--- filegate2*CLI> <-- SIP read from 216.143.130.36:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK138a7671;rport=5060 From: ;tag=as1051febd To: ;tag=68d238da7d1832978b1a894dbad8a35e.5d9d Call-ID: 681293625380292600bf2f4f510faa07@192.168.1.76 CSeq: 433 REGISTER Contact: ;expires=120 Server: OpenSer (1.0.0 (i386/freebsd)) Content-Length: 0 --- (9 headers 0 lines) --- Scheduling destruction of call '681293625380292600bf2f4f510faa07@192.168.1.76' in 32000 ms Mar 23 20:01:55 NOTICE[163848]: chan_sip.c:12279 handle_response_register: Outbound Registration: Expiry for sip.voipyourphone.com is 120 sec (Scheduling reregistration in 105 s) filegate2*CLI> <-- SIP read from 216.143.130.36:5060: NOTIFY sip:s@209.42.200.5:5060 SIP/2.0 Max-Forwards: 10 Record-Route: Via: SIP/2.0/UDP 216.143.130.36;branch=z9hG4bK4265.16330f61.0 Via: SIP/2.0/UDP resi-sip-01.rnktel.com:38896;received=216.143.130.47;rport=38897 From: "Unknown" To: Contact: Call-ID: 718111ed3c1ed2fb3Dx3Ow3Co4To4Bp7@216.143.130.47 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 36 Messages-Waiting: no Voicemail: 0/0 --- (14 headers 2 lines) --- Transmitting (NAT) to 216.143.130.36:5060: SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 216.143.130.36;branch=z9hG4bK4265.16330f61.0;received=216.143.130.36 Via: SIP/2.0/UDP resi-sip-01.rnktel.com:38896;received=216.143.130.47;rport=38897 From: "Unknown" To: ;tag=as46b030cc Call-ID: 718111ed3c1ed2fb3Dx3Ow3Co4To4Bp7@216.143.130.47 CSeq: 102 NOTIFY User-Agent: OpenPBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Content-Length: 0 --- filegate2*CLI> <-- SIP read from 192.168.1.147:5060: NOTIFY sip:192.168.1.76 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.147:5060;branch=z9hG4bK-65ac76ba From: 6001 ;tag=95f215bae45e8b41o0 To: Call-ID: 6c4ec45e-9ce70ced@192.168.1.147 CSeq: 1587 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Linksys/SPA2102-3.3.6 Content-Length: 0 --- (10 headers 0 lines) --- Transmitting (NAT) to 192.168.1.147:5060: SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 192.168.1.147:5060;branch=z9hG4bK-65ac76ba;received=192.168.1.147 From: 6001 ;tag=95f215bae45e8b41o0 To: ;tag=as445c32ac Call-ID: 6c4ec45e-9ce70ced@192.168.1.147 CSeq: 1587 NOTIFY User-Agent: OpenPBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Content-Length: 0 --- filegate2*CLI> <-- SIP read from 216.143.130.36:5060: INVITE sip:s@209.42.200.5:5060 SIP/2.0 Max-Forwards: 10 Record-Route: Via: SIP/2.0/UDP 216.143.130.36;branch=z9hG4bKaa66.05c5cd87.0 Via: SIP/2.0/UDP 216.143.130.95:5060;branch=z9hG4bK32b07293;rport=5060 From: "4150050473" ;tag=as3e769388 To: Contact: Call-ID: 21aafb7239a9f4101d6ad3bf4ad51310@216.143.130.95 CSeq: 102 INVITE ser-Agent: Asterisk PBX Date: Sat, 24 Mar 2007 00:02:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 319 v=0 o=root 4649 4649 IN IP4 216.143.130.95 s=session c=IN IP4 216.143.130.95 t=0 0 m=audio 48508 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 - - - - m=video 23128 RTP/AVP 31 34 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 --- (15 headers 14 lines) --- Using INVITE request as basis request - 21aafb7239a9f4101d6ad3bf4ad51310@216.143.130.95 Sending to 216.143.130.36 : 5060 (NAT) Found peer 'sip_proxy-out' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Mar 23 20:02:10 WARNING[163848]: chan_sip.c:4704 process_sdp: Unknown SDP media type in offer: video 23128 RTP/AVP 31 34 Peer audio RTP is at port 216.143.130.95:48508 Found description format PCMU Found description format PCMA Found description format telephone-event Found description format H261 Found description format H263 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for s in default (domain 209.42.200.5) list_route: hop: Transmitting (NAT) to 216.143.130.36:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 216.143.130.36;branch=z9hG4bKaa66.05c5cd87.0;received=216.143.130.36 Via: SIP/2.0/UDP 216.143.130.95:5060;branch=z9hG4bK32b07293;rport=5060 From: "4150050473" ;tag=as3e769388 To: Call-ID: 21aafb7239a9f4101d6ad3bf4ad51310@216.143.130.95 CSeq: 102 INVITE User-Agent: OpenPBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Length: 0 --- -- Executing FaxDetect("SIP/hillct-d3ac", "4000") in new stack We're at 192.168.1.76 port 12652 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 216.143.130.36:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 216.143.130.36;branch=z9hG4bKaa66.05c5cd87.0;received=216.143.130.36 Via: SIP/2.0/UDP 216.143.130.95:5060;branch=z9hG4bK32b07293;rport=5060 Record-Route: rom: "4150050473" ;tag=as3e769388 To: ;tag=as15686f6a Call-ID: 21aafb7239a9f4101d6ad3bf4ad51310@216.143.130.95 CSeq: 102 INVITE User-Agent: OpenPBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Type: application/sdp Content-Length: 261 v=0 o=root 2764 2764 IN IP4 192.168.1.76 s=session c=IN IP4 192.168.1.76 t=0 0 m=audio 12652 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- filegate2*CLI> <-- SIP read from 216.143.130.36:5060: ACK sip:s@209.42.200.5:5060 SIP/2.0 Max-Forwards: 10 Record-Route: Via: SIP/2.0/UDP 216.143.130.36;branch=0 Via: SIP/2.0/UDP 216.143.130.95:5060;branch=z9hG4bK08a58ec3;rport=5060 From: "4150050473" ;tag=as3e769388 To: ;tag=as15686f6a Contact: Call-ID: 21aafb7239a9f4101d6ad3bf4ad51310@216.143.130.95 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- (12 headers 0 lines) --- Mar 23 20:02:11 NOTICE[1441805]: app_faxdetect.c:306 detectfax_exec: Redirecting SIP/hillct-d3ac to fax extension [DTMF] -- Executing Dial("SIP/hillct-d3ac", "SIP/5000") in new stack We're at 192.168.1.76 port 13696 Adding codec 0x4 (ulaw) to SDP 13 headers, 8 lines Reliably Transmitting (NAT) to 192.168.1.147:5060: INVITE sip:5000@192.168.1.147:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK23175690;rport From: "4150050473" ;tag=as72d92084 To: Contact: Call-ID: 4f984df71b4ccfdf3fa56249233e7e6a@192.168.1.76 CSeq: 102 INVITE User-Agent: OpenPBX Max-Forwards: 70 Date: Sat, 24 Mar 2007 00:02:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 158 v=0 o=root 2764 2764 IN IP4 192.168.1.76 s=session c=IN IP4 192.168.1.76 t=0 0 m=audio 13696 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - --- -- Called 5000 Mar 23 20:02:11 NOTICE[1441805]: channel.c:1627 opbx_read: Dropping incompatible voice frame on SIP/hillct-d3ac of format slin since our native format has changed to ulaw filegate2*CLI> <-- SIP read from 192.168.1.147:5060: SIP/2.0 100 Trying To: From: "4150050473" ;tag=as72d92084 Call-ID: 4f984df71b4ccfdf3fa56249233e7e6a@192.168.1.76 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK23175690 Server: Linksys/SPA2102-3.3.6 Content-Length: 0 --- (8 headers 0 lines) --- filegate2*CLI> <-- SIP read from 192.168.1.147:5060: SIP/2.0 180 Ringing To: ;tag=cca40ac71da83e1ei1 From: "4150050473" ;tag=as72d92084 Call-ID: 4f984df71b4ccfdf3fa56249233e7e6a@192.168.1.76 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK23175690 Server: Linksys/SPA2102-3.3.6 Remote-Party-ID: 5000 ;screen=yes;party=called Content-Length: 0 --- (9 headers 0 lines) --- -- SIP/5000-b6c5 is ringing filegate2*CLI> <-- SIP read from 192.168.1.147:5060: NOTIFY sip:192.168.1.76 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.147:5060;branch=z9hG4bK-9fd3eaec From: 6001 ;tag=95f215bae45e8b41o0 To: Call-ID: 6c4ec45e-9ce70ced@192.168.1.147 CSeq: 1588 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Linksys/SPA2102-3.3.6 Content-Length: 0 --- (10 headers 0 lines) --- Transmitting (NAT) to 192.168.1.147:5060: SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 192.168.1.147:5060;branch=z9hG4bK-9fd3eaec;received=192.168.1.147 From: 6001 ;tag=95f215bae45e8b41o0 To: ;tag=as4ddcdd9e Call-ID: 6c4ec45e-9ce70ced@192.168.1.147 CSeq: 1588 NOTIFY User-Agent: OpenPBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Content-Length: 0 --- filegate2*CLI> <-- SIP read from 216.143.130.36:5060: NOTIFY sip:s@209.42.200.5:5060 SIP/2.0 Max-Forwards: 10 Record-Route: Via: SIP/2.0/UDP 216.143.130.36;branch=z9hG4bKcbfc.4d1d527.0 Via: SIP/2.0/UDP resi-sip-01.rnktel.com:39302;received=216.143.130.47;rport=39303 From: "Unknown" To: Contact: Call-ID: 718111ed3c1ed2fb3Lx3Tw5Ml5Op1Ax6@216.143.130.47 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 36 Messages-Waiting: no Voicemail: 0/0 --- (14 headers 2 lines) --- Transmitting (NAT) to 216.143.130.36:5060: SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 216.143.130.36;branch=z9hG4bKcbfc.4d1d527.0;received=216.143.130.36 Via: SIP/2.0/UDP resi-sip-01.rnktel.com:39302;received=216.143.130.47;rport=39303 From: "Unknown" To: ;tag=as1ae6b4d1 Call-ID: 718111ed3c1ed2fb3Lx3Tw5Ml5Op1Ax6@216.143.130.47 CSeq: 102 NOTIFY User-Agent: OpenPBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Content-Length: 0 --- filegate2*CLI> <-- SIP read from 192.168.1.147:5060: NOTIFY sip:192.168.1.76 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.147:5060;branch=z9hG4bK-7906da40 From: 6001 ;tag=95f215bae45e8b41o0 To: Call-ID: 6c4ec45e-9ce70ced@192.168.1.147 CSeq: 1589 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Linksys/SPA2102-3.3.6 Content-Length: 0 --- (10 headers 0 lines) --- Transmitting (NAT) to 192.168.1.147:5060: SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 192.168.1.147:5060;branch=z9hG4bK-7906da40;received=192.168.1.147 From: 6001 ;tag=95f215bae45e8b41o0 To: ;tag=as6d953122 Call-ID: 6c4ec45e-9ce70ced@192.168.1.147 CSeq: 1589 NOTIFY User-Agent: OpenPBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Content-Length: 0 --- Destroying call '681293625380292600bf2f4f510faa07@192.168.1.76' filegate2*CLI> <-- SIP read from 216.143.130.36:5060: --- (0 headers 0 lines) Nat keepalive --- filegate2*CLI> <-- SIP read from 192.168.1.147:5060: SIP/2.0 200 OK To: ;tag=cca40ac71da83e1ei1 From: "4150050473" ;tag=as72d92084 Call-ID: 4f984df71b4ccfdf3fa56249233e7e6a@192.168.1.76 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK23175690 Contact: 5000 Server: Linksys/SPA2102-3.3.6 Remote-Party-ID: 5000 ;screen=yes;party=called Content-Length: 257 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 2387898 2387898 IN IP4 192.168.1.147 s=- c=IN IP4 192.168.1.147 t=0 0 m=audio 16390 RTP/AVP 0 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv --- (13 headers 13 lines) --- Found RTP audio format 0 Found RTP audio format 100 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.147:16390 Found description format PCMU Found description format NSE Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.147, port 5060 Transmitting (NAT) to 192.168.1.147:5060: ACK sip:5000@192.168.1.147:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK42b6fc9e;rport From: "4150050473" ;tag=as72d92084 To: ;tag=cca40ac71da83e1ei1 Contact: Call-ID: 4f984df71b4ccfdf3fa56249233e7e6a@192.168.1.76 CSeq: 102 ACK User-Agent: OpenPBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/5000-b6c5 answered SIP/hillct-d3ac -- Attempting native bridge of SIP/hillct-d3ac and SIP/5000-b6c5 set_destination: Parsing for address/port to send to set_destination: set destination to 216.143.130.36, port 5060 We're at 192.168.1.76 port 12652 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 14 headers, 10 lines Reliably Transmitting (NAT) to 216.143.130.36:5060: INVITE sip:4150050473@216.143.130.95 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK05fc6954;rport Route: From: ;tag=as15686f6a To: "4150050473" ;tag=as3e769388 Contact: Call-ID: 21aafb7239a9f4101d6ad3bf4ad51310@216.143.130.95 CSeq: 102 INVITE User-Agent: OpenPBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-openpbx-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 216 v=0 o=root 2764 2765 IN IP4 192.168.1.147 s=session c=IN IP4 192.168.1.147 t=0 0 m=audio 16390 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.147, port 5060 We're at 192.168.1.76 port 13696 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 11 lines Reliably Transmitting (NAT) to 192.168.1.147:5060: INVITE sip:5000@192.168.1.147:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK09e9244d;rport From: "4150050473" ;tag=as72d92084 To: ;tag=cca40ac71da83e1ei1 Contact: Call-ID: 4f984df71b4ccfdf3fa56249233e7e6a@192.168.1.76 CSeq: 103 INVITE User-Agent: OpenPBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-openpbx-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 242 v=0 o=root 2764 2765 IN IP4 216.143.130.95 s=session c=IN IP4 216.143.130.95 t=0 0 m=audio 48508 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 - - - - --- filegate2*CLI> <-- SIP read from 192.168.1.147:5060: SIP/2.0 200 OK To: ;tag=cca40ac71da83e1ei1 From: "4150050473" ;tag=as72d92084 Call-ID: 4f984df71b4ccfdf3fa56249233e7e6a@192.168.1.76 CSeq: 103 INVITE Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK09e9244d Contact: 5000 Server: Linksys/SPA2102-3.3.6 Content-Length: 284 Content-Type: application/sdp v=0 o=- 2389749 2389749 IN IP4 192.168.1.147 s=- c=IN IP4 192.168.1.147 t=0 0 m=audio 16390 RTP/AVP 0 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=silenceSupp:off - - - - --- (10 headers 14 lines) --- Found RTP audio format 0 Found RTP audio format 100 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.147:16390 Found description format PCMU Found description format NSE Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Mar 23 20:02:30 WARNING[163848]: chan_sip.c:12037 handle_response_invite: Strange... The other side of the bridge don't have udptl struct set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.147, port 5060 Transmitting (NAT) to 192.168.1.147:5060: ACK sip:5000@192.168.1.147:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK588b5634;rport From: "4150050473" ;tag=as72d92084 To: ;tag=cca40ac71da83e1ei1 Contact: Call-ID: 4f984df71b4ccfdf3fa56249233e7e6a@192.168.1.76 CSeq: 103 ACK User-Agent: OpenPBX Max-Forwards: 70 Content-Length: 0 --- filegate2*CLI> <-- SIP read from 216.143.130.36:5060: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 209.42.200.5:5060;branch=z9hG4bK05fc6954;rport=5060 From: ;tag=as15686f6a To: "4150050473" ;tag=as3e769388 Call-ID: 21aafb7239a9f4101d6ad3bf4ad51310@216.143.130.95 CSeq: 102 INVITE Server: OpenSer (1.0.0 (i386/freebsd)) Content-Length: 0 --- (8 headers 0 lines) --- filegate2*CLI> <-- SIP read from 216.143.130.36:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 209.42.200.5:5060;branch=z9hG4bK05fc6954;rport=5060 Record-Route: From: ;tag=as15686f6a To: "4150050473" ;tag=as3e769388 Call-ID: 21aafb7239a9f4101d6ad3bf4ad51310@216.143.130.95 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 238 v=0 o=root 4649 4650 IN IP4 216.143.130.95 s=session c=IN IP4 216.143.130.95 t=0 0 m=audio 48508 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=direction:active --- (12 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 216.143.130.95:48508 Found description format PCMU Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: Mar 23 20:02:30 WARNING[163848]: chan_sip.c:12037 handle_response_invite: Strange... The other side of the bridge don't have udptl struct set_destination: Parsing for address/port to send to set_destination: set destination to 216.143.130.36, port 5060 Transmitting (NAT) to 216.143.130.36:5060: ACK sip:4150050473@216.143.130.95:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK231e5826;rport Route: From: ;tag=as15686f6a To: "4150050473" ;tag=as3e769388 Contact: Call-ID: 21aafb7239a9f4101d6ad3bf4ad51310@216.143.130.95 CSeq: 102 ACK User-Agent: OpenPBX Max-Forwards: 70 Content-Length: 0 --- filegate2*CLI> <-- SIP read from 216.143.130.36:5060: BYE sip:s@209.42.200.5:5060 SIP/2.0 Max-Forwards: 10 Record-Route: Via: SIP/2.0/UDP 216.143.130.36;branch=z9hG4bKba66.697031b4.0 Via: SIP/2.0/UDP 216.143.130.95:5060;branch=z9hG4bK5737f314;rport=5060 From: "4150050473" ;tag=as3e769388 To: ;tag=as15686f6a Contact: Call-ID: 21aafb7239a9f4101d6ad3bf4ad51310@216.143.130.95 CSeq: 103 BYE User-Agent: Asterisk PBX Content-Length: 0 --- (12 headers 0 lines) --- Sending to 216.143.130.36 : 5060 (NAT) Transmitting (NAT) to 216.143.130.36:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 216.143.130.36;branch=z9hG4bKba66.697031b4.0;received=216.143.130.36 Via: SIP/2.0/UDP 216.143.130.95:5060;branch=z9hG4bK5737f314;rport=5060 Record-Route: From: "4150050473" ;tag=as3e769388 To: ;tag=as15686f6a Call-ID: 21aafb7239a9f4101d6ad3bf4ad51310@216.143.130.95 CSeq: 103 BYE User-Agent: OpenPBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Length: 0 X-OpenPBX-HangupCause: Normal Clearing --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.147, port 5060 We're at 192.168.1.76 port 13696 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 10 lines Reliably Transmitting (NAT) to 192.168.1.147:5060: INVITE sip:5000@192.168.1.147:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK67e7b98c;rport From: "4150050473" ;tag=as72d92084 To: ;tag=cca40ac71da83e1ei1 Contact: Call-ID: 4f984df71b4ccfdf3fa56249233e7e6a@192.168.1.76 CSeq: 104 INVITE User-Agent: OpenPBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-openpbx-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 214 v=0 o=root 2764 2766 IN IP4 192.168.1.76 s=session c=IN IP4 192.168.1.76 t=0 0 m=audio 13696 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- == Spawn extension (default, fax, 1) exited non-zero on 'SIP/hillct-d3ac' filegate2*CLI> <-- SIP read from 192.168.1.147:5060: SIP/2.0 200 OK To: ;tag=cca40ac71da83e1ei1 From: "4150050473" ;tag=as72d92084 Call-ID: 4f984df71b4ccfdf3fa56249233e7e6a@192.168.1.76 CSeq: 104 INVITE Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK67e7b98c Contact: 5000 Server: Linksys/SPA2102-3.3.6 Content-Length: 284 Content-Type: application/sdp v=0 o=- 2390630 2390630 IN IP4 192.168.1.147 s=- c=IN IP4 192.168.1.147 t=0 0 m=audio 16390 RTP/AVP 0 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=silenceSupp:off - - - - --- (10 headers 14 lines) --- Found RTP audio format 0 Found RTP audio format 100 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.147:16390 Found description format PCMU Found description format NSE Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.147, port 5060 Transmitting (NAT) to 192.168.1.147:5060: ACK sip:5000@192.168.1.147:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK482af03f;rport From: "4150050473" ;tag=as72d92084 To: ;tag=cca40ac71da83e1ei1 Contact: Call-ID: 4f984df71b4ccfdf3fa56249233e7e6a@192.168.1.76 CSeq: 104 ACK User-Agent: OpenPBX Max-Forwards: 70 Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.147, port 5060 Reliably Transmitting (NAT) to 192.168.1.147:5060: BYE sip:5000@192.168.1.147:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK20e56adf;rport From: "4150050473" ;tag=as72d92084 To: ;tag=cca40ac71da83e1ei1 Contact: Call-ID: 4f984df71b4ccfdf3fa56249233e7e6a@192.168.1.76 CSeq: 105 BYE User-Agent: OpenPBX Max-Forwards: 70 Content-Length: 0 --- <-- SIP read from 192.168.1.147:5060: SIP/2.0 200 OK To: ;tag=cca40ac71da83e1ei1 From: "4150050473" ;tag=as72d92084 Call-ID: 4f984df71b4ccfdf3fa56249233e7e6a@192.168.1.76 CSeq: 105 BYE Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK20e56adf Server: Linksys/SPA2102-3.3.6 Content-Length: 0 --- (8 headers 0 lines) --- Destroying call '4f984df71b4ccfdf3fa56249233e7e6a@192.168.1.76' Destroying call '21aafb7239a9f4101d6ad3bf4ad51310@216.143.130.95' filegate2*CLI> <-- SIP read from 192.168.1.147:5060: NOTIFY sip:192.168.1.76 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.147:5060;branch=z9hG4bK-88a698d9 From: 6001 ;tag=95f215bae45e8b41o0 To: Call-ID: 6c4ec45e-9ce70ced@192.168.1.147 CSeq: 1590 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Linksys/SPA2102-3.3.6 Content-Length: 0 --- (10 headers 0 lines) --- Transmitting (NAT) to 192.168.1.147:5060: SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 192.168.1.147:5060;branch=z9hG4bK-88a698d9;received=192.168.1.147 From: 6001 ;tag=95f215bae45e8b41o0 To: ;tag=as096499c1 Call-ID: 6c4ec45e-9ce70ced@192.168.1.147 CSeq: 1590 NOTIFY User-Agent: OpenPBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Content-Length: 0 --- filegate2*CLI>