Destroying call '0970b6b72e5a98160ac252c2275e3571@195.38.96.5' We're at 195.38.96.5 port 12234 Answering/Requesting with root capability 0x8 (alaw) Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x400 (ilbc) 13 headers, 10 lines Reliably Transmitting (no NAT) to 192.168.50.171:5060: INVITE sip:15800100@192.168.50.171 SIP/2.0 Via: SIP/2.0/UDP 195.38.96.5:5060;branch=z9hG4bK70515c81 From: "THG520" ;tag=as2164c6af To: Contact: Call-ID: 409ce743313294156e4b14c746c6314b@195.38.96.5 CSeq: 102 INVITE User-Agent: Asterisk PBX Remote-Party-ID: "THG520" ;privacy=off;screen=yes Date: Tue, 27 Sep 2005 11:59:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 208 v=0 o=root 29714 29714 IN IP4 195.38.96.5 s=session c=IN IP4 195.38.96.5 t=0 0 m=audio 12234 RTP/AVP 8 0 97 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=silenceSupp:off - - - - --- pbx0*CLI> <-- SIP read from 192.168.50.171:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 195.38.96.5:5060;branch=z9hG4bK70515c81 From: "THG520" ;tag=as2164c6af To: Call-ID: 409ce743313294156e4b14c746c6314b@195.38.96.5 CSeq: 102 INVITE User-Agent: Grandstream BT110 1.0.7.11 Content-Length: 0 --- (8 headers 0 lines)--- pbx0*CLI> <-- SIP read from 192.168.50.171:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 195.38.96.5:5060;branch=z9hG4bK70515c81 From: "THG520" ;tag=as2164c6af To: ;tag=91560a3c928a2560 Call-ID: 409ce743313294156e4b14c746c6314b@195.38.96.5 CSeq: 102 INVITE User-Agent: Grandstream BT110 1.0.7.11 Content-Length: 0 --- (8 headers 0 lines)--- pbx0*CLI> <-- SIP read from 192.168.50.171:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 195.38.96.5:5060;branch=z9hG4bK70515c81 From: "THG520" ;tag=as2164c6af To: ;tag=91560a3c928a2560 Call-ID: 409ce743313294156e4b14c746c6314b@195.38.96.5 CSeq: 102 INVITE User-Agent: Grandstream BT110 1.0.7.11 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Supported: replaces Content-Length: 163 pbx0*CLI> v=0 o=15800100 8000 8000 IN IP4 192.168.50.171 s=SIP Call c=IN IP4 192.168.50.171 t=0 0 m=audio 5004 RTP/AVP 8 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 --- (12 headers 9 lines)--- Found RTP audio format 8 Peer audio RTP is at port 192.168.50.171:5004 Found description format PCMA Capabilities: us - 0x40c (ulaw|alaw|ilbc), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.50.171, port 5060 Transmitting (no NAT) to 192.168.50.171:5060: ACK sip:15800100@192.168.50.171 SIP/2.0 Via: SIP/2.0/UDP 195.38.96.5:5060;branch=z9hG4bK4ea92b8e From: "THG520" ;tag=as2164c6af To: ;tag=91560a3c928a2560 Contact: Call-ID: 409ce743313294156e4b14c746c6314b@195.38.96.5 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- pbx0*CLI> <-- SIP read from 192.168.50.171:5060: BYE sip:0615800115@195.38.96.5 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.171;branch=z9hG4bKb6c8d89ed06068d8 From: ;tag=91560a3c928a2560 To: "THG520" ;tag=as2164c6af Supported: replaces Call-ID: 409ce743313294156e4b14c746c6314b@195.38.96.5 CSeq: 17656 BYE User-Agent: Grandstream BT110 1.0.7.11 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 --- (11 headers 0 lines)--- Sending to 192.168.50.171 : 5060 (non-NAT) Transmitting (no NAT) to 192.168.50.171:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.50.171;branch=z9hG4bKb6c8d89ed06068d8;received=192.168.50.171 From: ;tag=91560a3c928a2560 To: "THG520" ;tag=as2164c6af Call-ID: 409ce743313294156e4b14c746c6314b@195.38.96.5 CSeq: 17656 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- Destroying call '409ce743313294156e4b14c746c6314b@195.38.96.5'