Summary:	ASTERISK-22853: SIP call hangup randomly during conversation due to MixMonitor
Reporter:	Cyril CONSTANTIN (cyrilc)	Labels:
Date Opened:	2013-11-13 05:53:24.000-0600	Date Closed:	2014-01-03 18:37:40.000-0600
Priority:	Critical	Regression?	No
Status:	Closed/Complete	Components:	Channels/chan_sip/General
Versions:	1.8.24.0	Frequency of Occurrence	Frequent
Related Issues:
Environment:	Debian 6	Attachments:	( 0) filtered_-_Asteriskooh323_to_AvayaClan.rar ( 1) logs.rar ( 2) tcpdump_sip_call.rar
Description:	Hi Team, I'm facing a random issue, when SIP user make call through ooh323 their call are hangup during conversation randomly, there is no specific duration where call hangup, it doesn't affect all calls but some per day per SIP user. System was working for several month without issue but since I have created a team working in queue and making lot of outbound calls with MIXMONITOR recording them it looks that they have started to get this issue, I'm not sure if it's related but issue started since I have introduced it apparently. I was working with version 1.8.15.1 and then I have upgraded to 1.8.24.0 but it didn't resolved the issue. I got two example this morning where two SIP user (not user from queue) where making outbound calls and were cut at the same time: 1st SIP user was calling number 0760399590 2nd SIP user was calling number 0659579568 Asterisk doesn't crash, SIP calls are just dropped, peer still registered to Asterisk but can't make any outbound calls for several second. I have joined all needed traces and below a link with full tcpdump traces: http://myaccount.dropsend.com/file/3bdc2d347254db5b Let me know if you need anything else. Best Regards
Comments:	By: Cyril CONSTANTIN (cyrilc) 2013-11-13 06:06:34.402-0600 Please find in attachment, cdr screenshot, debug, full, message, backtrace, console logs By: Joshua C. Colp (jcolp) 2013-11-13 06:17:41.797-0600 For anyone reading this issue the topology is not SIP<->SIP, it's SIP<->OOH323... By: Cyril CONSTANTIN (cyrilc) 2013-11-13 06:33:36.355-0600 Please find tcpdump traces for sip call between SIP peers and asterisk By: Cyril CONSTANTIN (cyrilc) 2013-11-13 06:34:22.624-0600 Traces from/to Asterisk to Avaya CLAN with ooH323 By: Cyril CONSTANTIN (cyrilc) 2013-11-18 10:58:48.168-0600 Hi, I made a test by stopping using MixMonitor for call distributed in queue during two days, and calls for others users were not disconnected anymore. After this test I've activated back MixMonitor on calls queued and all my users faced again the same issue and have their calls disconnected randomly. I just have again tried to deactivate MixMonitor today then since 6 hours all my SIP users are happy, I asked them if issue appears again but it didn't. Generally this problem occurs several times per hour and per user, so I'm getting a feedback quickly from them. See below what I'm using for MixMonitor, there is nothing special I'm just passing two ARGS to my perl script {noformat}same => n,Set(MIXMONITOR_FILENAME=/var/www/recording/${STRFTIME(${EPOCH},,%Y%m%d-%H-%M-%S)}-${var}) same => n,MixMonitor(${MIXMONITOR_FILENAME}.wav,v(4),/usr/local/sbin/getagentID.pl ^{UNIQUEID} ^{MIXMONITOR_FILENAME}) {noformat} Base on my different test I'm really suspecting that issue is coming from MixMonitor function which hangup calls. Best Regards By: Cyril CONSTANTIN (cyrilc) 2013-11-21 10:04:11.900-0600 From 2 days and half now the issue doesn't occurs since MixMonitor is deactivated, the problem is that I have to record my calls and reactivate MixMonitor... Any feedback will be appreciated Thanks a lot By: Rusty Newton (rnewton) 2013-11-22 15:36:12.853-0600 To know for sure what is going on here, we'll need full pcaps of working calls (where mixmonitor was used on their channels as well) As far as I can tell from your current pcaps, without going into various oddities that are probably unrelated, it looks like the phones are sending Asterisk a bye to hang up the call. If Asterisk is doing something bad that results in the phone sending the BYE... I don't know, but possibly comparing audio between pcaps of working and non working calls may help. Do the users on the phone report any audio abnormalities before the call is hung up? By: Cyril CONSTANTIN (cyrilc) 2013-11-23 01:27:44.500-0600 Hi, Users didn't notice any abnormalities before call hang up and it happens simultaneously to users having Avaya SIP hardphone, Bria softphone, Polycom sound station hardphone... Yesterday I have upgraded to Asterisk 11.6 and reactivated MixMonitor, users didn't complain for 5 hours, I'll let you now Monday if issue occurs again or not. Have a nice week-end! Best Regards By: Cyril CONSTANTIN (cyrilc) 2013-11-28 14:10:12.381-0600 Hi, Well since the upgrade users complain less than before I just got 3 complain when it was several per day, MixMonitor still activated for the moment. Hard to say if it's really MixMonitor causing it, I'll try to make others traces but as the issue happens rarely now it will be difficult to get it. Keep you posted. Best Regards By: Rusty Newton (rnewton) 2013-12-12 17:26:25.896-0600 Thanks. Since the frequency of the issue has greatly reduced and you have moved from 1.8 to 11, I'd say give it more time as you could encounter new issues just from moving across two LTS releases. Let us know in a couple weeks. By: Rusty Newton (rnewton) 2014-01-03 18:37:40.942-0600 I'm closing this out since we can't reproduce here and it is not clear yet if this is a bug or how to trigger it. If any one has further information about it you can ask a bug marshal to re-open the issue in the #asterisk-bugs IRC channel.