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

[OMS]

I just checked again to make sure. I am not seeing anything at all on
gateway on failed calls.
Again 2 out of 5 test calls were failed to reach gateway.



----- Original Message ----- 
From: "Paul Belanger" <pabelanger at codeslingers.ca>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Sent: Tuesday, August 09, 2005 5:33 PM
Subject: Re: [Asterisk-Users] SIP-Trunk problem, Please help!!!


> 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 - - - -
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