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jdixon at omniti.com
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 Posted: Fri May 09, 2008 10:52 am Post subject: [asterisk-users] Polycom causes conference to fail |
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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.
Thanks,
---
Jason Dixon
OmniTI Computer Consulting, Inc.
jdixon at omniti.com
443.325.1357 x.241 |
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philipp.kempgen at amo...
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| Posted: Fri May 09, 2008 11:27 am Post subject: [asterisk-users] Polycom causes conference to fail |
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Jason Dixon schrieb:
| Quote: | 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
Gr??e,
Philipp Kempgen
--
Asterisk-Tag.org 2008, 26.-27. Mai -> http://www.asterisk-tag.org
amooma GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de
Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 |
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jdixon at omniti.com
Guest
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| Posted: Fri May 09, 2008 11:36 am Post subject: [asterisk-users] Polycom causes conference to fail |
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On May 9, 2008, at 12:27 PM, Philipp Kempgen wrote:
| Quote: | Jason Dixon schrieb:
| Quote: | 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 |
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jdixon at omniti.com
Guest
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| Posted: Mon May 12, 2008 9:36 am Post subject: [asterisk-users] Polycom causes conference to fail |
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Sorry to be a pest, but does anyone have any ideas on this? I've
opened a bug, but I was hoping someone else on the list has
encountered this issue before.
Thanks,
Jason
On May 9, 2008, at 12:36 PM, Jason Dixon wrote:
| Quote: | | Quote: |
| Quote: | 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.
|
|
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'
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---
Jason Dixon
OmniTI Computer Consulting, Inc.
jdixon at omniti.com
443.325.1357 x.241 |
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andersen at mwdental.com
Guest
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| Posted: Mon May 12, 2008 10:49 am Post subject: [asterisk-users] Polycom causes conference to fail |
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| Quote: | Sorry to be a pest, but does anyone have any ideas on this? I've
opened a bug, but I was hoping someone else on the list has
encountered this issue before.
Jason
|
Does the Polycom have the Buddy List turned on? We had an IP601 that
would reboot (or lock up) about 60% of the time when IT started a
"Page All" (which is nothing but a meetme). All of the other phones
would send their "status" for the Buddy List to the IP601 and would
simply overwhelm it - causing a reboot.
Just a shot in the dark...
Bill |
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