[asterisk-users] PSTN connectivity

Sam Govind govoiper at gmail.com
Fri Sep 30 00:03:22 CDT 2011


Hey Warren I thought that these are the complete CLI logs for one call. It
started like "  == Using SIP RTP CoS mark 5" and from-internal priority-1
..So that seemed legit to me. Yeah I too suspect that dialing rules are not
being matched and thats why Gotoif's are failing.

On Thu, Sep 29, 2011 at 8:15 PM, Warren Selby <wcselby at selbytech.com> wrote:

>
> On Thu, Sep 29, 2011 at 7:51 AM, michael k <michael at inapp.com> wrote:
>
>> Thanks for the update. but how do i resolve this issue ? can you help me
>> please ?
>>
>
> You didn't provide a full CLI trace of the outgoing call, you only supplied
> the hangup portion of the call.  Please try again.
>
> Also, what are the dialing rules like in your country?  You only have
> outbound dial patterns setup to handle North American numbers (8+ NXXNXXXXXX
> or 8+ NXXXXXX).
> The Dial Pattern box in the Outbound Rules box is where you define what
> numbers you want to go out over this trunk.  If you dial a number that
> doesn't match one of these
> patterns, FreePBX is going to look internally for a dial pattern to match
> against, and if it doesn't find one there, it will end the call.
>
>
> --
> Thanks,
> --Warren Selby, dCAP
> http://www.SelbyTech.com <http://www.selbytech.com>
>
>
> --
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