*CLI> <-- SIP read from 69.1.236.33:5060: INVITE sip:s@12.206.2.116 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 69.1.236.33;branch=z9hG4bKec6.39b501d4.1 Via: SIP/2.0/UDP 209.247.16.5:5060;branch=z9hG4bK50603522629637-1157004452658 From: "PCS Phone IL" ;tag=VPSF50603522629637 To: Call-ID: DALMGC0320060930202747006680@209.244.62.68 CSeq: 1 INVITE Contact: Max-Forwards: 68 Content-Type: application/sdp Content-Length: 115 Remote-Party-ID: "PCS Phone IL" ;party=calling;screen=yes;privacy=off v=0 o=- 1159648067 1159648068 IN IP4 63.215.29.33 s=- c=IN IP4 66.128.8.234 t=0 0 m=audio 52890 RTP/AVP 0 18 --- (13 headers 6 lines)--- Using INVITE request as basis request - DALMGC0320060930202747006680@209.244.62.68 Sending to 69.1.236.33 : 5060 (non-NAT) Found peer 'sipmedia' Found RTP audio format 0 Found RTP audio format 18 Peer audio RTP is at port 66.128.8.234:52890 Capabilities: us - 0x4 (ulaw), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Looking for s in sipurain (domain 12.206.2.116) list_route: hop: Transmitting (no NAT) to 69.1.236.33:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 69.1.236.33;branch=z9hG4bKec6.39b501d4.1;received=69.1.236.33 Via: SIP/2.0/UDP 209.247.16.5:5060;branch=z9hG4bK50603522629637-1157004452658 From: "PCS Phone IL" ;tag=VPSF50603522629637 To: Call-ID: DALMGC0320060930202747006680@209.244.62.68 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing SIPGetHeader("SIP/2179034857-081b3010", "TO=To") in new stack -- SIPGetHeader: set variable TO to -- Executing GotoIf("SIP/2179034857-081b3010", "1?5") in new stack -- Goto (sipurain,s,5) -- Executing GotoIf("SIP/2179034857-081b3010", "0?10:6") in new stack -- Goto (sipurain,s,6) -- Executing GotoIf("SIP/2179034857-081b3010", "0?10:7") in new stack -- Goto (sipurain,s,7) -- Executing Dial("SIP/2179034857-081b3010", "sip/sipura1|30|r") in new stack We're at 172.21.100.2 port 19026 Adding codec 0x4 (ulaw) to SDP 13 headers, 8 lines Reliably Transmitting (no NAT) to 172.21.100.10:5060: INVITE sip:sipura1@172.21.100.10:5060 SIP/2.0 Via: SIP/2.0/UDP 172.21.100.2:5060;branch=z9hG4bK6509c353;rport From: "PCS Phone IL" ;tag=as6da68343 To: Contact: Call-ID: 1ec2c8160c1fa492786bde2b255de2b9@172.21.100.2 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sat, 30 Sep 2006 20:27:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 160 v=0 o=root 23965 23965 IN IP4 172.21.100.2 s=session c=IN IP4 172.21.100.2 t=0 0 m=audio 19026 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - --- -- Called sipura1 Transmitting (no NAT) to 69.1.236.33:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 69.1.236.33;branch=z9hG4bKec6.39b501d4.1;received=69.1.236.33 Via: SIP/2.0/UDP 209.247.16.5:5060;branch=z9hG4bK50603522629637-1157004452658 From: "PCS Phone IL" ;tag=VPSF50603522629637 To: ;tag=as430782dc Call-ID: DALMGC0320060930202747006680@209.244.62.68 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- <-- SIP read from 172.21.100.10:5060: SIP/2.0 100 Trying To: From: "PCS Phone IL" ;tag=as6da68343 Call-ID: 1ec2c8160c1fa492786bde2b255de2b9@172.21.100.2 CSeq: 102 INVITE Via: SIP/2.0/UDP 172.21.100.2:5060;branch=z9hG4bK6509c353 Server: Sipura/SPA2000-2.0.13(g) Content-Length: 0 --- (8 headers 0 lines)--- <-- SIP read from 172.21.100.10:5060: SIP/2.0 180 Ringing To: ;tag=28532ecf9c6858dci0 From: "PCS Phone IL" ;tag=as6da68343 Call-ID: 1ec2c8160c1fa492786bde2b255de2b9@172.21.100.2 CSeq: 102 INVITE Via: SIP/2.0/UDP 172.21.100.2:5060;branch=z9hG4bK6509c353 Server: Sipura/SPA2000-2.0.13(g) Content-Length: 0 --- (8 headers 0 lines)--- -- SIP/sipura1-081b8550 is ringing Destroying call '68ca75e52b6e5efa1364484e5bda5474@172.21.100.2'