<-- SIP read from 85.95.72.181:5060: INVITE sip:4912379@24.14.143.74 SIP/2.0 CSeq: 1 INVITE Via: SIP/2.0/UDP 85.95.72.181:5060 From: ;tag=1204540719451887921109 Call-ID: 1887921109@85.95.72.181 To: Contact: Content-Type: application/sdp Content-Length: 213 v=0 o=VoipSwitch 9522 9522 IN IP4 85.95.72.181 s=VoipSIP i=Audio Session c=IN IP4 85.95.72.181 t=0 0 m=audio 8522 RTP/AVP 18 101 a=rtpmap:18 G729/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --- (9 headers 10 lines) --- Using INVITE request as basis request - 1887921109@85.95.72.181 Sending to 85.95.72.181 : 5060 (NAT) Found peer 'Voicelink' Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 85.95.72.181:8522 Found description format G729 Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Transmitting (NAT) to 85.95.72.181:5060: SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 85.95.72.181:5060;received=85.95.72.181 From: ;tag=1204540719451887921109 To: ;tag=as715b02ef Call-ID: 1887921109@85.95.72.181 CSeq: 1 INVITE User-Agent: KillerPhone Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Destroying call '1887921109@85.95.72.181' rt-home*CLI> <-- SIP read from 85.95.72.181:5060: ACK sip:4912379@24.14.143.74 SIP/2.0 CSeq: 1 ACK Via: SIP/2.0/UDP 85.95.72.181:5060 From: ;tag=1204540719451887921109 Call-ID: 1887921109@85.95.72.181 To: ;tag=as715b02ef Contact: Content-Length: 0