[asterisk-users] how asterisk work with VoIP trunk?

[virendra bhati]

Hi Steve,

Thanks for reply. Is this method will follow on DID incoming calls too?

I mean when we call on DID then call will come to my server and then I want
to move this call to any SIP extension. But call will not come to extension
just got message *"device not in use". *But device already registered into
asterisk server.

But thanks you clear my concept into Voip Call routing too.

On Thu, Jun 9, 2011 at 12:15 AM, Steve Edwards <asterisk.org at sedwards.com>wrote:

> On Wed, 8 Jun 2011, virendra bhati wrote:
>
>  I have working experience of asterisk with PRI lines. Recently I have took
>> VoIP routes from my provider. My basic issue is that now how asterisk will
>> behave in such case. I mean in PRI call will come as below process
>>
>> PRI - -> Digium Card - -> Dadhi/Zap - ->  Extensions.conf
>>
>> What will be the VoIP calling call flow in Incoming and outgoing calls?
>>
>
> Eth[x] -> sip.conf -> extensions.conf
>
> --
> Thanks in advance,
> -------------------------------------------------------------------------
> Steve Edwards       sedwards at sedwards.com      Voice: +1-760-468-3867 PST
> Newline                                              Fax: +1-760-731-3000
> --
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-- 



-----
Thanks and regards

 Virendra Bhati
+91-9172341457
Asterisk Engineer
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