<--- SIP read from 84.8.191.13:5060 ---> INVITE sip:*7708452418360@84.8.129.188 SIP/2.0 Max-Forwards: 69 Session-Expires: 1800;refresher=uac Min-SE: 600 Supported: timer, 100rel To: From: "07976946209" ;tag=3463916609-626192 Remote-Party-Id: "07976946209" ;party=calling;screen=no;privacy=off Call-ID: 92483-3463916609-626185@aosbc1.alwaysongroup.com CSeq: 1 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Via: SIP/2.0/UDP 84.8.191.13:5060;branch=z9hG4bKcd4f579080fdc5ff8b2c68cd2e797670 Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 150 v=0 o=aosbc1 1615085 1615085 IN IP4 84.8.191.13 s=sip call c=IN IP4 84.8.191.14 t=0 0 m=audio 13566 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=sendrecv <-------------> --- (16 headers 8 lines) --- Sending to 84.8.191.13 : 5060 (no NAT) Using INVITE request as basis request - 92483-3463916609-626185@aosbc1.alwaysongroup.com Found peer 'nextpoint-sbc' Found RTP audio format 0 Peer audio RTP is at port 84.8.191.14:13566 Found audio description format PCMU for ID 0 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 84.8.191.14:13566 Looking for *7708452418360 in default (domain 84.8.129.188) list_route: hop: aovastest01*CLI> <--- Transmitting (no NAT) to 84.8.191.13:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 84.8.191.13:5060;branch=z9hG4bKcd4f579080fdc5ff8b2c68cd2e797670;received=84.8.191.13 From: "07976946209" ;tag=3463916609-626192 To: Call-ID: 92483-3463916609-626185@aosbc1.alwaysongroup.com CSeq: 1 INVITE User-Agent: alwaysON Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Audio is at 84.8.129.188 port 17646 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 195.219.133.219:5065: INVITE sip:08452418360@195.219.133.219:5065 SIP/2.0 Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK07e4ffc5;rport From: "07976946209" ;tag=as7e2d6104 To: Contact: Call-ID: 3ff001e64ecb77313ff3c10432647719@84.8.129.188 CSeq: 102 INVITE User-Agent: alwaysON Max-Forwards: 70 Remote-Party-ID: "07976946209" ;privacy=off;screen=no Date: Wed, 07 Oct 2009 14:50:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 264 v=0 o=root 16668 16668 IN IP4 84.8.129.188 s=session c=IN IP4 84.8.129.188 t=0 0 m=audio 17646 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:10 a=sendrecv --- aovastest01*CLI> <--- SIP read from 195.219.133.219:5065 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK07e4ffc5;rport CSeq: 102 INVITE Call-ID: 3ff001e64ecb77313ff3c10432647719@84.8.129.188 From: "07976946209" ;tag=as7e2d6104 To: Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from 195.219.133.219:5065 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK07e4ffc5;rport CSeq: 102 INVITE Call-ID: 3ff001e64ecb77313ff3c10432647719@84.8.129.188 From: "07976946209" ;tag=as7e2d6104 To: ;tag=92a4f9e8-1dd2-11b2-bd8c-b03162323164+92a4f9e8 Contact: Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- Transmitting (no NAT) to 84.8.191.13:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 84.8.191.13:5060;branch=z9hG4bKcd4f579080fdc5ff8b2c68cd2e797670;received=84.8.191.13 From: "07976946209" ;tag=3463916609-626192 To: ;tag=as6880918b Call-ID: 92483-3463916609-626185@aosbc1.alwaysongroup.com CSeq: 1 INVITE User-Agent: alwaysON Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> aovastest01*CLI> <--- SIP read from 195.219.133.219:5065 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK07e4ffc5;rport CSeq: 102 INVITE Call-ID: 3ff001e64ecb77313ff3c10432647719@84.8.129.188 From: "07976946209" ;tag=as7e2d6104 To: ;tag=92a4f9e8-1dd2-11b2-bd8c-b03162323164+92a4f9e8 Contact: Content-Type: application/sdp Allow: INVITE,BYE,ACK,CANCEL,PRACK,REFER,OPTIONS,REGISTER,NOTIFY Content-Length: 161 v=0 o=- 797361286 797361286 IN IP4 195.219.133.219 s=Polycom IP Phone c=IN IP4 84.8.129.140 t=0 0 m=audio 2222 RTP/AVP 0 a=sendrecv a=rtpmap:0 PCMU/8000 <-------------> --- (10 headers 8 lines) --- Found RTP audio format 0 Peer audio RTP is at port 84.8.129.140:2222 Found audio description format PCMU for ID 0 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 84.8.129.140:2222 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 195.219.133.219, port 5065 Transmitting (no NAT) to 195.219.133.219:5065: ACK sip:08452418360@195.219.133.219:5065;transport=udp SIP/2.0 Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK7b54ce41;rport From: "07976946209" ;tag=as7e2d6104 To: ;tag=92a4f9e8-1dd2-11b2-bd8c-b03162323164+92a4f9e8 Contact: Call-ID: 3ff001e64ecb77313ff3c10432647719@84.8.129.188 CSeq: 102 ACK User-Agent: alwaysON Max-Forwards: 70 Remote-Party-ID: "07976946209" ;privacy=off;screen=no Content-Length: 0 --- Audio is at 84.8.129.188 port 12702 Adding codec 0x4 (ulaw) to SDP <--- Reliably Transmitting (no NAT) to 84.8.191.13:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 84.8.191.13:5060;branch=z9hG4bKcd4f579080fdc5ff8b2c68cd2e797670;received=84.8.191.13 From: "07976946209" ;tag=3463916609-626192 To: ;tag=as6880918b Call-ID: 92483-3463916609-626185@aosbc1.alwaysongroup.com CSeq: 1 INVITE User-Agent: alwaysON Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 184 v=0 o=root 16668 16668 IN IP4 84.8.129.188 s=session c=IN IP4 84.8.129.188 t=0 0 m=audio 12702 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:10 a=sendrecv <------------> set_destination: Parsing for address/port to send to set_destination: set destination to 195.219.133.219, port 5065 Audio is at 84.8.129.188 port 17646 Adding codec 0x4 (ulaw) to SDP Reliably Transmitting (no NAT) to 195.219.133.219:5065: INVITE sip:08452418360@195.219.133.219:5065;transport=udp SIP/2.0 Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK7db684b4;rport From: "07976946209" ;tag=as7e2d6104 To: ;tag=92a4f9e8-1dd2-11b2-bd8c-b03162323164+92a4f9e8 Contact: Call-ID: 3ff001e64ecb77313ff3c10432647719@84.8.129.188 CSeq: 103 INVITE User-Agent: alwaysON Max-Forwards: 70 Remote-Party-ID: "07976946209" ;privacy=off;screen=no 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: 182 v=0 o=root 16668 16669 IN IP4 84.8.191.14 s=session c=IN IP4 84.8.191.14 t=0 0 m=audio 13566 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:10 a=sendrecv --- aovastest01*CLI> <--- SIP read from 195.219.133.219:5065 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK7db684b4;rport CSeq: 103 INVITE Call-ID: 3ff001e64ecb77313ff3c10432647719@84.8.129.188 From: "07976946209" ;tag=as7e2d6104 To: ;tag=92a4f9e8-1dd2-11b2-bd8c-b03162323164+92a4f9e8 Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from 84.8.191.13:5060 ---> ACK sip:*7708452418360@84.8.129.188 SIP/2.0 Max-Forwards: 69 To: ;tag=as6880918b From: "07976946209" ;tag=3463916609-626192 Call-ID: 92483-3463916609-626185@aosbc1.alwaysongroup.com CSeq: 1 ACK Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Via: SIP/2.0/UDP 84.8.191.13:5060;branch=z9hG4bKab9c9ab779f59133bc2d3347e73e84df Contact: Content-Length: 0 <-------------> --- (10 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 84.8.191.13, port 5060 Audio is at 84.8.129.188 port 12702 Adding codec 0x4 (ulaw) to SDP Reliably Transmitting (no NAT) to 84.8.191.13:5060: INVITE sip:07976946209@84.8.191.13:5060 SIP/2.0 Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK74156918;rport From: ;tag=as6880918b To: "07976946209" ;tag=3463916609-626192 Contact: Call-ID: 92483-3463916609-626185@aosbc1.alwaysongroup.com CSeq: 102 INVITE User-Agent: alwaysON 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: 183 v=0 o=root 16668 16669 IN IP4 84.8.129.140 s=session c=IN IP4 84.8.129.140 t=0 0 m=audio 2222 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:10 a=sendrecv --- aovastest01*CLI> <--- SIP read from 84.8.191.13:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK74156918;rport From: ;tag=as6880918b To: "07976946209" ;tag=3463916609-626192 Call-ID: 92483-3463916609-626185@aosbc1.alwaysongroup.com CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from 195.219.133.219:5065 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK7db684b4;rport CSeq: 103 INVITE Call-ID: 3ff001e64ecb77313ff3c10432647719@84.8.129.188 From: "07976946209" ;tag=as7e2d6104 To: ;tag=92a4f9e8-1dd2-11b2-bd8c-b03162323164+92a4f9e8 Contact: Content-Type: application/sdp Allow: INVITE,BYE,ACK,CANCEL,PRACK,REFER,OPTIONS,REGISTER,NOTIFY Content-Length: 161 v=0 o=- 797361286 797361286 IN IP4 195.219.133.219 s=Polycom IP Phone c=IN IP4 84.8.129.140 t=0 0 m=audio 2222 RTP/AVP 0 a=sendrecv a=rtpmap:0 PCMU/8000 <-------------> --- (10 headers 8 lines) --- Found RTP audio format 0 Peer audio RTP is at port 84.8.129.140:2222 Found audio description format PCMU for ID 0 Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 84.8.129.140:2222 set_destination: Parsing for address/port to send to set_destination: set destination to 195.219.133.219, port 5065 Transmitting (no NAT) to 195.219.133.219:5065: ACK sip:08452418360@195.219.133.219:5065;transport=udp SIP/2.0 Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK7ef1ce25;rport From: "07976946209" ;tag=as7e2d6104 To: ;tag=92a4f9e8-1dd2-11b2-bd8c-b03162323164+92a4f9e8 Contact: Call-ID: 3ff001e64ecb77313ff3c10432647719@84.8.129.188 CSeq: 103 ACK User-Agent: alwaysON Max-Forwards: 70 Remote-Party-ID: "07976946209" ;privacy=off;screen=no Content-Length: 0 --- aovastest01*CLI> <--- SIP read from 84.8.191.13:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK74156918;rport To: "07976946209" ;tag=3463916609-626192 From: ;tag=as6880918b Call-ID: 92483-3463916609-626185@aosbc1.alwaysongroup.com CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 138 v=0 o=aosbc1 0 1 IN IP4 84.8.191.13 s=sip call c=IN IP4 84.8.191.14 t=0 0 m=audio 13566 RTP/AVP 0 a=sendrecv a=rtpmap:0 PCMU/8000 <-------------> --- (11 headers 8 lines) --- Found RTP audio format 0 Peer audio RTP is at port 84.8.191.14:13566 Found audio description format PCMU for ID 0 Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 84.8.191.14:13566 set_destination: Parsing for address/port to send to set_destination: set destination to 84.8.191.13, port 5060 Transmitting (no NAT) to 84.8.191.13:5060: ACK sip:07976946209@84.8.191.13:5060 SIP/2.0 Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK1d4fed28;rport From: ;tag=as6880918b To: "07976946209" ;tag=3463916609-626192 Contact: Call-ID: 92483-3463916609-626185@aosbc1.alwaysongroup.com CSeq: 102 ACK User-Agent: alwaysON Max-Forwards: 70 Content-Length: 0 --- aovastest01*CLI> <--- SIP read from 195.219.133.219:5065 ---> INVITE sip:07976946209@84.8.129.188 SIP/2.0 From: ;tag=92a4f9e8-1dd2-11b2-bd8c-b03162323164+92a4f9e8 To: "07976946209" ;tag=as7e2d6104 Call-ID: 3ff001e64ecb77313ff3c10432647719@84.8.129.188 CSeq: 1 INVITE Via: SIP/2.0/UDP 195.219.133.219:5065;branch=z9hG4bK341783651385733 Contact: Max-Forwards: 70 Content-Type: application/sdp Remote-Party-ID: "Mansoor,Jehanzeb" ;party=calling;id-type=subscriber;privacy=off P-Asserted-Identity: "Mansoor,Jehanzeb" Content-Length: 142 v=0 o=- 797361286 797361287 IN IP4 195.219.133.219 s=SIP Call c=IN IP4 195.219.151.7 t=0 0 m=audio 1128 RTP/AVP 0 a=rtpmap:0 pcmu/8000 <-------------> --- (12 headers 7 lines) --- Sending to 195.219.133.219 : 5065 (no NAT) Found RTP audio format 0 Peer audio RTP is at port 195.219.151.7:1128 Found audio description format pcmu for ID 0 Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 195.219.151.7:1128 <--- Transmitting (no NAT) to 195.219.133.219:5065 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 195.219.133.219:5065;branch=z9hG4bK341783651385733;received=195.219.133.219 From: ;tag=92a4f9e8-1dd2-11b2-bd8c-b03162323164+92a4f9e8 To: "07976946209" ;tag=as7e2d6104 Call-ID: 3ff001e64ecb77313ff3c10432647719@84.8.129.188 CSeq: 1 INVITE User-Agent: alwaysON Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Audio is at 84.8.129.188 port 17646 Adding codec 0x4 (ulaw) to SDP <--- Reliably Transmitting (no NAT) to 195.219.133.219:5065 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 195.219.133.219:5065;branch=z9hG4bK341783651385733;received=195.219.133.219 From: ;tag=92a4f9e8-1dd2-11b2-bd8c-b03162323164+92a4f9e8 To: "07976946209" ;tag=as7e2d6104 Call-ID: 3ff001e64ecb77313ff3c10432647719@84.8.129.188 CSeq: 1 INVITE User-Agent: alwaysON Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 182 v=0 o=root 16668 16670 IN IP4 84.8.191.14 s=session c=IN IP4 84.8.191.14 t=0 0 m=audio 13566 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:10 a=sendrecv <------------> aovastest01*CLI> <--- SIP read from 195.219.133.219:5065 ---> ACK sip:07976946209@84.8.129.188 SIP/2.0 From: ;tag=92a4f9e8-1dd2-11b2-bd8c-b03162323164+92a4f9e8 To: "07976946209" ;tag=as7e2d6104 Call-ID: 3ff001e64ecb77313ff3c10432647719@84.8.129.188 CSeq: 1 ACK Via: SIP/2.0/UDP 195.219.133.219:5065;branch=z9hG4bK1737219700415489 Max-Forwards: 70 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 195.219.133.219:5065 ---> INVITE sip:07976946209@84.8.129.188 SIP/2.0 From: ;tag=92a4f9e8-1dd2-11b2-bd8c-b03162323164+92a4f9e8 To: "07976946209" ;tag=as7e2d6104 Call-ID: 3ff001e64ecb77313ff3c10432647719@84.8.129.188 CSeq: 2 INVITE Via: SIP/2.0/UDP 195.219.133.219:5065;branch=z9hG4bK725197768102465 Contact: Max-Forwards: 70 Content-Type: application/sdp Remote-Party-ID: "NA" ;party=calling;id-type=subscriber;privacy=off P-Asserted-Identity: "08452418338" Content-Length: 161 v=0 o=- 797361286 797361288 IN IP4 195.219.133.219 s=Polycom IP Phone c=IN IP4 84.8.129.143 t=0 0 m=audio 2250 RTP/AVP 0 a=sendrecv a=rtpmap:0 PCMU/8000 <-------------> --- (12 headers 8 lines) --- Sending to 195.219.133.219 : 5065 (no NAT) Found RTP audio format 0 Peer audio RTP is at port 84.8.129.143:2250 Found audio description format PCMU for ID 0 Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 84.8.129.143:2250 <--- Transmitting (no NAT) to 195.219.133.219:5065 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 195.219.133.219:5065;branch=z9hG4bK725197768102465;received=195.219.133.219 From: ;tag=92a4f9e8-1dd2-11b2-bd8c-b03162323164+92a4f9e8 To: "07976946209" ;tag=as7e2d6104 Call-ID: 3ff001e64ecb77313ff3c10432647719@84.8.129.188 CSeq: 2 INVITE User-Agent: alwaysON Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Audio is at 84.8.129.188 port 17646 Adding codec 0x4 (ulaw) to SDP <--- Reliably Transmitting (no NAT) to 195.219.133.219:5065 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 195.219.133.219:5065;branch=z9hG4bK725197768102465;received=195.219.133.219 From: ;tag=92a4f9e8-1dd2-11b2-bd8c-b03162323164+92a4f9e8 To: "07976946209" ;tag=as7e2d6104 Call-ID: 3ff001e64ecb77313ff3c10432647719@84.8.129.188 CSeq: 2 INVITE User-Agent: alwaysON Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 182 v=0 o=root 16668 16671 IN IP4 84.8.191.14 s=session c=IN IP4 84.8.191.14 t=0 0 m=audio 13566 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:10 a=sendrecv <------------> aovastest01*CLI> <--- SIP read from 195.219.133.219:5065 ---> ACK sip:07976946209@84.8.129.188 SIP/2.0 From: ;tag=92a4f9e8-1dd2-11b2-bd8c-b03162323164+92a4f9e8 To: "07976946209" ;tag=as7e2d6104 Call-ID: 3ff001e64ecb77313ff3c10432647719@84.8.129.188 CSeq: 2 ACK Via: SIP/2.0/UDP 195.219.133.219:5065;branch=z9hG4bK686323010129616 Max-Forwards: 70 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 195.219.133.219:5065 ---> BYE sip:07976946209@84.8.129.188 SIP/2.0 From: ;tag=92a4f9e8-1dd2-11b2-bd8c-b03162323164+92a4f9e8 To: "07976946209" ;tag=as7e2d6104 Call-ID: 3ff001e64ecb77313ff3c10432647719@84.8.129.188 CSeq: 3 BYE Via: SIP/2.0/UDP 195.219.133.219:5065;branch=z9hG4bK66691639975282 Max-Forwards: 70 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Sending to 195.219.133.219 : 5065 (no NAT) <--- Transmitting (no NAT) to 195.219.133.219:5065 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 195.219.133.219:5065;branch=z9hG4bK66691639975282;received=195.219.133.219 From: ;tag=92a4f9e8-1dd2-11b2-bd8c-b03162323164+92a4f9e8 To: "07976946209" ;tag=as7e2d6104 Call-ID: 3ff001e64ecb77313ff3c10432647719@84.8.129.188 CSeq: 3 BYE User-Agent: alwaysON Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> set_destination: Parsing for address/port to send to set_destination: set destination to 84.8.191.13, port 5060 Audio is at 84.8.129.188 port 12702 Adding codec 0x4 (ulaw) to SDP Reliably Transmitting (no NAT) to 84.8.191.13:5060: INVITE sip:07976946209@84.8.191.13:5060 SIP/2.0 Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK7e792027;rport From: ;tag=as6880918b To: "07976946209" ;tag=3463916609-626192 Contact: Call-ID: 92483-3463916609-626185@aosbc1.alwaysongroup.com CSeq: 103 INVITE User-Agent: alwaysON 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: 184 v=0 o=root 16668 16670 IN IP4 84.8.129.188 s=session c=IN IP4 84.8.129.188 t=0 0 m=audio 12702 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:10 a=sendrecv --- Scheduling destruction of SIP dialog '92483-3463916609-626185@aosbc1.alwaysongroup.com' in 32000 ms (Method: ACK) aovastest01*CLI> <--- SIP read from 84.8.191.13:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK7e792027;rport From: ;tag=as6880918b To: "07976946209" ;tag=3463916609-626192 Call-ID: 92483-3463916609-626185@aosbc1.alwaysongroup.com CSeq: 103 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '3ff001e64ecb77313ff3c10432647719@84.8.129.188' Method: BYE aovastest01*CLI> <--- SIP read from 84.8.191.13:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK7e792027;rport To: "07976946209" ;tag=3463916609-626192 From: ;tag=as6880918b Call-ID: 92483-3463916609-626185@aosbc1.alwaysongroup.com CSeq: 103 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 138 v=0 o=aosbc1 0 2 IN IP4 84.8.191.13 s=sip call c=IN IP4 84.8.191.14 t=0 0 m=audio 13566 RTP/AVP 0 a=sendrecv a=rtpmap:0 PCMU/8000 <-------------> --- (11 headers 8 lines) --- Found RTP audio format 0 Peer audio RTP is at port 84.8.191.14:13566 Found audio description format PCMU for ID 0 Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 84.8.191.14:13566 set_destination: Parsing for address/port to send to set_destination: set destination to 84.8.191.13, port 5060 Transmitting (no NAT) to 84.8.191.13:5060: ACK sip:07976946209@84.8.191.13:5060 SIP/2.0 Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK5bea24f0;rport From: ;tag=as6880918b To: "07976946209" ;tag=3463916609-626192 Contact: Call-ID: 92483-3463916609-626185@aosbc1.alwaysongroup.com CSeq: 103 ACK User-Agent: alwaysON Max-Forwards: 70 Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 84.8.191.13, port 5060 Reliably Transmitting (no NAT) to 84.8.191.13:5060: BYE sip:07976946209@84.8.191.13:5060 SIP/2.0 Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK331cbad2;rport From: ;tag=as6880918b To: "07976946209" ;tag=3463916609-626192 Call-ID: 92483-3463916609-626185@aosbc1.alwaysongroup.com CSeq: 104 BYE User-Agent: alwaysON Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '92483-3463916609-626185@aosbc1.alwaysongroup.com' in 32000 ms (Method: ACK) aovastest01*CLI> <--- SIP read from 84.8.191.13:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK331cbad2;rport To: "07976946209" ;tag=3463916609-626192 From: ;tag=as6880918b Call-ID: 92483-3463916609-626185@aosbc1.alwaysongroup.com CSeq: 104 BYE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '92483-3463916609-626185@aosbc1.alwaysongroup.com' Method: ACK