<--- SIP read from 81.201.82.39:5060 ---> INVITE sip:155469877445@Asterisk_IP SIP/2.0 Call-ID: 85a1ec6b7dc998cd3b985869ef87f806@81.201.82.39 CSeq: 102 INVITE From: "anonymous" ;tag=70665 To: Via: SIP/2.0/UDP 81.201.82.39:5060;branch=z9hG4bKd865aecb62fe5bc6e374f67b61eabce7 Max-Forwards: 69 Content-Type: application/sdp Contact: User-Agent: Vox Callcontrol Content-Length: 311 v=0 o=root 16790 16790 IN IP4 81.201.82.23 s=session c=IN IP4 81.201.82.23 t=0 0 m=audio 11564 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (11 headers 15 lines) --- Sending to 81.201.82.39 : 5060 (no NAT) Using INVITE request as basis request - 85a1ec6b7dc998cd3b985869ef87f806@81.201.82.39 Found peer 'sip_proxy1' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 81.201.82.23:11564 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 81.201.82.23:11564 Looking for 155469877445 in stations (domain Asterisk_IP) list_route: hop: localhost*CLI> <--- Transmitting (no NAT) to 81.201.82.39:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 81.201.82.39:5060;branch=z9hG4bKd865aecb62fe5bc6e374f67b61eabce7;received=81.201.82.39 From: "anonymous" ;tag=70665 To: Call-ID: 85a1ec6b7dc998cd3b985869ef87f806@81.201.82.39 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 < <--- Transmitting (no NAT) to 81.201.82.39:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 81.201.82.39:5060;branch=z9hG4bKd865aecb62fe5bc6e374f67b61eabce7;received=81.201.82.39 From: "anonymous" ;tag=70665 To: ;tag=as78e4c405 Call-ID: 85a1ec6b7dc998cd3b985869ef87f806@81.201.82.39 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 238 v=0 o=root 637 637 IN IP4 Asterisk_IP s=session c=IN IP4 Asterisk_IP t=0 0 m=audio 17750 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv localhost*CLI> <--- SIP read from 81.201.82.39:5060 ---> CANCEL sip:155469877445@Asterisk_IP SIP/2.0 Call-ID: 85a1ec6b7dc998cd3b985869ef87f806@81.201.82.39 CSeq: 102 CANCEL From: "anonymous" ;tag=70665 To: Via: SIP/2.0/UDP 81.201.82.39:5060;branch=z9hG4bKd865aecb62fe5bc6e374f67b61eabce7 Max-Forwards: 69 User-Agent: Vox Callcontrol Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Sending to 81.201.82.39 : 5060 (no NAT) <--- Reliably Transmitting (no NAT) to 81.201.82.39:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 81.201.82.39:5060;branch=z9hG4bKd865aecb62fe5bc6e374f67b61eabce7;received=81.201.82.39 From: "anonymous" ;tag=70665 To: ;tag=as78e4c405 Call-ID: 85a1ec6b7dc998cd3b985869ef87f806@81.201.82.39 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> <--- Transmitting (no NAT) to 81.201.82.39:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 81.201.82.39:5060;branch=z9hG4bKd865aecb62fe5bc6e374f67b61eabce7;received=81.201.82.39 From: "anonymous" ;tag=70665 To: ;tag=as78e4c405 Call-ID: 85a1ec6b7dc998cd3b985869ef87f806@81.201.82.39 CSeq: 102 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 localhost*CLI> <--- SIP read from 81.201.82.39:5060 ---> ACK sip:155469877445@Asterisk_IP SIP/2.0 Call-ID: 85a1ec6b7dc998cd3b985869ef87f806@81.201.82.39 CSeq: 102 ACK From: "anonymous" ;tag=70665 To: ;tag=as78e4c405 Via: SIP/2.0/UDP 81.201.82.39:5060;branch=z9hG4bKd865aecb62fe5bc6e374f67b61eabce7 Max-Forwards: 69 User-Agent: Vox Callcontrol Content-Length: 0