Summary:ASTERISK-21912: Call hang-up when issuing mixmonitor start
Reporter:John Covert (jcovert)Labels:
Date Opened:2013-06-14 16:12:39Date Closed:2013-06-16 21:37:33
Status:Closed/CompleteComponents:Applications/app_mixmonitor Channels/chan_sip/General
Versions:10.7.1 10.12.2 Frequency of
is the original version of this clone:ASTERISK-18863 Call hang-up when issuing "mixmonitor start" just after "bridge" through AMI
Environment:Darwin PowerPC Attachments:
Description:Call drops immediately after issuing mixmonitor command:

asterisk*CLI> mixmonitor start SIP/x37-000005dd 2013-06-14-16:30:16.59.ulaw
   -- Native bridging SIP/x37-000005dd and SIP/von-g-000005de ended
 == Begin MixMonitor Recording SIP/x37-000005dd
 == Spawn extension (macro-vonout, s, 4) exited non-zero on 'SIP/x37-000005dd' in macro 'vonout'
 == Spawn extension (dialstation, 824313034997111, 1) exited non-zero on 'SIP/x37-000005dd'
 == MixMonitor close filestream (mixed)
 == End MixMonitor Recording SIP/x37-000005dd

I have just noticed that this failure occurs on an INCOMING call dialed from SIP/x37 (or any other sip device) to context dialstation37, but does not occur if the call is set up with "originate sip/x37 extension 824313034997111@dialstation37".


It seems that the following creates the conditions which cause the failure:
Executing [s@macro-vonout:2] Set("SIP/x37-0000000c", "SIP_CODEC=ulaw") in new stack
[Jun 14 22:32:58] NOTICE[21167]: chan_sip.c:6905 try_suggested_sip_codec: Changing codec to 'ulaw' for this call because of ${SIP_CODEC} variable

Even though this is executed in both the phone-originated (failing) and asterisk-originated case, it creates the necessary conditions for the failure only in the phone-originated case.

It was inserted in the outgoing macro for this trunk group (which doesn't do g.722) so that the phone would set up the call in ulaw mode and not require transcoding by asterisk.
Comments:By: Michael L. Young (elguero) 2013-06-16 21:37:12.482-0500

Per the Asterisk maintenance timeline page at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions (bug) support for the 10 branch has ended. For continued maintenance support please move to the 11 branch which is a long term support (LTS) branch.  After testing with Asterisk 11, if you find this problem has not been resolved, please open a new issue against Asterisk 11 and include full debug logs.