Summary: | ASTERISK-12004: MeetMe conference freezes with Polycom | ||
Reporter: | Jason Dixon (fuzzyping) | Labels: | |
Date Opened: | 2008-05-09 11:58:34 | Date Closed: | 2011-06-07 14:03:11 |
Priority: | Major | Regression? | No |
Status: | Closed/Complete | Components: | Applications/app_meetme |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ||
Description: | We have a remote office that's having problems with their Polycom. Sometime after they start a conference, the audio will halt and the Polycom will become unresponsive. The only recourse is to kill the Polycom meetme. Symptoms include a flood of RTP packets from the Asterisk server to the Polycom, a loss of audio for all participants, and the Polycom console becomes frozen. It appears to be isolated to this particular device; we routinely have conference bridges with other offices and Polycoms without issue. We are using Asterisk 1.4.10. Here is the output from the channels and "meetme kick". ****** ADDITIONAL INFORMATION ****** pbx*CLI> meetme list 642696 User #: 01 293 Conference Channel: SIP/seattleconference-08a1fc68 (Admin) (unmonitored) 00:05:23 User #: 02 205 Todd Channel: SIP/205-08a211e8 (Admin) (unmonitored) 00:02:39 User #: 03 441452712698 <no name> Channel: Zap/6-1 (Admin) (unmonitored) 00:01:38 User #: 04 219 Alec Channel: SIP/219-08a6a0f0 (Admin) (unmonitored) 00:00:31 4 users in that conference. pbx*CLI> core show channel SIP/seattleconference-08a1fc68 -- General -- Name: SIP/seattleconference-08a1fc68 Type: SIP UniqueID: 1210346914.429 Caller ID: 293 Caller ID Name: Conference DNID Digits: 7000 State: Up (6) Rings: 0 NativeFormats: 0x4 (ulaw) WriteFormat: 0x40 (slin) ReadFormat: 0x40 (slin) WriteTranscode: Yes ReadTranscode: Yes 1st File Descriptor: 62 Frames in: 12330 Frames out: 21899 Time to Hangup: 0 Elapsed Time: 0h7m23s Direct Bridge: <none> Indirect Bridge: <none> -- PBX -- Context: internal Extension: 7000 Priority: 1 Call Group: 0 Pickup Group: 0 Application: MeetMe Data: 642696|aciAsdpr| Blocking in: ast_waitfor_nandfds Variables: MEETME_RECORDINGFILE=conf-recordings/642696-160 AstVar=0 SIPCALLID=481448f4-a728d931-ee37cd72@192.168.250.51 SIPUSERAGENT=PolycomSoundStationIP-SSIP_4000-UA/2.0.3.0127 SIPDOMAIN=192.168.100.1 SIPURI=sip:seattleconference@192.168.250.51 CDR Variables: level 1: clid="Conference" <293> level 1: src=293 level 1: dst=7000 level 1: dcontext=internal level 1: channel=SIP/seattleconference-08a1fc68 level 1: lastapp=MeetMe level 1: lastdata=642696|aciAsdpr| level 1: start=2008-05-09 11:28:34 level 1: answer=2008-05-09 11:28:39 level 1: end=2008-05-09 11:28:39 level 1: duration=0 level 1: billsec=0 level 1: disposition=ANSWERED level 1: amaflags=DOCUMENTATION level 1: uniqueid=1210346914.429 pbx*CLI> core show channel SIP/205-08a211e8 -- General -- Name: SIP/205-08a211e8 Type: SIP UniqueID: 1210347074.433 Caller ID: 205 Caller ID Name: Todd DNID Digits: 7000 State: Up (6) Rings: 0 NativeFormats: 0x4 (ulaw) WriteFormat: 0x40 (slin) ReadFormat: 0x40 (slin) WriteTranscode: Yes ReadTranscode: Yes 1st File Descriptor: 77 Frames in: 14819 Frames out: 14375 Time to Hangup: 0 Elapsed Time: 0h4m54s Direct Bridge: <none> Indirect Bridge: <none> -- PBX -- Context: internal Extension: 7000 Priority: 1 Call Group: 0 Pickup Group: 0 Application: MeetMe Data: 642696|aciAsdpr| Blocking in: ast_waitfor_nandfds Variables: MEETME_RECORDINGFILE=conf-recordings/642696-160 AstVar=1 SIPCALLID=00164705-14aa0041-6a87b425-52e6b4c1@192.168.100.210 SIPUSERAGENT=Cisco-CP7940G/8.0 SIPDOMAIN=192.168.100.1 SIPURI=sip:205@192.168.100.210:5060 CDR Variables: level 1: clid="Todd" <205> level 1: src=205 level 1: dst=7000 level 1: dcontext=internal level 1: channel=SIP/205-08a211e8 level 1: lastapp=MeetMe level 1: lastdata=642696|aciAsdpr| level 1: start=2008-05-09 11:31:14 level 1: answer=2008-05-09 11:31:23 level 1: end=2008-05-09 11:31:23 level 1: duration=0 level 1: billsec=0 level 1: disposition=ANSWERED level 1: amaflags=DOCUMENTATION level 1: uniqueid=1210347074.433 pbx*CLI> core show channel Zap/6-1 -- General -- Name: Zap/6-1 Type: Zap UniqueID: 1210347122.434 Caller ID: 441452712698 Caller ID Name: (N/A) DNID Digits: 4436563323 State: Up (6) Rings: 1 NativeFormats: 0x44 (ulaw|slin) WriteFormat: 0x40 (slin) ReadFormat: 0x40 (slin) WriteTranscode: No ReadTranscode: No 1st File Descriptor: 18 Frames in: 12747 Frames out: 1694 Time to Hangup: 0 Elapsed Time: 0h4m18s Direct Bridge: <none> Indirect Bridge: <none> -- PBX -- Context: conferencing Extension: s Priority: 4 Call Group: 0 Pickup Group: 0 Application: MeetMe Data: 642696|aciAsdpr| Blocking in: ast_waitfor_nandfds Variables: MEETME_RECORDINGFILE=conf-recordings/642696-160 AstVar=2 PLAYBACKSTATUS=SUCCESS CALLEDTON=33 ANI2=0 TRANSFERCAPABILITY=3K1AUDIO CDR Variables: level 1: clid=441452712698 level 1: src=441452712698 level 1: dst=s level 1: dcontext=conferencing level 1: channel=Zap/6-1 level 1: lastapp=MeetMe level 1: lastdata=642696|aciAsdpr| level 1: start=2008-05-09 11:32:02 level 1: answer=2008-05-09 11:32:02 level 1: end=2008-05-09 11:32:02 level 1: duration=0 level 1: billsec=0 level 1: disposition=ANSWERED level 1: amaflags=DOCUMENTATION level 1: uniqueid=1210347122.434 pbx*CLI> core show channel SIP/219-08a6a0f0 -- General -- Name: SIP/219-08a6a0f0 Type: SIP UniqueID: 1210347205.436 Caller ID: 219 Caller ID Name: Alec DNID Digits: 7000 State: Up (6) Rings: 0 NativeFormats: 0x4 (ulaw) WriteFormat: 0x40 (slin) ReadFormat: 0x40 (slin) WriteTranscode: Yes ReadTranscode: Yes 1st File Descriptor: 82 Frames in: 9441 Frames out: 9079 Time to Hangup: 0 Elapsed Time: 0h3m6s Direct Bridge: <none> Indirect Bridge: <none> -- PBX -- Context: internal Extension: 7000 Priority: 1 Call Group: 0 Pickup Group: 0 Application: MeetMe Data: 642696|aciAsdpr| Blocking in: ast_waitfor_nandfds Variables: MEETME_RECORDINGFILE=conf-recordings/642696-160 AstVar=3 SIPCALLID=001e4a5f-5da40010-075145e2-2c6dbb82@192.168.250.50 SIPUSERAGENT=CSCO/7 SIPDOMAIN=192.168.100.1 SIPURI=sip:219@192.168.250.50:5060 CDR Variables: level 1: clid="Alec" <219> level 1: src=219 level 1: dst=7000 level 1: dcontext=internal level 1: channel=SIP/219-08a6a0f0 level 1: lastapp=MeetMe level 1: lastdata=642696|aciAsdpr| level 1: start=2008-05-09 11:33:25 level 1: answer=2008-05-09 11:33:31 level 1: end=2008-05-09 11:33:31 level 1: duration=0 level 1: billsec=0 level 1: disposition=ANSWERED level 1: amaflags=DOCUMENTATION level 1: uniqueid=1210347205.436 -- <Zap/pseudo-47320381> Playing '/var/spool/asterisk/meetme/meetme-username-642696-4' (language 'en') -- <Zap/pseudo-47320381> Playing 'conf-hasleft' (language 'en') == Spawn extension (internal, 7000, 1) exited non-zero on 'SIP/219-08a6a0f0' -- <Zap/pseudo-47320381> Playing '/var/spool/asterisk/meetme/meetme-username-642696-2' (language 'en') -- Channel 0/6, span 1 got hangup request, cause 16 -- <Zap/pseudo-47320381> Playing 'conf-hasleft' (language 'en') == Spawn extension (internal, 7000, 1) exited non-zero on 'SIP/205-08a211e8' -- <Zap/pseudo-47320381> Playing '/var/spool/asterisk/meetme/meetme-username-642696-3' (language 'en') -- Channel 0/6, span 1 got hangup, cause 102 -- <Zap/pseudo-47320381> Playing 'conf-hasleft' (language 'en') == Spawn extension (conferencing, s, 4) exited non-zero on 'Zap/6-1' -- Hungup 'Zap/6-1' pbx*CLI> meetme kick 642696 all 1 pbx*CLI> meetme kick 642696 1 -- <SIP/seattleconference-08a1fc68> Playing 'conf-kicked' (language 'en') -- Hungup 'Zap/pseudo-1440941539' -- Hungup 'Zap/pseudo-47320381' == Spawn extension (internal, 7000, 1) exited non-zero on 'SIP/seattleconference-08a1fc68' | ||
Comments: | By: Jason Parker (jparker) 2008-05-14 12:06:47 I believe that this has already been fixed. Please always test the latest version of Asterisk before reporting bugs. If after upgrading to the latest version, you can reproduce this issue, feel free to reopen. By: Jason Dixon (fuzzyping) 2008-05-14 12:59:16 Sorry for re-opening, but I see no other way to ask questions or add comments to this ticket. Do you have any idea which version might have fixed this? I'd like to narrow down the problem and see what was fixed. Please feel free to close this ticket again. Thanks, Jason By: Jason Parker (jparker) 2008-05-14 13:54:59 You are using a version that was released 9 months ago. There have been 1129 changes since 1.4.10 - your request is quite unreasonable. By: Jason Dixon (fuzzyping) 2008-06-18 10:52:13 We have upgraded to the following bits, this problem is still reproducible. asterisk-1.4.21 asterisk-addons-1.4.7 libpmi-1.4.4 zaptel-1.4.10.1 Here is the meetme information from the most recent lockup: pbx*CLI> meetme list 642696 User #: 02 293 Conference Channel: SIP/seattleconference-08477840 (Admin) (unmonitored) 00:22:58 1 users in that conference. pbx*CLI> meetme kick 642696 all -- <SIP/seattleconference-08477840> Playing 'conf-kicked' (language 'en') -- Hungup 'Zap/pseudo-1280319490' -- Hungup 'Zap/pseudo-785979283' == Spawn extension (internal, 7000, 1) exited non-zero on 'SIP/seattleconference-08477840' By: Tilghman Lesher (tilghman) 2008-09-08 14:29:26 Could you upload the output of a 'sip set debug on' of this phone sending an INVITE (preferably one where you're causing this problem to occur)? By: Leif Madsen (lmadsen) 2009-02-02 16:37:42.000-0600 Suspended due to lack of activity. Please request a bug marshal on #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested. Thanks! |