Re: [asterisk-users] Asterisk 1.6 + Cisco AS5300
Le 7/03/13 11:12, Mickael Monsieur a écrit : Hello, I have a Cisco AS5300 connected to Asterisk (1.6.2.9) Between 15-16 minutes, the call is disconnected without reason. Here is what is displayed in the debug: Received an SDES from 10.4.0.10:17399 -- Got SIP response 420 "Bad Extension" back from 10.4.0.10 -- Stopped music on hold on SIP/as5300-1-004d == Spawn extension (dialin, 065939191, 2) exited non-zero on 'SIP/as5300-1-004d' Do you have an explanation? Best regards, Mickael Ok i solved : https://issues.asterisk.org/jira/browse/ASTERISK-15787 -- _ -- 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
Re: [asterisk-users] Asterisk 1.6 + Cisco AS5300
Can u debug on AS ? On Thu, Mar 7, 2013 at 9:20 AM, Mickael Monsieur 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 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: ;tag=as12acaefb > To: ;tag=36CA05C-167B > Contact: > 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: ;tag=as12acaefb > To: ;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 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: ;tag=as12acaefb > To: ;tag=36CA05C-167B > Contact: > 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/as5300-1-0050 > == Spawn extension (dialin, 065939191, 2) exited non-zero on > 'SIP/as5300-1-0050' > 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" ;tag=as4eb3efa7 > To: > Contact: > 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" ;tag=as4eb3efa7 > To: ;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: ;tag=36CA05C-167B > To: ;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: ;tag=36CA05C-167B > To: ;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 -- >> N
Re: [asterisk-users] Asterisk 1.6 + Cisco AS5300
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 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: ;tag=as12acaefb To: ;tag=36CA05C-167B Contact: 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: ;tag=as12acaefb To: ;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 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: ;tag=as12acaefb To: ;tag=36CA05C-167B Contact: 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/as5300-1-0050 == Spawn extension (dialin, 065939191, 2) exited non-zero on 'SIP/as5300-1-0050' 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" ;tag=as4eb3efa7 To: Contact: 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" ;tag=as4eb3efa7 To: ;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: ;tag=36CA05C-167B To: ;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: ;tag=36CA05C-167B To: ;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? 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Joi
Re: [asterisk-users] Asterisk 1.6 + Cisco AS5300
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 -- _ -- 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
[asterisk-users] Asterisk 1.6 + Cisco AS5300
Hello, I have a Cisco AS5300 connected to Asterisk (1.6.2.9) Between 15-16 minutes, the call is disconnected without reason. Here is what is displayed in the debug: Received an SDES from 10.4.0.10:17399 -- Got SIP response 420 "Bad Extension" back from 10.4.0.10 -- Stopped music on hold on SIP/as5300-1-004d == Spawn extension (dialin, 065939191, 2) exited non-zero on 'SIP/as5300-1-004d' Do you have an explanation? Best regards, Mickael -- _ -- 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
