[asterisk-users] Polycom causes conference to fail

[Jason Dixon]

On May 9, 2008, at 12:27 PM, Philipp Kempgen wrote:
> Jason Dixon schrieb:
>> 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.
>>
>> I've managed to get the output from "meetme list", "core show  
>> channel"
>> and "meetme kick <id> 1" from today, but it's a lot of output.  Can I
>> forward this to the list?  I'm not very experienced with Asterisk, so
>> any assistance would be greatly appreciated.
>
> If you think Asterisk may somehow be a part of the problem
> (... "loss of audio for all participants" ...) the best thing
> would probably be to open a bug in the bug tracker.
> http://bugs.digium.com


Considering we have other Polycoms (same model) operating successfully  
in bridges, I'm hesitant to put all of the blame on an Asterisk bug.   
But I guess it couldn't hurt, worst case is they smack me down and  
tell me what we fudged up.  :)

For the sake of curiosity (if anyone is), here is the channel  
information for the Polycom while it's in the frozen state.  Just  
below that is the output from kicking it.

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 at 192.168.250.51
SIPUSERAGENT=PolycomSoundStationIP-SSIP_4000-UA/2.0.3.0127
SIPDOMAIN=192.168.100.1
SIPURI=sip:seattleconference at 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> 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'


Thanks,

---
Jason Dixon
OmniTI Computer Consulting, Inc.
jdixon at omniti.com
443.325.1357 x.241