[App_rpt-users] Dial plan question

Sun Nov 17 19:52:12 UTC 2013

I think I will, like you say, have to put a series of dial commands with different extension masks to do it. The following should work from what I see in examples but I have not been able to get it to work....

[pbx_server]
switch => IAX2/user:pw at 192.168.0.2/default    

Of course with proper username and pw, ip, and context for the server.

But when I do this I only get the following trace at the server and an incomplete call. Perhaps I am missing how this works. My understanding was that if a call hit this context it would send it to the remote server for processing with its dial plan.

slate*CLI>
    -- Accepted AUTHENTICATED TBD call from 192.168.0.124

Anyone ever used switch?
73 Doug
WA3DSP
http://www.crompton.com/hamradio

From: mike at midnighteng.com
To: doug at crompton.com
Subject: RE: [App_rpt-users] Dial plan question
Date: Sat, 16 Nov 2013 01:15:58 -0700

I use much more than this in routing calls but I think this is what you are after.There are several ways to skin this cat but here is one.

[autopatch]
exten=_NXXNXXNXXX,1,Dial,IAX2/PBX3/${EXTEN}  ;PSTN outside dial on pbx3
exten=_NXXX,1,Dial,IAX2/PBX3/${EXTEN}        ;dial my voicemail/funtions/ivr on pbx3
exten=_NXXNXXNXXX,2,Congestion
Of course, using mine as example, I have already weeded out/routed other types of calls before it gets to this script including stripping off a dialing prefix that was used to separate internal and external calls on the radiopbx. Be aware of "digit timeout" but if you have no long digit descriptions, it will not look for more.
...mike/kb8jnm

-------- Original Message --------

Subject: RE: [App_rpt-users] Dial plan question

From: Doug Crompton <doug at crompton.com>

Date: Sat, November 16, 2013 1:21 am

To: "mike at midnighteng.com" <mike at midnighteng.com>

Cc: "app_rpt-users at ohnosec.org" <app_rpt-users at ohnosec.org>

   Mike,

 Yes, the iax connection works and I can dial 3 digit extensions with the dial plan I showed. I just did not want to have to recreate all of the dial plans I have on the pbx server on the Allstar server. I wanted to wholesale send everything that went in the radio patch there for dial plan analysis. Since some extensions are 3 digits and others are more I was stymied as to how to do that. Possibly SIP would do a better job at this. Early dial is a sip feature I believe. It would not make much difference if I used IAX or SIP on the local circuit from a loading and routing standpoint.

If app_rpt did not send the digits to the dial plan until the radio carrier detect was released this would work. The digits would not be processed until they were all present. Then a dial plan of  _x!  would work. It would just except it as sent and then dial the pbx server with the extension of whatever length it was. Perhaps there is a way to do this???  Maybe we need another option in autopatch to do this!

73 Doug
WA3DSP
http://www.crompton.com/hamradio

From: mike at midnighteng.com
To: doug at crompton.com
Subject: RE: [App_rpt-users] Dial plan question
Date: Fri, 15 Nov 2013 22:04:57 -0700

You must route the incoming digits to the other server.
I used this method for my phone patch. I trunked it to my phone pbx (asterisk) for outgoing calls.You just need to make a iax connection to/from each system and check that it is functional with a route in your dialplan to the other server.As far as early dial ? -not sure if you can achieve what I think you mean by that without a bunch of script but you could start off with a prefix dial for route to the other server. That is how I make phone patches and give myself access to the radiopbx via phonepbx/inbound or sip.
...mike/kb8jnm

  -------- Original Message --------
 Subject: Re: [App_rpt-users] Dial plan question
 From: Doug Crompton <doug at crompton.com>
 Date: Fri, November 15, 2013 11:30 pm
 To: "app_rpt-users at ohnosec.org" <app_rpt-users at ohnosec.org>

  Further info...

What I want to achieve here is called "early dial" when connecting a sip phone to Asterisk. Each digit is sent to the server and if it does not match a dial plan it looks for the next digit until it finally finds a match. Is there a way to do this between servers to achieve what I asked in the prior message?
73 Doug
WA3DSP
http://www.crompton.com/hamradio

From: doug at crompton.com
To: app_rpt-users at ohnosec.org
Date: Fri, 15 Nov 2013 22:04:33 -0500
Subject: [App_rpt-users] Dial plan question

  I have been using Asterisk for years but I just ran into a problem that I am sure there is a way to fix but I can't figure it out.

I am connecting my Allstar server to my PBX asterisk server via IAX2. The connection works fine but I need some help on passing the extension. I want to pass the variable length extension to my pbx server. Extensions could be any combination of 3 digits up to a full phone number length. The actual extensions are processed in the PBX.

Here is the Allstar extension snippet -

[pbx_server]
exten => _xxx,1,Answer
exten => _xxx,n,noop("${EXTEN}")
exten => _xxx,n,Wait(2)
exten => _xxx,n,Dial(IAX2/pbx/${EXTEN})
exten => _xxx,n,Playback(vm-nobodyavail)
exten => ,n,Hangup

in rpt.conf 

61=autopatchup,context=pbx_server,noct=1,farenddisconnect=1,dialtime=20000,quiet=1   ; Autopatch up
0=autopatchdn                           ; Autopatch down

In the example I have put a 3 digit extension.  So if I dial *61522  -  522 is passed to the pbx server. This works fine but I want to pass any length extension. If I put something like  _x!  -  it just hits on the first digit. Maybe I could use waitexten() to capture it and pass to the pbx. Any ideas?

The Allstar Asterisk and pbx Asterisk are obviously different machines on the same network. 

73 Doug
WA3DSP
http://www.crompton.com/hamradio

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