Hello, i have a problem with the hangup cause received from the AMI in the Hangup events. All calls that arent answered after ringing are returning hangup cause 16 (normal clearing) instead 19.
Im running asterisk 1.4.11, the calls are generated to a SIP peer using the AMI originate command. This is the 'sip debug' output: Reliably Transmitting (no NAT) to 192.168.0.70:5060: INVITE sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK286113b7;rport From: "123" <sip:[EMAIL PROTECTED]>;tag=as0cd1aab0 To: <sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 27 Aug 2007 05:53:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 238 v=0 o=root 21676 21676 IN IP4 192.168.0.1 s=session c=IN IP4 192.168.0.1 t=0 0 m=audio 15274 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- gw*CLI> <--- SIP read from 192.168.0.70:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK286113b7;rport From: "123" <sip:[EMAIL PROTECTED]>;tag=as0cd1aab0 To: <sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp>;tag=2035093099 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Server: Cisco ATA 186 v3.2.1 atasip (050616A) Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Content-Length: 0 <-------------> --- (9 headers 0 lines) --- gw*CLI> <--- SIP read from 192.168.0.70:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK286113b7;rport From: "123" <sip:[EMAIL PROTECTED]>;tag=as0cd1aab0 To: <sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp>;tag=2035093099 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Contact: 1 <sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp> Server: Cisco ATA 186 v3.2.1 atasip (050616A) Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Reliably Transmitting (no NAT) to 192.168.0.70:5060: OPTIONS sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK49966ec7;rport From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as0916f4ed To: <sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 27 Aug 2007 05:53:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- gw*CLI> <--- SIP read from 192.168.0.70:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK49966ec7;rport From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as0916f4ed To: <sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp>;tag=3724167432 Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS Server: Cisco ATA 186 v3.2.1 atasip (050616A) Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces Content-Length: 250 Content-Type: application/sdp v=0 o=2 19680158 19680158 IN IP4 192.168.0.70 s=ATA186 Call c=IN IP4 192.168.0.70 t=0 0 m=audio 16386 RTP/AVP 0 8 4 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:4 G723/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (11 headers 11 lines) --- Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS Reliably Transmitting (no NAT) to 192.168.0.70:5060: OPTIONS sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK02be2790;rport From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as6ba5f9aa To: <sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 27 Aug 2007 05:53:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- gw*CLI> <--- SIP read from 192.168.0.70:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK02be2790;rport From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as6ba5f9aa To: <sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp>;tag=2035093099 Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS Server: Cisco ATA 186 v3.2.1 atasip (050616A) Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces Content-Length: 250 Content-Type: application/sdp v=0 o=1 19680166 19680166 IN IP4 192.168.0.70 s=ATA186 Call c=IN IP4 192.168.0.70 t=0 0 m=audio 16384 RTP/AVP 0 8 4 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:4 G723/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (11 headers 11 lines) --- Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 6400 ms (Method: INVITE) Reliably Transmitting (no NAT) to 192.168.0.70:5060: CANCEL sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK286113b7;rport From: "123" <sip:[EMAIL PROTECTED]>;tag=as0cd1aab0 To: <sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp> Call-ID: [EMAIL PROTECTED] CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 6400 ms (Method: INVITE) [Aug 27 02:53:44] NOTICE[21820]: pbx_spool.c:341 attempt_thread: Call failed to go through, reason (3) Remote end Ringing gw*CLI> <--- SIP read from 192.168.0.70:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK286113b7;rport From: "123" <sip:[EMAIL PROTECTED]>;tag=as0cd1aab0 To: <sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp>;tag=2035093099 Call-ID: [EMAIL PROTECTED] CSeq: 102 CANCEL Server: Cisco ATA 186 v3.2.1 atasip (050616A) Supported: replaces Content-Length: 0 <-------------> --- (9 headers 0 lines) --- gw*CLI> <--- SIP read from 192.168.0.70:5060 ---> SIP/2.0 487 Request Cancelled Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK286113b7;rport From: "123" <sip:[EMAIL PROTECTED]>;tag=as0cd1aab0 To: <sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp>;tag=2035093099 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Server: Cisco ATA 186 v3.2.1 atasip (050616A) Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Transmitting (no NAT) to 192.168.0.70:5060: ACK sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK286113b7;rport From: "123" <sip:[EMAIL PROTECTED]>;tag=as0cd1aab0 To: <sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp>;tag=2035093099 Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- These are the events received from the AMI: Event: Newchannel Privilege: call,all Timestamp: 1188194254.782040 Channel: SIP/1-081d3ba0 State: Down CallerIDNum: <unknown> CallerIDName: <unknown> Uniqueid: 1188194254.9 Event: Newcallerid Privilege: call,all Timestamp: 1188194254.782548 Channel: SIP/1-081d3ba0 CallerID: 123 CallerIDName: 123 Uniqueid: 1188194254.9 CID-CallingPres: 0 (Presentation Allowed, Not Screened) Event: Newcallerid Privilege: call,all Timestamp: 1188194254.782694 Channel: SIP/1-081d3ba0 CallerID: 123 CallerIDName: 123 Uniqueid: 1188194254.9 CID-CallingPres: 0 (Presentation Allowed, Not Screened) Event: Newstate Privilege: call,all Timestamp: 1188194254.811535 Channel: SIP/1-081d3ba0 State: Ringing CallerID: 123 CallerIDName: 123 Uniqueid: 1188194254.9 Event: Hangup Privilege: call,all Timestamp: 1188194264.781755 Channel: SIP/1-081d3ba0 Uniqueid: 1188194254.9 Cause: 16 Cause-txt: Normal Clearing Thanks in advance, Francisco. ____________________________________________________________________________________ ¡Sé un mejor ambientalista! 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