SIP Debugging enabled bmy*CLI> <-- SIP read from 67.84.215.54:9168: --- (0 headers 0 lines) Nat keepalive --- bmy*CLI> <-- SIP read from 198.65.166.131:5060: --- (0 headers 0 lines) Nat keepalive --- bmy*CLI> <-- SIP read from 67.84.215.54:9168: INVITE sip:12016180234@bmy.gotdns.org SIP/2.0 To: From: ;tag=d33b6915 Via: SIP/2.0/UDP 10.0.0.100:9168;branch=z9hG4bK-d87543-130614794-1--d87543-;rport Call-ID: 5615f77f064f0558 CSeq: 1 INVITE Contact: Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 3004w stamp 16863 Content-Length: 265 v=0 o=- 10621336 10621370 IN IP4 10.0.0.100 s=eyeBeam c=IN IP4 10.0.0.100 t=0 0 m=audio 9170 RTP/AVP 100 6 0 8 3 18 5 101 a=alt:1 1 : 97678153 DB683B18 10.0.0.100 9170 a=fmtp:101 0-15 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv --- (12 headers 11 lines)--- Sending to 67.84.215.54 : 9168 (NAT) Using INVITE request as basis request - 5615f77f064f0558 Reliably Transmitting (NAT) to 67.84.215.54:9168: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.0.0.100:9168;branch=z9hG4bK-d87543-130614794-1--d87543-;received=67.84.215.54;rport=9168 From: ;tag=d33b6915 To: ;tag=as3c692f9f Call-ID: 5615f77f064f0558 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1f9ec4a9" Content-Length: 0 --- Scheduling destruction of SIP dialog '5615f77f064f0558' in 32000 ms (Method: INVITE) Found user 'RemoteBrian' bmy*CLI> <-- SIP read from 67.84.215.54:9168: ACK sip:12016180234@bmy.gotdns.org SIP/2.0 To: ;tag=as3c692f9f From: ;tag=d33b6915 Via: SIP/2.0/UDP 10.0.0.100:9168;branch=z9hG4bK-d87543-130614794-1--d87543-;rport Call-ID: 5615f77f064f0558 CSeq: 1 ACK Content-Length: 0 --- (7 headers 0 lines)--- bmy*CLI> <-- SIP read from 67.84.215.54:9168: INVITE sip:12016180234@bmy.gotdns.org SIP/2.0 To: From: ;tag=d33b6915 Via: SIP/2.0/UDP 10.0.0.100:9168;branch=z9hG4bK-d87543-377315927-1--d87543-;rport Call-ID: 5615f77f064f0558 CSeq: 2 INVITE Contact: Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Proxy-Authorization: Digest username="RemoteBrian",realm="asterisk",nonce="1f9ec4a9",uri="sip:12016180234@bmy.gotdns.org",response="56a8d5af038526da41fb51b4108811eb",algorithm=MD5 User-Agent: eyeBeam release 3004w stamp 16863 Content-Length: 265 v=0 o=- 10621336 10621370 IN IP4 10.0.0.100 s=eyeBeam c=IN IP4 10.0.0.100 t=0 0 m=audio 9170 RTP/AVP 100 6 0 8 3 18 5 101 a=alt:1 1 : 97678153 DB683B18 10.0.0.100 9170 a=fmtp:101 0-15 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv --- (13 headers 11 lines)--- Sending to 67.84.215.54 : 9168 (NAT) Using INVITE request as basis request - 5615f77f064f0558 Found user 'RemoteBrian' Found RTP audio format 100 Found RTP audio format 6 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 5 Found RTP audio format 101 Peer audio RTP is at port 10.0.0.100:9170 Found description format speex for ID 100 Found description format telephone-event for ID 101 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x32e (gsm|ulaw|alaw|adpcm|g729|speex)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.0.0.100:9170 Peer video RTP is at port 10.0.0.100:54710 Looking for 12016180234 in superuser (domain bmy.gotdns.org) list_route: hop: Transmitting (NAT) to 67.84.215.54:9168: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.100:9168;branch=z9hG4bK-d87543-377315927-1--d87543-;received=67.84.215.54;rport=9168 From: ;tag=d33b6915 To: Call-ID: 5615f77f064f0558 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- -- Executing [12016180234@superuser:1] SetCallerID("SIP/RemoteBrian-0820e840", "<2016180234>") in new stack -- Executing [12016180234@superuser:2] Dial("SIP/RemoteBrian-0820e840", "iax2/espike@nufone/12016180234") in new stack -- Called espike@nufone/12016180234 -- Call accepted by 66.225.202.80 (format ulaw) -- Format for call is ulaw Audio is at 127.0.0.1 port 13202 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Transmitting (NAT) to 67.84.215.54:9168: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.0.0.100:9168;branch=z9hG4bK-d87543-377315927-1--d87543-;received=67.84.215.54;rport=9168 From: ;tag=d33b6915 To: ;tag=as06d5e6bb Call-ID: 5615f77f064f0558 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 267 v=0 o=root 8032 8032 IN IP4 127.0.0.1 s=session c=IN IP4 127.0.0.1 t=0 0 m=audio 13202 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:3 GSM/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv --- -- IAX2/nufone-5 is making progress passing it to SIP/RemoteBrian-0820e840 -- IAX2/nufone-5 is making progress passing it to SIP/RemoteBrian-0820e840 bmy*CLI> <-- SIP read from 67.84.215.54:9168: --- (0 headers 0 lines) Nat keepalive --- -- IAX2/nufone-5 answered SIP/RemoteBrian-0820e840 Audio is at 127.0.0.1 port 13202 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 67.84.215.54:9168: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.100:9168;branch=z9hG4bK-d87543-377315927-1--d87543-;received=67.84.215.54;rport=9168 From: ;tag=d33b6915 To: ;tag=as06d5e6bb Call-ID: 5615f77f064f0558 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 267 v=0 o=root 8032 8033 IN IP4 127.0.0.1 s=session c=IN IP4 127.0.0.1 t=0 0 m=audio 13202 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:3 GSM/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv --- Retransmitting #1 (NAT) to 67.84.215.54:9168: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.100:9168;branch=z9hG4bK-d87543-377315927-1--d87543-;received=67.84.215.54;rport=9168 From: ;tag=d33b6915 To: ;tag=as06d5e6bb Call-ID: 5615f77f064f0558 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 267 v=0 o=root 8032 8033 IN IP4 127.0.0.1 s=session c=IN IP4 127.0.0.1 t=0 0 m=audio 13202 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:3 GSM/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv --- Retransmitting #2 (NAT) to 67.84.215.54:9168: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.100:9168;branch=z9hG4bK-d87543-377315927-1--d87543-;received=67.84.215.54;rport=9168 From: ;tag=d33b6915 To: ;tag=as06d5e6bb Call-ID: 5615f77f064f0558 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 267 v=0 o=root 8032 8033 IN IP4 127.0.0.1 s=session c=IN IP4 127.0.0.1 t=0 0 m=audio 13202 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:3 GSM/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv --- Retransmitting #3 (NAT) to 67.84.215.54:9168: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.100:9168;branch=z9hG4bK-d87543-377315927-1--d87543-;received=67.84.215.54;rport=9168 From: ;tag=d33b6915 To: ;tag=as06d5e6bb Call-ID: 5615f77f064f0558 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 267 v=0 o=root 8032 8033 IN IP4 127.0.0.1 s=session c=IN IP4 127.0.0.1 t=0 0 m=audio 13202 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:3 GSM/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv --- bmy*CLI> <-- SIP read from 67.84.215.54:9168: --- (0 headers 0 lines) Nat keepalive --- -- Hungup 'IAX2/nufone-5' == Spawn extension (superuser, 12016180234, 2) exited non-zero on 'SIP/RemoteBrian-0820e840' Scheduling destruction of SIP dialog '5615f77f064f0558' in 32000 ms (Method: INVITE) Retransmitting #4 (NAT) to 67.84.215.54:9168: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.100:9168;branch=z9hG4bK-d87543-377315927-1--d87543-;received=67.84.215.54;rport=9168 From: ;tag=d33b6915 To: ;tag=as06d5e6bb Call-ID: 5615f77f064f0558 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 267 v=0 o=root 8032 8033 IN IP4 127.0.0.1 s=session c=IN IP4 127.0.0.1 t=0 0 m=audio 13202 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:3 GSM/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv --- Retransmitting #5 (NAT) to 67.84.215.54:9168: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.100:9168;branch=z9hG4bK-d87543-377315927-1--d87543-;received=67.84.215.54;rport=9168 From: ;tag=d33b6915 To: ;tag=as06d5e6bb Call-ID: 5615f77f064f0558 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 267 v=0 o=root 8032 8033 IN IP4 127.0.0.1 s=session c=IN IP4 127.0.0.1 t=0 0 m=audio 13202 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:3 GSM/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv --- bmy*CLI> <-- SIP read from 67.84.215.54:9168: --- (0 headers 0 lines) Nat keepalive --- bmy*CLI> <-- SIP read from 198.65.166.131:5060: --- (0 headers 0 lines) Nat keepalive --- Retransmitting #6 (NAT) to 67.84.215.54:9168: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.100:9168;branch=z9hG4bK-d87543-377315927-1--d87543-;received=67.84.215.54;rport=9168 From: ;tag=d33b6915 To: ;tag=as06d5e6bb Call-ID: 5615f77f064f0558 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 267 v=0 o=root 8032 8033 IN IP4 127.0.0.1 s=session c=IN IP4 127.0.0.1 t=0 0 m=audio 13202 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:3 GSM/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv --- [Nov 1 04:10:41] WARNING[8157]: chan_sip.c:1847 retrans_pkt: Maximum retries exceeded on transmission 5615f77f064f0558 for seqno 2 (Critical Response) Really destroying SIP dialog '5615f77f064f0558' Method: INVITE bmy*CLI> <-- SIP read from 67.84.215.54:9168: --- (0 headers 0 lines) Nat keepalive --- bmy*CLI> <-- SIP read from 67.84.215.54:9168: --- (0 headers 0 lines) Nat keepalive --- bmy*CLI> <-- SIP read from 67.84.215.54:9168: --- (0 headers 0 lines) Nat keepalive --- bmy*CLI> <-- SIP read from 198.65.166.131:5060: --- (0 headers 0 lines) Nat keepalive --- bmy*CLI> <-- SIP read from 67.84.215.54:9168: --- (0 headers 0 lines) Nat keepalive --- [Nov 1 04:11:11] NOTICE[8157]: chan_sip.c:6853 sip_reregister: -- Re-registration for 17476016765@proxy01.sipphone.com REGISTER 13 headers, 0 lines Reliably Transmitting (NAT) to 198.65.166.131:5060: REGISTER sip:proxy01.sipphone.com SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK2d2dabd5;rport From: ;tag=as0a08979f To: Call-ID: 2f3118f0525176d848a1a5a1250aaa13@proxy01.sipphone.com CSeq: 104 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="17476016765", realm="proxy01.sipphone.com", algorithm=MD5, uri="sip:proxy01.sipphone.com", nonce="4548651e9701dae1e90c3d21edfdf7284d7e4744", response="b615ce6c17abc0f4347c040c8d97120e", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 --- bmy*CLI> <-- SIP read from 198.65.166.131:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK2d2dabd5;rport=5060;received=67.83.110.168 From: ;tag=as0a08979f To: ;tag=21a483426c2cd5d9b85bffe6bba40a2e.c0df Call-ID: 2f3118f0525176d848a1a5a1250aaa13@proxy01.sipphone.com CSeq: 104 REGISTER P-Behind-NAT: Yes Contact: ;expires=120 Content-Length: 0 --- (9 headers 0 lines)--- Scheduling destruction of SIP dialog '2f3118f0525176d848a1a5a1250aaa13@proxy01.sipphone.com' in 32000 ms (Method: REGISTER) [Nov 1 04:11:11] NOTICE[8157]: chan_sip.c:11675 handle_response_register: Outbound Registration: Expiry for proxy01.sipphone.com is 120 sec (Scheduling reregistration in 105 s) bmy*CLI> <-- SIP read from 67.84.215.54:9168: --- (0 headers 0 lines) Nat keepalive --- [Nov 1 04:11:25] WARNING[8144]: chan_iax2.c:3788 iax2_send: No private structure for packet? bmy*CLI> <-- SIP read from 67.84.215.54:9168: --- (0 headers 0 lines) Nat keepalive --- bmy*CLI> <-- SIP read from 198.65.166.131:5060: --- (0 headers 0 lines) Nat keepalive --- bmy*CLI> <-- SIP read from 67.84.215.54:9168: --- (0 headers 0 lines) Nat keepalive --- Really destroying SIP dialog '2f3118f0525176d848a1a5a1250aaa13@proxy01.sipphone.com' Method: REGISTER bmy*CLI> <-- SIP read from 67.84.215.54:9168: --- (0 headers 0 lines) Nat keepalive --- bmy*CLI> <-- SIP read from 67.84.215.54:9168: --- (0 headers 0 lines) Nat keepalive --- bmy*CLI> <-- SIP read from 198.65.166.131:5060: --- (0 headers 0 lines) Nat keepalive --- bmy*CLI> <-- SIP read from 67.84.215.54:9168: --- (0 headers 0 lines) Nat keepalive --- [Nov 1 04:12:05] WARNING[8148]: chan_iax2.c:3788 iax2_send: No private structure for packet? bmy*CLI> <-- SIP read from 67.84.215.54:9168: --- (0 headers 0 lines) Nat keepalive --- bmy*CLI> <-- SIP read from 67.84.215.54:9168: --- (0 headers 0 lines) Nat keepalive --- [Nov 1 04:12:25] WARNING[8141]: chan_iax2.c:3788 iax2_send: No private structure for packet? bmy*CLI> <-- SIP read from 198.65.166.131:5060: --- (0 headers 0 lines) Nat keepalive --- bmy*CLI> <-- SIP read from 67.84.215.54:9168: --- (0 headers 0 lines) Nat keepalive --- bmy*CLI> <-- SIP read from 67.84.215.54:9168: --- (0 headers 0 lines) Nat keepalive --- bmy*CLI> <-- SIP read from 67.84.215.54:9168: --- (0 headers 0 lines) Nat keepalive --- bmy*CLI> <-- SIP read from 198.65.166.131:5060: --- (0 headers 0 lines) Nat keepalive --- [Nov 1 04:12:56] NOTICE[8157]: chan_sip.c:6853 sip_reregister: -- Re-registration for 17476016765@proxy01.sipphone.com REGISTER 13 headers, 0 lines Reliably Transmitting (NAT) to 198.65.166.131:5060: REGISTER sip:proxy01.sipphone.com SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK0d5c90a4;rport From: ;tag=as14d3782b To: Call-ID: 2f3118f0525176d848a1a5a1250aaa13@proxy01.sipphone.com CSeq: 105 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="17476016765", realm="proxy01.sipphone.com", algorithm=MD5, uri="sip:proxy01.sipphone.com", nonce="4548651e9701dae1e90c3d21edfdf7284d7e4744", response="b615ce6c17abc0f4347c040c8d97120e", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0