asterisk1*CLI> sip debug peer providertrunk SIP Debugging Enabled for IP: 209.183.xxx.xxx:5060 asterisk1*CLI> <-- SIP read from 209.183.xxx.xxx:5060: INVITE sip:479242xxxx@72.18.xxx.xxx:5060;transport=udp SIP/2.0 To: From: "LASTNAME,FIRSTNAME";tag=c9cdc102 Via: SIP/2.0/UDP 209.183.xxx.xxx:5060;branch=z9hG4bK-d87543-fddb482d9dc4ff50dbac-1-cHAwMTYwYjYzYWIzMTAwNTVhOGIzYg..-d87543- Call-ID: 7513591450075b58a823083aa4abaf77 CSeq: 369683301 INVITE Contact: "LASTNAME,FIRSTNAME" Expires: 180 Max-Forwards: 69 Content-Type: application/sdp Organization: Supported: 100rel Allow-Events: message-summary, refer, dialog, line-seize, presence Content-Length: 171 v=0 o=- 1107660232 1079319116 IN IP4 209.183.xxx.xxx s=- c=IN IP4 209.183.xxx.xxx t=0 0 m=audio 35442 RTP/AVP 2 0 101 a=ptime:20 a=rtpmap:101 telephone-event/8000 --- (14 headers 8 lines) --- Using INVITE request as basis request - 7513591450075b58a823083aa4abaf77 Sending to 209.183.xxx.xxx : 5060 (NAT) Found peer 'providerTrunk' Found RTP audio format 2 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 209.183.xxx.xxx:35442 Found description format telephone-event Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x0 (nothing), peer - 0x1 (telephone-event), combined - 0x0 (nothing) Looking for 479242xxxx in from-sip-external (domain 72.18.xxx.xxx) list_route: hop: Transmitting (no NAT) to 209.183.xxx.xxx:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 209.183.xxx.xxx:5060;branch=z9hG4bK-d87543-fddb482d9dc4ff50dbac-1-cHAwMTYwYjYzYWIzMTAwNTVhOGIzYg..-d87543-;received=209.183.xxx.xxx From: "LASTNAME,FIRSTNAME";tag=c9cdc102 To: Call-ID: 7513591450075b58a823083aa4abaf77 CSeq: 369683301 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- dialparties.agi: Starting New Dialparties.agi dialparties.agi: Caller ID name is 'LASTNAME,FIRSTNAME' number is '479652xxxx' dialparties.agi: Methodology of ring is 'ringall' Transmitting (no NAT) to 209.183.xxx.xxx:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 209.183.xxx.xxx:5060;branch=z9hG4bK-d87543-fddb482d9dc4ff50dbac-1-cHAwMTYwYjYzYWIzMTAwNTVhOGIzYg..-d87543-;received=209.183.xxx.xxx From: "LASTNAME,FIRSTNAME";tag=c9cdc102 To: ;tag=as55f1d89b Call-ID: 7513591450075b58a823083aa4abaf77 CSeq: 369683301 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- dialparties.agi: Starting New Dialparties.agi dialparties.agi: Caller ID name is 'LASTNAME,FIRSTNAME' number is '479652xxxx' dialparties.agi: Methodology of ring is 'ringall' Transmitting (no NAT) to 209.183.xxx.xxx:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 209.183.xxx.xxx:5060;branch=z9hG4bK-d87543-fddb482d9dc4ff50dbac-1-cHAwMTYwYjYzYWIzMTAwNTVhOGIzYg..-d87543-;received=209.183.xxx.xxx From: "LASTNAME,FIRSTNAME";tag=c9cdc102 To: ;tag=as55f1d89b Call-ID: 7513591450075b58a823083aa4abaf77 CSeq: 369683301 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- We're at 72.18.xxx.xxx port 16758 Adding codec 0x4 (ulaw) to SDP Reliably Transmitting (no NAT) to 209.183.xxx.xxx:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 209.183.xxx.xxx:5060;branch=z9hG4bK-d87543-fddb482d9dc4ff50dbac-1-cHAwMTYwYjYzYWIzMTAwNTVhOGIzYg..-d87543-;received=209.183.xxx.xxx From: "LASTNAME,FIRSTNAME";tag=c9cdc102 To: ;tag=as55f1d89b Call-ID: 7513591450075b58a823083aa4abaf77 CSeq: 369683301 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 158 v=0 o=root 3039 3039 IN IP4 72.18.xxx.xxx s=session c=IN IP4 72.18.xxx.xxx t=0 0 m=audio 16758 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - --- asterisk1*CLI> <-- SIP read from 209.183.xxx.xxx:5060: ACK sip:479242xxxx@72.18.xxx.xxx SIP/2.0 To: ;tag=as55f1d89b From: "LASTNAME,FIRSTNAME";tag=c9cdc102 Via: SIP/2.0/UDP 209.183.xxx.xxx:5060;branch=z9hG4bK-d87543-eaee5b198d7d897e-1-cHA0MmVkMWQxZWE4OWI2MDYyMTQ3Yg..-d87543- Call-ID: 7513591450075b58a823083aa4abaf77 CSeq: 369683301 ACK Contact: "LASTNAME,FIRSTNAME" Max-Forwards: 69 Organization: Allow-Events: message-summary, refer, dialog, line-seize, presence Content-Length: 0 --- (11 headers 0 lines) --- 12 headers, 0 lines Reliably Transmitting (no NAT) to 209.183.xxx.xxx:5060: OPTIONS sip:voip.myprovider.com SIP/2.0 Via: SIP/2.0/UDP 72.18.xxx.xxx:5060;branch=z9hG4bK5d4a93d9;rport From: "Unknown" ;tag=as548d1adb To: Contact: Call-ID: 1c159e7356e093f413c6225415506193@72.18.xxx.xxx CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 22 Feb 2008 16:41:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Retransmitting #1 (no NAT) to 209.183.xxx.xxx:5060: OPTIONS sip:voip.myprovider.com SIP/2.0 Via: SIP/2.0/UDP 72.18.xxx.xxx:5060;branch=z9hG4bK5d4a93d9;rport From: "Unknown" ;tag=as548d1adb To: Contact: Call-ID: 1c159e7356e093f413c6225415506193@72.18.xxx.xxx CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 22 Feb 2008 16:41:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 209.183.xxx.xxx:5060: SIP/2.0 200 OK To: ;tag=metaswitch.providertelecom.net+1+0+6e023f86 From: "Unknown";tag=as548d1adb Via: SIP/2.0/UDP 72.18.xxx.xxx:5060;branch=z9hG4bK5d4a93d9 Call-ID: 1c159e7356e093f413c6225415506193@72.18.xxx.xxx CSeq: 102 OPTIONS Accept: application/sdp, application/simple-message-summary, message/sipfrag, application/isup, application/x-simple-call-service-info, multipart/mixed Accept-Encoding: identity Allow: INVITE, ACK, CANCEL, BYE, REGISTER, OPTIONS, PRACK, UPDATE, SUBSCRIBE, NOTIFY, REFER Organization: Server: DC-SIP/2.0 Supported: 100rel Allow-Events: message-summary, refer, dialog, line-seize, presence Content-Length: 0 --- (14 headers 0 lines) --- Destroying call '1c159e7356e093f413c6225415506193@72.18.xxx.xxx' REGISTER 12 headers, 0 lines Reliably Transmitting (NAT) to 209.183.xxx.xxx:5060: REGISTER sip:voip.myprovider.com SIP/2.0 Via: SIP/2.0/UDP 72.18.xxx.xxx:5060;branch=z9hG4bK2bc2c28a;rport From: ;tag=as18f07382 To: Call-ID: 52cc5f9158fdf085350ac5b341b833f9@127.0.0.1 CSeq: 116 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: Event: registration Content-Length: 0 --- Retransmitting #1 (NAT) to 209.183.xxx.xxx:5060: REGISTER sip:voip.myprovider.com SIP/2.0 Via: SIP/2.0/UDP 72.18.xxx.xxx:5060;branch=z9hG4bK2bc2c28a;rport From: ;tag=as18f07382 To: Call-ID: 52cc5f9158fdf085350ac5b341b833f9@127.0.0.1 CSeq: 116 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: Event: registration Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 209.183.xxx.xxx:5060: SIP/2.0 200 OK To: ;tag=182a807b From: ;tag=as18f07382 Via: SIP/2.0/UDP 72.18.xxx.xxx:5060;branch=z9hG4bK2bc2c28a Call-ID: 52cc5f9158fdf085350ac5b341b833f9@127.0.0.1 CSeq: 116 REGISTER Contact: ;expires=30 Content-Length: 0 --- (8 headers 0 lines) --- Scheduling destruction of call '52cc5f9158fdf085350ac5b341b833f9@127.0.0.1' in 32000 ms Scheduling destruction of call '7513591450075b58a823083aa4abaf77' in 32000 ms set_destination: Parsing for address/port to send to set_destination: set destination to 209.183.xxx.xxx, port 5060 Reliably Transmitting (no NAT) to 209.183.xxx.xxx:5060: BYE sip:7aJA2wDjHwat39SzpfB2OApBBg7IEJLp0roXJgh8w@209.183.xxx.xxx:5060 SIP/2.0 Via: SIP/2.0/UDP 72.18.xxx.xxx:5060;branch=z9hG4bK76008b77;rport From: ;tag=as55f1d89b To: "LASTNAME,FIRSTNAME";tag=c9cdc102 Call-ID: 7513591450075b58a823083aa4abaf77 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 209.183.xxx.xxx:5060: SIP/2.0 200 OK To: "LASTNAME,FIRSTNAME";tag=c9cdc102 From: ;tag=as55f1d89b Via: SIP/2.0/UDP 72.18.xxx.xxx:5060;branch=z9hG4bK76008b77 Call-ID: 7513591450075b58a823083aa4abaf77 CSeq: 102 BYE Contact: "LASTNAME,FIRSTNAME" Organization: Server: DC-SIP/2.0 Supported: 100rel Allow-Events: message-summary, refer, dialog, line-seize, presence Content-Length: 0 --- (12 headers 0 lines) --- Destroying call '7513591450075b58a823083aa4abaf77' asterisk1*CLI> sip no debug