* You are viewing the archive for August 10th, 2012

Asterisk 11.0.0-beta1 Now Available!

The Asterisk Development Team is pleased to announce the first beta release 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:

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


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

* 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:


A full list of all new features can also 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!

Debian 7/Asterisk TLS Bug And Others

Debian 7 is currently in the `freeze’ status with 1.8.13 – that means Debian 7 is very likely to release 1.8.13 and be carrying it for the next 2-3 years (typical lifetime of a Debian release)

I run 1.8.8. TLS has a bug: it fails to receive BYE over the TLS
connection from my Polycom phone.

I tried 1.8.13, the version in Debian 7, and found a more severe bug:


The TLS clients can’t connect at all, this looks like a really bad regression from 1.8.8

I’ve looked at 1.8.(14, 15, 16-rc1) and their changelogs don’t mention any fix.

Debian is very conservative about accepting updates during the `freeze’
process – they will most likely want to see a release with ONLY
the most essential fixes

a) is anyone else aware of these bugs?

b) what essential changes should go into for Debian?

ICall Service Any Good?

Hi everyone,

We are getting cotinueous error messages over the past few days from iCall:

— Called iCall/01144xxxxxxxx
— Got SIP response 500 “Server internal failure” back from

Is this something everyone else is getting? They are very bad at support and I am not sure if it’s their servers or my Asterisk server that is causing the issue.


Question On App_confbridge

I have a profile in confbridge
and more…

Asterisk is reading it at startup.
[1;30m == ^[[0mParsing ‘/etc/asterisk/confbridge.conf': ^[[1;30m =^[[0mFound
^[[1;30m ^[[0mapp_confbridge.so => (^[[0;33mConference Bridge Application^[[0m)

When I try to use it I get a warning about
WARNING[20678] app_confbridge.c: Conference bridge profile
MessageNetConfBridge does not exist

My dialplan looks like:
exten => new_app_confbridge,1,ConfBridge(${agi_pa_meetme}, MessageNetConfBridge)

Not sure how its not picking it up?

What did I miss?


Asterisk And Meetme

I just downloaded and compiled from source asterisk 10.7.0
after installing and running I tried to do a meetme, did not work. I looked in the apps/app_meetme* and there is only the C file, there is no .o seems like it did not compile.

Is that a new default behavior?

Looking for the trick to get it compiled?

DAHDI is installed my system and works fine.

I dont see anything meetme related in configure.

What did I miss?


Chan_dahdi.c: No D-channels Available! Using Primary Channel 16 As D-channel Anyway!

Hi all. I have this problem with my Digium 2E1 card and PRI, for hours It works well, with some meesages…

[Aug 10 09:20:31] NOTICE[32270] chan_dahdi.c: PRI got event: HDLC Abort (6)
on Primary D-channel of span 1

But PRI continue up….hours later… PRI go down. I thought the problem was in the telco, but the strange thing is that I
have a loop cable in the second E1 and when I scan both E1 are with alarms=LMFA/OK and i have only the first E1 connected to the telco!!

[Aug 10 09:20:31] NOTICE[32270] chan_dahdi.c: PRI got event: HDLC Abort (6)
on Primary D-channel of span 1
[Aug 10 09:20:32] NOTICE[32270] chan_dahdi.c: PRI got event: HDLC Abort (6)
on Primary D-channel of span 1
[Aug 10 09:20:32] NOTICE[32270] chan_dahdi.c: PRI got event: HDLC Abort (6)
on Primary D-channel of span 1
[Aug 10 09:20:32] NOTICE[32270] chan_dahdi.c: PRI got event: HDLC Abort (6)
on Primary D-channel of span 1
[Aug 10 09:20:32] NOTICE[32270] chan_dahdi.c: PRI got event: HDLC Abort (6)
on Primary D-channel of span 1
[Aug 10 09:20:36] WARNING[32270] chan_dahdi.c: No D-channels available!
Using Primary channel 16 as D-channel anyway!

gentoo1 ~ # dahdi_scan
active=yes alarms=LMFA/OK
description=T2XXP (PCI) Card 0 Span 1
manufacturer=Digium devicetype=Wildcard TE220 (5th Gen)
location=Board ID Switch 0
lbo=0 db (CSU)/0-133 feet (DSX-1)