Can u debug on AS ? On Thu, Mar 7, 2013 at 9:20 AM, Mickael Monsieur <mickael.monsi...@gmail.com> wrote: > Le 7/03/13 11:21, Steven Howes a écrit : > >> On 7 Mar 2013, at 10:12, Mickael Monsieur wrote: >>> >>> Do you have an explanation? >> >> Put a SIP debug on and we may be able to find one.. >> >> Steve > > Hello Steve, > After checking, I confirm that the call is cut precisely to 900 seconds (15 > min). > > 10.4.0.1 = Asterisk > 10.4.0.10 = Cisco AS 5300 > > Info : debug start at 14min30sec > > set_destination: Parsing <sip:0032487997160@10.4.0.10:5060> for address/port > to send to > set_destination: set destination to 10.4.0.10, port 5060 > Audio is at 10.4.0.1 port 11842 > Adding codec 0x8 (alaw) to SDP > Adding codec 0x4 (ulaw) to SDP > Reliably Transmitting (NAT) to 10.4.0.10:54789: > INVITE sip:0032487997160@10.4.0.10:5060 SIP/2.0 > Via: SIP/2.0/UDP 10.4.0.1:5060;branch=z9hG4bK41af21b3;rport > Max-Forwards: 70 > From: <sip:65939191@10.4.0.1>;tag=as12acaefb > To: <sip:0032487997160@10.4.0.10>;tag=36CA05C-167B > Contact: <sip:65939191@10.4.0.1> > Call-ID: FA122D28-BF9A11D3-83D393BC-25F3EF03@10.4.0.10 > CSeq: 102 INVITE > User-Agent: isdnbox1.1 > Require: timer > Session-Expires: 1800;refresher=uas > Min-SE: 90 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO > Supported: replaces, timer > X-asterisk-Info: SIP re-invite (Session-Timers) > Content-Type: application/sdp > Content-Length: 207 > > v=0 > o=root 1538728127 1538728127 IN IP4 10.4.0.1 > s=Asterisk PBX 1.6.2.9-2+squeeze8 > c=IN IP4 10.4.0.1 > t=0 0 > m=audio 11842 RTP/AVP 8 0 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=ptime:20 > a=sendrecv > > --- > > <--- SIP read from UDP:10.4.0.10:5060 ---> > SIP/2.0 420 Bad Extension > Via: SIP/2.0/UDP 10.4.0.1:5060;branch=z9hG4bK41af21b3;rport > From: <sip:65939191@10.4.0.1>;tag=as12acaefb > To: <sip:0032487997160@10.4.0.10>;tag=36CA05C-167B > Call-ID: FA122D28-BF9A11D3-83D393BC-25F3EF03@10.4.0.10 > CSeq: 102 INVITE > Unsupported: timer > Content-Length: 0 > > > <-------------> > --- (8 headers 0 lines) --- > > -- Got SIP response 420 "Bad Extension" back from 10.4.0.10 > set_destination: Parsing <sip:0032487997160@10.4.0.10:5060> for address/port > to send to > set_destination: set destination to 10.4.0.10, port 5060 > Transmitting (NAT) to 10.4.0.10:5060: > ACK sip:0032487997160@10.4.0.10:5060 SIP/2.0 > Via: SIP/2.0/UDP 10.4.0.1:5060;branch=z9hG4bK41af21b3;rport > Max-Forwards: 70 > From: <sip:65939191@10.4.0.1>;tag=as12acaefb > To: <sip:0032487997160@10.4.0.10>;tag=36CA05C-167B > Contact: <sip:65939191@10.4.0.1> > Call-ID: FA122D28-BF9A11D3-83D393BC-25F3EF03@10.4.0.10 > CSeq: 102 ACK > User-Agent: isdnbox1.1 > Content-Length: 0 > > > --- > -- Stopped music on hold on SIP/as5300-1-00000050 > == Spawn extension (dialin, 065939191, 2) exited non-zero on > 'SIP/as5300-1-00000050' > Reliably Transmitting (NAT) to 10.4.0.10:5060: > OPTIONS sip:10.4.0.10 SIP/2.0 > Via: SIP/2.0/UDP 10.4.0.1:5060;branch=z9hG4bK4d8b5654;rport > Max-Forwards: 70 > From: "asterisk" <sip:asterisk@10.4.0.1>;tag=as4eb3efa7 > To: <sip:10.4.0.10> > Contact: <sip:asterisk@10.4.0.1> > Call-ID: 6a43ad4b27d870d048e8425077bcc075@10.4.0.1 > CSeq: 102 OPTIONS > User-Agent: isdnbox1.1 > Date: Thu, 07 Mar 2013 11:17:44 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO > Supported: replaces, timer > Content-Length: 0 > > > --- > > <--- SIP read from UDP:10.4.0.10:5060 ---> > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.4.0.1:5060;branch=z9hG4bK4d8b5654;rport > From: "asterisk" <sip:asterisk@10.4.0.1>;tag=as4eb3efa7 > To: <sip:10.4.0.10>;tag=37A724C-211C > Date: Sat, 01 Jan 2000 16:12:32 GMT > Call-ID: 6a43ad4b27d870d048e8425077bcc075@10.4.0.1 > Server: Cisco-SIPGateway/IOS-12.x > Content-Type: application/sdp > CSeq: 102 OPTIONS > Supported: 100rel > Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, > NOTIFY, INFO > Accept: application/sdp > Allow-Events: telephone-event > Content-Length: 154 > > v=0 > o=CiscoSystemsSIP-GW-UserAgent 793 5073 IN IP4 10.4.0.10 > s=SIP Call > c=IN IP4 10.4.0.10 > t=0 0 > m=audio 0 RTP/AVP 18 0 8 4 2 15 3 > c=IN IP4 10.4.0.10 > > <-------------> > --- (14 headers 7 lines) --- > Really destroying SIP dialog '6a43ad4b27d870d048e8425077bcc075@10.4.0.1' > Method: OPTIONS > > <--- SIP read from UDP:10.4.0.10:54336 ---> > BYE sip:65939191@10.4.0.1:5060 SIP/2.0 > Via: SIP/2.0/UDP 10.4.0.10:5060 > From: <sip:0032487997160@10.4.0.10>;tag=36CA05C-167B > To: <sip:65939191@10.4.0.1>;tag=as12acaefb > Date: Sat, 01 Jan 2000 16:12:26 GMT > Call-ID: FA122D28-BF9A11D3-83D393BC-25F3EF03@10.4.0.10 > User-Agent: Cisco-SIPGateway/IOS-12.x > Max-Forwards: 6 > Timestamp: 946743153 > CSeq: 102 BYE > Content-Length: 0 > > > <-------------> > --- (11 headers 0 lines) --- > > <--- Transmitting (NAT) to 10.4.0.10:54336 ---> > SIP/2.0 481 Call leg/transaction does not exist > Via: SIP/2.0/UDP 10.4.0.10:5060;received=10.4.0.10 > From: <sip:0032487997160@10.4.0.10>;tag=36CA05C-167B > To: <sip:65939191@10.4.0.1>;tag=as12acaefb > Call-ID: FA122D28-BF9A11D3-83D393BC-25F3EF03@10.4.0.10 > CSeq: 102 BYE > Server: isdnbox1.1 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO > Supported: replaces, timer > Content-Length: 0 > > > <------------> > > 15 min (call ended) > > > >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? 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-- Eduardo A. Muñoz GPG Key fingerprint = 175E 6AEB AD23 8EFE 0FC3 F558 9AB1 7885 40A4 ABBB CCNA - CCNP -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users