<--- SIP read from 195.113.224.138:5066 ---> INVITE sip:3063@195.113.224.138 SIP/2.0 Via: SIP/2.0/UDP 195.113.224.138:5066;rport;branch=z9hG4bK9U5j6X8BrKQ0S Max-Forwards: 70 From: OptimTalk ;tag=yQFFHyetr1tjD To: Call-ID: 3242c20b-8a73-122c-5c8c@195.113.224.138 CSeq: 112356585 INVITE Contact: User-Agent: OptimTalk / 1.7 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, INFO, REGISTER Supported: timer, 100rel Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 213 v=0 o=- 562026619064333415 5593743237115887572 IN IP4 195.113.224.138 s=- c=IN IP4 195.113.224.138 t=0 0 m=audio 16800 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (15 headers 9 lines) --- Sending to 195.113.224.138 : 5066 (NAT) Using INVITE request as basis request - 3242c20b-8a73-122c-5c8c@195.113.224.138 Found peer 'optimtalk' -- Real user that calls like is trusty -- Peer's context was changed from to . Context should have been Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 195.113.224.138:16800 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event), combined - 0x0 (nothing) Peer audio RTP is at port 195.113.224.138:16800 Looking for 3063 in optimtalkCtx (domain 195.113.224.138) list_route: hop: <--- Transmitting (no NAT) to 195.113.224.138:5066 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 195.113.224.138:5066;branch=z9hG4bK9U5j6X8BrKQ0S;received=195.113.224.138;rport=5066 From: OptimTalk ;tag=yQFFHyetr1tjD To: Call-ID: 3242c20b-8a73-122c-5c8c@195.113.224.138 CSeq: 112356585 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> asterisk*CLI> <--- Transmitting (no NAT) to 195.113.224.138:5066 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 195.113.224.138:5066;branch=z9hG4bK9U5j6X8BrKQ0S;received=195.113.224.138;rport=5066 From: OptimTalk ;tag=yQFFHyetr1tjD To: ;tag=as183c2ea0 Call-ID: 3242c20b-8a73-122c-5c8c@195.113.224.138 CSeq: 112356585 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Audio is at 195.113.224.138 port 12826 Adding codec 0x4 (ulaw) to SDP <--- Reliably Transmitting (no NAT) to 195.113.224.138:5066 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 195.113.224.138:5066;branch=z9hG4bK9U5j6X8BrKQ0S;received=195.113.224.138;rport=5066 From: OptimTalk ;tag=yQFFHyetr1tjD To: ;tag=as183c2ea0 Call-ID: 3242c20b-8a73-122c-5c8c@195.113.224.138 CSeq: 112356585 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 190 v=0 o=root 21836 21836 IN IP4 195.113.224.138 s=session c=IN IP4 195.113.224.138 t=0 0 m=audio 12826 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> asterisk*CLI> <--- SIP read from 195.113.224.138:5066 ---> ACK sip:3063@195.113.224.138 SIP/2.0 Via: SIP/2.0/UDP 195.113.224.138:5066;rport;branch=z9hG4bKa5yB8rSFNvDKN Max-Forwards: 70 From: OptimTalk ;tag=yQFFHyetr1tjD To: ;tag=as183c2ea0 Call-ID: 3242c20b-8a73-122c-5c8c@195.113.224.138 CSeq: 112356585 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 195.113.224.138:5066 ---> INVITE sip:3082@195.113.224.138 SIP/2.0 Via: SIP/2.0/UDP 195.113.224.138:5066;rport;branch=z9hG4bKBer49KaKj535g Max-Forwards: 70 From: 3063 ;tag=Z087jSZXNaH5r To: Call-ID: 33548c8a-8a73-122c-5c8c@195.113.224.138 CSeq: 112356585 INVITE Contact: User-Agent: OptimTalk / 1.7 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, INFO, REGISTER Supported: timer, 100rel Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 214 v=0 o=- 4080987691406513291 6100440639209746259 IN IP4 195.113.224.138 s=- c=IN IP4 195.113.224.138 t=0 0 m=audio 16802 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (15 headers 9 lines) --- Sending to 195.113.224.138 : 5066 (NAT) Using INVITE request as basis request - 33548c8a-8a73-122c-5c8c@195.113.224.138 -- Real user that calls like <3063> is trusty -- User's context was changed from to . Context should have been Found user '3063' Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 195.113.224.138:16802 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/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 195.113.224.138:16802 Looking for 3082 in optimtalkCtx (domain 195.113.224.138) list_route: hop: <--- Transmitting (no NAT) to 195.113.224.138:5066 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 195.113.224.138:5066;branch=z9hG4bKBer49KaKj535g;received=195.113.224.138;rport=5066 From: 3063 ;tag=Z087jSZXNaH5r To: Call-ID: 33548c8a-8a73-122c-5c8c@195.113.224.138 CSeq: 112356585 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> asterisk*CLI> <--- Transmitting (no NAT) to 195.113.224.138:5066 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 195.113.224.138:5066;branch=z9hG4bKBer49KaKj535g;received=195.113.224.138;rport=5066 From: 3063 ;tag=Z087jSZXNaH5r To: ;tag=as3dbdf26d Call-ID: 33548c8a-8a73-122c-5c8c@195.113.224.138 CSeq: 112356585 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Audio is at 195.113.224.138 port 11286 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 195.113.224.138:5066 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 195.113.224.138:5066;branch=z9hG4bKBer49KaKj535g;received=195.113.224.138;rport=5066 From: 3063 ;tag=Z087jSZXNaH5r To: ;tag=as3dbdf26d Call-ID: 33548c8a-8a73-122c-5c8c@195.113.224.138 CSeq: 112356585 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 246 v=0 o=root 21836 21836 IN IP4 195.113.224.138 s=session c=IN IP4 195.113.224.138 t=0 0 m=audio 11286 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 <------------> asterisk*CLI> <--- SIP read from 195.113.224.138:5066 ---> ACK sip:3082@195.113.224.138 SIP/2.0 Via: SIP/2.0/UDP 195.113.224.138:5066;rport;branch=z9hG4bKcQHXBFUpFetrc Max-Forwards: 70 From: 3063 ;tag=Z087jSZXNaH5r To: ;tag=as3dbdf26d Call-ID: 33548c8a-8a73-122c-5c8c@195.113.224.138 CSeq: 112356585 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- [Mar 13 14:10:48] WARNING[24590]: res_musiconhold.c:660 get_mohbyname: Music on Hold class 'default' not found [Mar 13 14:10:48] WARNING[24590]: res_musiconhold.c:660 get_mohbyname: Music on Hold class 'default' not found [Mar 13 14:10:59] WARNING[24595]: res_musiconhold.c:660 get_mohbyname: Music on Hold class 'default' not found [Mar 13 14:10:59] WARNING[24595]: res_musiconhold.c:660 get_mohbyname: Music on Hold class 'default' not found Scheduling destruction of SIP dialog '3242c20b-8a73-122c-5c8c@195.113.224.138' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 195.113.224.138, port 5066 Reliably Transmitting (no NAT) to 195.113.224.138:5066: BYE sip:195.113.224.138:5066 SIP/2.0 Via: SIP/2.0/UDP 195.113.224.138:5060;branch=z9hG4bK226d2ec5;rport From: ;tag=as183c2ea0 To: OptimTalk ;tag=yQFFHyetr1tjD Call-ID: 3242c20b-8a73-122c-5c8c@195.113.224.138 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- asterisk*CLI> <--- SIP read from 195.113.224.138:5066 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 195.113.224.138:5060;branch=z9hG4bK226d2ec5;rport=5060 From: ;tag=as183c2ea0 To: OptimTalk ;tag=yQFFHyetr1tjD Call-ID: 3242c20b-8a73-122c-5c8c@195.113.224.138 CSeq: 102 BYE User-Agent: OptimTalk / 1.7 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, INFO, REGISTER Supported: timer, 100rel Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '3242c20b-8a73-122c-5c8c@195.113.224.138' Method: ACK <--- SIP read from 195.113.224.138:5066 ---> BYE sip:3082@195.113.224.138 SIP/2.0 Via: SIP/2.0/UDP 195.113.224.138:5066;rport;branch=z9hG4bKD0apDactcQgBr Max-Forwards: 70 From: 3063 ;tag=Z087jSZXNaH5r To: ;tag=as3dbdf26d Call-ID: 33548c8a-8a73-122c-5c8c@195.113.224.138 CSeq: 112356586 BYE User-Agent: OptimTalk / 1.7 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, INFO, REGISTER Supported: timer, 100rel Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 195.113.224.138 : 5066 (NAT) <--- Transmitting (NAT) to 195.113.224.138:5066 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 195.113.224.138:5066;branch=z9hG4bKD0apDactcQgBr;received=195.113.224.138;rport=5066 From: 3063 ;tag=Z087jSZXNaH5r To: ;tag=as3dbdf26d Call-ID: 33548c8a-8a73-122c-5c8c@195.113.224.138 CSeq: 112356586 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Really destroying SIP dialog '33548c8a-8a73-122c-5c8c@195.113.224.138' Method: BYE