RE: [Asterisk-Users] Unable to accept incoming PSTN calls

[Johnny Stork]

I actually tried that before but it didnt seem to work. I tried once again and still nothing rings, whether I set the destination to a single extension, or a ring group. But the suggestion from another user below did work, but wont go to voicemail yet when its not answered.

 

[from-pstn]

include => from-pstn-custom ; create this context in

extensions_custom.conf to include customizations

include => ext-did

exten => _.,1,Wait(1)

exten => _.,2,Goto(from-pstn,s,1)

exten => s,1,Answer

exten => s,2,Dial(SIP/100,20)

-----Original Message-----
From: Alex Robar [mailto:[EMAIL PROTECTED]
Sent: Thursday, April 27, 2006 7:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Unable to accept incoming PSTN calls

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 <[EMAIL PROTECTED]> wrote:

Since I am using [EMAIL PROTECTED] 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:[EMAIL PROTECTED] ]
> 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 [EMAIL PROTECTED], 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
[EMAIL PROTECTED]

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