[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:alex.robar at thegoldfish.net]
   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 < 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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