-- Retrying gatekeeper registration. REGISTER 12 headers, 0 lines REGISTER attempt 1 to 1123456@XXX.XXX.XXX.154 Reliably Transmitting (no NAT) to XXX.XXX.XXX.154:5060: REGISTER sip:XXX.XXX.XXX.154 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.242:5060;branch=z9hG4bK24e727e3;rport From: ;tag=as231833b2 To: Call-ID: 1141ed40687873a7234dd4dd7a220a83@127.0.0.1 CSeq: 112 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: Event: registration Content-Length: 0 --- asterisk04*CLI> <-- SIP read from XXX.XXX.XXX.154:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP XXX.XXX.XXX.242:5060;branch=z9hG4bK24e727e3;rport=5060 From: ;tag=as231833b2 To: Call-ID: 1141ed40687873a7234dd4dd7a220a83@127.0.0.1 CSeq: 112 REGISTER Contact: user-agent: Asterisk PBX expires: 120 event: registration Allow: REFER, INFO, BYE, CANCEL, INVITE Content-Length: 0 --- (12 headers 0 lines)--- Scheduling destruction of call '1141ed40687873a7234dd4dd7a220a83@127.0.0.1' in 32000 ms asterisk04*CLI> <-- SIP read from XXX.XXX.XXX.154:5060: INVITE sip:7676@XXX.XXX.XXX.242:5060 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.154:5060;branch=z9hG4bK5ea76e33c01c5d22a54038179c51738b From: "12345678" ;tag=GR52RWG346-34 To: "7676" Call-ID: 210920149@XXX.XXX.XXX.154 CSeq: 1 INVITE Contact: max-forwards: 70 Content-Type: application/sdp Content-Length: 180 v=0 o=Clarent 211785 211786 IN IP4 XXX.XXX.XXX.161 s=Clarent C5CM c=IN IP4 XXX.XXX.XXX.161 t=0 0 m=audio 5106 RTP/AVP 4 8 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=SendRecv --- (10 headers 9 lines)--- Using INVITE request as basis request - 210920149@XXX.XXX.XXX.154 Sending to XXX.XXX.XXX.154 : 5060 (non-NAT) Found no matching peer or user for 'XXX.XXX.XXX.154:5060' Found RTP audio format 4 Found RTP audio format 8 Peer audio RTP is at port XXX.XXX.XXX.161:5106 Found description format G723 Found description format PCMA Capabilities: us - 0x8000f (g723|gsm|ulaw|alaw|h263), peer - audio=0x9 (g723|alaw)/video=0x0 (nothing), combined - 0x9 (g723|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Looking for 7676 in from-sip-external (domain XXX.XXX.XXX.242) list_route: hop: Transmitting (no NAT) to XXX.XXX.XXX.154:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP XXX.XXX.XXX.154:5060;branch=z9hG4bK5ea76e33c01c5d22a54038179c51738b;received=XXX.XXX.XXX.154 From: "12345678" ;tag=GR52RWG346-34 To: "7676" Call-ID: 210920149@XXX.XXX.XXX.154 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing Wait("SIP/5060-08cad548", "2") in new stack asterisk04*CLI> <-- SIP read from 83.98.215.12:5060: OPTIONS sip:XXX.XXX.XXX.242 SIP/2.0 Via: SIP/2.0/UDP 83.98.215.12:5060;branch=z9hG4bK7952bbcf;rport From: "asterisk" ;tag=as440fa7c3 To: Contact: Call-ID: 060034281605057e268eb5c22027e643@83.98.215.12 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 05 Apr 2007 19:27:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- (12 headers 0 lines)--- Looking for s in from-sip-external (domain XXX.XXX.XXX.242) Transmitting (no NAT) to 83.98.215.12:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 83.98.215.12:5060;branch=z9hG4bK7952bbcf;rport;received=83.98.215.12 From: "asterisk" ;tag=as440fa7c3 To: ;tag=as1dd282e7 Call-ID: 060034281605057e268eb5c22027e643@83.98.215.12 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Destroying call '060034281605057e268eb5c22027e643@83.98.215.12' -- Executing Dial("SIP/5060-08cad548", "SIP/33XXXXXXXXX@XXX.XXX.XXX.154") in new stack We're at XXX.XXX.XXX.242 port 46272 Adding codec 0x1 (g723) to SDP 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 13 headers, 13 lines Reliably Transmitting (no NAT) to XXX.XXX.XXX.154:5060: INVITE sip:33XXXXXXXXX@XXX.XXX.XXX.154 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.242:5060;branch=z9hG4bK7156dbea;rport From: "12345678" ;tag=as623e29a0 To: Contact: Call-ID: 2a572a8e151917e57bdc18a57eaa4af5@XXX.XXX.XXX.242 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 05 Apr 2007 20:00:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 291 v=0 o=root 18566 18566 IN IP4 XXX.XXX.XXX.242 s=session c=IN IP4 XXX.XXX.XXX.242 t=0 0 m=audio 46272 RTP/AVP 4 3 0 8 101 a=rtpmap:4 G723/8000 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 - - - - --- -- Called 33XXXXXXXXX@XXX.XXX.XXX.154 asterisk04*CLI> <-- SIP read from XXX.XXX.XXX.154:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP XXX.XXX.XXX.242:5060;branch=z9hG4bK7156dbea;rport=5060 From: "12345678" ;tag=as623e29a0 To: ;tag=GR52RWG346-34 Call-ID: 2a572a8e151917e57bdc18a57eaa4af5@XXX.XXX.XXX.242 CSeq: 102 INVITE Contact: user-agent: Asterisk PBX date: Thu, 05 Apr 2007 20:00:27 GMT Content-Length: 0 --- (10 headers 0 lines)--- asterisk04*CLI> <-- SIP read from XXX.XXX.XXX.154:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP XXX.XXX.XXX.242:5060;branch=z9hG4bK7156dbea;rport=5060 From: "12345678" ;tag=as623e29a0 To: ;tag=GR52RWG346-34 Call-ID: 2a572a8e151917e57bdc18a57eaa4af5@XXX.XXX.XXX.242 CSeq: 102 INVITE Contact: user-agent: Asterisk PBX date: Thu, 05 Apr 2007 20:00:27 GMT Content-Type: application/sdp Content-Length: 160 v=0 o=Clarent 211860 211861 IN IP4 XXX.XXX.XXX.100 s=Clarent C5CM c=IN IP4 XXX.XXX.XXX.100 t=0 0 m=audio 5198 RTP/AVP 4 a=rtpmap:4 G723/8000 a=SendRecv --- (11 headers 8 lines)--- Found RTP audio format 4 Peer audio RTP is at port XXX.XXX.XXX.100:5198 Found description format G723 Capabilities: us - 0x8000f (g723|gsm|ulaw|alaw|h263), peer - audio=0x1 (g723)/video=0x0 (nothing), combined - 0x1 (g723) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) -- SIP/XXX.XXX.XXX.154-5bad is making progress passing it to SIP/5060-08cad548 We're at XXX.XXX.XXX.242 port 48778 Adding codec 0x1 (g723) to SDP Adding codec 0x8 (alaw) to SDP Transmitting (no NAT) to XXX.XXX.XXX.154:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP XXX.XXX.XXX.154:5060;branch=z9hG4bK5ea76e33c01c5d22a54038179c51738b;received=XXX.XXX.XXX.154 From: "12345678" ;tag=GR52RWG346-34 To: "7676" ;tag=as20e9275a Call-ID: 210920149@XXX.XXX.XXX.154 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 188 v=0 o=root 18566 18566 IN IP4 XXX.XXX.XXX.242 s=session c=IN IP4 XXX.XXX.XXX.242 t=0 0 m=audio 48778 RTP/AVP 4 8 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- asterisk04*CLI> <-- SIP read from XXX.XXX.XXX.154:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP XXX.XXX.XXX.242:5060;branch=z9hG4bK7156dbea;rport=5060 From: "12345678" ;tag=as623e29a0 To: ;tag=GR52RWG346-34 Call-ID: 2a572a8e151917e57bdc18a57eaa4af5@XXX.XXX.XXX.242 CSeq: 102 INVITE Contact: "Verso C5CM" user-agent: Asterisk PBX date: Thu, 05 Apr 2007 20:00:27 GMT Content-Type: application/sdp Content-Length: 150 v=0 o=Clarent 1 0 IN IP4 XXX.XXX.XXX.100 s=Clarent C5CM c=IN IP4 XXX.XXX.XXX.100 t=0 0 m=audio 5198 RTP/AVP 4 a=rtpmap:4 G723/8000 a=SendRecv --- (11 headers 8 lines)--- Found RTP audio format 4 Peer audio RTP is at port XXX.XXX.XXX.100:5198 Found description format G723 Capabilities: us - 0x8000f (g723|gsm|ulaw|alaw|h263), peer - audio=0x1 (g723)/video=0x0 (nothing), combined - 0x1 (g723) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to XXX.XXX.XXX.154, port 5060 Transmitting (no NAT) to XXX.XXX.XXX.154:5060: ACK sip:XXX.XXX.XXX.154:5060 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.242:5060;branch=z9hG4bK188e3c70;rport From: "12345678" ;tag=as623e29a0 To: ;tag=GR52RWG346-34 Contact: Call-ID: 2a572a8e151917e57bdc18a57eaa4af5@XXX.XXX.XXX.242 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/XXX.XXX.XXX.154-5bad answered SIP/5060-08cad548 We're at XXX.XXX.XXX.242 port 48778 Adding codec 0x1 (g723) to SDP Adding codec 0x8 (alaw) to SDP Reliably Transmitting (no NAT) to XXX.XXX.XXX.154:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP XXX.XXX.XXX.154:5060;branch=z9hG4bK5ea76e33c01c5d22a54038179c51738b;received=XXX.XXX.XXX.154 From: "12345678" ;tag=GR52RWG346-34 To: "7676" ;tag=as20e9275a Call-ID: 210920149@XXX.XXX.XXX.154 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 188 v=0 o=root 18566 18567 IN IP4 XXX.XXX.XXX.242 s=session c=IN IP4 XXX.XXX.XXX.242 t=0 0 m=audio 48778 RTP/AVP 4 8 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- -- Attempting native bridge of SIP/5060-08cad548 and SIP/XXX.XXX.XXX.154-5bad set_destination: Parsing for address/port to send to set_destination: set destination to XXX.XXX.XXX.154, port 5060 We're at XXX.XXX.XXX.242 port 46272 Adding codec 0x1 (g723) to SDP Adding codec 0x8 (alaw) to SDP 13 headers, 9 lines Reliably Transmitting (no NAT) to XXX.XXX.XXX.154:5060: INVITE sip:XXX.XXX.XXX.154:5060 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.242:5060;branch=z9hG4bK119f2144;rport From: "12345678" ;tag=as623e29a0 To: ;tag=GR52RWG346-34 Contact: Call-ID: 2a572a8e151917e57bdc18a57eaa4af5@XXX.XXX.XXX.242 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 185 v=0 o=root 18566 18567 IN IP4 XXX.XXX.XXX.161 s=session c=IN IP4 XXX.XXX.XXX.161 t=0 0 m=audio 5106 RTP/AVP 4 8 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- asterisk04*CLI> <-- SIP read from XXX.XXX.XXX.154:5060: ACK sip:7676@XXX.XXX.XXX.242 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.154:5060;branch=z9hG4bKc17ca9f02a2cba731344e9e8877b7ea6 From: "12345678" ;tag=GR52RWG346-34 To: "7676" ;tag=as20e9275a Call-ID: 210920149@XXX.XXX.XXX.154 CSeq: 1 ACK Contact: max-forwards: 70 --- (8 headers 0 lines)--- set_destination: Parsing for address/port to send to set_destination: set destination to XXX.XXX.XXX.154, port 5060 We're at XXX.XXX.XXX.242 port 48778 Adding codec 0x1 (g723) to SDP 13 headers, 8 lines Reliably Transmitting (no NAT) to XXX.XXX.XXX.154:5060: INVITE sip:XXX.XXX.XXX.154:5060 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.242:5060;branch=z9hG4bK53ee057e;rport From: "7676" ;tag=as20e9275a To: "12345678" ;tag=GR52RWG346-34 Contact: Call-ID: 210920149@XXX.XXX.XXX.154 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 165 v=0 o=root 18566 18568 IN IP4 XXX.XXX.XXX.100 s=session c=IN IP4 XXX.XXX.XXX.100 t=0 0 m=audio 5198 RTP/AVP 4 a=rtpmap:4 G723/8000 a=silenceSupp:off - - - - --- Retransmitting #1 (no NAT) to XXX.XXX.XXX.154:5060: INVITE sip:XXX.XXX.XXX.154:5060 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.242:5060;branch=z9hG4bK119f2144;rport From: "12345678" ;tag=as623e29a0 To: ;tag=GR52RWG346-34 Contact: Call-ID: 2a572a8e151917e57bdc18a57eaa4af5@XXX.XXX.XXX.242 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 185 v=0 o=root 18566 18567 IN IP4 XXX.XXX.XXX.161 s=session c=IN IP4 XXX.XXX.XXX.161 t=0 0 m=audio 5106 RTP/AVP 4 8 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- Retransmitting #1 (no NAT) to XXX.XXX.XXX.154:5060: INVITE sip:XXX.XXX.XXX.154:5060 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.242:5060;branch=z9hG4bK53ee057e;rport From: "7676" ;tag=as20e9275a To: "12345678" ;tag=GR52RWG346-34 Contact: Call-ID: 210920149@XXX.XXX.XXX.154 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 165 v=0 o=root 18566 18568 IN IP4 XXX.XXX.XXX.100 s=session c=IN IP4 XXX.XXX.XXX.100 t=0 0 m=audio 5198 RTP/AVP 4 a=rtpmap:4 G723/8000 a=silenceSupp:off - - - - --- Retransmitting #2 (no NAT) to XXX.XXX.XXX.154:5060: INVITE sip:XXX.XXX.XXX.154:5060 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.242:5060;branch=z9hG4bK119f2144;rport From: "12345678" ;tag=as623e29a0 To: ;tag=GR52RWG346-34 Contact: Call-ID: 2a572a8e151917e57bdc18a57eaa4af5@XXX.XXX.XXX.242 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 185 v=0 o=root 18566 18567 IN IP4 XXX.XXX.XXX.161 s=session c=IN IP4 XXX.XXX.XXX.161 t=0 0 m=audio 5106 RTP/AVP 4 8 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- Retransmitting #2 (no NAT) to XXX.XXX.XXX.154:5060: INVITE sip:XXX.XXX.XXX.154:5060 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.242:5060;branch=z9hG4bK53ee057e;rport From: "7676" ;tag=as20e9275a To: "12345678" ;tag=GR52RWG346-34 Contact: Call-ID: 210920149@XXX.XXX.XXX.154 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 165 v=0 o=root 18566 18568 IN IP4 XXX.XXX.XXX.100 s=session c=IN IP4 XXX.XXX.XXX.100 t=0 0 m=audio 5198 RTP/AVP 4 a=rtpmap:4 G723/8000 a=silenceSupp:off - - - - --- Retransmitting #3 (no NAT) to XXX.XXX.XXX.154:5060: INVITE sip:XXX.XXX.XXX.154:5060 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.242:5060;branch=z9hG4bK119f2144;rport From: "12345678" ;tag=as623e29a0 To: ;tag=GR52RWG346-34 Contact: Call-ID: 2a572a8e151917e57bdc18a57eaa4af5@XXX.XXX.XXX.242 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 185 v=0 o=root 18566 18567 IN IP4 XXX.XXX.XXX.161 s=session c=IN IP4 XXX.XXX.XXX.161 t=0 0 m=audio 5106 RTP/AVP 4 8 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- Retransmitting #3 (no NAT) to XXX.XXX.XXX.154:5060: INVITE sip:XXX.XXX.XXX.154:5060 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.242:5060;branch=z9hG4bK53ee057e;rport From: "7676" ;tag=as20e9275a To: "12345678" ;tag=GR52RWG346-34 Contact: Call-ID: 210920149@XXX.XXX.XXX.154 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 165 v=0 o=root 18566 18568 IN IP4 XXX.XXX.XXX.100 s=session c=IN IP4 XXX.XXX.XXX.100 t=0 0 m=audio 5198 RTP/AVP 4 a=rtpmap:4 G723/8000 a=silenceSupp:off - - - - --- Retransmitting #4 (no NAT) to XXX.XXX.XXX.154:5060: INVITE sip:XXX.XXX.XXX.154:5060 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.242:5060;branch=z9hG4bK119f2144;rport From: "12345678" ;tag=as623e29a0 To: ;tag=GR52RWG346-34 Contact: Call-ID: 2a572a8e151917e57bdc18a57eaa4af5@XXX.XXX.XXX.242 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 185 v=0 o=root 18566 18567 IN IP4 XXX.XXX.XXX.161 s=session c=IN IP4 XXX.XXX.XXX.161 t=0 0 m=audio 5106 RTP/AVP 4 8 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- Retransmitting #4 (no NAT) to XXX.XXX.XXX.154:5060: INVITE sip:XXX.XXX.XXX.154:5060 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.242:5060;branch=z9hG4bK53ee057e;rport From: "7676" ;tag=as20e9275a To: "12345678" ;tag=GR52RWG346-34 Contact: Call-ID: 210920149@XXX.XXX.XXX.154 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 165 v=0 o=root 18566 18568 IN IP4 XXX.XXX.XXX.100 s=session c=IN IP4 XXX.XXX.XXX.100 t=0 0 m=audio 5198 RTP/AVP 4 a=rtpmap:4 G723/8000 a=silenceSupp:off - - - - --- asterisk04*CLI> <-- SIP read from XXX.XXX.XXX.154:5060: BYE sip:7676@XXX.XXX.XXX.242 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.154:5060;branch=z9hG4bKfa5a9280c4ec8eb5114988b6bdde7ee2 From: "12345678" ;tag=GR52RWG346-34 To: "7676" ;tag=as20e9275a Call-ID: 210920149@XXX.XXX.XXX.154 CSeq: 2 BYE user-agent: Asterisk PBX x-asterisk-info: SIP re-invite (RTP bridge) max-forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- (11 headers 0 lines)--- Sending to XXX.XXX.XXX.154 : 5060 (non-NAT) Transmitting (no NAT) to XXX.XXX.XXX.154:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP XXX.XXX.XXX.154:5060;branch=z9hG4bKfa5a9280c4ec8eb5114988b6bdde7ee2;received=XXX.XXX.XXX.154 From: "12345678" ;tag=GR52RWG346-34 To: "7676" ;tag=as20e9275a Call-ID: 210920149@XXX.XXX.XXX.154 CSeq: 2 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- -- Executing Hangup("SIP/5060-08cad548", "") in new stack Retransmitting #5 (no NAT) to XXX.XXX.XXX.154:5060: INVITE sip:XXX.XXX.XXX.154:5060 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.242:5060;branch=z9hG4bK119f2144;rport From: "12345678" ;tag=as623e29a0 To: ;tag=GR52RWG346-34 Contact: Call-ID: 2a572a8e151917e57bdc18a57eaa4af5@XXX.XXX.XXX.242 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 185 v=0 o=root 18566 18567 IN IP4 XXX.XXX.XXX.161 s=session c=IN IP4 XXX.XXX.XXX.161 t=0 0 m=audio 5106 RTP/AVP 4 8 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- Retransmitting #5 (no NAT) to XXX.XXX.XXX.154:5060: INVITE sip:XXX.XXX.XXX.154:5060 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.242:5060;branch=z9hG4bK53ee057e;rport From: "7676" ;tag=as20e9275a To: "12345678" ;tag=GR52RWG346-34 Contact: Call-ID: 210920149@XXX.XXX.XXX.154 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 165 v=0 o=root 18566 18568 IN IP4 XXX.XXX.XXX.100 s=session c=IN IP4 XXX.XXX.XXX.100 t=0 0 m=audio 5198 RTP/AVP 4 a=rtpmap:4 G723/8000 a=silenceSupp:off - - - - --- asterisk04*CLI> <-- SIP read from XXX.XXX.XXX.154:5060: SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP XXX.XXX.XXX.242:5060;branch=z9hG4bK53ee057e;rport=5060 From: "7676" ;tag=as20e9275a To: "12345678" ;tag=GR52RWG346-34 Call-ID: 210920149@XXX.XXX.XXX.154 CSeq: 102 INVITE Contact: user-agent: Asterisk PBX x-asterisk-info: SIP re-invite (RTP bridge) Content-Length: 0 --- (10 headers 0 lines)--- -- Got SIP response 500 "Internal Server Error" back from XXX.XXX.XXX.154 set_destination: Parsing for address/port to send to set_destination: set destination to XXX.XXX.XXX.154, port 5060 Transmitting (no NAT) to XXX.XXX.XXX.154:5060: ACK 2.0 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.242:5060;branch=z9hG4bK53ee057e;rport From: "12345678" ;tag=GR52RWG346-34 To: "7676" ;tag=as20e9275a Contact: Call-ID: 210920149@XXX.XXX.XXX.154 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Destroying call '210920149@XXX.XXX.XXX.154' -- Retrying gatekeeper registration. asterisk04*CLI> <-- SIP read from XXX.XXX.XXX.154:5060: SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP XXX.XXX.XXX.242:5060;branch=z9hG4bK53ee057e;rport=5060 From: "7676" ;tag=as20e9275a To: "12345678" ;tag=GR52RWG346-34 Call-ID: 210920149@XXX.XXX.XXX.154 CSeq: 102 INVITE Contact: user-agent: Asterisk PBX x-asterisk-info: SIP re-invite (RTP bridge) Content-Length: 0 --- (10 headers 0 lines)--- Destroying call '210920149@XXX.XXX.XXX.154' Destroying call '1141ed40687873a7234dd4dd7a220a83@127.0.0.1' asterisk04*CLI> <-- SIP read from XXX.XXX.XXX.154:5060: SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP XXX.XXX.XXX.242:5060;branch=z9hG4bK53ee057e;rport=5060 From: "7676" ;tag=as20e9275a To: "12345678" ;tag=GR52RWG346-34 Call-ID: 210920149@XXX.XXX.XXX.154 CSeq: 102 INVITE Contact: user-agent: Asterisk PBX x-asterisk-info: SIP re-invite (RTP bridge) Content-Length: 0 --- (10 headers 0 lines)--- Destroying call '210920149@XXX.XXX.XXX.154' asterisk04*CLI> <-- SIP read from XXX.XXX.XXX.154:5060: BYE sip:12345678@XXX.XXX.XXX.242 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.154:5060;branch=z9hG4bK0f699e6c23fbbd6902f93247b5004b59 From: ;tag=GR52RWG346-34 To: "12345678" ;tag=as623e29a0 Call-ID: 2a572a8e151917e57bdc18a57eaa4af5@XXX.XXX.XXX.242 CSeq: 1 BYE user-agent: Asterisk PBX x-asterisk-info: SIP re-invite (RTP bridge) max-forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- (11 headers 0 lines)--- Sending to XXX.XXX.XXX.154 : 5060 (non-NAT) Transmitting (no NAT) to XXX.XXX.XXX.154:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP XXX.XXX.XXX.154:5060;branch=z9hG4bK0f699e6c23fbbd6902f93247b5004b59;received=XXX.XXX.XXX.154 From: ;tag=GR52RWG346-34 To: "12345678" ;tag=as623e29a0 Call-ID: 2a572a8e151917e57bdc18a57eaa4af5@XXX.XXX.XXX.242 CSeq: 1 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Retransmitting #6 (no NAT) to XXX.XXX.XXX.154:5060: INVITE sip:XXX.XXX.XXX.154:5060 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.242:5060;branch=z9hG4bK119f2144;rport From: "12345678" ;tag=as623e29a0 To: ;tag=GR52RWG346-34 Contact: Call-ID: 2a572a8e151917e57bdc18a57eaa4af5@XXX.XXX.XXX.242 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 185 v=0 o=root 18566 18567 IN IP4 XXX.XXX.XXX.161 s=session c=IN IP4 XXX.XXX.XXX.161 t=0 0 m=audio 5106 RTP/AVP 4 8 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- asterisk04*CLI> <-- SIP read from XXX.XXX.XXX.154:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP XXX.XXX.XXX.242:5060;branch=z9hG4bK119f2144;rport=5060 From: "12345678" ;tag=as623e29a0 To: ;tag=GR52RWG346-34 Call-ID: 2a572a8e151917e57bdc18a57eaa4af5@XXX.XXX.XXX.242 CSeq: 103 INVITE Contact: user-agent: Asterisk PBX x-asterisk-info: SIP re-invite (RTP bridge) Content-Length: 0 --- (10 headers 0 lines)--- asterisk04*CLI> <-- SIP read from XXX.XXX.XXX.154:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP XXX.XXX.XXX.242:5060;branch=z9hG4bK119f2144;rport=5060 From: "12345678" ;tag=as623e29a0 To: ;tag=GR52RWG346-34 Call-ID: 2a572a8e151917e57bdc18a57eaa4af5@XXX.XXX.XXX.242 CSeq: 103 INVITE Contact: user-agent: Asterisk PBX x-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 160 v=0 o=Clarent 211031 211032 IN IP4 XXX.XXX.XXX.100 s=Clarent C5CM c=IN IP4 XXX.XXX.XXX.100 t=0 0 m=audio 5194 RTP/AVP 4 a=rtpmap:4 G723/8000 a=SendRecv --- (11 headers 8 lines)--- asterisk04*CLI> <-- SIP read from XXX.XXX.XXX.154:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP XXX.XXX.XXX.242:5060;branch=z9hG4bK119f2144;rport=5060 From: "12345678" ;tag=as623e29a0 To: ;tag=GR52RWG346-34 Call-ID: 2a572a8e151917e57bdc18a57eaa4af5@XXX.XXX.XXX.242 CSeq: 103 INVITE Contact: "Verso C5CM" user-agent: Asterisk PBX x-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 150 v=0 o=Clarent 1 0 IN IP4 XXX.XXX.XXX.100 s=Clarent C5CM c=IN IP4 XXX.XXX.XXX.100 t=0 0 m=audio 5194 RTP/AVP 4 a=rtpmap:4 G723/8000 a=SendRecv --- (11 headers 8 lines)--- Destroying call '2a572a8e151917e57bdc18a57eaa4af5@XXX.XXX.XXX.242' asterisk04*CLI> <-- SIP read from XXX.XXX.XXX.154:5060: SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP XXX.XXX.XXX.242:5060;branch=z9hG4bK119f2144;rport=5060 From: "12345678" ;tag=as623e29a0 To: ;tag=GR52RWG346-34 Call-ID: 2a572a8e151917e57bdc18a57eaa4af5@XXX.XXX.XXX.242 CSeq: 103 INVITE Contact: "Verso C5CM" user-agent: Asterisk PBX x-asterisk-info: SIP re-invite (RTP bridge) Content-Length: 0 --- (10 headers 0 lines)--- Destroying call '2a572a8e151917e57bdc18a57eaa4af5@XXX.XXX.XXX.242' asterisk04*CLI>