Summary:ASTERISK-07700: SIP or IAX redirect from AGI script to meetme room results in no audio
Reporter:Matt Florell (mflorell)Labels:
Date Opened:2006-09-08 14:19:18Date Closed:2006-09-13 09:53:30
Versions:Frequency of
Environment:Attachments:( 0) channel.c-42600.patch
Description:Starting in release 1.2.11 of Asterisk(not affected by zaptel version) when a call is sent from an AGI script to a meetme room there is no audio upon the SIP/IAX channel entering the conference except for the double-tone on entry which all parties can hear.

Zap channels operate normally. app_conference also functions normally with SIP/IAX.

This seems to be an issue with meetme, although app_meetme.c is unchanged from 1.2.10 to 1.2.11 so it must be that something else is causing this to happen somehow.

If a SIP/IAX call is initiated from the Manager API as calling from within the meetme room the audio is fine on SIP/IAX.

This issue affects the VICIDIAL project which is how this came to my attention. For anyone using SIP or IAX trunks and VICIDIAL, Asterisk 1.2.11 is unusable.

Any suggestions on where in the code to look in the code would be greatly appreciated.


In the messages log file the only difference I see between the two versions is that the Local/ channel leg that was masq'd for the call is hungup before the channel joins the meetme in 1.2.11 whereas in 1.2.10 the Local/ channel is hungup after the channel joins the meetme. No idea why that would affect things since a Zap channel doesn't have these problems.
Comments:By: Matt Florell (mflorell) 2006-09-09 00:19:01

Looking at the SVN change logs it looks like a couple of changes made by kpfleming may be the cause: 38310 and 38347

38310 - don't do useless translation destroy/build when the channel is already in the correct format, July 26, 2006

38347 - do a better job avoiding translation path teardown/setup when not needed, July 27, 2006

The problem may be that meetme is wanting to translate the channels to slin and these changes don't think that needs to happen for some reason.

I will look into this more later.

By: Matt Florell (mflorell) 2006-09-11 11:49:59

it was kpfleming's changes in 38310 and several other changes involving channel transcoding in the channel.c code that caused this.

The issue seems to be fixed in SVN asterisk 1.2 branch as of 42600

Asterisk release 1.2.12 does have this bug as well.

related to bug 7803
related to bug 7887

By: Matt Florell (mflorell) 2006-09-11 11:51:22

posted the channel.c-42600.patch file for those that want to patch their 1.2.12 release.

Bug is fixed in SVN 1.2 asterisk so this bug can be closed as well.

Thanks file!

By: Serge Vecher (serge-v) 2006-09-13 09:53:30

or simply update to, which contains the aforementioned fix.