Summary: | ASTERISK-00128: Asterisk doesn't respond to 407 for BYE | ||
Reporter: | linxu01 (linxu01) | Labels: | |
Date Opened: | 2003-08-19 16:39:37 | Date Closed: | 2004-09-25 02:40:14 |
Priority: | Major | Regression? | No |
Status: | Closed/Complete | Components: | Core/General |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ||
Description: | When an incoming call from a sip proxy to an extension on *, if * hangs up first and * will send BYE to the proxy. The proxy will reply with a 407 for BYE, but * doesn't know how to handle it. Here is the register entry in sip.conf: register=9988:123456@sip.mydomain.net/1207 Here is the sip debug message: (1204 on the sip proxy calling 9988, which routed to 1207 on the *. When 1207 hangs up first, * closes the sip session between * and 1207 first, then sending a BYE to proxy. proxy return a 407, causing * not handling. Finally 1204 hangs up.) INVITE sip:1207@10.1.3.17 SIP/2.0 Via: SIP/2.0/UDP 10.1.2.15:5060;branch=1_gZ2M2HjLZ7L5HjfdWDl10Q99 Via: SIP/2.0/UDP 10.1.1.1:5060 From: "1204" <sip:1204@sip.mydomain.net>;tag=5250070017d68010a86c32-6fe0d44d To: sip:9988@sip.mydomain.net Call-ID: 00075052-80d62845-28b3aed2-71155263@10.1.1.1 Date:Tue, 19 Aug 2003 20:14:34 GMT User-Agent:Cisco-SIP-IP-Phone/2 CSeq: 102 INVITE Expires: 1500 Contact:sip:1204@10.1.1.1:5060 Content-Type:application/sdp Content-Length: 169 Record-Route: <sip:9988@sip.mydomain.net:5060;maddr=10.1.2.15> v=0 o=CiscoSystemsSIP-IPPhone-UserAgent 4482 20697 IN IP4 10.1.1.1 s=SIP Call c=IN IP4 10.1.1.1 t=0 0 m=audio 23660 RTP/AVP 0 8 18 a=rtpmap:0 pcmu/8000 14 headers, 7 lines Using latest request as basis request Sending to 10.1.2.15 : 5060 (non-NAT) Found audio format UNKN Found audio format ALAW Found audio format UNKN Found description format pcmu Capabilities: us - 524300, them - 268/0, combined - 12 Non-codec capabilities: us - 1, them - 0, combined - 0 DEBUG[147466]: File chan_sip.c, Line 4628 (handle_request): Check for res DEBUG[147466]: File chan_sip.c, Line 906 (find_user): is not a local user Looking for 1207 in sip-incoming DEBUG[147466]: File chan_sip.c, Line 3131 (build_route): build_route: Record-Route hop: <sip:9988@sip.mydomain.net:5060;maddr=10.1.2.15> DEBUG[147466]: File chan_sip.c, Line 3156 (build_route): build_route: Contact hop: sip:1204@10.1.1.1:5060 list_route: hop: <sip:9988@sip.mydomain.net:5060;maddr=10.1.2.15> list_route: hop: <sip:1204@10.1.1.1:5060> Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.2.15:5060;branch=1_gZ2M2HjLZ7L5HjfdWDl10Q99 Via: SIP/2.0/UDP 10.1.1.1:5060 From: "1204" <sip:1204@sip.mydomain.net>;tag=5250070017d68010a86c32-6fe0d44d To: sip:9988@sip.mydomain.net;tag=as4c6fc279 Call-ID: 00075052-80d62845-28b3aed2-71155263@10.1.1.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:1207@10.1.3.17> Content-Length: 0 to 10.1.2.15:5060 -- Executing Dial("SIP/-081135f0", "SIP/1207|10") in new stack DEBUG[311314]: File chan_sip.c, Line 622 (create_addr): Setting NAT on RTP to -1 We're at 10.1.3.17 port 19358 Answering with preferred capability 4 Answering with capability 8 Answering with non-codec capability 1 11 headers, 10 lines Reliably Transmitting: INVITE sip:1207@172.17.252.27 SIP/2.0 Via: SIP/2.0/UDP 10.1.3.17:5060;branch=z9hG4bK1310c593 From: "1204" <sip:1204@10.1.3.17>;tag=as0c7a9c14 To: <sip:1207@172.17.252.27> Contact: <sip:1204@10.1.3.17> Call-ID: 38028e7d6d49f8d75a12940a76aeec04@10.1.3.17 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 211 v=0 o=root 2368 2368 IN IP4 10.1.3.17 s=session c=IN IP4 10.1.3.17 t=0 0 m=audio 19358 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 (NAT) to 172.17.252.27:5060 -- Called 1207 Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.3.17:5060;branch=z9hG4bK1310c593 From: "1204" <sip:1204@10.1.3.17>;tag=as0c7a9c14 To: <sip:1207@172.17.252.27>;tag=468530005584d5594c7654-272b913f Call-ID: 38028e7d6d49f8d75a12940a76aeec04@10.1.3.17 Server: Cisco-SIP-IP-Phone/2 Date: Tue, 19 Aug 2003 20:14:34 GMT CSeq: 102 INVITE Content-Length: 0 9 headers, 0 lines DEBUG[147466]: File chan_sip.c, Line 515 (__sip_ack): Acked pending invite 102 DEBUG[147466]: File chan_sip.c, Line 533 (__sip_ack): Stopping retransmission on '38028e7d6d49f8d75a12940a76aeec04@10.1.3.17' of Request 102: Found Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.1.3.17:5060;branch=z9hG4bK1310c593 From: "1204" <sip:1204@10.1.3.17>;tag=as0c7a9c14 To: <sip:1207@172.17.252.27>;tag=468530005584d5594c7654-272b913f Call-ID: 38028e7d6d49f8d75a12940a76aeec04@10.1.3.17 Server: Cisco-SIP-IP-Phone/2 Date: Tue, 19 Aug 2003 20:14:34 GMT CSeq: 102 INVITE Content-Length: 0 9 headers, 0 lines DEBUG[147466]: File chan_sip.c, Line 533 (__sip_ack): Stopping retransmission on '38028e7d6d49f8d75a12940a76aeec04@10.1.3.17' of Request 102: Not Found -- SIP/1207-1ca8 is ringing Transmitting (no NAT): SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.1.2.15:5060;branch=1_gZ2M2HjLZ7L5HjfdWDl10Q99 Via: SIP/2.0/UDP 10.1.1.1:5060 From: "1204" <sip:1204@sip.mydomain.net>;tag=5250070017d68010a86c32-6fe0d44d To: sip:9988@sip.mydomain.net;tag=as4c6fc279 Call-ID: 00075052-80d62845-28b3aed2-71155263@10.1.1.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:1207@10.1.3.17> Content-Length: 0 to 10.1.2.15:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.3.17:5060;branch=z9hG4bK1310c593 From: "1204" <sip:1204@10.1.3.17>;tag=as0c7a9c14 To: <sip:1207@172.17.252.27>;tag=468530005584d5594c7654-272b913f Call-ID: 38028e7d6d49f8d75a12940a76aeec04@10.1.3.17 Server: Cisco-SIP-IP-Phone/2 Contact: sip:1207@172.17.252.27:5060 Date: Tue, 19 Aug 2003 20:14:36 GMT CSeq: 102 INVITE Content-Type: application/sdp Content-Length: 219 v=0 o=CiscoSystemsSIP-IPPhone-UserAgent 7907 8427 IN IP4 172.17.252.27 s=SIP Call c=IN IP4 172.17.252.27 t=0 0 m=audio 28606 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 11 headers, 9 lines DEBUG[147466]: File chan_sip.c, Line 533 (__sip_ack): Stopping retransmission on '38028e7d6d49f8d75a12940a76aeec04@10.1.3.17' of Request 102: Not Found Found audio format UNKN Found audio format UNKN Found description format pcmu Found description format telephone-event Capabilities: us - 524300, them - 4/0, combined - 4 Non-codec capabilities: us - 1, them - 1, combined - 1 DEBUG[147466]: File chan_sip.c, Line 3156 (build_route): build_route: Contact hop: sip:1207@172.17.252.27:5060 list_route: hop: <sip:1207@172.17.252.27:5060> set_destination: Parsing <sip:1207@172.17.252.27:5060> for address/port to send to set_destination: set destination to 172.17.252.27, port 5060 Transmitting: ACK sip:1207@172.17.252.27 SIP/2.0 Via: SIP/2.0/UDP 10.1.3.17:5060;branch=z9hG4bK1310c593 From: "1204" <sip:1204@10.1.3.17>;tag=as0c7a9c14 To: <sip:1207@172.17.252.27>;tag=468530005584d5594c7654-272b913f Contact: <sip:1204@10.1.3.17> Call-ID: 38028e7d6d49f8d75a12940a76aeec04@10.1.3.17 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 172.17.252.27:5060 -- SIP/1207-1ca8 answered SIP/-081135f0 We're at 10.1.3.17 port 19770 Answering with preferred capability 4 Answering with capability 8 Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.2.15:5060;branch=1_gZ2M2HjLZ7L5HjfdWDl10Q99 Via: SIP/2.0/UDP 10.1.1.1:5060 Record-Route: <sip:9988@sip.mydomain.net:5060;maddr=10.1.2.15> From: "1204" <sip:1204@sip.mydomain.net>;tag=5250070017d68010a86c32-6fe0d44d To: sip:9988@sip.mydomain.net;tag=as4c6fc279 Call-ID: 00075052-80d62845-28b3aed2-71155263@10.1.1.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:1207@10.1.3.17> Content-Type: application/sdp Content-Length: 155 v=0 o=root 2368 2368 IN IP4 10.1.3.17 s=session c=IN IP4 10.1.3.17 t=0 0 m=audio 19770 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 to 10.1.2.15:5060 -- Attempting native bridge of SIP/-081135f0 and SIP/1207-1ca8 Sip read: ACK sip:1207@10.1.3.17:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.2.15:5060;branch=1_gZ2M2HjLZ7L5HjfdWDl10Q99 Via: SIP/2.0/UDP 10.1.1.1:5060 From: "1204" <sip:1204@sip.mydomain.net>;tag=5250070017d68010a86c32-6fe0d44d To: sip:9988@sip.mydomain.net;tag=as4c6fc279 Call-ID: 00075052-80d62845-28b3aed2-71155263@10.1.1.1 Date:Tue, 19 Aug 2003 20:14:35 GMT CSeq: 102 ACK Content-Length: 0 9 headers, 0 lines DEBUG[147466]: File chan_sip.c, Line 533 (__sip_ack): Stopping retransmission on '00075052-80d62845-28b3aed2-71155263@10.1.1.1' of Response 102: Found DEBUG[311314]: File rtp.c, Line 1007 (ast_rtp_write): Ooh, format changed from UNKN to ULAW DEBUG[311314]: File rtp.c, Line 1007 (ast_rtp_write): Ooh, format changed from UNKN to ULAW Sip read: BYE sip:1204@10.1.3.17:5060 SIP/2.0 Via: SIP/2.0/UDP 172.17.252.27:5060 From: <sip:1207@172.17.252.27>;tag=468530005584d5594c7654-272b913f To: "1204" <sip:1204@10.1.3.17>;tag=as0c7a9c14 Call-ID: 38028e7d6d49f8d75a12940a76aeec04@10.1.3.17 Date: Tue, 19 Aug 2003 20:14:39 GMT User-Agent: Cisco-SIP-IP-Phone/2 CSeq: 101 BYE Content-Length: 0 9 headers, 0 lines Sending to 172.17.252.27 : 5060 (NAT) Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.252.27:5060;received=172.17.252.27 From: <sip:1207@172.17.252.27>;tag=468530005584d5594c7654-272b913f To: "1204" <sip:1204@10.1.3.17>;tag=as0c7a9c14 Call-ID: 38028e7d6d49f8d75a12940a76aeec04@10.1.3.17 CSeq: 101 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:1204@10.1.3.17> Content-Length: 0 to 172.17.252.27:5060 DEBUG[311314]: File channel.c, Line 2176 (ast_channel_bridge): Didn't get a frame from channel: SIP/1207-1ca8 DEBUG[311314]: File channel.c, Line 2244 (ast_channel_bridge): Bridge stops bridging channels SIP/-081135f0 and SIP/1207-1ca8 DEBUG[311314]: File chan_sip.c, Line 953 (sip_hangup): find_user(1207) == Spawn extension (sip-incoming, 1207, 1) exited non-zero on 'SIP/-081135f0' DEBUG[311314]: File chan_sip.c, Line 953 (sip_hangup): find_user() DEBUG[311314]: File chan_sip.c, Line 906 (find_user): is not a local user set_destination: Parsing <sip:9988@sip.mydomain.net:5060;maddr=10.1.2.15> for address/port to send to set_destination: set destination to 10.1.2.15, port 5060 Reliably Transmitting: BYE sip:1204@sip.mydomain.net SIP/2.0 Via: SIP/2.0/UDP 10.1.3.17:5060;branch=z9hG4bK65f4060b Route: <sip:1204@10.1.1.1:5060> From: sip:9988@sip.mydomain.net;tag=as4c6fc279 To: "1204" <sip:1204@sip.mydomain.net>;tag=5250070017d68010a86c32-6fe0d44d Contact: <sip:1207@10.1.3.17> Call-ID: 00075052-80d62845-28b3aed2-71155263@10.1.1.1 CSeq: 102 BYE User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 10.1.2.15:5060 Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.3.17:5060;branch=z9hG4bK65f4060b From: sip:9988@sip.mydomain.net;tag=as4c6fc279 To: "1204" <sip:1204@sip.mydomain.net>;tag=5250070017d68010a86c32-6fe0d44d Call-ID: 00075052-80d62845-28b3aed2-71155263@10.1.1.1 CSeq: 102 BYE Content-Length: 0 7 headers, 0 lines DEBUG[147466]: File chan_sip.c, Line 533 (__sip_ack): Stopping retransmission on '00075052-80d62845-28b3aed2-71155263@10.1.1.1' of Request 102: Found Message is BYE DEBUG[147466]: File chan_sip.c, Line 835 (__sip_destroy): Destorying call '38028e7d6d49f8d75a12940a76aeec04@10.1.3.17' DEBUG[147466]: File chan_sip.c, Line 835 (__sip_destroy): Destorying call '00075052-80d62845-28b3aed2-71155263@10.1.1.1' Sip read: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.1.3.17:5060;branch=z9hG4bK65f4060b From: sip:9988@sip.mydomain.net;tag=as4c6fc279 To: "1204" <sip:1204@sip.mydomain.net>;tag=5250070017d68010a86c32-6fe0d44d Call-ID: 00075052-80d62845-28b3aed2-71155263@10.1.1.1 CSeq: 102 BYE Content-Length: 0 Proxy-Authenticate: Digest realm="mydomain.net",domain="sip:1204@sip.mydomain.net",nonce="qPcMY3W+jHMdEkzj/LXmyg==",algorithm=MD5 8 headers, 0 lines DEBUG[147466]: File chan_sip.c, Line 4550 (handle_request): That's odd... Got a response on a call we dont know about. DEBUG[147466]: File chan_sip.c, Line 835 (__sip_destroy): Destorying call '00075052-80d62845-28b3aed2-71155263@10.1.1.1' Sip read: BYE sip:1207@10.1.3.17:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.2.15:5060;branch=1_ffS0vy16062B1IuupR8p5w99 Via: SIP/2.0/UDP 10.1.1.1:5060 From: "1204" <sip:1204@sip.mydomain.net>;tag=5250070017d68010a86c32-6fe0d44d To: sip:9988@sip.mydomain.net;tag=as4c6fc279 Call-ID: 00075052-80d62845-28b3aed2-71155263@10.1.1.1 Date:Tue, 19 Aug 2003 20:14:42 GMT User-Agent:Cisco-SIP-IP-Phone/2 CSeq: 103 BYE Content-Length: 0 10 headers, 0 lines Sending to 10.1.2.15 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.2.15:5060;branch=1_ffS0vy16062B1IuupR8p5w99 Via: SIP/2.0/UDP 10.1.1.1:5060 From: "1204" <sip:1204@sip.mydomain.net>;tag=5250070017d68010a86c32-6fe0d44d To: sip:9988@sip.mydomain.net;tag=as4c6fc279 Call-ID: 00075052-80d62845-28b3aed2-71155263@10.1.1.1 CSeq: 103 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 10.1.2.15:5060 | ||
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