Call hanging up after 30 minutes (exactly)
Hello, I have problems with calls being hung up after 30 minutes. I do not know if the problem is with FreeSWITCH or my sip provider. I've got a log here: http://pastebin.freeswitch.org/11962 Can you please point me in the right direction? Also, why is a second invite sent after 15 minutes? Regards, Jonas
Ok. And it's my sip provider and not FreeSWITCH that do not refresh properly? On Wed, Jan 27, 2010 at 8:40 PM, Brian West <brian@freeswitch.org> wrote:
Its a session timer.
/b
On Jan 27, 2010, at 1:26 PM, Jonas Gauffin wrote:
Also, why is a second invite sent after 15 minutes?
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
We are experiencing an odd issue. We have many calls that don't drop, but some do after being up a minute or two. The reason code is NORMAL_TEMPORARY_FAILURE and the sip status that is triggering that is 503 (Service Unavailable). With only one or two calls up at a time, I don't think it's a session limit issue (set to 1000). Here is the console log from just before the 503 status - any help is greatly appreciated! 2010-01-27 12:49:11.879251 [DEBUG] sofia.c:4003 Channel sofia/internal/400@192.168.0.31 entering state [calling][0] 2010-01-27 12:49:11.899334 [INFO] sofia.c:597 Update Callee ID to "400" <400> 2010-01-27 12:49:11.919018 [DEBUG] sofia.c:4003 Channel sofia/internal/400@192.168.0.31 entering state [ready][200] 2010-01-27 12:49:11.919018 [DEBUG] sofia.c:4011 Duplicate SDP v=0 o=- 1264621687 1264621687 IN IP4 192.168.0.46 s=Polycom IP Phone c=IN IP4 192.168.0.46 t=0 0 a=sendrecv m=audio 2222 RTP/AVP 0 8 18 127 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 2010-01-27 12:50:06.068999 [DEBUG] sofia.c:4003 Channel sofia/internal/400@192.168.0.31 entering state [calling][0] 2010-01-27 12:50:06.108877 [DEBUG] sofia.c:4003 Channel sofia/internal/400@192.168.0.31 entering state [ready][200] 2010-01-27 12:50:06.108877 [DEBUG] sofia.c:4011 Duplicate SDP v=0 o=- 1264621687 1264621687 IN IP4 192.168.0.46 s=Polycom IP Phone c=IN IP4 192.168.0.46 t=0 0 a=sendrecv m=audio 2222 RTP/AVP 0 8 18 127 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 2010-01-27 12:51:05.259614 [DEBUG] sofia.c:4003 Channel sofia/internal/400@192.168.0.31 entering state [calling][0] 2010-01-27 12:51:05.298972 [DEBUG] sofia.c:4003 Channel sofia/internal/400@192.168.0.31 entering state [ready][200] 2010-01-27 12:51:05.298972 [DEBUG] sofia.c:4011 Duplicate SDP v=0 o=- 1264621687 1264621687 IN IP4 192.168.0.46 s=Polycom IP Phone c=IN IP4 192.168.0.46 t=0 0 a=sendrecv m=audio 2222 RTP/AVP 0 8 18 127 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 2010-01-27 12:52:05.369138 [DEBUG] sofia.c:4003 Channel sofia/internal/400@192.168.0.31 entering state [calling][0] 2010-01-27 12:52:05.399251 [DEBUG] sofia.c:4003 Channel sofia/internal/400@192.168.0.31 entering state [ready][200] 2010-01-27 12:52:05.399251 [DEBUG] sofia.c:4011 Duplicate SDP v=0 o=- 1264621687 1264621687 IN IP4 192.168.0.46 s=Polycom IP Phone c=IN IP4 192.168.0.46 t=0 0 a=sendrecv m=audio 2222 RTP/AVP 0 8 18 127 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 2010-01-27 12:53:09.599069 [DEBUG] sofia.c:4003 Channel sofia/internal/400@192.168.0.31 entering state [calling][0] 2010-01-27 12:53:09.599069 [DEBUG] sofia.c:4003 Channel sofia/internal/400@192.168.0.31 entering state [terminating][503] 2010-01-27 12:53:09.599069 [NOTICE] sofia.c:4647 Hangup sofia/internal/400@192.168.0.31 [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] 2010-01-27 12:53:09.599069 [DEBUG] switch_ivr_bridge.c:466 sofia/internal/400@192.168.0.31 ending bridge by request from write function
try turning on sip trace as well to see the sip traffic sofia profile internal siptrace on (from cli) probably its something that said it could do session timers but was lying On Wed, Jan 27, 2010 at 2:05 PM, Troy Anderson <troy@tlainvestments.com>wrote:
We are experiencing an odd issue. We have many calls that don't drop, but some do after being up a minute or two.
The reason code is NORMAL_TEMPORARY_FAILURE and the sip status that is triggering that is 503 (Service Unavailable). With only one or two calls up at a time, I don't think it's a session limit issue (set to 1000).
Here is the console log from just before the 503 status - any help is greatly appreciated!
2010-01-27 12:49:11.879251 [DEBUG] sofia.c:4003 Channel sofia/internal/ 400@192.168.0.31 entering state [calling][0] 2010-01-27 12:49:11.899334 [INFO] sofia.c:597 Update Callee ID to "400" <400> 2010-01-27 12:49:11.919018 [DEBUG] sofia.c:4003 Channel sofia/internal/ 400@192.168.0.31 entering state [ready][200] 2010-01-27 12:49:11.919018 [DEBUG] sofia.c:4011 Duplicate SDP v=0 o=- 1264621687 1264621687 IN IP4 192.168.0.46 s=Polycom IP Phone c=IN IP4 192.168.0.46 t=0 0 a=sendrecv m=audio 2222 RTP/AVP 0 8 18 127 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000
2010-01-27 12:50:06.068999 [DEBUG] sofia.c:4003 Channel sofia/internal/ 400@192.168.0.31 entering state [calling][0] 2010-01-27 12:50:06.108877 [DEBUG] sofia.c:4003 Channel sofia/internal/ 400@192.168.0.31 entering state [ready][200] 2010-01-27 12:50:06.108877 [DEBUG] sofia.c:4011 Duplicate SDP v=0 o=- 1264621687 1264621687 IN IP4 192.168.0.46 s=Polycom IP Phone c=IN IP4 192.168.0.46 t=0 0 a=sendrecv m=audio 2222 RTP/AVP 0 8 18 127 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000
2010-01-27 12:51:05.259614 [DEBUG] sofia.c:4003 Channel sofia/internal/ 400@192.168.0.31 entering state [calling][0] 2010-01-27 12:51:05.298972 [DEBUG] sofia.c:4003 Channel sofia/internal/ 400@192.168.0.31 entering state [ready][200] 2010-01-27 12:51:05.298972 [DEBUG] sofia.c:4011 Duplicate SDP v=0 o=- 1264621687 1264621687 IN IP4 192.168.0.46 s=Polycom IP Phone c=IN IP4 192.168.0.46 t=0 0 a=sendrecv m=audio 2222 RTP/AVP 0 8 18 127 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000
2010-01-27 12:52:05.369138 [DEBUG] sofia.c:4003 Channel sofia/internal/ 400@192.168.0.31 entering state [calling][0] 2010-01-27 12:52:05.399251 [DEBUG] sofia.c:4003 Channel sofia/internal/ 400@192.168.0.31 entering state [ready][200] 2010-01-27 12:52:05.399251 [DEBUG] sofia.c:4011 Duplicate SDP v=0 o=- 1264621687 1264621687 IN IP4 192.168.0.46 s=Polycom IP Phone c=IN IP4 192.168.0.46 t=0 0 a=sendrecv m=audio 2222 RTP/AVP 0 8 18 127 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000
2010-01-27 12:53:09.599069 [DEBUG] sofia.c:4003 Channel sofia/internal/ 400@192.168.0.31 entering state [calling][0] 2010-01-27 12:53:09.599069 [DEBUG] sofia.c:4003 Channel sofia/internal/ 400@192.168.0.31 entering state [terminating][503] 2010-01-27 12:53:09.599069 [NOTICE] sofia.c:4647 Hangup sofia/internal/ 400@192.168.0.31 [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] 2010-01-27 12:53:09.599069 [DEBUG] switch_ivr_bridge.c:466 sofia/internal/ 400@192.168.0.31 ending bridge by request from write function
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
Of course the error didn't show up in the 4 hours I had the sip trace on... I downgraded the firmware on the Polycom 301's to 3.3.1RevB in stead of 3.2.2 and don't seem to be having the problem any more. If it comes back, we'll break out sip trance again to see what's up. Thanks! -Troy On Jan 27, 2010, at 3:31 PM, Anthony Minessale wrote:
try turning on sip trace as well to see the sip traffic
sofia profile internal siptrace on (from cli) probably its something that said it could do session timers but was lying
On Wed, Jan 27, 2010 at 2:05 PM, Troy Anderson <troy@tlainvestments.com> wrote: We are experiencing an odd issue. We have many calls that don't drop, but some do after being up a minute or two.
The reason code is NORMAL_TEMPORARY_FAILURE and the sip status that is triggering that is 503 (Service Unavailable). With only one or two calls up at a time, I don't think it's a session limit issue (set to 1000).
Here is the console log from just before the 503 status - any help is greatly appreciated!
2010-01-27 12:49:11.879251 [DEBUG] sofia.c:4003 Channel sofia/internal/400@192.168.0.31 entering state [calling][0] 2010-01-27 12:49:11.899334 [INFO] sofia.c:597 Update Callee ID to "400" <400> 2010-01-27 12:49:11.919018 [DEBUG] sofia.c:4003 Channel sofia/internal/400@192.168.0.31 entering state [ready][200] 2010-01-27 12:49:11.919018 [DEBUG] sofia.c:4011 Duplicate SDP v=0 o=- 1264621687 1264621687 IN IP4 192.168.0.46 s=Polycom IP Phone c=IN IP4 192.168.0.46 t=0 0 a=sendrecv m=audio 2222 RTP/AVP 0 8 18 127 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000
2010-01-27 12:50:06.068999 [DEBUG] sofia.c:4003 Channel sofia/internal/400@192.168.0.31 entering state [calling][0] 2010-01-27 12:50:06.108877 [DEBUG] sofia.c:4003 Channel sofia/internal/400@192.168.0.31 entering state [ready][200] 2010-01-27 12:50:06.108877 [DEBUG] sofia.c:4011 Duplicate SDP v=0 o=- 1264621687 1264621687 IN IP4 192.168.0.46 s=Polycom IP Phone c=IN IP4 192.168.0.46 t=0 0 a=sendrecv m=audio 2222 RTP/AVP 0 8 18 127 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000
2010-01-27 12:51:05.259614 [DEBUG] sofia.c:4003 Channel sofia/internal/400@192.168.0.31 entering state [calling][0] 2010-01-27 12:51:05.298972 [DEBUG] sofia.c:4003 Channel sofia/internal/400@192.168.0.31 entering state [ready][200] 2010-01-27 12:51:05.298972 [DEBUG] sofia.c:4011 Duplicate SDP v=0 o=- 1264621687 1264621687 IN IP4 192.168.0.46 s=Polycom IP Phone c=IN IP4 192.168.0.46 t=0 0 a=sendrecv m=audio 2222 RTP/AVP 0 8 18 127 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000
2010-01-27 12:52:05.369138 [DEBUG] sofia.c:4003 Channel sofia/internal/400@192.168.0.31 entering state [calling][0] 2010-01-27 12:52:05.399251 [DEBUG] sofia.c:4003 Channel sofia/internal/400@192.168.0.31 entering state [ready][200] 2010-01-27 12:52:05.399251 [DEBUG] sofia.c:4011 Duplicate SDP v=0 o=- 1264621687 1264621687 IN IP4 192.168.0.46 s=Polycom IP Phone c=IN IP4 192.168.0.46 t=0 0 a=sendrecv m=audio 2222 RTP/AVP 0 8 18 127 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000
2010-01-27 12:53:09.599069 [DEBUG] sofia.c:4003 Channel sofia/internal/400@192.168.0.31 entering state [calling][0] 2010-01-27 12:53:09.599069 [DEBUG] sofia.c:4003 Channel sofia/internal/400@192.168.0.31 entering state [terminating][503] 2010-01-27 12:53:09.599069 [NOTICE] sofia.c:4647 Hangup sofia/internal/400@192.168.0.31 [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] 2010-01-27 12:53:09.599069 [DEBUG] switch_ivr_bridge.c:466 sofia/internal/400@192.168.0.31 ending bridge by request from write function
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (4)
-
Anthony Minessale
-
Brian West
-
Jonas Gauffin
-
Troy Anderson