[asterisk-users] Re: Odd queue issue

[Andrew Joakimsen]

Could you explain which devices have what IP and what is behind NAT between
what?

On 12/4/06, Matt <mhoppes at gmail.com> wrote:
>
> Debug of the sip peer 126 shows:
>
>     -- Called 126
>     -- Agent/9999 is ringing
> Retransmitting #1 (NAT) to 63.174.244.196:5060:
> INVITE sip:126 at 63.174.244.196 SIP/2.0
> Via: SIP/2.0/UDP 63.174.244.175:5060;branch=z9hG4bK598573e4;rport
> From: "Test VoIP Accounts"" <sip:5706016716 at 63.174.244.175>;tag=as1a3a38f5
> To: <sip:126 at 63.174.244.196>
> Contact: <sip:5706016716 at 63.174.244.175>
> Call-ID: 1882bae616cf60be14d2436e73c7026a at 63.174.244.175
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 04 Dec 2006 20:42:26 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Content-Type: application/sdp
> Content-Length: 275
>
> v=0
> o=root 3555 3555 IN IP4 63.174.244.175
> s=session
> c=IN IP4 63.174.244.175
> t=0 0
> m=audio 19720 RTP/AVP 0 97 111 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:97 iLBC/8000
> a=rtpmap:111 G726-32/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
>
> ---
> Retransmitting #2 (NAT) to 63.174.244.196:5060:
> INVITE sip:126 at 63.174.244.196 SIP/2.0
> Via: SIP/2.0/UDP 63.174.244.175:5060;branch=z9hG4bK598573e4;rport
> From: "Test VoIP Accounts"" <sip:5706016716 at 63.174.244.175>;tag=as1a3a38f5
> To: <sip:126 at 63.174.244.196>
> Contact: <sip:5706016716 at 63.174.244.175>
> Call-ID: 1882bae616cf60be14d2436e73c7026a at 63.174.244.175
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 04 Dec 2006 20:42:26 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Content-Type: application/sdp
> Content-Length: 275
>
> v=0
> o=root 3555 3555 IN IP4 63.174.244.175
> s=session
> c=IN IP4 63.174.244.175
> t=0 0
> m=audio 19720 RTP/AVP 0 97 111 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:97 iLBC/8000
> a=rtpmap:111 G726-32/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
>
> ---
> Retransmitting #3 (NAT) to 63.174.244.196:5060:
> INVITE sip:126 at 63.174.244.196 SIP/2.0
> Via: SIP/2.0/UDP 63.174.244.175:5060;branch=z9hG4bK598573e4;rport
> From: "Test VoIP Accounts"" <sip:5706016716 at 63.174.244.175>;tag=as1a3a38f5
> To: <sip:126 at 63.174.244.196>
> Contact: <sip:5706016716 at 63.174.244.175>
> Call-ID: 1882bae616cf60be14d2436e73c7026a at 63.174.244.175
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 04 Dec 2006 20:42:26 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Content-Type: application/sdp
> Content-Length: 275
>
> v=0
> o=root 3555 3555 IN IP4 63.174.244.175
> s=session
> c=IN IP4 63.174.244.175
> t=0 0
> m=audio 19720 RTP/AVP 0 97 111 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:97 iLBC/8000
> a=rtpmap:111 G726-32/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
>
> ---
> Retransmitting #4 (NAT) to 63.174.244.196:5060:
> INVITE sip:126 at 63.174.244.196 SIP/2.0
> Via: SIP/2.0/UDP 63.174.244.175:5060;branch=z9hG4bK598573e4;rport
> From: "Test VoIP Accounts"" <sip:5706016716 at 63.174.244.175>;tag=as1a3a38f5
> To: <sip:126 at 63.174.244.196>
> Contact: <sip:5706016716 at 63.174.244.175>
> Call-ID: 1882bae616cf60be14d2436e73c7026a at 63.174.244.175
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 04 Dec 2006 20:42:26 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Content-Type: application/sdp
> Content-Length: 275
>
> v=0
> o=root 3555 3555 IN IP4 63.174.244.175
> s=session
> c=IN IP4 63.174.244.175
> t=0 0
> m=audio 19720 RTP/AVP 0 97 111 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:97 iLBC/8000
> a=rtpmap:111 G726-32/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
>
> ---
> Retransmitting #5 (NAT) to 63.174.244.196:5060:
> INVITE sip:126 at 63.174.244.196 SIP/2.0
> Via: SIP/2.0/UDP 63.174.244.175:5060;branch=z9hG4bK598573e4;rport
> From: "Test VoIP Accounts"" <sip:5706016716 at 63.174.244.175>;tag=as1a3a38f5
> To: <sip:126 at 63.174.244.196>
> Contact: <sip:5706016716 at 63.174.244.175>
> Call-ID: 1882bae616cf60be14d2436e73c7026a at 63.174.244.175
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 04 Dec 2006 20:42:26 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Content-Type: application/sdp
> Content-Length: 275
>
> v=0
> o=root 3555 3555 IN IP4 63.174.244.175
> s=session
> c=IN IP4 63.174.244.175
> t=0 0
> m=audio 19720 RTP/AVP 0 97 111 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:97 iLBC/8000
> a=rtpmap:111 G726-32/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
>
> What is doing this?
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