Reliably Transmitting (no NAT) to 192.168.0.254:5060: INVITE sip:16562226071845@192.168.0.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK31d5c8c5;rport From: "1656222" ;tag=as7369da55 To: Contact: Call-ID: 6816d1ce4fa1f89624fe4f6d3a805830@192.168.0.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 28 Jun 2006 15:37:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 212 v=0 o=root 3131 3131 IN IP4 192.168.0.1 s=session c=IN IP4 192.168.0.1 t=0 0 m=audio 19030 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #1 (no NAT) to 192.168.0.254:5060: INVITE sip:16562226071845@192.168.0.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK31d5c8c5;rport From: "1656222" ;tag=as7369da55 To: Contact: Call-ID: 6816d1ce4fa1f89624fe4f6d3a805830@192.168.0.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 28 Jun 2006 15:37:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 212 v=0 o=root 3131 3131 IN IP4 192.168.0.1 s=session c=IN IP4 192.168.0.1 t=0 0 m=audio 19030 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- asterisk1*CLI> <-- SIP read from 192.168.0.254:5060: SIP/2.0 100 Trying Call-ID: 6816d1ce4fa1f89624fe4f6d3a805830@192.168.0.1 CSeq: 102 INVITE From: "1656222";tag=as7369da55 To: ;tag=c0a800fe-ebb User-Agent: Quintum/1.0.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK31d5c8c5;rport Quintum: 0b023335 --- (8 headers 0 lines)--- asterisk1*CLI> <-- SIP read from 192.168.0.254:5060: SIP/2.0 100 Trying Call-ID: 6816d1ce4fa1f89624fe4f6d3a805830@192.168.0.1 CSeq: 102 INVITE From: "1656222";tag=as7369da55 To: ;tag=c0a800fe-ebb User-Agent: Quintum/1.0.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK31d5c8c5;rport Quintum: 0b023335 --- (8 headers 0 lines)--- asterisk1*CLI> <-- SIP read from 192.168.0.254:5060: SIP/2.0 183 Session Progress Call-ID: 6816d1ce4fa1f89624fe4f6d3a805830@192.168.0.1 Content-Length: 160 Content-Type: application/sdp CSeq: 102 INVITE From: "1656222";tag=as7369da55 To: ;tag=c0a800fe-ebb User-Agent: Quintum/1.0.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK31d5c8c5;rport Quintum: 070e00000003008f6506001e03808081 v=0 o=Quintum 4 3131 IN IP4 192.168.0.254 s=VoipCall c=IN IP4 192.168.0.254 t=0 0 m=audio 10240 RTP/AVP 0 c=IN IP4 192.168.0.254 a=rtpmap:0 pcmu/8000/1 --- (10 headers 8 lines)--- Found RTP audio format 0 Peer audio RTP is at port 192.168.0.254:10240 Found description format pcmu Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) asterisk1*CLI> <-- SIP read from 192.168.0.254:5060: SIP/2.0 180 Ringing Call-ID: 6816d1ce4fa1f89624fe4f6d3a805830@192.168.0.1 Content-Length: 160 Content-Type: application/sdp CSeq: 102 INVITE From: "1656222";tag=as7369da55 To: ;tag=c0a800fe-ebb User-Agent: Quintum/1.0.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK31d5c8c5;rport v=0 o=Quintum 5 3131 IN IP4 192.168.0.254 s=VoipCall c=IN IP4 192.168.0.254 t=0 0 m=audio 10240 RTP/AVP 0 c=IN IP4 192.168.0.254 a=rtpmap:0 pcmu/8000/1 --- (9 headers 8 lines)--- Found RTP audio format 0 Peer audio RTP is at port 192.168.0.254:10240 Found description format pcmu Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Reliably Transmitting (no NAT) to 192.168.0.254:5060: CANCEL sip:16562226071845@192.168.0.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK31d5c8c5;rport From: "1656222" ;tag=as7369da55 To: Contact: Call-ID: 6816d1ce4fa1f89624fe4f6d3a805830@192.168.0.1 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of call '6816d1ce4fa1f89624fe4f6d3a805830@192.168.0.1' in 32000 ms asterisk1*CLI> <-- SIP read from 192.168.0.254:5060: SIP/2.0 200 OK Call-ID: 6816d1ce4fa1f89624fe4f6d3a805830@192.168.0.1 CSeq: 102 CANCEL From: "1656222";tag=as7369da55 To: Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK31d5c8c5;rport --- (6 headers 0 lines)--- 12 headers, 0 lines Reliably Transmitting (no NAT) to 192.168.0.254:5060: OPTIONS sip:192.168.0.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK7ce6aaf6;rport From: "Unknown" ;tag=as32b17854 To: Contact: Call-ID: 0725474745878aad152aaf22033d9997@192.168.0.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 28 Jun 2006 15:38:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 192.168.0.254:5060: SIP/2.0 200 OK Call-ID: 0725474745878aad152aaf22033d9997@192.168.0.1 CSeq: 102 OPTIONS From: "Unknown";tag=as32b17854 To: Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK7ce6aaf6;rport