DevilFish
2005-Jan-15 16:14 UTC
[Asterisk-Users] Polycom IP600 - Bridge stops because we're zombie or need a soft hangup
I'm having trouble with both my Polycom IP600 and IP500 disconnecting calls to the PSTN after about 1 hour. The below log is of a phone call that lasted 1hr 39mins which is my record so far. I cannot figure out what is causing the call to terminate and I am hoping somone on this list can help me. In this example both the phone and the asterisk server have public IP addresses so NAT shoul not be an issue whatsoever. Any ideas or help would be greatly appreciated. -DevilFish Asterisk Version: Asterisk CVS-v1-0-01/13/05 Call Start: Jan 15 12:46:59 VERBOSE[4290]: -- Executing SetGroup("SIP/302-928e", "302") in new stack Jan 15 12:46:59 VERBOSE[4290]: -- Executing Dial("SIP/302-928e", "SIP/12699264242@sip_proxy-out|30") in new stack Jan 15 12:46:59 VERBOSE[4290]: -- SIP/sip_proxy-out-f201 is making progress passing it to SIP/302-928e Jan 15 12:47:08 VERBOSE[4290]: -- SIP/sip_proxy-out-f201 answered SIP/302-928e Jan 15 12:47:08 VERBOSE[4290]: -- Attempting native bridge of SIP/302-928e and SIP/sip_proxy-out-f201 Call Terminating Jan 15 14:25:42 DEBUG[4290]: Stopping retransmission on '1a59ffd871845b5d27c9b56824a47d5b@voip.acd.net' of Response 661161239: Found Jan 15 14:25:57 DEBUG[4290]: Stopping retransmission on '1a59ffd871845b5d27c9b56824a47d5b@voip.acd.net' of Response 661161240: Found Jan 15 14:26:15 DEBUG[4290]: Failed to grab lock, trying again... Jan 15 14:26:15 VERBOSE[4290]: -- Started music on hold, class 'default', on SIP/302-928e Jan 15 14:26:16 DEBUG[4290]: Bridge stops because we're zombie or need a soft hangup: c0=SIP/302-928e, c1=SIP/sip_proxy-out-f201, flags: No,No,No,Yes Jan 15 14:26:16 DEBUG[4290]: Bridge stops bridging channels SIP/302-928e and SIP/sip_proxy-out-f201 Jan 15 14:26:16 DEBUG[4290]: Ignoring too old packet packet 661161242 (expecting >= 661161243) Jan 15 14:26:16 DEBUG[4290]: update_user_counter(12699264242) - decrement inUse counter Jan 15 14:26:16 DEBUG[4290]: 12699264242 is not a local user Jan 15 14:26:16 DEBUG[4290]: Exiting with DIALSTATUS=ANSWER. Call length 1hr 39mins -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050115/5409ba00/attachment.htm