<-- SIP read from 83.136.32.179:54942: INVITE sip:371@mediaserver2.at43.at SIP/2.0 Via: SIP/2.0/UDP 83.136.32.80:54941;branch=z9hG4bK.72bcdcc0;rport Route: Max-Forwards: 16 Contact: To: From: "Klaus Darilion enum eyebea";tag=f1614469 Call-ID: 03614d34b41caf24@a2xhdXNwYw.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: sipsak Content-Length: 210 v=0 o=- 23084192 23084377 IN IP4 10.10.0.50 s=eyeBeam c=IN IP4 83.136.32.83 t=0 0 m=audio 35062 RTP/AVP 0 8 3 98 101 a=fmtp:101 0-15 a=rtpmap:98 ilbc/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv --- (13 headers 10 lines)--- Using INVITE request as basis request - 03614d34b41caf24@a2xhdXNwYw.. Sending to 83.136.32.80 : 54941 (non-NAT) Found no matching peer or user for '83.136.32.179:54942' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 98 Found RTP audio format 101 Peer audio RTP is at port 83.136.32.83:35062 Found description format ilbc Found description format telephone-event Capabilities: us - 0x8060e (gsm|ulaw|alaw|speex|ilbc|h263), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=0x0 (nothing), combined - 0x40e (gsm|ulaw|alaw|ilbc) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 371 in sipanon list_route: hop: Transmitting (no NAT) to 83.136.32.80:54941: SIP/2.0 100 Trying Via: SIP/2.0/UDP 83.136.32.80:54941;branch=z9hG4bK.72bcdcc0 From: "Klaus Darilion enum eyebea";tag=f1614469 To: Call-ID: 03614d34b41caf24@a2xhdXNwYw.. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0