Summary:ASTERISK-09757: MixMonitor does not work in MeetMe using Zap Channels
Reporter:mparker (mparker)Labels:
Date Opened:2007-06-25 18:17:14Date Closed:2007-10-31 20:53:54
Versions:Frequency of
Environment:Attachments:( 0) digium_debug.txt
Description:If you start a mixmonitor on a zap channel and put that channel into a MeetMe conference, you will only hear outgoing audio from the channel and no incoming.  This does work for SIP channels however.  I need it working for Zap channels also.  
Comments:By: Joshua C. Colp (jcolp) 2007-06-26 19:05:23

Can you please enable debug in logger.conf, do set debug 9 on the CLI, and go through this call flow again and paste the output? Thanks.

By: Keshav (keshav) 2007-06-27 10:58:51

I have tried MixMonitor on asterisk 1.2.17, 1.2.19 and 1.4.5 and found it working properly.
can you send the cli log, that what error r u getting on that.

By: mparker (mparker) 2007-06-27 12:26:50

I'll get some debug attached today.  To respond to Keshav, there are no errors that are showing up in the cli log.  The MixMonitor works great up until I put the channel that I'm monitoring into a MeetMe conference.  Once I put the channel into a MeetMe conference then only outgoing audio is heard in the recording.  Keshav, Have you tried the MixMonitor while in a MeetMe conference?

By: mparker (mparker) 2007-06-27 13:25:34

I've added the requested debug.  It's found in the attached file: digium_debug.txt.

Let me give you the sequence of events so that the debug makes more sense.  The manager api is used to originate the first channel Zap/95.  Once Zap/95 is answered it goes into the dialplan and executes a Dial command and Zap/94 is created.  MixMonitor is started on Zap/95 while Zap/94 is ringing.  Once Zap/95 and Zap/94 are bridged and conversation is proceeding between the two, Zap/95 puts Zap/94 into a MeetMe conference.  Zap/95 proceeds in the dialplan to execute another Dial command and Zap/93 is created.  Zap/93 is then hungup and Zap/95 goes into the MeetMe conference to talk with Zap/94.

The MixMonitor works great up until Zap/95 and Zap/94 are in the MeetMe conference together.  In the resulting recording, only the outgoing audio is heard on Zap/95 when it is in conference with Zap/94.

Thanks for help

By: Keshav (keshav) 2007-06-29 03:50:21

I thing you will be using MixMonitor for the mixing pupose of recording. The mixmonitor has issues in it self, and it kill the asterisk when u are using more calls. Better will be if you go for Monitor using with "m", it will help u . I am writing the line here:------
exten => XXX,n,Monitor(wav,/recording/${callfilename},m)

By: mparker (mparker) 2007-06-29 17:14:59

I tried the Monitor application and the same problem exists.  It doesn't grab any of the conference audio, only the outgoing audio from the channel that I'm monitoring.  It works with SIP channels but not ZAP channels.  Thanks for the suggestion Keshav.  I was hoping it would work.

By: Eldad Ran (eldadran) 2007-07-01 17:22:34

I have the same problem, I had to work around it using local channel, if you need someone to test a patch I'm up to it.

By: Jason Parker (jparker) 2007-08-03 18:22:22

During an internal Digium training session on MeetMe this morning, the topic that this issue describes came up.

It was explained that the audio going to Zap channels from a MeetMe conference does not actually go through Asterisk - this is done on purpose.  It is "optimized" to stay inside of Zaptel, rather than coming back to Asterisk, and then through Zaptel a second time.  The workaround we discussed, as eldadran suggested, was to use a Local channel (with the 'n' option).

I'm not really sure what we want to do here.  I guess if something like MixMonitor is attached to the channel, it shouldn't try to optimize Asterisk out of the audio path.

By: Digium Subversion (svnbot) 2007-10-31 20:52:15

Repository: asterisk
Revision: 87970

U   branches/1.4/apps/app_meetme.c

r87970 | file | 2007-10-31 20:52:12 -0500 (Wed, 31 Oct 2007) | 4 lines

If a Zap channel contains a spy or a spy is added take it out of the conference in kernel space and make it go through Asterisk so the spy gets audio from both sides.
(closes issue ASTERISK-9757)
Reported by: mparker


By: Digium Subversion (svnbot) 2007-10-31 20:53:54

Repository: asterisk
Revision: 87971

_U  trunk/
U   trunk/apps/app_meetme.c

r87971 | file | 2007-10-31 20:53:53 -0500 (Wed, 31 Oct 2007) | 12 lines

Merged revisions 87970 via svnmerge from

r87970 | file | 2007-10-31 22:53:55 -0300 (Wed, 31 Oct 2007) | 4 lines

If a Zap channel contains a spy or a spy is added take it out of the conference in kernel space and make it go through Asterisk so the spy gets audio from both sides.
(closes issue ASTERISK-9757)
Reported by: mparker