[*] Getting past Demo-Congrats - Help Appreciated

Sean Walberg sean at ertw.com
Thu Dec 1 13:35:08 CST 2005


Hi, Brent.  Have you looked at 
http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11

It's the pdf of a book that O'Reilly just released, it might be the guided 
approach you're looking for.

As for the extensions.conf, it sounds like you're on the right track with 
your incoming context.  Generally you have a default context that defines 
all your extensions:

[default]

# Voice mail access
exten => 8500,1,VoicemailMain
exten => 8500,n,Hangup
# My extension
exten => 2001,1,Macro(stdexten,2001,H323/${EXTEN}@192.168.1.95)

Then you define a series of contexts that have increasing levels of 
privilege and that include the ones below it:

[local-free]
# include some other contexts that people in this group can use
include => default
include => outbound-fwd
include => outbound-isn
# also define toll free numbers and IAXtel here
exten => _1700NXXXXXX,1,Dial(IAX2/${EXTEN}@iaxtel)
exten => _1888NXXXXXX,1,Dial(IAX2/${EXTEN}@iaxtel)
exten => _1877NXXXXXX,1,Dial(IAX2/${EXTEN}@iaxtel)
exten => _1866NXXXXXX,1,Dial(IAX2/${EXTEN}@iaxtel)
exten => _1800NXXXXXX,1,Dial(IAX2/${EXTEN}@iaxtel)

[local-out]
# These people can dial 9 to get out using the IAX gateway
exten => _9NXXXXXX,1,Dial(IAX2/USERNAME at GW/${EXTEN})
include => local-free

And when you define the soft phone or FXS port, you put it in the 
appropriate context.

For incoming stuff, you can either include the default context:

[inbound-pots]
include => default

or what I do is define some mapping stuff

[inbound-sip]
exten => _X.,1,Macro(remap,${EXTEN},sipextensions)

which basically does some database lookups to figure out what extension 
and context to send the call to, rather than including the default 
context.

HTH,

Sean

-- 
Sean Walberg <sean at ertw.com>                    http://ertw.com 


On Thu, 1 Dec 2005, Brent Hawryluk wrote:

> Thanks very much for your prompt reply Sean.
> 
> At this point, with my boss out of town on business, I'll take a stab at 
> it...
> 
> 
> Voice Mail?
>         - YES
> Conferencing?
>         - PREFERRABLY
> IVR?
>         - WE CURRENTLY HAVE AN EXISTING SYSTEM WHICH LEADS ME TO BELIEVE 
> THIS WOULD SERVE AS A LOW-COST REPLACEMENT
> 
> 
> Anything that can get me on track (in regards to configuration etc) to 
> getting the Analog phone to ring and make calls between the Analog phone 
> plugged into the PBX and my desk phone at another extension, would be 
> fantastic.  In regards to the dialplan, I have been playing around with it a 
> bit in the Extensions.conf file, added some at the bottom in a group named 
> [Incoming] based on an example I've found.  Still I don't know if I should 
> blow away what's above it, most is commented out anyways.  And I'm figuring 
> the conf file acts like an INI, just jumps down to whatever's called, which 
> would only be what falls in the [Incoming] block.
> 
> Thanks and looking forward to your and anyone else's reply.
> 
> Brent
> 
> 
> _______________________________________________
> Asterisk mailing list
> Asterisk at muug.mb.ca
> http://www.muug.mb.ca/mailman/listinfo/asterisk
> 

-- 
Sean Walberg <sean at ertw.com>                    http://ertw.com 


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