[asterisk-users] PSTN connectivity

[John Novack]

michael k wrote:
> Thanks for the update. but how do i resolve this issue ? can you help me please ?
>
>
Can you PLEASE take this to the FreePBX support group?

It seems obvious to most that therein lies the problem
You are thinking you wish to dial out through the X100, but Asterisk is attempting to dial out on a non existent SIP connection
Something isn't right in your dialplan, created by FreePBX


Also, no echo canceller on the X100 card isn't wise, but you will not realize that until you are able to use it!

John Novack

>
> On Thu, Sep 29, 2011 at 1:00 PM, Sam Govind <govoiper at gmail.com <mailto:govoiper at gmail.com>> wrote:
>
>     Actually its easier. I haven't worked on FreePBX lately so what I remember is here: You've created a Zap trunk. Focus on the Dial Rule - You can keep it empty as well. Then you've created an outbound route its dial-rule is important.
>
>     But the funny thing which I didn't mention before is that you've ZAP defined in FreePBX but actually its DAHDI so I remember they've this cute parameter in amportal.conf which tells FreePBX to convert ZAP into DAHDI.
>
<snip>

-- 

Dog is my Co-pilot

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