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Summary:ASTERISK-18194: Asterisk 1.6. No audio when using MixMonitor on sip channels
Reporter:Alexander Tatarinov (xanders)Labels:
Date Opened:2011-07-27 08:04:08Date Closed:2011-09-12 15:35:16
Priority:MinorRegression?
Status:Closed/CompleteComponents:Applications/app_mixmonitor
Versions:1.6.2.15 1.6.2.18 Frequency of
Occurrence		Frequent
Related
Issues:
Environment:Reproduced this behavior on Fedora 15 (asterisk ver. 1.6.2.18) and CentOS 5.5 (asterisk ver. 1.6.2.15), using 3CXPhone softphones as sip clientsAttachments:
Description:With the following extensions.conf configuration there is no audio on both sides (problem persists in 90% of all calls) and no monitor file generated.

exten => 1000,1,Answer()
exten => 1000,n,MixMonitor(${UNIQUEID}.wav); <--- no problem when this lines commented out
exten => 1000,n,Dial(Sip/${EXTEN},30)
exten => 1000,n,StopMixMonitor(); <--- no problem when this lines commented out
exten => 1000,n,Hangup()

When I comment out MixMonitor lines no problems with audio found, everything works perfectly;

Asterisk CLI output during "no-audio" call:
 == Using SIP RTP CoS mark 5
   -- Executing [1000@default:1] Answer("SIP/1001-00000032", "") in new stack
   -- Executing [1000@default:2] MixMonitor("SIP/1001-00000032", "1311771205.50.wav") in new stack
   -- Executing [1000@default:3] Dial("SIP/1001-00000032", "Sip/1000,30") in new stack
 == Begin MixMonitor Recording SIP/1001-00000032
 == Using SIP RTP CoS mark 5
   -- Called 1000
   -- SIP/1000-00000033 is ringing
   -- SIP/1000-00000033 answered SIP/1001-00000032
 == Spawn extension (default, 1000, 3) exited non-zero on 'SIP/1001-00000032'
 == End MixMonitor Recording SIP/1001-00000032

Both softphones computers are on the same ip subnetwork as asterisk server.

Sip configuration for used softphones:

[1000]
type=friend
regexten=1000
callerid="1000"
secret=1000
host=dynamic
nat=no
disallow=all
allow=gsm
allow=ulaw
allow=alaw

[1001]
type=friend
regexten=1001
callerid="1001"
secret=1001
host=dynamic
nat=no
disallow=all
allow=gsm
allow=ulaw
allow=alaw

Comments:By: Alexander Tatarinov (xanders) 2011-08-12 06:19:52.085-0500

Found out that the problem happens only with 3CXPhone sofphones when they set in auto-answer mode
By: Leif Madsen (lmadsen) 2011-09-12 15:35:08.437-0500

Per the Asterisk maintenance timeline page at http://www.asterisk.org/asterisk-versions maintenance (bug) support for the 1.4 and 1.6.x branches has ended. For continued maintenance support please move to the 1.8 branch which is a long term support (LTS) branch. For more information about branch support, please see https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions.  After testing with Asterisk 1.8, if you find this problem has not been resolved, please open a new issue against Asterisk 1.8.