[Asterisk-Users] Unable to accept incoming PSTN calls

[Alex Robar]

Johnny,

You need to setup an Inbound Route that matches all DIDs and all CIDs. In
FreePBX, click on Inbound Routes, create a new route with blank CID and DID,
and point it where you want it to go. It should work after that.

Alex

On 4/27/06, Johnny Stork <stork at openenterprise.ca> wrote:
>
> Since I am using A at H 2.8 which now uses freePBX, there does not seem to be
> a menu area/settings for "Incoming Calls"?
>
> If you have a similiar setup, or know what the settings should be, could
> you possibly post them? If I were to create a dial group
> to ring all extensions, could that be used in place of "s"?
>
> Thanks kindly
>
> > -----Original Message-----
> > From: Time Bandit [mailto:timebandit001 at gmail.com]
> > Sent: Thursday, April 27, 2006 6:19 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] Unable to accept incoming PSTN calls
> >
> >
> > > [from-pstn]
> > > include => from-pstn-custom ; create this context in
> > extensions_custom.conf to include customizations
> > > include => ext-did
> > > ;exten => fax,1,Goto(ext-fax,in_fax,1)
> > > exten => _.,1,Wait(1)
> > > exten => _.,2,Goto(from-pstn,s,1)
> >
> > Here is what is happening :
> >
> > Your ZAP channels are in the context "from-pstn"
> > Since there is no "s" extension defined, it goes to "_."
> > (which match anything)
> >
> > So, like seen in the log, Asterisk wait a second, then execute
> > "Goto(from-pstr,s,1)" which brings it back to "_.,1". It just loop
> > there until the caller hangup
> >
> > Since you're using A at H, you have to go into AMP (or FreePBX) and click
> > on Setup -> Incoming Calls and define something to do with incoming
> > calls
> >
> > hth
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--
Alex Robar
alex.robar at gmail.com
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