[asterisk-users] IAX or SIP termination provider that
reaches6421xxxxxxx?
Mojo with Horan & Company, LLC
mojo at horanappraisals.com
Thu Oct 5 17:15:19 MST 2006
Confirmed, I removed the r from my dial command and it connects now.
Thank you for the tip!
Luki wrote:
>> I'm interested if anyone else in the Asterisk list can get
>> through to +1-907-747-8633 via voip
>
> Sure, no problem. A nice friendly female voice tells you the time and
> temp, indeed. The thing is that the call never connects -- that info
> is sent via call progress, so a misconfigured server (i.e. one that
> uses the "r" option in dial() or equivalent) would just give you
> ringing and ringing...
>
> [Sep 21 17:49:45] -- Called 9077478633 at trunks
> [Sep 21 17:49:45] -- SIP/trunks-094da090 is making progress
> passing it to SIP/1001-b7a030f8
> [Sep 21 17:49:48] -- Ringing
> [Sep 21 17:49:48] -- Progress
> [Sep 21 17:49:48] -- Peer audio RTP is at port 1.2.3.4:12345
>
> etc.
>
> --Luki
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--
Mojo <mojo at horanappraisals.com>
Office Manager, Horan & Company, LLC
(907) 747-6666 x112
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