[Asterisk-Users] SIP behind NAT (*sigh*)

[Florian Overkamp]

*talking to myself*

My issues with the ATA are still unresolved. However, I am noticing things 
in the SIP dumps:

>vectra*CLI>
>Sip read: >
>INVITE sip:0534280105 at 217.114.97.249;user=phone SIP/2.0
>Via: SIP/2.0/UDP 130.89.224.240:5060
>From: sip:ata1-1 at 217.114.97.249;tag=2733832243
>To:  <sip:0534280105 at 217.114.97.249;user=phone>
>Call-ID: 855024110 at 130.89.224.240
>CSeq: 1 INVITE
>Contact:  <sip:ata1-1 at 130.89.224.0:5060;transport=udp>
>User-Agent: Cisco ATA  v2.15 ata18x (020927a)
>Expires: 300
>Content-Length: 253
>Content-Type: application/sdp
>
>v=0
>o=ata1-1 33968 33968 IN IP4 130.89.224.240
>s=ATA186 Call
>c=IN IP4 130.89.224.0

Hmm, does my ATA request the sdp goes to 130.89.224.0 instead of 
130.89.224.240 ? Did it read the Via header back wrong ?? Can this be 
causing the issues ? Simply setting a NATIP address makes it go wild and 
bang on the * server like a maniac :-(

>t=0 0
>m=audio 16384 RTP/AVP 0 4 8 101
>a=rtpmap:0 PCMU/8000/1
>a=rtpmap:4 G723/8000/1
>a=rtpmap:8 PCMA/8000/1
>a=rtpmap:101 telephone-event/8000
>a=fmtp:101 0-15
>
>11 headers, 11 lines
>Using latest request as basis request
>Sending to 130.89.224.240 : 5060 (non-NAT)
>Capabilities: us - 14, them - 13, combined - 12
>Non-codec capabilities: us - 1, them - 1, combined - 1
>Reliably Transmitting (NAT):
>SIP/2.0 407 Proxy Authentication Required
>Via: SIP/2.0/UDP 130.89.224.240:5060;received=130.89.224.240
>From: sip:ata1-1 at 217.114.97.249;tag=2733832243
>To: <sip:0534280105 at 217.114.97.249;user=phone>;tag=as7503b8bb
>Call-ID: 855024110 at 130.89.224.240
>CSeq: 1 INVITE
>User-Agent: Asterisk PBX
>Contact:
>Proxy-Authenticate: Digest realm="asterisk", nonce="1e537e10"
>Content-Length: 0
>
>
>  to 130.89.224.240:5060

Florian