voipathon.org for the details including a permatime link to allow to to know at exactly what time you can join us via, SIP, Skype or PSTN.

IRC: #vuc on Freenode.net 24/7

Happy 3rd Birthday to us.

/r

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Zoa has posted details of the release of Attrafax under GPL2 and commercial licensing:

On friday we finally released Attrafax under a GPL2 license. It comes with its own set of modems and built in transparent gatewaying. The solution should be quite stable as long as the line quality is ok. (Some tools for measuring the line quality are included in the release, as well as some fax2mail scripts).

There is an example implementation included for Asterisk 1.4, if someone wants to porting it to the new fax backend or more recent asterisk versions and needs some help, let us know.

The full press release can be found here:
http://www.zoiper.com/downloads/opensource_fax_stack_PR.pdf the project homepage can be found at www.zoiper.com/foip/

Cheers,

Zoa

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Håkon Nessjøen has posted a patch he has written to show all call attempts including failing ones for Asterisk Queues:

Hi,

I've just uploaded a patch here: https://issues.asterisk.org/view.php?id=16925

This patch introduces a new parameter; "congestion" to both RINGNOANSWER in queue_log and AgentRingNoAnswer AMI event, which is set to 1 if the call failed to go through because of technical difficulties.

And it also is more verbose than app_queue has been earlier, since app_queue usually silently ignores channel problems with its agent members.

With this patch, it is easier to make statistics out of queue_log with information about problems with an agent. For example if an agent has a faulty line, or your telco/dahdi connection is having problems.

Please come with comments about this patch, and help test it if you agree with the idea.

Regards,
Håkon Nessjøen

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The Asterisk Development Team has announced release candidates of Asterisk for versions 1.4.30, 1.6.0.26, 1.6.1.18, and 1.6.2.6.

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

These release candidates are a continuation of the release candidate process started prior to the Asterisk security releases over the last month. The following is a rundown of release versions over the last two months:

1.4.29 full release
1.4.29.1 security release based on 1.4.29
1.4.30-rc1 release candidate
1.4.30-rc2 release candidate
1.4.30-rc3 release candidate (NEW)

1.6.0.22 full release
1.6.0.23-rc1 release candidate
1.6.0.23-rc2 release candidate
1.6.0.23 version skipped due to security release
1.6.0.24 security release based on 1.6.0.22
1.6.0.25 security release based on 1.6.0.22 (with changes from 1.6.0.24)
1.6.0.26-rc1 release candidate (NEW)

1.6.1.14 full release
1.6.1.15-rc1 release candidate
1.6.1.15-rc2 release candidate
1.6.1.15 version skipped due to security release
1.6.1.16 security release based on 1.6.1.14
1.6.1.17 security release based on 1.6.1.14 (with changes from 1.6.1.16)
1.6.1.18-rc1 release candidate (NEW)

1.6.2.2 full release
1.6.2.3-rc1 release candidate
1.6.2.3-rc2 release candidate
1.6.2.3 version skipped due to security release
1.6.2.4 security release based on 1.6.2.2
1.6.2.5 security release based on 1.6.2.2 (with changes from 1.6.2.2)
1.6.2.6-rc1 release candidate (NEW)

These release candidates address issues that were reported by the community and resolved since the last round of bug fix releases.

Below is a list of issues resolved by the previous release candidates, and are included in the latest round of release candidates:

* When a transferer hangs up during an attended transfer BEFORE the transfer is answered, don't stop playing MOH.
(Closes issue #16513. Reported, tested by litnimax. Patched by gknispel proformatique.)

* Extend announcement URL used with Queue from 80 chars to PATH_MAX.
(Closes issue #16488. Reported, patched by syspert.)

* Fix bug with channel receiving wrong privileges after call parking.
(Closes issue #16429. Reported, patched by Yasuhiro Konishi.)

* Fix a memory leak in pbx_spool when using SetVar in a call file.
(Closes issue #16554. Reported, tested by mav3rick. Patched by seanbright.)

* Fix regression for timed out parked call returning to caller.
(Closes issue #15459. Reported by djrodman. Patched by mnick, jpeeler.)


Below is a list of issues resolved since the last round of release candidates:

* Make sure to clear red alarm after polarity reversal
(Closes issue #14163. Reported, patched by jedi98. Tested by mattbrown, Chainsaw, mikeeccleston.)

* Fix problem with duplicate TXREQ packets in chan_iax2
(Closes issue #16904. Reported, patched by: rain. Tested by rain, dvossel.)

* Fix crash in app_voicemail related to message counting.
(Closes issue #16921. Reported by whardier. Patched by seanbright. Tested by whardier.)

* Fix attended transfers with Local channels. Fix for regression introduced in revision 244785.
(Closes issue #16816. Reported by jamhed. Tested by jamhed, corruptor.)

* Remove color code sequences from verbose messages that go to logfiles.
(Closes issue #16786. Reported, patched by: dodo. Tested by tilghman.)

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

ChangeLog-1.4.30-rc3
ChangeLog-1.6.0.26-rc1
ChangeLog-1.6.1.18-rc1
ChangeLog-1.6.2.6-rc1

Issues found in any of these release candidates should be reported to the Asterisk issue tracker at https://issues.asterisk.org

Thank you for your continued support of Asterisk!

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