Reliably Transmitting (NAT) to 192.168.1.8:5060: INVITE sip:123456789@sip.provider SIP/2.0 Via: SIP/2.0/UDP 192.168.1.71:5080;branch=z9hG4bK32b6ba03;rport Max-Forwards: 70 From: "Wolfgang Liegel" ;tag=as3a27951e To: Contact: Call-ID: 11d89ee404306ad264a922ed6af00a99@sip.provider CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Thu, 23 Feb 2012 09:13:15 GMT Session-Expires: 1800 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 308 v=0 o=root 618482267 618482267 IN IP4 192.168.1.71 s=Asterisk PBX 1.8.9.2 c=IN IP4 192.168.1.71 t=0 0 m=audio 12332 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called SIP/123456789@sip.provider <--- SIP read from UDP:192.168.1.8:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.168.1.71:5080;branch=z9hG4bK32b6ba03;rport=5080 From: "Wolfgang Liegel" ;tag=as3a27951e To: Call-ID: 11d89ee404306ad264a922ed6af00a99@sip.provider CSeq: 102 INVITE Server: Sip EXpress router (0.9.6 (i386/linux)) Content-Length: 0 Warning: 392 192.168.1.8:5060 "Noisy feedback tells: pid=25384 req_src_ip=192.168.1.71 req_src_port=5080 in_uri=sip:123456789@sip.provider out_uri=sip:+123456789@192.168.97.118:5060 via_cnt==1" <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP:192.168.1.8:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.1.71:5080;branch=z9hG4bK32b6ba03;rport=5080 From: "VG1006" ;tag=as3a27951e To: ;tag=00E0F510089D384679127F3528ED Call-ID: 11d89ee404306ad264a922ed6af00a99@sip.provider CSeq: 102 INVITE Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY, SUBSCRIBE, UPDATE Content-Type: application/sdp Content-Length: 185 v=0 o=- 3012094665 0 IN IP4 192.168.97.107 s=session t=0 0 m=audio 5200 RTP/AVP 0 101 c=IN IP4 192.168.97.123 a=rtpmap:0 PCMU/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 <-------------> --- (9 headers 9 lines) --- list_route: no route Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=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 192.168.97.123:5200 -- SIP/sip.provider-0000002d is making progress passing it to SIP/11-0000002c <--- SIP read from UDP:192.168.1.8:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.71:5080;branch=z9hG4bK32b6ba03;rport=5080 From: "VG1006" ;tag=as3a27951e To: ;tag=00E0F510089D384679127F3528ED Call-ID: 11d89ee404306ad264a922ed6af00a99@sip.provider CSeq: 102 INVITE Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY, SUBSCRIBE, UPDATE Content-Type: application/sdp Content-Length: 185 v=0 o=- 3012094665 1 IN IP4 192.168.97.107 s=session t=0 0 m=audio 5200 RTP/AVP 0 101 c=IN IP4 192.168.97.123 a=rtpmap:0 PCMU/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 <-------------> --- (9 headers 9 lines) --- list_route: no route Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=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 192.168.97.123:5200 -- SIP/sip.provider-0000002d is ringing -- SIP/sip.provider-0000002d is making progress passing it to SIP/11-0000002c <--- SIP read from UDP:192.168.1.8:5060 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.1.71:5080;branch=z9hG4bK32b6ba03;rport=5080 Record-Route: Record-Route: From: "VG1006" ;tag=as3a27951e To: ;tag=00E0F510089D384679127F3528ED Call-ID: 11d89ee404306ad264a922ed6af00a99@sip.provider CSeq: 102 INVITE Contact: Allow-Events: refer Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY, SUBSCRIBE, UPDATE Content-Type: application/sdp Supported: timer, replaces Content-Length: 185 v=0 o=- 3012094665 2 IN IP4 192.168.97.107 s=session t=0 0 m=audio 5200 RTP/AVP 0 101 c=IN IP4 192.168.97.123 a=rtpmap:0 PCMU/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 <-------------> --- (14 headers 9 lines) --- Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=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 192.168.97.123:5200 list_route: hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.8:5060 Transmitting (NAT) to 192.168.1.8:5060: ACK sip:192.168.97.107:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.71:5080;branch=z9hG4bK5ecead5e;rport Route: , Max-Forwards: 70 From: "Wolfgang Liegel" ;tag=as3a27951e To: ;tag=00E0F510089D384679127F3528ED Contact: Call-ID: 11d89ee404306ad264a922ed6af00a99@sip.provider CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- -- SIP/sip.provider-0000002d answered SIP/11-0000002c -- Locally bridging SIP/11-0000002c and SIP/sip.provider-0000002d <--- SIP read from UDP:192.168.1.8:5060 ---> INVITE sip:111111@192.168.1.71:5080 SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 192.168.1.8;branch=z9hG4bK6dae.4c31af37.0 Via: SIP/2.0/UDP 192.168.97.118;branch=z9hG4bK6dae.06ee3945.0 Via: SIP/2.0/UDP 192.168.97.107:5060;branch=z9hG4bK00E0F510089D38467D16758335A1 From: ;tag=00E0F510089D384679127F3528ED To: "VG1006" ;tag=as3a27951e Call-ID: 11d89ee404306ad264a922ed6af00a99@sip.provider CSeq: 12052 INVITE Contact: Max-Forwards: 68 Content-Type: application/sdp Supported: timer, replaces Content-Length: 310 v=0 o=- 3012094665 3 IN IP4 192.168.97.107 s=session t=0 0 m=audio 5200 RTP/AVP 8 0 2 18 18 3 96 c=IN IP4 192.168.97.123 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:18 G729/8000 a=rtpmap:18 G729A/8000 a=rtpmap:3 GSM/8000 a=sendonly a=rtpmap:96 telephone-event/8000 <-------------> --- (15 headers 14 lines) --- Sending to 192.168.1.8:5060 (NAT) Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 2 Found RTP audio format 18 Found RTP audio format 18 Found RTP audio format 3 Found RTP audio format 96 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G726-32 for ID 2 Found audio description format G729 for ID 18 Found audio description format G729A for ID 18 Found audio description format GSM for ID 3 Found audio description format telephone-event for ID 96 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x90e (gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.97.123:5200 <--- Transmitting (NAT) to 192.168.1.8:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.8;branch=z9hG4bK6dae.4c31af37.0;received=192.168.1.8;rport=5060 Via: SIP/2.0/UDP 192.168.97.118;branch=z9hG4bK6dae.06ee3945.0 Via: SIP/2.0/UDP 192.168.97.107:5060;branch=z9hG4bK00E0F510089D38467D16758335A1 Record-Route: Record-Route: From: ;tag=00E0F510089D384679127F3528ED To: "VG1006" ;tag=as3a27951e Call-ID: 11d89ee404306ad264a922ed6af00a99@sip.provider CSeq: 12052 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> Audio is at 12332 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 192.168.1.8:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.8;branch=z9hG4bK6dae.4c31af37.0;received=192.168.1.8;rport=5060 Via: SIP/2.0/UDP 192.168.97.118;branch=z9hG4bK6dae.06ee3945.0 Via: SIP/2.0/UDP 192.168.97.107:5060;branch=z9hG4bK00E0F510089D38467D16758335A1 Record-Route: Record-Route: From: ;tag=00E0F510089D384679127F3528ED To: "VG1006" ;tag=as3a27951e Call-ID: 11d89ee404306ad264a922ed6af00a99@sip.provider CSeq: 12052 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 305 v=0 o=root 618482267 618482268 IN IP4 192.168.1.71 s=Asterisk PBX 1.8.9.2 c=IN IP4 192.168.1.71 t=0 0 m=audio 12332 RTP/AVP 0 8 3 96 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> -- Started music on hold, class 'default', on SIP/11-0000002c <--- SIP read from UDP:192.168.1.8:5060 ---> ACK sip:111111@192.168.1.71:5080 SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 192.168.1.8;branch=0 Via: SIP/2.0/UDP 192.168.97.118;branch=z9hG4bK6dae.06ee3945.2 Via: SIP/2.0/UDP 192.168.97.107:5060;branch=z9hG4bK00E0F510089D38467D185EBB13C6 From: ;tag=00E0F510089D384679127F3528ED To: "VG1006" ;tag=as3a27951e Call-ID: 11d89ee404306ad264a922ed6af00a99@sip.provider CSeq: 12052 ACK Contact: Max-Forwards: 68 Content-Length: 0 <--- SIP read from UDP:192.168.1.8:5060 ---> INVITE sip:111111@192.168.1.71:5080 SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 192.168.1.8;branch=z9hG4bK7dae.9123f973.0 Via: SIP/2.0/UDP 192.168.97.118;branch=z9hG4bK7dae.68efafb3.0 Via: SIP/2.0/UDP 192.168.97.107:5060;branch=z9hG4bK00E0F510089D38467FBF8FEB1458 From: ;tag=00E0F510089D384679127F3528ED To: "VG1006" ;tag=as3a27951e Call-ID: 11d89ee404306ad264a922ed6af00a99@sip.provider CSeq: 12053 INVITE Contact: Max-Forwards: 68 Content-Type: application/sdp Supported: timer, replaces Content-Length: 310 v=0 o=- 3012094665 4 IN IP4 192.168.97.107 s=session t=0 0 m=audio 5200 RTP/AVP 8 0 2 18 18 3 96 c=IN IP4 192.168.97.123 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:18 G729/8000 a=rtpmap:18 G729A/8000 a=rtpmap:3 GSM/8000 a=sendrecv a=rtpmap:96 telephone-event/8000 <-------------> --- (15 headers 14 lines) --- Sending to 192.168.1.8:5060 (NAT) Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 2 Found RTP audio format 18 Found RTP audio format 18 Found RTP audio format 3 Found RTP audio format 96 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G726-32 for ID 2 Found audio description format G729 for ID 18 Found audio description format G729A for ID 18 Found audio description format GSM for ID 3 Found audio description format telephone-event for ID 96 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x90e (gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.97.123:5200 <--- Transmitting (NAT) to 192.168.1.8:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.8;branch=z9hG4bK7dae.9123f973.0;received=192.168.1.8;rport=5060 Via: SIP/2.0/UDP 192.168.97.118;branch=z9hG4bK7dae.68efafb3.0 Via: SIP/2.0/UDP 192.168.97.107:5060;branch=z9hG4bK00E0F510089D38467FBF8FEB1458 Record-Route: Record-Route: From: ;tag=00E0F510089D384679127F3528ED To: "VG1006" ;tag=as3a27951e Call-ID: 11d89ee404306ad264a922ed6af00a99@sip.provider CSeq: 12053 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> Audio is at 12332 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 192.168.1.8:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.8;branch=z9hG4bK7dae.9123f973.0;received=192.168.1.8;rport=5060 Via: SIP/2.0/UDP 192.168.97.118;branch=z9hG4bK7dae.68efafb3.0 Via: SIP/2.0/UDP 192.168.97.107:5060;branch=z9hG4bK00E0F510089D38467FBF8FEB1458 Record-Route: Record-Route: From: ;tag=00E0F510089D384679127F3528ED To: "VG1006" ;tag=as3a27951e Call-ID: 11d89ee404306ad264a922ed6af00a99@sip.provider CSeq: 12053 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 305 v=0 o=root 618482267 618482269 IN IP4 192.168.1.71 s=Asterisk PBX 1.8.9.2 c=IN IP4 192.168.1.71 t=0 0 m=audio 12332 RTP/AVP 0 8 3 96 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Stopped music on hold on SIP/11-0000002c <--- SIP read from UDP:192.168.1.8:5060 ---> ACK sip:111111@192.168.1.71:5080 SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 192.168.1.8;branch=0 Via: SIP/2.0/UDP 192.168.97.118;branch=z9hG4bK7dae.68efafb3.2 Via: SIP/2.0/UDP 192.168.97.107:5060;branch=z9hG4bK00E0F510089D38467FC1FB5AA9B1 From: ;tag=00E0F510089D384679127F3528ED To: "VG1006" ;tag=as3a27951e Call-ID: 11d89ee404306ad264a922ed6af00a99@sip.provider CSeq: 12053 ACK Contact: Max-Forwards: 68 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- <--- SIP read from UDP:192.168.1.8:5060 ---> INVITE sip:111111@192.168.1.71:5080 SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 192.168.1.8;branch=z9hG4bK4dae.02edd842.0 Via: SIP/2.0/UDP 192.168.97.118;branch=z9hG4bK4dae.21a64fc5.0 Via: SIP/2.0/UDP 192.168.97.107:5060;branch=z9hG4bK00E0F510089D384681373E8CA5FC From: ;tag=00E0F510089D384679127F3528ED To: "VG1006" ;tag=as3a27951e Call-ID: 11d89ee404306ad264a922ed6af00a99@sip.provider CSeq: 12054 INVITE Contact: Max-Forwards: 68 Content-Type: application/sdp Supported: timer, replaces Content-Length: 310 v=0 o=- 3012094665 5 IN IP4 192.168.97.107 s=session t=0 0 m=audio 5200 RTP/AVP 8 0 2 18 18 3 96 c=IN IP4 192.168.97.123 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:18 G729/8000 a=rtpmap:18 G729A/8000 a=rtpmap:3 GSM/8000 a=sendonly a=rtpmap:96 telephone-event/8000 <-------------> --- (15 headers 14 lines) --- Sending to 192.168.1.8:5060 (NAT) Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 2 Found RTP audio format 18 Found RTP audio format 18 Found RTP audio format 3 Found RTP audio format 96 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G726-32 for ID 2 Found audio description format G729 for ID 18 Found audio description format G729A for ID 18 Found audio description format GSM for ID 3 Found audio description format telephone-event for ID 96 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x90e (gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.97.123:5200 <--- Transmitting (NAT) to 192.168.1.8:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.8;branch=z9hG4bK4dae.02edd842.0;received=192.168.1.8;rport=5060 Via: SIP/2.0/UDP 192.168.97.118;branch=z9hG4bK4dae.21a64fc5.0 Via: SIP/2.0/UDP 192.168.97.107:5060;branch=z9hG4bK00E0F510089D384681373E8CA5FC Record-Route: Record-Route: From: ;tag=00E0F510089D384679127F3528ED To: "VG1006" ;tag=as3a27951e Call-ID: 11d89ee404306ad264a922ed6af00a99@sip.provider CSeq: 12054 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> Audio is at 12332 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 192.168.1.8:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.8;branch=z9hG4bK4dae.02edd842.0;received=192.168.1.8;rport=5060 Via: SIP/2.0/UDP 192.168.97.118;branch=z9hG4bK4dae.21a64fc5.0 Via: SIP/2.0/UDP 192.168.97.107:5060;branch=z9hG4bK00E0F510089D384681373E8CA5FC Record-Route: Record-Route: From: ;tag=00E0F510089D384679127F3528ED To: "VG1006" ;tag=as3a27951e Call-ID: 11d89ee404306ad264a922ed6af00a99@sip.provider CSeq: 12054 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 305 v=0 o=root 618482267 618482270 IN IP4 192.168.1.71 s=Asterisk PBX 1.8.9.2 c=IN IP4 192.168.1.71 t=0 0 m=audio 12332 RTP/AVP 0 8 3 96 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> -- Started music on hold, class 'default', on SIP/11-0000002c -- Locally bridging SIP/11-0000002c and SIP/sip.provider-0000002d <--- SIP read from UDP:192.168.1.8:5060 ---> ACK sip:111111@192.168.1.71:5080 SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 192.168.1.8;branch=0 Via: SIP/2.0/UDP 192.168.97.118;branch=z9hG4bK4dae.21a64fc5.2 Via: SIP/2.0/UDP 192.168.97.107:5060;branch=z9hG4bK00E0F510089D38468139D95DA485 From: ;tag=00E0F510089D384679127F3528ED To: "VG1006" ;tag=as3a27951e Call-ID: 11d89ee404306ad264a922ed6af00a99@sip.provider CSeq: 12054 ACK Contact: Max-Forwards: 68 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- <--- SIP read from UDP:192.168.1.8:5060 ---> INVITE sip:111111@192.168.1.71:5080 SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 192.168.1.8;branch=z9hG4bK5dae.96af088.0 Via: SIP/2.0/UDP 192.168.97.118;branch=z9hG4bK5dae.5eaa4523.0 Via: SIP/2.0/UDP 192.168.97.107:5060;branch=z9hG4bK00E0F510089D3846839C55D9ECDB From: ;tag=00E0F510089D384679127F3528ED To: "VG1006" ;tag=as3a27951e Call-ID: 11d89ee404306ad264a922ed6af00a99@sip.provider CSeq: 12055 INVITE Contact: Max-Forwards: 68 Content-Type: application/sdp Supported: timer, replaces Content-Length: 310 v=0 o=- 3012094665 6 IN IP4 192.168.97.107 s=session t=0 0 m=audio 5200 RTP/AVP 8 0 2 18 18 3 96 c=IN IP4 192.168.97.123 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:18 G729/8000 a=rtpmap:18 G729A/8000 a=rtpmap:3 GSM/8000 a=sendrecv a=rtpmap:96 telephone-event/8000 <-------------> --- (15 headers 14 lines) --- Sending to 192.168.1.8:5060 (NAT) Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 2 Found RTP audio format 18 Found RTP audio format 18 Found RTP audio format 3 Found RTP audio format 96 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G726-32 for ID 2 Found audio description format G729 for ID 18 Found audio description format G729A for ID 18 Found audio description format GSM for ID 3 Found audio description format telephone-event for ID 96 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x90e (gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.97.123:5200 <--- Transmitting (NAT) to 192.168.1.8:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.8;branch=z9hG4bK5dae.96af088.0;received=192.168.1.8;rport=5060 Via: SIP/2.0/UDP 192.168.97.118;branch=z9hG4bK5dae.5eaa4523.0 Via: SIP/2.0/UDP 192.168.97.107:5060;branch=z9hG4bK00E0F510089D3846839C55D9ECDB Record-Route: Record-Route: From: ;tag=00E0F510089D384679127F3528ED To: "VG1006" ;tag=as3a27951e Call-ID: 11d89ee404306ad264a922ed6af00a99@sip.provider CSeq: 12055 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> Audio is at 12332 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 192.168.1.8:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.8;branch=z9hG4bK5dae.96af088.0;received=192.168.1.8;rport=5060 Via: SIP/2.0/UDP 192.168.97.118;branch=z9hG4bK5dae.5eaa4523.0 Via: SIP/2.0/UDP 192.168.97.107:5060;branch=z9hG4bK00E0F510089D3846839C55D9ECDB Record-Route: Record-Route: From: ;tag=00E0F510089D384679127F3528ED To: "VG1006" ;tag=as3a27951e Call-ID: 11d89ee404306ad264a922ed6af00a99@sip.provider CSeq: 12055 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 305 v=0 o=root 618482267 618482271 IN IP4 192.168.1.71 s=Asterisk PBX 1.8.9.2 c=IN IP4 192.168.1.71 t=0 0 m=audio 12332 RTP/AVP 0 8 3 96 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Stopped music on hold on SIP/11-0000002c <--- SIP read from UDP:192.168.1.8:5060 ---> ACK sip:111111@192.168.1.71:5080 SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 192.168.1.8;branch=0 Via: SIP/2.0/UDP 192.168.97.118;branch=z9hG4bK5dae.5eaa4523.2 Via: SIP/2.0/UDP 192.168.97.107:5060;branch=z9hG4bK00E0F510089D3846839DC056C378 From: ;tag=00E0F510089D384679127F3528ED To: "VG1006" ;tag=as3a27951e Call-ID: 11d89ee404306ad264a922ed6af00a99@sip.provider CSeq: 12055 ACK Contact: Max-Forwards: 68 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- <--- SIP read from UDP:192.168.1.8:5060 ---> BYE sip:111111@192.168.1.71:5080 SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 192.168.1.8;branch=z9hG4bK2dae.a9bbd0d4.0 Via: SIP/2.0/UDP 192.168.97.118;branch=z9hG4bK2dae.1d47839.0 Via: SIP/2.0/UDP 192.168.97.107:5060;branch=z9hG4bK00E0F510089D3846846E7466A289 From: ;tag=00E0F510089D384679127F3528ED To: "VG1006" ;tag=as3a27951e Call-ID: 11d89ee404306ad264a922ed6af00a99@sip.provider CSeq: 12056 BYE Contact: Max-Forwards: 68 Reason: Q.850;cause=16;text="Normal call clearing" Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Sending to 192.168.1.8:5060 (NAT) Scheduling destruction of SIP dialog '11d89ee404306ad264a922ed6af00a99@sip.provider' in 6400 ms (Method: BYE) <--- Transmitting (NAT) to 192.168.1.8:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.8;branch=z9hG4bK2dae.a9bbd0d4.0;received=192.168.1.8;rport=5060 Via: SIP/2.0/UDP 192.168.97.118;branch=z9hG4bK2dae.1d47839.0 Via: SIP/2.0/UDP 192.168.97.107:5060;branch=z9hG4bK00E0F510089D3846846E7466A289 Record-Route: Record-Route: From: ;tag=00E0F510089D384679127F3528ED To: "VG1006" ;tag=as3a27951e Call-ID: 11d89ee404306ad264a922ed6af00a99@sip.provider CSeq: 12056 BYE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0