--- -- Called 23003 voip*CLI> <--- SIP read from 89.35.79.69:16088 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 89.35.79.5:5060;branch=z9hG4bK201b6b8c;rport=5060 Contact: To: ;tag=b47ae525 From: "ORAS";tag=as5b05e2af Call-ID: 1a0bbfa340b8ead330884abf6a4044fe@89.35.79.5 CSeq: 102 INVITE User-Agent: X-Lite release 1006e stamp 34025 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- -- SIP/23003-08228400 is ringing <--- Transmitting (no NAT) to 10.0.0.1:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-15da6-555ee58-48bb;received=10.0.0.1 From: ORAS;tag=a000001-13c4-15da6-555ee58-14a To: ;tag=as03e3b4ce Call-ID: 97949c8c-a000001-13c4-15da6-555ee58-42ec@10.0.0.1 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Audio is at 10.0.0.2 port 14776 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (no NAT) to 10.0.0.1:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-15da6-555ee58-48bb;received=10.0.0.1 From: ORAS;tag=a000001-13c4-15da6-555ee58-14a To: ;tag=as03e3b4ce Call-ID: 97949c8c-a000001-13c4-15da6-555ee58-42ec@10.0.0.1 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 300 v=0 o=root 24756 24756 IN IP4 10.0.0.2 s=session c=IN IP4 10.0.0.2 t=0 0 m=audio 14776 RTP/AVP 8 0 18 97 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> voip*CLI> <--- SIP read from 89.35.79.69:16088 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 89.35.79.5:5060;branch=z9hG4bK201b6b8c;rport=5060 Contact: To: ;tag=b47ae525 From: "ORAS";tag=as5b05e2af Call-ID: 1a0bbfa340b8ead330884abf6a4044fe@89.35.79.5 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp ser-Agent: X-Lite release 1006e stamp 34025 Content-Length: 181 v=0 o=- 9 2 IN IP4 89.35.79.69 s=CounterPath X-Lite 3.0 c=IN IP4 89.35.79.69 t=0 0 m=audio 16992 RTP/AVP 8 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> --- (11 headers 9 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 89.35.79.69:16992 Found description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 89.35.79.69:16992 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 89.35.79.69, port 16088 Transmitting (no NAT) to 89.35.79.69:16088: ACK sip:23003@89.35.79.69:16088;rinstance=7ed7b782ad482eab SIP/2.0 Via: SIP/2.0/UDP 89.35.79.5:5060;branch=z9hG4bK47a7f252;rport From: "ORAS" ;tag=as5b05e2af To: ;tag=b47ae525 Contact: Call-ID: 1a0bbfa340b8ead330884abf6a4044fe@89.35.79.5 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- Call on SIP/23003-08228400 left from hold -- SIP/23003-08228400 answered SIP/5060-08232388 Audio is at 10.0.0.2 port 14776 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-15da6-555ee58-48bb;received=10.0.0.1 From: ORAS;tag=a000001-13c4-15da6-555ee58-14a To: ;tag=as03e3b4ce Call-ID: 97949c8c-a000001-13c4-15da6-555ee58-42ec@10.0.0.1 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 300 v=0 o=root 24756 24757 IN IP4 10.0.0.2 s=session c=IN IP4 10.0.0.2 t=0 0 m=audio 14776 RTP/AVP 8 0 18 97 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Packet2Packet bridging SIP/5060-08232388 and SIP/23003-08228400 voip*CLI> <--- SIP read from 10.0.0.1:5060 ---> ACK sip:23003@10.0.0.2 SIP/2.0 From: ORAS;tag=a000001-13c4-15da6-555ee58-14a To: ;tag=as03e3b4ce Call-ID: 97949c8c-a000001-13c4-15da6-555ee58-42ec@10.0.0.1 CSeq: 1 ACK Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-15dae-5560ed0-21d8 Max-Forwards: 70 Contact: Content-Length:0 <-------------> --- (9 headers 0 lines) --- voip*CLI> <--- SIP read from 10.0.0.1:5060 ---> BYE sip:23003@10.0.0.2;TCID-27 SIP/2.0 From: ORAS;tag=a000001-13c4-15da6-555ee58-14a To: ;tag=as03e3b4ce Call-ID: 97949c8c-a000001-13c4-15da6-555ee58-42ec@10.0.0.1 CSeq: 2 BYE Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-15dae-5560ed6-6816 Max-Forwards: 70 Content-Length:0 <-------------> --- (8 headers 0 lines) --- Sending to 10.0.0.1 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-15dae-5560ed6-6816;received=10.0.0.1 From: ORAS;tag=a000001-13c4-15da6-555ee58-14a To: ;tag=as03e3b4ce Call-ID: 97949c8c-a000001-13c4-15da6-555ee58-42ec@10.0.0.1 CSeq: 2 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Scheduling destruction of SIP dialog '1a0bbfa340b8ead330884abf6a4044fe@89.35.79.5' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 89.35.79.69, port 16088 Reliably Transmitting (no NAT) to 89.35.79.69:16088: BYE sip:23003@89.35.79.69:16088;rinstance=7ed7b782ad482eab SIP/2.0 Via: SIP/2.0/UDP 89.35.79.5:5060;branch=z9hG4bK2bb66b0b;rport From: "ORAS" ;tag=as5b05e2af To: ;tag=b47ae525 Call-ID: 1a0bbfa340b8ead330884abf6a4044fe@89.35.79.5 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- == Spawn extension (default, 23003, 1) exited non-zero on 'SIP/5060-08232388' voip*CLI> <--- SIP read from 89.35.79.69:16088 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 89.35.79.5:5060;branch=z9hG4bK2bb66b0b;rport=5060 Contact: To: ;tag=b47ae525 From: "ORAS";tag=as5b05e2af Call-ID: 1a0bbfa340b8ead330884abf6a4044fe@89.35.79.5 CSeq: 103 BYE User-Agent: X-Lite release 1006e stamp 34025 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '1a0bbfa340b8ead330884abf6a4044fe@89.35.79.5' Method: INVITE Really destroying SIP dialog '97949c8c-a000001-13c4-15da6-555ee58-42ec@10.0.0.1' Method: BYE Really destroying SIP dialog 'XfLu8uaWvRwQUWiM@89.35.79.247' Method: REGISTER voip*CLI> <--- SIP read from 89.35.79.69:16088 ---> <-------------> --- (0 headers 1 lines) --- voip*CLI>