--- (12 headers 20 lines)--- Using INVITE request as basis request - 0bcdee0973f7618803ee258d5426da3e@66.250.69.11 Sending to 66.250.69.11 : 5060 (non-NAT) Found peer 'switch2' Found RTP audio format 10 Found RTP audio format 4 Found RTP audio format 3 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 111 Found RTP audio format 5 Found RTP audio format 7 Found RTP audio format 18 Found RTP audio format 110 Found RTP audio format 97 Found RTP audio format 101 Peer audio RTP is at port 66.250.69.11:16622 Found description format L16 Found description format G723 Found description format GSM Found description format PCMU Found description format PCMA Found description format G726-32 Found description format DVI4 Found description format LPC Found description format G729 Found description format speex Found description format iLBC Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x7ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 8008831849 in incoming Reliably Transmitting (no NAT) to 66.250.69.11:5060: SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 66.250.69.11:5060;branch=z9hG4bK65fc5a1c From: "7542351245" ;tag=as738fe796 To: ;tag=as2c0b43bb Call-ID: 0bcdee0973f7618803ee258d5426da3e@66.250.69.11 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- 11 headers, 0 lines Reliably Transmitting (no NAT) to 64.135.20.204:5060: OPTIONS sip:64.135.20.204 SIP/2.0 Via: SIP/2.0/UDP 64.135.90.5:5060;branch=z9hG4bK778a4220 From: "asterisk" ;tag=as4c3f77ea To: Contact: Call-ID: 7d04ad223bcf19c3128becea003617a8@64.135.90.5 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 31 Aug 2005 00:06:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Length: 0 --- 11 headers, 0 lines Reliably Transmitting (no NAT) to 64.135.20.204:5060: OPTIONS sip:64.135.20.204 SIP/2.0 Via: SIP/2.0/UDP 64.135.90.5:5060;branch=z9hG4bK6076e601 From: "asterisk" ;tag=as413cad04 To: Contact: Call-ID: 42af110744577fb66d3b43ab1bb42bd4@64.135.90.5 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 31 Aug 2005 00:06:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Length: 0 --- taiko*CLI> <-- SIP read from 66.250.69.11:5060: ACK sip:8008831849@64.135.90.5 SIP/2.0 Via: SIP/2.0/UDP 66.250.69.11:5060;branch=z9hG4bK65fc5a1c;rport From: "7542351245" ;tag=as738fe796 To: ;tag=as2c0b43bb Contact: Call-ID: 0bcdee0973f7618803ee258d5426da3e@66.250.69.11 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- (9 headers 0 lines)--- Destroying call '0bcdee0973f7618803ee258d5426da3e@66.250.69.11' Retransmitting #1 (no NAT) to 64.135.20.204:5060: OPTIONS sip:64.135.20.204 SIP/2.0 Via: SIP/2.0/UDP 64.135.90.5:5060;branch=z9hG4bK778a4220 From: "asterisk" ;tag=as4c3f77ea To: Contact: Call-ID: 7d04ad223bcf19c3128becea003617a8@64.135.90.5 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 31 Aug 2005 00:06:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Length: 0 --- Retransmitting #1 (no NAT) to 64.135.20.204:5060: OPTIONS sip:64.135.20.204 SIP/2.0 Via: SIP/2.0/UDP 64.135.90.5:5060;branch=z9hG4bK6076e601 From: "asterisk" ;tag=as413cad04 To: Contact: Call-ID: 42af110744577fb66d3b43ab1bb42bd4@64.135.90.5 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 31 Aug 2005 00:06:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Length: 0 --- 11 headers, 0 lines Reliably Transmitting (no NAT) to 64.135.11.85:5060: OPTIONS sip:64.135.11.85 SIP/2.0 Via: SIP/2.0/UDP 64.135.90.5:5060;branch=z9hG4bK3d0b3fdb From: "asterisk" ;tag=as284686f6 To: Contact: Call-ID: 6cd4126a3f3c7f333921997036644be8@64.135.90.5 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 31 Aug 2005 00:06:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Length: 0 --- taiko*CLI> <-- SIP read from 64.135.11.85:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 64.135.90.5:5060;branch=z9hG4bK3d0b3fdb From: "asterisk" ;tag=as284686f6 To: ;tag=as7a9798d8 Call-ID: 6cd4126a3f3c7f333921997036644be8@64.135.90.5 CSeq: 102 OPTIONS User-Agent: InnerVoIP-GW Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Accept: application/sdp Content-Length: 0 --- (11 headers 0 lines)--- Destroying call '6cd4126a3f3c7f333921997036644be8@64.135.90.5' 11 headers, 0 lines Reliably Transmitting (no NAT) to 64.135.11.85:5060: OPTIONS sip:64.135.11.85 SIP/2.0 Via: SIP/2.0/UDP 64.135.90.5:5060;branch=z9hG4bK23f0da7b From: "asterisk" ;tag=as5ca2a7c7 To: Contact: Call-ID: 35adee4b59dd6c91275b994900765796@64.135.90.5 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 31 Aug 2005 00:06:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Length: 0 --- taiko*CLI> <-- SIP read from 64.135.11.85:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 64.135.90.5:5060;branch=z9hG4bK23f0da7b From: "asterisk" ;tag=as5ca2a7c7 To: ;tag=as29a90f91 Call-ID: 35adee4b59dd6c91275b994900765796@64.135.90.5 CSeq: 102 OPTIONS User-Agent: InnerVoIP-GW Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Accept: application/sdp Content-Length: 0 --- (11 headers 0 lines)--- Destroying call '35adee4b59dd6c91275b994900765796@64.135.90.5' Retransmitting #2 (no NAT) to 64.135.20.204:5060: OPTIONS sip:64.135.20.204 SIP/2.0 Via: SIP/2.0/UDP 64.135.90.5:5060;branch=z9hG4bK778a4220 From: "asterisk" ;tag=as4c3f77ea To: Contact: Call-ID: 7d04ad223bcf19c3128becea003617a8@64.135.90.5 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 31 Aug 2005 00:06:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Length: 0 --- Destroying call '7d04ad223bcf19c3128becea003617a8@64.135.90.5' Retransmitting #2 (no NAT) to 64.135.20.204:5060: OPTIONS sip:64.135.20.204 SIP/2.0 Via: SIP/2.0/UDP 64.135.90.5:5060;branch=z9hG4bK6076e601 From: "asterisk" ;tag=as413cad04 To: Contact: Call-ID: 42af110744577fb66d3b43ab1bb42bd4@64.135.90.5 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 31 Aug 2005 00:06:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Length: 0 --- Destroying call '42af110744577fb66d3b43ab1bb42bd4@64.135.90.5' taiko*CLI> <-- SIP read from 64.135.113.4:5060: --- (0 headers 0 lines) Nat keepalive --- taiko*CLI>