Found RTP audio format 3 Found RTP audio format 98 Found RTP audio format 97 Found RTP audio format 101 Peer RTP is at port 81.101.182.51:0 Found description format pcmu Found description format pcma Found description format gsm Found description format iLBC Found description format speex Found description format telephone-event Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - audio=0x60e(GSM|ULAW|ALAW|SPEEX|ILBC)/video=0x0(EMPTY), combined - 0xe(GSM|ULAW|ALAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Reliably Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 81.101.182.51:5060;rport;branch=z9hG4bK24F468818DC84C7FA726D0518830D107;received=81.101.182.51;rport=5060 From: Phil ;tag=2170766058 To: ;tag=as5c5441da Call-ID: 0D39E5C8-5411-4C86-82FF-FE0FFEC2D645@192.168.1.100 CSeq: 58567 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="26363f45" Content-Length: 0 to 81.101.182.51:5060 Scheduling destruction of call '0D39E5C8-5411-4C86-82FF-FE0FFEC2D645@192.168.1.100' in 15000 ms Found user '2004' localhost*CLI> Sip read: ACK sip:2001@rekcutmnehpets.co.uk SIP/2.0 Via: SIP/2.0/UDP 81.101.182.51:5060;rport;branch=z9hG4bK24F468818DC84C7FA726D0518830D107 From: Phil ;tag=2170766058 To: ;tag=as5c5441da Contact: Call-ID: 0D39E5C8-5411-4C86-82FF-FE0FFEC2D645@192.168.1.100 CSeq: 58567 ACK Max-Forwards: 70 Content-Length: 0 9 headers, 0 lines localhost*CLI> Sip read: INVITE sip:2001@rekcutmnehpets.co.uk SIP/2.0 Via: SIP/2.0/UDP 81.101.182.51:5060;rport;branch=z9hG4bK7CF09C3FA06744A3B85657B45A89D830 From: Phil ;tag=2170766058 To: Contact: Call-ID: 0D39E5C8-5411-4C86-82FF-FE0FFEC2D645@192.168.1.100 CSeq: 58568 INVITE Proxy-Authorization: Digest username="2004",realm="asterisk",nonce="26363f45",response="f8e914100e07dc2265f9f2795539229a",uri="sip:2001@rekcutmnehpets.co.uk" Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1103m Content-Length: 295 v=0 o=2004 81776781 81776796 IN IP4 81.101.182.51 s=X-Lite c=IN IP4 81.101.182.51 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 12 headers, 13 lines Using latest request as basis request Sending to 81.101.182.51 : 5060 (NAT) 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 RTP is at port 81.101.182.51:0 Found description format pcmu Found description format pcma Found description format gsm Found description format iLBC Found description format speex Found description format telephone-event Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - audio=0x60e(GSM|ULAW|ALAW|SPEEX|ILBC)/video=0x0(EMPTY), combined - 0xe(GSM|ULAW|ALAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Found user '2004' Looking for 2001 in sip list_route: hop: Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 81.101.182.51:5060;rport;branch=z9hG4bK7CF09C3FA06744A3B85657B45A89D830;received=81.101.182.51;rport=5060 From: Phil ;tag=2170766058 To: ;tag=as460e436e Call-ID: 0D39E5C8-5411-4C86-82FF-FE0FFEC2D645@192.168.1.100 CSeq: 58568 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 81.101.182.51:5060 -- Executing Wait("SIP/2004-a438", "1") in new stack -- Executing Dial("SIP/2004-a438", "SIP/2001|20|tr") in new stack We're at 192.168.1.100 port 14906 Answering/Requesting with root capability 4 Answering with capability 0x2(GSM) Answering with capability 0x8(ALAW) Answering with non-codec capability 0x1(G723) 12 headers, 12 lines Reliably Transmitting: INVITE sip:2001@192.168.1.101 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK5d175a95;rport From: "Phil" ;tag=as764db2d4 To: Contact: Call-ID: 4d24d06d5c6bfd4418e2786449cd8598@192.168.1.100 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 07 Jul 2004 19:53:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 265 v=0 o=root 26949 26949 IN IP4 192.168.1.100 s=session c=IN IP4 192.168.1.100 t=0 0 m=audio 14906 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (NAT) to 192.168.1.101:5060 -- Called 2001 Transmitting (NAT): SIP/2.0 180 Ringing Via: SIP/2.0/UDP 81.101.182.51:5060;rport;branch=z9hG4bK7CF09C3FA06744A3B85657B45A89D830;received=81.101.182.51;rport=5060 From: Phil ;tag=2170766058 To: ;tag=as460e436e Call-ID: 0D39E5C8-5411-4C86-82FF-FE0FFEC2D645@192.168.1.100 CSeq: 58568 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 81.101.182.51:5060 localhost*CLI> Sip read: SIP/2.0 100 Trying To: From: "Phil" ;tag=as764db2d4 Call-ID: 4d24d06d5c6bfd4418e2786449cd8598@192.168.1.100 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK5d175a95;rport Server: Sipura/SPA2000-1.0.33 Content-Length: 0 8 headers, 0 lines Sip read: SIP/2.0 180 Ringing To: ;tag=745e600bfc19decf From: "Phil" ;tag=as764db2d4 Call-ID: 4d24d06d5c6bfd4418e2786449cd8598@192.168.1.100 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK5d175a95;rport Server: Sipura/SPA2000-1.0.33 Content-Length: 0 8 headers, 0 lines -- SIP/2001-7299 is ringing 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:192.168.1.101 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK036a5d0b From: "asterisk" ;tag=as612a6e10 To: Contact: Call-ID: 62b210a3782cf2d5779b299e3f06527e@192.168.1.100 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 07 Jul 2004 19:54:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 192.168.1.101:5060 localhost*CLI> Sip read: SIP/2.0 404 Not Found To: From: "asterisk" ;tag=as612a6e10 Call-ID: 62b210a3782cf2d5779b299e3f06527e@192.168.1.100 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK036a5d0b Server: Sipura/SPA2000-1.0.33 Content-Length: 0 8 headers, 0 lines Destroying call '62b210a3782cf2d5779b299e3f06527e@192.168.1.100' localhost*CLI> Sip read: SIP/2.0 200 OK To: ;tag=745e600bfc19decf From: "Phil" ;tag=as764db2d4 Call-ID: 4d24d06d5c6bfd4418e2786449cd8598@192.168.1.100 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK5d175a95;rport Contact: 2001 Server: Sipura/SPA2000-1.0.33 Content-Length: 212 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Content-Type: application/sdp v=0 o=- 25309479 25309479 IN IP4 192.168.1.101 s=- c=IN IP4 192.168.1.101 t=0 0 m=audio 16420 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv 11 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer RTP is at port 192.168.1.101:0 Found description format PCMU Found description format telephone-event Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - audio=0x4(ULAW)/video=0x0(EMPTY), combined - 0x4(ULAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.101, port 5060 Transmitting: ACK sip:2001@192.168.1.101:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK60be3799;rport From: "Phil" ;tag=as764db2d4 To: ;tag=745e600bfc19decf Contact: Call-ID: 4d24d06d5c6bfd4418e2786449cd8598@192.168.1.100 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 192.168.1.101:5060 -- SIP/2001-7299 answered SIP/2004-a438 We're at 192.168.1.100 port 16832 Answering with capability 0x2(GSM) Answering with capability 0x4(ULAW) Answering with capability 0x8(ALAW) Answering with non-codec capability 0x1(G723) Reliably Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 81.101.182.51:5060;rport;branch=z9hG4bK7CF09C3FA06744A3B85657B45A89D830;received=81.101.182.51;rport=5060 From: Phil ;tag=2170766058 To: ;tag=as460e436e Call-ID: 0D39E5C8-5411-4C86-82FF-FE0FFEC2D645@192.168.1.100 CSeq: 58568 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 265 v=0 o=root 26949 26949 IN IP4 192.168.1.100 s=session c=IN IP4 192.168.1.100 t=0 0 m=audio 16832 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 - - - - to 81.101.182.51:5060 -- Attempting native bridge of SIP/2004-a438 and SIP/2001-7299 -- Attempting native bridge of SIP/2004-a438 and SIP/2001-7299 Retransmitting #1 (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 81.101.182.51:5060;rport;branch=z9hG4bK7CF09C3FA06744A3B85657B45A89D830;received=81.101.182.51;rport=5060 From: Phil ;tag=2170766058 To: ;tag=as460e436e Call-ID: 0D39E5C8-5411-4C86-82FF-FE0FFEC2D645@192.168.1.100 CSeq: 58568 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 265 v=0 o=root 26949 26949 IN IP4 192.168.1.100 s=session c=IN IP4 192.168.1.100 t=0 0 m=audio 16832 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 - - - - E-9E8D- to 81.101.182.51:5060 Retransmitting #2 (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 81.101.182.51:5060;rport;branch=z9hG4bK7CF09C3FA06744A3B85657B45A89D830;received=81.101.182.51;rport=5060 From: Phil ;tag=2170766058 To: ;tag=as460e436e Call-ID: 0D39E5C8-5411-4C86-82FF-FE0FFEC2D645@192.168.1.100 CSeq: 58568 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 265 v=0 o=root 26949 26949 IN IP4 192.168.1.100 s=session c=IN IP4 192.168.1.100 t=0 0 m=audio 16832 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 - - - - E-9E8D-9 to 81.101.182.51:5060 Retransmitting #3 (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 81.101.182.51:5060;rport;branch=z9hG4bK7CF09C3FA06744A3B85657B45A89D830;received=81.101.182.51;rport=5060 From: Phil ;tag=2170766058 To: ;tag=as460e436e Call-ID: 0D39E5C8-5411-4C86-82FF-FE0FFEC2D645@192.168.1.100 CSeq: 58568 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 265 v=0 o=root 26949 26949 IN IP4 192.168.1.100 s=session c=IN IP4 192.168.1.100 t=0 0 m=audio 16832 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 - - - - E-9E8D-9 to 81.101.182.51:5060 Retransmitting #4 (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 81.101.182.51:5060;rport;branch=z9hG4bK7CF09C3FA06744A3B85657B45A89D830;received=81.101.182.51;rport=5060 From: Phil ;tag=2170766058 To: ;tag=as460e436e Call-ID: 0D39E5C8-5411-4C86-82FF-FE0FFEC2D645@192.168.1.100 CSeq: 58568 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 265 v=0 o=root 26949 26949 IN IP4 192.168.1.100 s=session c=IN IP4 192.168.1.100 t=0 0 m=audio 16832 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 - - - - E-9E8D-9 to 81.101.182.51:5060 Retransmitting #5 (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 81.101.182.51:5060;rport;branch=z9hG4bK7CF09C3FA06744A3B85657B45A89D830;received=81.101.182.51;rport=5060 From: Phil ;tag=2170766058 To: ;tag=as460e436e Call-ID: 0D39E5C8-5411-4C86-82FF-FE0FFEC2D645@192.168.1.100 CSeq: 58568 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 265 v=0 o=root 26949 26949 IN IP4 192.168.1.100 s=session c=IN IP4 192.168.1.100 t=0 0 m=audio 16832 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 - - - - E-9E8D-9 to 81.101.182.51:5060 Jul 7 20:54:12 WARNING[-214266960]: chan_sip.c:674 retrans_pkt: Maximum retries exceeded on call 0D39E5C8-5411-4C86-82FF-FE0FFEC2D645@192.168.1.100 for seqno 58568 (Non-critical Response) localhost*CLI> Sip read: BYE sip:2004@192.168.1.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK-f01bc233 From: ;tag=745e600bfc19decf To: "Phil" ;tag=as764db2d4 Call-ID: 4d24d06d5c6bfd4418e2786449cd8598@192.168.1.100 CSeq: 101 BYE Max-Forwards: 70 User-Agent: Sipura/SPA2000-1.0.33 Content-Length: 0 9 headers, 0 lines Sending to 192.168.1.101 : 5060 (NAT) Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK-f01bc233;received=192.168.1.101;rport=5060 From: ;tag=745e600bfc19decf To: "Phil" ;tag=as764db2d4 Call-ID: 4d24d06d5c6bfd4418e2786449cd8598@192.168.1.100 CSeq: 101 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.1.101:5060 == Spawn extension (sip, 2001, 2) exited non-zero on 'SIP/2004-a438' Destroying call '4d24d06d5c6bfd4418e2786449cd8598@192.168.1.100' Destroying call '0D39E5C8-5411-4C86-82FF-FE0FFEC2D645@192.168.1.100' localhost*CLI> sip no debug