SIP Debugging re-enabled bath-office-pbx1*CLI> <--- SIP read from 193.111.200.11:5082 ---> INVITE sip:101@195.112.25.179 SIP/2.0 To: From: "Keith Sudbury";tag=3875437f Via: SIP/2.0/UDP 193.111.200.11:5082;branch=z9hG4bK-d87543-35a2da6c6112924d72a3-1-cHBmMmYzNzY0YTZjZWM3NjcwOTk4Nw..-d87543- Call-ID: 83114487914ab0bf433a1e053dbbc91e CSeq: 98564564 INVITE Contact: ;flow-id=1 Max-Forwards: 69 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, MESSAGE, INFO Content-Type: application/sdp Supported: timer, replaces, callerid User-Agent: snom360/6.5.8 Allow-Events: talk, hold, refer Content-Length: 453 P-Key-Flags: resolution="31x13", keys="4" Session-Expires: 3600;refresher=uas Min-SE: 90 v=0 o=root 48522265 1728620696 IN IP4 193.111.200.11 s=call c=IN IP4 193.111.200.11 t=0 0 m=audio 10508 RTP/AVP 0 8 18 2 3 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:sybe9dg6cvzOc9nmhfJhkkro/s3XdP2Dxsnqfe1s a=encryption:optional a=fmtp:101 0-16 a=ptime:20 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:18 g729/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> --- (18 headers 18 lines) --- Sending to 193.111.200.11 : 5082 (no NAT) Using INVITE request as basis request - 83114487914ab0bf433a1e053dbbc91e Found no matching peer or user for '193.111.200.11:5082' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 2 Found RTP audio format 3 Found RTP audio format 4 Found RTP audio format 101 Peer audio RTP is at port 193.111.200.11:10508 Found description format pcmu for ID 0 Found description format pcma for ID 8 Found description format g729 for ID 18 Found description format g726-32 for ID 2 Found description format gsm for ID 3 Found description format g723 for ID 4 Found description format telephone-event for ID 101 Capabilities: us - 0x40e (gsm|ulaw|alaw|ilbc), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 193.111.200.11:10508 Looking for 101 in default (domain 195.112.25.179) <--- Reliably Transmitting (no NAT) to 193.111.200.11:5082 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 193.111.200.11:5082;branch=z9hG4bK-d87543-35a2da6c6112924d72a3-1-cHBmMmYzNzY0YTZjZWM3NjcwOTk4Nw..-d87543-;received=193.111.200.11 From: "Keith Sudbury";tag=3875437f To: ;tag=as4c82a484 Call-ID: 83114487914ab0bf433a1e053dbbc91e CSeq: 98564564 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> [Oct 29 16:24:01] NOTICE[9866]: chan_sip.c:13669 handle_request_invite: Call from '' to extension '101' rejected because extension not found. Scheduling destruction of SIP dialog '83114487914ab0bf433a1e053dbbc91e' in 32000 ms (Method: INVITE) bath-office-pbx1*CLI> <--- SIP read from 193.111.200.11:5082 ---> ACK sip:101@195.112.25.179 SIP/2.0 To: ;tag=as4c82a484 From: "Keith Sudbury";tag=3875437f Via: SIP/2.0/UDP 193.111.200.11:5082;branch=z9hG4bK-d87543-35a2da6c6112924d72a3-1-cHBmMmYzNzY0YTZjZWM3NjcwOTk4Nw..-d87543- Call-ID: 83114487914ab0bf433a1e053dbbc91e CSeq: 98564564 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '83114487914ab0bf433a1e053dbbc91e' Method: ACK