asteriskRecorder*CLI> <--- SIP read from 192.168.1.104:5060 ---> REGISTER sip:192.168.1.27 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.104:5060;rport;branch=z9hG4bK7C8A4951775D4C2799C9EEFBB72AEDBA From: Roderick ;tag=1848079261 To: Roderick Contact: "Roderick" Call-ID: 100FD5584507403DA28BC927854DD24F@192.168.1.27 CSeq: 27961 REGISTER Expires: 60 Max-Forwards: 70 User-Agent: X-Lite release 1103m Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.1.104 : 5060 (NAT) <--- Transmitting (no NAT) to 192.168.1.104:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK7C8A4951775D4C2799C9EEFBB72AEDBA;received=192.168.1.104;rport=5060 From: Roderick ;tag=1848079261 To: Roderick Call-ID: 100FD5584507403DA28BC927854DD24F@192.168.1.27 CSeq: 27961 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> asteriskRecorder*CLI> <--- Transmitting (no NAT) to 192.168.1.104:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK7C8A4951775D4C2799C9EEFBB72AEDBA;received=192.168.1.104;rport=5060 From: Roderick ;tag=1848079261 To: Roderick ;tag=as5ff8a78b Call-ID: 100FD5584507403DA28BC927854DD24F@192.168.1.27 CSeq: 27961 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Tue, 26 Jun 2007 19:37:06 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '100FD5584507403DA28BC927854DD24F@192.168.1.27' in 32000 ms (Method: REGISTER) asteriskRecorder*CLI> <--- SIP read from 192.168.1.100:5060 ---> INVITE sip:123456@192.168.1.27:5065 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bK7614d594a11d3d7d From: ;tag=ddfe4836d6e8ad6e To: Contact: Supported: replaces Call-ID: 5367e8b3e8a75b26@192.168.1.100 CSeq: 6184 INVITE User-Agent: Grandstream HT286 1.0.7.19 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 156 v=0 o=279 8000 8000 IN IP4 192.168.1.100 s=SIP Call c=IN IP4 192.168.1.100 t=0 0 m=audio 8002 RTP/AVP 0 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:40 <-------------> --- (13 headers 9 lines) --- Sending to 192.168.1.100 : 5060 (no NAT) Using INVITE request as basis request - 5367e8b3e8a75b26@192.168.1.100 Found peer '279' Found RTP audio format 0 Peer audio RTP is at port 192.168.1.100:8002 Found description format PCMU for ID 0 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.1.100:8002 Looking for 123456 in default (domain 192.168.1.27) list_route: hop: <--- Transmitting (no NAT) to 192.168.1.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bK7614d594a11d3d7d;received=192.168.1.100 From: ;tag=ddfe4836d6e8ad6e To: Call-ID: 5367e8b3e8a75b26@192.168.1.100 CSeq: 6184 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [123456@default:1] Set("SIP/279-081dcad0", "MONITOR_EXEC=/etc/asterisk/soxmixwav.sh") in new stack -- Executing [123456@default:2] Playback("SIP/279-081dcad0", "sera_grabada") in new stack Audio is at 192.168.1.27 port 10652 Adding codec 0x4 (ulaw) to SDP <--- Reliably Transmitting (no NAT) to 192.168.1.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bK7614d594a11d3d7d;received=192.168.1.100 From: ;tag=ddfe4836d6e8ad6e To: ;tag=as5ea476c4 Call-ID: 5367e8b3e8a75b26@192.168.1.100 CSeq: 6184 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 182 v=0 o=root 6675 6675 IN IP4 192.168.1.27 s=session c=IN IP4 192.168.1.27 t=0 0 m=audio 10652 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Playing 'sera_grabada' (language 'en') asteriskRecorder*CLI> <--- SIP read from 192.168.1.100:5060 ---> ACK sip:123456@192.168.1.27:5065 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bK5baea0d15ebefa6d From: ;tag=ddfe4836d6e8ad6e To: ;tag=as5ea476c4 Contact: Call-ID: 5367e8b3e8a75b26@192.168.1.100 CSeq: 6184 ACK User-Agent: Grandstream HT286 1.0.7.19 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 <-------------> --- (11 headers 0 lines) --- -- Executing [123456@default:3] Answer("SIP/279-081dcad0", "") in new stack -- Executing [123456@default:4] Monitor("SIP/279-081dcad0", "wav|123456") in new stack -- Executing [123456@default:5] Dial("SIP/279-081dcad0", "SIP/123456") in new stack Audio is at 192.168.1.27 port 16866 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Reliably Transmitting (no NAT) to 192.168.1.27:5060: INVITE sip:123456@192.168.1.27:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.27:5065;branch=z9hG4bK30e306a3;rport From: "279" ;tag=as3b74adbb To: Contact: Call-ID: 1a50079e121d5a025a2a25124067b0de@192.168.1.27 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 26 Jun 2007 19:37:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 229 v=0 o=root 6675 6675 IN IP4 192.168.1.27 s=session c=IN IP4 192.168.1.27 t=0 0 m=audio 16866 RTP/AVP 0 3 8 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 123456 asteriskRecorder*CLI> <--- SIP read from 192.168.1.27:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.27:5065;branch=z9hG4bK30e306a3;rport From: "279" ;tag=as3b74adbb To: ;tag=1984039473 Contact: Call-ID: 1a50079e121d5a025a2a25124067b0de@192.168.1.27 CSeq: 102 INVITE Server: X-Lite release 1105d Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from 192.168.1.27:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.27:5065;branch=z9hG4bK30e306a3;rport From: "279" ;tag=as3b74adbb To: ;tag=1984039473 Contact: Call-ID: 1a50079e121d5a025a2a25124067b0de@192.168.1.27 CSeq: 102 INVITE Server: X-Lite release 1105d Content-Length: 0 <-------------> --- (9 headers 0 lines) --- -- SIP/123456-081d7480 is ringing set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.100, port 5060 Audio is at 192.168.1.27 port 10652 Adding codec 0x4 (ulaw) to SDP Reliably Transmitting (no NAT) to 192.168.1.100:5060: INVITE sip:279@192.168.1.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.27:5065;branch=z9hG4bK7f7dc7bc;rport From: ;tag=as5ea476c4 To: ;tag=ddfe4836d6e8ad6e Contact: Call-ID: 5367e8b3e8a75b26@192.168.1.100 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: 182 v=0 o=root 6675 6676 IN IP4 192.168.1.27 s=session c=IN IP4 192.168.1.27 t=0 0 m=audio 10652 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- <--- SIP read from 192.168.1.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.27:5065;branch=z9hG4bK7f7dc7bc;rport From: ;tag=as5ea476c4 To: ;tag=ddfe4836d6e8ad6e Call-ID: 5367e8b3e8a75b26@192.168.1.100 CSeq: 102 INVITE User-Agent: Grandstream HT286 1.0.7.19 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Supported: replaces Content-Length: 156 v=0 o=279 8000 8001 IN IP4 192.168.1.100 s=SIP Call c=IN IP4 192.168.1.100 t=0 0 m=audio 8002 RTP/AVP 0 =sendrecvecorder*CLI> a=rtpmap:0 PCMU/8000 a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 192.168.1.100:8002 Found description format PCMU for ID 0 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.1.100:8002 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.100, port 5060 Transmitting (no NAT) to 192.168.1.100:5060: ACK sip:279@192.168.1.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.27:5065;branch=z9hG4bK580b5810;rport From: ;tag=as5ea476c4 To: ;tag=ddfe4836d6e8ad6e Contact: Call-ID: 5367e8b3e8a75b26@192.168.1.100 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- asteriskRecorder*CLI> <--- SIP read from 192.168.1.27:5060 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.1.27:5065;branch=z9hG4bK30e306a3;rport From: "279" ;tag=as3b74adbb To: ;tag=1984039473 Contact: Call-ID: 1a50079e121d5a025a2a25124067b0de@192.168.1.27 CSeq: 102 INVITE Content-Type: application/sdp Server: X-Lite release 1105d Content-Length: 311 v=0 o=123456 1770631733 1770638493 IN IP4 192.168.1.27 s=X-Lite c=IN IP4 192.168.1.27 t=0 0 m=audio 8000 RTP/AVP 0 8 3 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> --- (10 headers 14 lines) --- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 98 Found RTP audio format 97 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.27:8000 Found description format pcmu for ID 0 Found description format pcma for ID 8 Found description format gsm for ID 3 Found description format iLBC for ID 98 Found description format speex for ID 97 Found description format telephone-event for ID 101 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.1.27:8000 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.27, port 5060 Transmitting (no NAT) to 192.168.1.27:5060: ACK sip:123456@192.168.1.27:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.27:5065;branch=z9hG4bK689c9fce;rport From: "279" ;tag=as3b74adbb To: ;tag=1984039473 Contact: Call-ID: 1a50079e121d5a025a2a25124067b0de@192.168.1.27 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/123456-081d7480 answered SIP/279-081dcad0 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.100, port 5060 Audio is at 192.168.1.27 port 10652 Adding codec 0x4 (ulaw) to SDP Reliably Transmitting (no NAT) to 192.168.1.100:5060: INVITE sip:279@192.168.1.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.27:5065;branch=z9hG4bK24ebc30f;rport From: ;tag=as5ea476c4 To: ;tag=ddfe4836d6e8ad6e Contact: Call-ID: 5367e8b3e8a75b26@192.168.1.100 CSeq: 103 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: 181 v=0 o=root 6675 6677 IN IP4 192.168.1.27 s=session c=IN IP4 192.168.1.27 t=0 0 m=audio 8000 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- asteriskRecorder*CLI> <--- SIP read from 192.168.1.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.27:5065;branch=z9hG4bK24ebc30f;rport From: ;tag=as5ea476c4 To: ;tag=ddfe4836d6e8ad6e Call-ID: 5367e8b3e8a75b26@192.168.1.100 CSeq: 103 INVITE User-Agent: Grandstream HT286 1.0.7.19 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Supported: replaces Content-Length: 156 v=0 o=279 8000 8002 IN IP4 192.168.1.100 s=SIP Call c=IN IP4 192.168.1.100 t=0 0 m=audio 8002 RTP/AVP 0 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 192.168.1.100:8002 Found description format PCMU for ID 0 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.1.100:8002 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.100, port 5060 Transmitting (no NAT) to 192.168.1.100:5060: ACK sip:279@192.168.1.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.27:5065;branch=z9hG4bK76774c8a;rport From: ;tag=as5ea476c4 To: ;tag=ddfe4836d6e8ad6e Contact: Call-ID: 5367e8b3e8a75b26@192.168.1.100 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Really destroying SIP dialog '306fcaeb1fe9e2e3@192.168.1.100' Method: REGISTER Really destroying SIP dialog '1A58AF22A0D97CE7EDD9112B17A83AF3@192.168.1.27' Method: REGISTER asteriskRecorder*CLI> <--- SIP read from 192.168.1.100:5060 ---> REGISTER sip:192.168.1.27:5065 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bKbd03fda823d73b54 From: ;tag=7ac086505cddff9f To: Contact: Supported: replaces Call-ID: 306fcaeb1fe9e2e3@192.168.1.100 CSeq: 241 REGISTER Expires: 60 User-Agent: Grandstream HT286 1.0.7.19 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 <------------->r*CLI> --- (13 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.1.100 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.1.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bKbd03fda823d73b54;received=192.168.1.100 From: ;tag=7ac086505cddff9f To: Call-ID: 306fcaeb1fe9e2e3@192.168.1.100 CSeq: 241 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> asteriskRecorder*CLI> <--- Transmitting (no NAT) to 192.168.1.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bKbd03fda823d73b54;received=192.168.1.100 From: ;tag=7ac086505cddff9f To: ;tag=as343ae754 Call-ID: 306fcaeb1fe9e2e3@192.168.1.100 CSeq: 241 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Tue, 26 Jun 2007 19:37:38 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '306fcaeb1fe9e2e3@192.168.1.100' in 32000 ms (Method: REGISTER) Really destroying SIP dialog '100FD5584507403DA28BC927854DD24F@192.168.1.27' Method: REGISTER asteriskRecorder*CLI> <--- SIP read from 192.168.1.100:5060 ---> BYE sip:123456@192.168.1.27:5065 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bK604d7c9453a2dab4 From: ;tag=ddfe4836d6e8ad6e To: ;tag=as5ea476c4 Supported: replaces Call-ID: 5367e8b3e8a75b26@192.168.1.100 CSeq: 6185 BYE User-Agent: Grandstream HT286 1.0.7.19 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 192.168.1.100 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.1.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bK604d7c9453a2dab4;received=192.168.1.100 From: ;tag=ddfe4836d6e8ad6e To: ;tag=as5ea476c4 Call-ID: 5367e8b3e8a75b26@192.168.1.100 CSeq: 6185 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 '1a50079e121d5a025a2a25124067b0de@192.168.1.27' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.27, port 5060 Reliably Transmitting (no NAT) to 192.168.1.27:5060: BYE sip:123456@192.168.1.27:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.27:5065;branch=z9hG4bK705351c1;rport From: "279" ;tag=as3b74adbb To: ;tag=1984039473 Call-ID: 1a50079e121d5a025a2a25124067b0de@192.168.1.27 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- == Spawn extension (default, 123456, 5) exited non-zero on 'SIP/279-081dcad0' asteriskRecorder*CLI> <--- SIP read from 192.168.1.27:5060 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.1.27:5065;branch=z9hG4bK705351c1;rport From: "279" ;tag=as3b74adbb To: ;tag=1984039473 Contact: Call-ID: 1a50079e121d5a025a2a25124067b0de@192.168.1.27 CSeq: 103 BYE Server: X-Lite release 1105d Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '1a50079e121d5a025a2a25124067b0de@192.168.1.27' Method: INVITE Really destroying SIP dialog '5367e8b3e8a75b26@192.168.1.100' Method: BYE asteriskRecorder*CLI> <--- SIP read from 192.168.1.27:5060 ---> REGISTER sip:192.168.1.27 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.27:5060;rport;branch=z9hG4bK1889F1F1070BEF509A4D33B19C5FE963 From: 1234566 ;tag=1358852701 To: 1234566 Contact: "1234566" Call-ID: 1A58AF22A0D97CE7EDD9112B17A83AF3@192.168.1.27 CSeq: 56613 REGISTER Expires: 60 Max-Forwards: 70 User-Agent: X-Lite release 1105d Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.1.27 : 5060 (NAT) <--- Transmitting (no NAT) to 192.168.1.27:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.27:5060;branch=z9hG4bK1889F1F1070BEF509A4D33B19C5FE963;received=192.168.1.27;rport=5060 From: 1234566 ;tag=1358852701 To: 1234566 Call-ID: 1A58AF22A0D97CE7EDD9112B17A83AF3@192.168.1.27 CSeq: 56613 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- Transmitting (no NAT) to 192.168.1.27:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.27:5060;branch=z9hG4bK1889F1F1070BEF509A4D33B19C5FE963;received=192.168.1.27;rport=5060 From: 1234566 ;tag=1358852701 To: 1234566 ;tag=as3ecdfa30 Call-ID: 1A58AF22A0D97CE7EDD9112B17A83AF3@192.168.1.27 CSeq: 56613 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Tue, 26 Jun 2007 19:37:59 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '1A58AF22A0D97CE7EDD9112B17A83AF3@192.168.1.27' in 32000 ms (Method: REGISTER) asteriskRecorder*CLI> <--- SIP read from 192.168.1.104:5060 ---> REGISTER sip:192.168.1.27 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.104:5060;rport;branch=z9hG4bK7FCA4317695A49B8ACB8F480819C5D02 From: Roderick ;tag=1848079261 To: Roderick Contact: "Roderick" Call-ID: 100FD5584507403DA28BC927854DD24F@192.168.1.27 CSeq: 27962 REGISTER Expires: 60 Max-Forwards: 70 User-Agent: X-Lite release 1103m Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.1.104 : 5060 (NAT) <--- Transmitting (no NAT) to 192.168.1.104:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK7FCA4317695A49B8ACB8F480819C5D02;received=192.168.1.104;rport=5060 From: Roderick ;tag=1848079261 To: Roderick Call-ID: 100FD5584507403DA28BC927854DD24F@192.168.1.27 CSeq: 27962 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- Transmitting (no NAT) to 192.168.1.104:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK7FCA4317695A49B8ACB8F480819C5D02;received=192.168.1.104;rport=5060 From: Roderick ;tag=1848079261 To: Roderick ;tag=as37123fdd Call-ID: 100FD5584507403DA28BC927854DD24F@192.168.1.27 CSeq: 27962 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Tue, 26 Jun 2007 19:38:07 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '100FD5584507403DA28BC927854DD24F@192.168.1.27' in 32000 ms (Method: REGISTER) Really destroying SIP dialog '306fcaeb1fe9e2e3@192.168.1.100' Method: REGISTER asteriskRecorder*CLI> <--- SIP read from 192.168.1.27:5060 ---> <-------------> --- (0 headers 0 lines) Nat keepalive --- asteriskRecorder*CLI> <--- SIP read from 192.168.1.100:5060 ---> REGISTER sip:192.168.1.27:5065 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bK4453d80c67348e02 From: ;tag=7ac086505cddff9f To: Contact: Supported: replaces Call-ID: 306fcaeb1fe9e2e3@192.168.1.100 CSeq: 242 REGISTER Expires: 60 User-Agent: Grandstream HT286 1.0.7.19 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 <------------->r*CLI> --- (13 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.1.100 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.1.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bK4453d80c67348e02;received=192.168.1.100 From: ;tag=7ac086505cddff9f To: Call-ID: 306fcaeb1fe9e2e3@192.168.1.100 CSeq: 242 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- Transmitting (no NAT) to 192.168.1.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bK4453d80c67348e02;received=192.168.1.100 From: ;tag=7ac086505cddff9f To: ;tag=as080494b8 Call-ID: 306fcaeb1fe9e2e3@192.168.1.100 CSeq: 242 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Tue, 26 Jun 2007 19:38:23 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '306fcaeb1fe9e2e3@192.168.1.100' in 32000 ms (Method: REGISTER) Really destroying SIP dialog '1A58AF22A0D97CE7EDD9112B17A83AF3@192.168.1.27' Method: REGISTER Really destroying SIP dialog '100FD5584507403DA28BC927854DD24F@192.168.1.27' Method: REGISTER asteriskRecorder*CLI> <--- SIP read from 192.168.1.104:5060 ---> <-------------> --- (0 headers 0 lines) Nat keepalive --- asteriskRecorder*CLI>