Call end after 15 minutes

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Call end after 15 minutes

Postby ikenny » Tue Feb 11, 2014 4:35 pm

Hello.
I have the following problem.
Incoming call from trunk(Cisco as) ends after 15 minutes of conversation.Call comes to queue.
Alfter 15 minutes this happens:
Code: Select all
[Feb 12 00:21:47] VERBOSE[32049] chan_sip.c: set_destination: Parsing <sip:93741948@cisco.ip:5060> for address/port to send to
[Feb 12 00:21:47] VERBOSE[32049] chan_sip.c: set_destination: set destination to cisco.ip:5060
[Feb 12 00:21:47] VERBOSE[32049] chan_sip.c: Audio is at 15924
[Feb 12 00:21:47] VERBOSE[32049] chan_sip.c: Adding codec 100003 (ulaw) to SDP
[Feb 12 00:21:47] VERBOSE[32049] chan_sip.c: Reliably Transmitting (no NAT) to cisco.ip:5060:
�INVITE sip:93741948@cisco.ip:5060 SIP/2.0
�Via: SIP/2.0/UDP asterisk.ip:5060;branch=z9hG4bK0c70f79b
�Max-Forwards: 70
�From: <sip:13472975@asterisk.ip>;tag=as596dc234
�To: <sip:93741948@asterisk.ip>;tag=F639CD5C-1674
�Contact: <sip:13472975@asterisk.ip:5060>
�Call-ID: A259898F-929F11E3-9FE9F57B-15CB26EA@cisco.ip
�CSeq: 102 INVITE
�User-Agent: FPBX-2.8.1(11.7.0)
�Session-Expires: 1800;refresher=uac
�Min-SE: 90
�Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
�Supported: replaces, timer
�X-asterisk-Info: SIP re-invite (Session-Timers)
�Content-Type: application/sdp
�Content-Length: 179

�v=0
�o=root 601906171 601906171 IN IP4 asterisk.ip
�s=Asterisk PBX 11.7.0
�c=IN IP4 asterisk.ip
�t=0 0
�m=audio 15924 RTP/AVP 0
�a=rtpmap:0 PCMU/8000
�a=ptime:20
�a=sendrecv

�---
[Feb 12 00:21:47] VERBOSE[32049] chan_sip.c:
�<--- SIP read from UDP:cisco.ip:5060 --->
�SIP/2.0 100 Trying
�Via: SIP/2.0/UDP asterisk.ip:5060;branch=z9hG4bK0c70f79b
�From: <sip:13472975@asterisk.ip>;tag=as596dc234
�To: <sip:93741948@asterisk.ip>;tag=F639CD5C-1674
�Date: Tue, 11 Feb 2014 22:21:46 GMT
�Call-ID: A259898F-929F11E3-9FE9F57B-15CB26EA@cisco.ip
�Server: Cisco-SIPGateway/IOS-12.x
�CSeq: 102 INVITE
�Allow-Events: telephone-event
�Remote-Party-ID: <sip:93741948@cisco.ip>;party=called;screen=yes;privacy=off
�Content-Length: 0

�<------------->
[Feb 12 00:21:47] VERBOSE[32049] chan_sip.c: --- (11 headers 0 lines) ---
[Feb 12 00:21:47] VERBOSE[32049] chan_sip.c:
�<--- SIP read from UDP:cisco.ip:5060 --->
�SIP/2.0 200 OK
�Via: SIP/2.0/UDP asterisk.ip:5060;branch=z9hG4bK0c70f79b
�From: <sip:13472975@asterisk.ip>;tag=as596dc234
�To: <sip:93741948@asterisk.ip>;tag=F639CD5C-1674
�Date: Tue, 11 Feb 2014 22:21:46 GMT
�Call-ID: A259898F-929F11E3-9FE9F57B-15CB26EA@cisco.ip
�Server: Cisco-SIPGateway/IOS-12.x
�CSeq: 102 INVITE
�Session-Expires: 1800;refresher=uac
�Require: timer
�Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
�Supported: replaces
�Allow-Events: telephone-event
�Remote-Party-ID: <sip:93741948@cisco.ip>;party=called;screen=yes;privacy=off
�Contact: <sip:93741948@cisco.ip:5060>
�Content-Type: application/sdp
�Content-Length: 194

�v=0
�o=CiscoSystemsSIP-GW-UserAgent 4550 6702 IN IP4 cisco.ip
�s=SIP Call
�c=IN IP4 cisco.ip
�t=0 0
�m=audio 17962 RTP/AVP 0
�c=IN IP4 cisco.ip
�a=rtpmap:0 PCMU/8000
�a=ptime:20
�<------------->
[Feb 12 00:21:47] VERBOSE[32049] chan_sip.c: --- (17 headers 9 lines) ---
[Feb 12 00:21:47] VERBOSE[32049][C-00001849] chan_sip.c: set_destination: Parsing <sip:93741948@cisco.ip:5060> for address/port to send to
[Feb 12 00:21:47] VERBOSE[32049][C-00001849] chan_sip.c: set_destination: set destination to cisco.ip:5060
[Feb 12 00:21:47] VERBOSE[32049][C-00001849] chan_sip.c: Transmitting (no NAT) to cisco.ip:5060:
�ACK sip:93741948@cisco.ip:5060 SIP/2.0
�Via: SIP/2.0/UDP asterisk.ip:5060;branch=z9hG4bK4c1b8d6b
�Max-Forwards: 70
�From: <sip:13472975@asterisk.ip>;tag=as596dc234
�To: <sip:93741948@asterisk.ip>;tag=F639CD5C-1674
�Contact: <sip:13472975@asterisk.ip:5060>
�Call-ID: A259898F-929F11E3-9FE9F57B-15CB26EA@cisco.ip
�CSeq: 102 ACK
�User-Agent: FPBX-2.8.1(11.7.0)
�Content-Length: 0


�---
[Feb 12 00:22:06] VERBOSE[32049] chan_sip.c:
�<--- SIP read from UDP:cisco.ip:52223 --->
�BYE sip:13472975@asterisk.ip:5060 SIP/2.0
�Via: SIP/2.0/UDP cisco.ip:5060;branch=z9hG4bK7DAF5850A
�From: <sip:93741948@asterisk.ip>;tag=F639CD5C-1674
�To: <sip:13472975@asterisk.ip>;tag=as596dc234
�Date: Tue, 11 Feb 2014 22:21:47 GMT
�Call-ID: A259898F-929F11E3-9FE9F57B-15CB26EA@cisco.ip
�User-Agent: Cisco-SIPGateway/IOS-12.x
�Max-Forwards: 70
�Timestamp: 1392157325
�CSeq: 102 BYE
�Content-Length: 0

�<------------->
[Feb 12 00:22:06] VERBOSE[32049] chan_sip.c: --- (11 headers 0 lines) ---
[Feb 12 00:22:06] VERBOSE[32049][C-00001849] chan_sip.c: Sending to cisco.ip:5060 (no NAT)
[Feb 12 00:22:06] VERBOSE[32049][C-00001849] chan_sip.c: Scheduling destruction of SIP dialog 'A259898F-929F11E3-9FE9F57B-15CB26EA@cisco.ip' in 6400 ms (Method: BYE)
[Feb 12 00:22:06] VERBOSE[32049][C-00001849] chan_sip.c:
�<--- Transmitting (no NAT) to cisco.ip:5060 --->
�SIP/2.0 200 OK
�Via: SIP/2.0/UDP cisco.ip:5060;branch=z9hG4bK7DAF5850A;received=cisco.ip
�From: <sip:93741948@asterisk.ip>;tag=F639CD5C-1674
�To: <sip:13472975@asterisk.ip>;tag=as596dc234
�Call-ID: A259898F-929F11E3-9FE9F57B-15CB26EA@cisco.ip
�CSeq: 102 BYE
�Server: FPBX-2.8.1(11.7.0)
�Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
�Supported: replaces, timer
�Content-Length: 0


�<------------>
[Feb 12 00:22:06] VERBOSE[17066][C-00001849] pbx.c:     -- Executing [h@ext-queues:1] Macro("SIP/Cisco-00003497", "hangupcall,") in new stack
[Feb 12 00:22:06] VERBOSE[17066][C-00001849] pbx.c:     -- Executing [s@macro-hangupcall:1] GotoIf("SIP/Cisco-00003497", "1?endmixmoncheck") in new stack
[Feb 12 00:22:06] VERBOSE[17066][C-00001849] pbx.c:     -- Goto (macro-hangupcall,s,9)
[Feb 12 00:22:06] VERBOSE[17066][C-00001849] pbx.c:     -- Executing [s@macro-hangupcall:9] NoOp("SIP/Cisco-00003497", "End of MIXMON check") in new stack
[Feb 12 00:22:06] VERBOSE[17066][C-00001849] pbx.c:     -- Executing [s@macro-hangupcall:10] GotoIf("SIP/Cisco-00003497", "1?nomeetmemon") in new stack
[Feb 12 00:22:06] VERBOSE[17066][C-00001849] pbx.c:     -- Goto (macro-hangupcall,s,28)
[Feb 12 00:22:06] VERBOSE[17066][C-00001849] pbx.c:     -- Executing [s@macro-hangupcall:28] NoOp("SIP/Cisco-00003497", "End of MEETME check") in new stack
[Feb 12 00:22:06] VERBOSE[17066][C-00001849] pbx.c:     -- Executing [s@macro-hangupcall:29] GotoIf("SIP/Cisco-00003497", "1?noautomon") in new stack
[Feb 12 00:22:06] VERBOSE[17066][C-00001849] pbx.c:     -- Goto (macro-hangupcall,s,34)
[Feb 12 00:22:06] VERBOSE[17066][C-00001849] pbx.c:     -- Executing [s@macro-hangupcall:34] NoOp("SIP/Cisco-00003497", "TOUCH_MONITOR_OUTPUT=") in new stack
[Feb 12 00:22:06] VERBOSE[17066][C-00001849] pbx.c:     -- Executing [s@macro-hangupcall:35] GotoIf("SIP/Cisco-00003497", "0?noautomon2") in new stack
[Feb 12 00:22:06] VERBOSE[17066][C-00001849] pbx.c:     -- Executing [s@macro-hangupcall:36] System("SIP/Cisco-00003497", "test -e /var/spool/asterisk/monitor/q502-20140212-000647-1392156407.13477*") in new stack
[Feb 12 00:22:06] VERBOSE[17066][C-00001849] pbx.c:     -- Executing [s@macro-hangupcall:37] NoOp("SIP/Cisco-00003497", "SYSTEMSTATUS = SUCCESS") in new stack
[Feb 12 00:22:06] VERBOSE[17066][C-00001849] pbx.c:     -- Executing [s@macro-hangupcall:38] GotoIf("SIP/Cisco-00003497", "0?errornoautomon") in new stack
[Feb 12 00:22:06] VERBOSE[17066][C-00001849] pbx.c:     -- Executing [s@macro-hangupcall:39] Set("SIP/Cisco-00003497", "CDR(userfield)=audio:/var/spool/asterisk/monitor/q502-20140212-000647-1392156407.13477") in new stack
[Feb 12 00:22:06] VERBOSE[17066][C-00001849] pbx.c:     -- Executing [s@macro-hangupcall:40] NoOp("SIP/Cisco-00003497", "End of MONITOR QUEUE check") in new stack
[Feb 12 00:22:06] VERBOSE[17066][C-00001849] pbx.c:     -- Executing [s@macro-hangupcall:41] NoOp("SIP/Cisco-00003497", "MONITOR_FILENAME=/var/spool/asterisk/monitor/q502-20140212-000647-1392156407.13477") in new stack
[Feb 12 00:22:06] VERBOSE[17066][C-00001849] pbx.c:     -- Executing [s@macro-hangupcall:42] GotoIf("SIP/Cisco-00003497", "1?skiprg") in new stack
[Feb 12 00:22:06] VERBOSE[17066][C-00001849] pbx.c:     -- Goto (macro-hangupcall,s,45)
[Feb 12 00:22:06] VERBOSE[17066][C-00001849] pbx.c:     -- Executing [s@macro-hangupcall:45] GotoIf("SIP/Cisco-00003497", "0?skipblkvm") in new stack
[Feb 12 00:22:06] VERBOSE[17066][C-00001849] pbx.c:     -- Executing [s@macro-hangupcall:46] NoOp("SIP/Cisco-00003497", "Cleaning Up Block VM Flag: BLKVM/502/SIP/Cisco-00003497") in new stack
[Feb 12 00:22:06] VERBOSE[17066][C-00001849] pbx.c:     -- Executing [s@macro-hangupcall:47] NoOp("SIP/Cisco-00003497", "Deleting: BLKVM/502/SIP/Cisco-00003497 TRUE") in new stack
[Feb 12 00:22:06] VERBOSE[17066][C-00001849] pbx.c:     -- Executing [s@macro-hangupcall:48] GotoIf("SIP/Cisco-00003497", "1?theend") in new stack
[Feb 12 00:22:06] VERBOSE[17066][C-00001849] pbx.c:     -- Goto (macro-hangupcall,s,50)
[Feb 12 00:22:06] VERBOSE[17066][C-00001849] pbx.c:     -- Executing [s@macro-hangupcall:50] AGI("SIP/Cisco-00003497", "hangup.agi") in new stack
[Feb 12 00:22:06] VERBOSE[17066][C-00001849] res_agi.c:     -- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.agi
[Feb 12 00:22:06] VERBOSE[17066][C-00001849] res_agi.c:     -- <SIP/Cisco-00003497>AGI Script hangup.agi completed, returning 0
[Feb 12 00:22:06] VERBOSE[17066][C-00001849] pbx.c:     -- Executing [s@macro-hangupcall:51] Hangup("SIP/Cisco-00003497", "") in new stack
[Feb 12 00:22:06] VERBOSE[17066][C-00001849] app_macro.c:   == Spawn extension (macro-hangupcall, s, 51) exited non-zero on 'SIP/Cisco-00003497' in macro 'hangupcall'
[Feb 12 00:22:06] VERBOSE[17066][C-00001849] pbx.c:   == Spawn extension (ext-queues, h, 1) exited non-zero on 'SIP/Cisco-00003497'
[Feb 12 00:22:06] VERBOSE[17066][C-00001849] pbx.c:   == Spawn extension (ext-queues, 502, 12) exited non-zero on 'SIP/Cisco-00003497'
[Feb 12 00:22:06] VERBOSE[21958][C-00001849] app_mixmonitor.c:   == MixMonitor close filestream (mixed)
[Feb 12 00:22:06] VERBOSE[21958][C-00001849] app_mixmonitor.c:   == End MixMonitor Recording SIP/Cisco-00003497



My trunk settings
Code: Select all
host=ip
context=from-trunk
canreinvite=yes
qualify=yes
type=peer
insecure=very
disallow=all
allow=ulaw&alaw
ikenny
Newsterisk
 
Posts: 11
Joined: Mon Apr 08, 2013 8:17 am

Re: Call end after 15 minutes

Postby ikenny » Wed Feb 12, 2014 7:44 am

Found the problem
in sip.conf in general section
session-timers=refuse

Works fine
ikenny
Newsterisk
 
Posts: 11
Joined: Mon Apr 08, 2013 8:17 am


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