Here is more of a SIP debug log: As you can see Asterisk retries four times but I assume the softphone is not responding?
--- Really destroying SIP dialog '[email protected]'<mailto:'[email protected]'> Method: OPTIONS Reliably Transmitting (no NAT) to 172.31.254.53:9653: OPTIONS sip:[email protected]:9653 SIP/2.0 Via: SIP/2.0/UDP 172.31.253.4:5060;branch=z9hG4bK6953e396;rport From: "asterisk" <sip:[email protected]>;tag=as78f74756 To: <sip:[email protected]:9653> Contact: <sip:[email protected]> Call-ID: [email protected]<mailto:[email protected]> CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 29 Dec 2011 04:22:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Retransmitting #1 (no NAT) to 172.31.254.53:9653: OPTIONS sip:[email protected]:9653 SIP/2.0 Via: SIP/2.0/UDP 172.31.253.4:5060;branch=z9hG4bK6953e396;rport From: "asterisk" <sip:[email protected]>;tag=as78f74756 To: <sip:[email protected]:9653> Contact: <sip:[email protected]> Call-ID: [email protected]<mailto:[email protected]> CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 29 Dec 2011 04:22:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Retransmitting #2 (no NAT) to 172.31.254.53:9653: OPTIONS sip:[email protected]:9653 SIP/2.0 Via: SIP/2.0/UDP 172.31.253.4:5060;branch=z9hG4bK6953e396;rport From: "asterisk" <sip:[email protected]>;tag=as78f74756 To: <sip:[email protected]:9653> Contact: <sip:[email protected]> Call-ID: [email protected]<mailto:[email protected]> CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 29 Dec 2011 04:22:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- -- Remote UNIX connection -- Remote UNIX connection disconnected Retransmitting #3 (no NAT) to 172.31.254.53:9653: OPTIONS sip:[email protected]:9653 SIP/2.0 Via: SIP/2.0/UDP 172.31.253.4:5060;branch=z9hG4bK6953e396;rport From: "asterisk" <sip:[email protected]>;tag=as78f74756 To: <sip:[email protected]:9653> Contact: <sip:[email protected]> Call-ID: [email protected]<mailto:[email protected]> CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 29 Dec 2011 04:22:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Retransmitting #4 (no NAT) to 172.31.254.53:9653: OPTIONS sip:[email protected]:9653 SIP/2.0 Via: SIP/2.0/UDP 172.31.253.4:5060;branch=z9hG4bK6953e396;rport From: "asterisk" <sip:[email protected]>;tag=as78f74756 To: <sip:[email protected]:9653> Contact: <sip:[email protected]> Call-ID: [email protected]<mailto:[email protected]> CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 29 Dec 2011 04:22:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0
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