ip-68-178-167-75*CLI> <-- SIP read from 190.54.21.68:5060: INVITE sip:25804759@68.178.167.75 SIP/2.0 Via: SIP/2.0/UDP 190.54.21.68:5060;branch=z9hG4bK58a70004;rport From: "28551194" ;tag=as34c2d0a8 To: Contact: Call-ID: 10795fa35afac7375342f809302070a0@190.54.21.68 CSeq: 102 INVITE User-Agent: CallMe LatinIP.com Max-Forwards: 70 Date: Tue, 29 May 2007 22:39:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 239 v=0 o=root 27824 27824 IN IP4 190.54.21.68 s=session c=IN IP4 190.54.21.68 t=0 0 m=audio 16400 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (13 headers 11 lines)--- Using INVITE request as basis request - 10795fa35afac7375342f809302070a0@190.54.21.68 Sending to 190.54.21.68 : 5060 (NAT) Found peer '25804759' Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 190.54.21.68:16400 Found description format G729 Found description format telephone-event Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 25804759 in from-sip-external (domain 68.178.167.75) list_route: hop: Transmitting (no NAT) to 190.54.21.68:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 190.54.21.68:5060;branch=z9hG4bK58a70004;rport;received=190.54.21.68 From: "28551194" ;tag=as34c2d0a8 To: Call-ID: 10795fa35afac7375342f809302070a0@190.54.21.68 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing NoOp("SIP/25804759-7d17", "Received incoming SIP connection from unknown peer to 25804759") in new stack -- Executing Set("SIP/25804759-7d17", "DID=25804759") in new stack -- Executing Goto("SIP/25804759-7d17", "s|1") in new stack -- Goto (from-sip-external,s,1) -- Executing Ringing("SIP/25804759-7d17", "") in new stack Transmitting (no NAT) to 190.54.21.68:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 190.54.21.68:5060;branch=z9hG4bK58a70004;rport;received=190.54.21.68 From: "28551194" ;tag=as34c2d0a8 To: ;tag=as4b73edcd Call-ID: 10795fa35afac7375342f809302070a0@190.54.21.68 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing GotoIf("SIP/25804759-7d17", "0?from-trunk|25804759|1") in new stack -- Executing Set("SIP/25804759-7d17", "TIMEOUT(absolute)=15") in new stack -- Channel will hangup at 2007-05-29 22:42:22 UTC. -- Executing Answer("SIP/25804759-7d17", "") in new stack We're at 68.178.167.75 port 15646 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 190.54.21.68:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 190.54.21.68:5060;branch=z9hG4bK58a70004;rport;received=190.54.21.68 From: "28551194" ;tag=as34c2d0a8 To: ;tag=as4b73edcd Call-ID: 10795fa35afac7375342f809302070a0@190.54.21.68 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 241 v=0 o=root 26036 26036 IN IP4 68.178.167.75 s=session c=IN IP4 68.178.167.75 t=0 0 m=audio 15646 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Executing Wait("SIP/25804759-7d17", "2") in new stack ip-68-178-167-75*CLI> <-- SIP read from 190.54.21.68:5060: ACK sip:25804759@68.178.167.75 SIP/2.0 Via: SIP/2.0/UDP 190.54.21.68:5060;branch=z9hG4bK76fae3b6;rport From: "28551194" ;tag=as34c2d0a8 To: ;tag=as4b73edcd Contact: Call-ID: 10795fa35afac7375342f809302070a0@190.54.21.68 CSeq: 102 ACK User-Agent: CallMe LatinIP.com Max-Forwards: 70 Content-Length: 0 --- (10 headers 0 lines)--- -- Executing Playback("SIP/25804759-7d17", "ss-noservice") in new stack -- Playing 'ss-noservice' (language 'en') -- Executing PlayTones("SIP/25804759-7d17", "congestion") in new stack -- Executing Congestion("SIP/25804759-7d17", "5") in new stack We're at 68.178.167.75 port 15646 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Transmitting (no NAT) to 190.54.21.68:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 190.54.21.68:5060;branch=z9hG4bK58a70004;rport;received=190.54.21.68 From: "28551194" ;tag=as34c2d0a8 To: ;tag=as4b73edcd Call-ID: 10795fa35afac7375342f809302070a0@190.54.21.68 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 241 v=0 o=root 26036 26037 IN IP4 68.178.167.75 s=session c=IN IP4 68.178.167.75 t=0 0 m=audio 15646 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- ip-68-178-167-75*CLI>