[Asterisk-Users] SIP-Trunk problem, Please help!!!

[Paul Belanger]

Can you see the INVITE if you put up a trace on your gateway 
(209.XXX.XXX.113)?  Asterisk is not getting anything back that is why it 
retransmits 5 times.

PB

OMS wrote:
> INVITE sip:1512484XXX2 at 209.XXX.XXX.113 SIP/2.0
> Via: SIP/2.0/UDP 24.XX.XXX.101:5060;branch=z9hG4bK6adcaf4f
> From: "512538XXX" <sip:512538XXXX at 24.XX.XXX.101>;tag=as5329d8fe
> To: <sip:1512484XXX2 at 209.XXX.XXX.113>
> Contact: <sip:512538XXX at 24.XX.XXX.101>
> Call-ID: 03896a42587e3f973b42daad031ea6a3 at 24.XX.XXX.101
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Date: Tue, 09 Aug 2005 20:36:43 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Content-Type: application/sdp
> Content-Length: 242
> 
> v=0
> o=root 2251 2251 IN IP4 24.XX.XXX.101
> s=session
> c=IN IP4 24.XX.XXX.101
> t=0 0
> m=audio 15202 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -