[Asterisk-Users] The same SIP problems...SORRY!

[Stephen Davies]

On Mon, 16 Jun 2003, michelle matis litio wrote:

> to 229.159.241.112:5060
> Retransmitting #5 (no NAT):
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 229.159.241.112:5060 ;branch=z9hG4bK-3a5246f7-
> 8c6b606-10eb
> From: ;tag=0-13c4-3a5246f7-8c6b604-c3a
> To: ;tag=as52ed0a6a
> Call-ID: f93b00-0-13c4-3a5246f7-8c6b602-3765 at 188.208.12.37
> CSeq: 1 INVITE
> User-Agent: Asterisk PBX
> Contact: 
> Content-Type: application/sdp
> Content-Length: 135
> 
> v=0
> o=root 11673 11673 IN IP4 188.208.12.237
> s=session
> c=IN IP4 188.208.12.237
> t=0 0
> =audio 13532 RTP/AVP 0
> a=rtpmap:0 PCMU/8000

Hi,

Its being sent to that IP address, because that is that the
originating SIP device put in its Via header.

Also, your SIP device didn't put any From or To in its INVITE.

Perhaps you could send a sip debug from the start of a SIP call
attempt.

But I'm sure that the trouble is with your SIP Gateway device's
setup.

Steve