a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- voip*CLI> <--- SIP read from 10.0.0.1:5060 ---> ACK sip:23003@10.0.0.2 SIP/2.0 From: ORAS;tag=a000001-13c4-1632d-56b8808-6fa2 To: ;tag=as7085b194 Call-ID: 97949dbc-a000001-13c4-1632d-56b8808-384@10.0.0.1 CSeq: 1 ACK Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-16335-56ba6ea-53f0 Max-Forwards: 70 Contact: Content-Length:0 <-------------> --- (9 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.0.1, port 5060 Audio is at 10.0.0.2 port 11728 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: INVITE sip:748881658@10.0.0.1:5060;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK195f8746;rport From: ;tag=as7085b194 To: ORAS;tag=a000001-13c4-1632d-56b8808-6fa2 Contact: Call-ID: 97949dbc-a000001-13c4-1632d-56b8808-384@10.0.0.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 306 v=0 o=root 24756 24757 IN IP4 89.35.79.69 s=session c=IN IP4 89.35.79.69 t=0 0 m=audio 18638 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 10.0.0.1:5060 ---> BYE sip:23003@10.0.0.2;TCID-28 SIP/2.0 From: ORAS;tag=a000001-13c4-1632d-56b8808-6fa2 To: ;tag=as7085b194 Call-ID: 97949dbc-a000001-13c4-1632d-56b8808-384@10.0.0.1 CSeq: 2 BYE Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-16335-56ba6f2-63ee Max-Forwards: 70 Content-Length:0 <-------------> --- (8 headers 0 lines) --- Sending to 10.0.0.1 : 5060 (no NAT) voip*CLI> <--- 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-16335-56ba6f2-63ee;received=10.0.0.1 From: ORAS;tag=a000001-13c4-1632d-56b8808-6fa2 To: ;tag=as7085b194 Call-ID: 97949dbc-a000001-13c4-1632d-56b8808-384@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 '5cef322324cadbc0372e39e17409242f@89.35.79.5' in 32000 ms (Method: INVITE) == Spawn extension (default, 23003, 1) exited non-zero on 'SIP/5060-0822ee68' voip*CLI> <--- SIP read from 10.0.0.1:5060 ---> SIP/2.0 400 Bad Request From: ;tag=as7085b194 To: ORAS;tag=a000001-13c4-1632d-56b8808-6fa2 Call-ID: 97949dbc-a000001-13c4-1632d-56b8808-384@10.0.0.1 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.0.0.2:5060;received=10.0.0.2;branch=z9hG4bK195f8746;rport Content-Length:0 <-------------> --- (7 headers 0 lines) --- -- Got SIP response 400 "Bad Request" back from 10.0.0.1 set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.0.1, port 5060 Transmitting (no NAT) to 10.0.0.1:5060: ACK sip:748881658@10.0.0.1:5060;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK195f8746;rport From: ORAS;tag=a000001-13c4-1632d-56b8808-6fa2 To: ;tag=as7085b194 Contact: Call-ID: 97949dbc-a000001-13c4-1632d-56b8808-384@10.0.0.1 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Really destroying SIP dialog '97949dbc-a000001-13c4-1632d-56b8808-384@10.0.0.1' Method: BYE 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=z9hG4bK0714407e;rport=5060 Contact: To: ;tag=8d028934 From: "ORAS";tag=as79772655 Call-ID: 5cef322324cadbc0372e39e17409242f@89.35.79.5 CSeq: 103 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: 178 v=0 o=- 7 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 18638 RTP/AVP 8 97 a=fmtp:97 0-15 a=rtpmap:97 telephone-event/8000 a=sendrecv <-------------> --- (11 headers 9 lines) --- Found RTP audio format 8 Found RTP audio format 97 Peer audio RTP is at port 89.35.79.69:18638 Found description format telephone-event for ID 97 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 89.35.79.69:18638 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=z9hG4bK717e4fe7;rport From: "ORAS" ;tag=as79772655 To: ;tag=8d028934 Contact: Call-ID: 5cef322324cadbc0372e39e17409242f@89.35.79.5 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- 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=z9hG4bK3dd48e88;rport From: "ORAS" ;tag=as79772655 To: ;tag=8d028934 Call-ID: 5cef322324cadbc0372e39e17409242f@89.35.79.5 CSeq: 104 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '5cef322324cadbc0372e39e17409242f@89.35.79.5' in 32000 ms (Method: INVITE) 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=z9hG4bK3dd48e88;rport=5060 Contact: To: ;tag=8d028934 From: "ORAS";tag=as79772655 Call-ID: 5cef322324cadbc0372e39e17409242f@89.35.79.5 CSeq: 104 BYE User-Agent: X-Lite release 1006e stamp 34025 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '5cef322324cadbc0372e39e17409242f@89.35.79.5' Method: INVITE voip*CLI> <--- SIP read from 89.35.79.69:16088 ---> <-------------> --- (0 headers 1 lines) --- Scheduling destruction of SIP dialog '323002043e33719a3a7fed292556809b@89.35.79.5' in 32000 ms (Method: NOTIFY) Reliably Transmitting (no NAT) to 89.35.79.247:5060: NOTIFY sip:23001@89.35.79.247:5060 SIP/2.0 Via: SIP/2.0/UDP 89.35.79.5:5060;branch=z9hG4bK3ea55859;rport From: "asterisk" ;tag=as78ea5c15 To: Contact: Call-ID: 323002043e33719a3a7fed292556809b@89.35.79.5 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 91 Messages-Waiting: yes Message-Account: sip:asterisk@89.35.79.5 Voice-Message: 1/0 (0/0) --- voip*CLI> <--- SIP read from 89.35.79.247:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 89.35.79.5:5060;branch=z9hG4bK3ea55859;rport Call-ID: 323002043e33719a3a7fed292556809b@89.35.79.5 CSeq: 102 NOTIFY From: "asterisk" ;tag=as78ea5c15 To: ;tag=uq6IKSs1wDgqoZpt Contact: Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '323002043e33719a3a7fed292556809b@89.35.79.5' Method: NOTIFY voip*CLI>