asterisk users - Jul 2005 - Dial 9 to PBX to PSTN pattern question

My question: How do I configure AAH via AMP to make a connection through our
legacy PBX to the PSTN?
 
Details:
We're trying out Asterisk through Asterisk @ Home.
 
Our legacy PBX has a modem type dial tone port that we hooked a Digium FXO
to.
 
Now I can dial from the XTEN client on my computer to any legacy PBX
extension.
 
If I connect a regular phone to the modem dial tone port, I can dial 9 to
get an outside line.
 
Replacing the phone line back to the Digium FXO port, I cannot dial 9 and
the phone number to route my call to the PSTN through the legacy PBX.
 
Looking at the AMP (Asterisk Management Portal)=>Outbound Route, I have two
routes created:
PBX=>to several legacy PBX extensions: 250, 270, 280 (these are the dial
patterns)
9_outside=>to the default dial pattern included with AAH: 9|. (that is the
sole dial pattern)
 
I wonder if the digits get dialed too fast to connect to the PSTN? Can I put
a pause in somehow?
 
I can see that Asterisk does grab the outside line.
When I dial, I get the following message, which I think is coming from the
PSTN:
"We're sorry your call did not go through, Will you please hang up and
try
your call again. This is a recording."
Any ideas anyone?
 
Thanks,
--Bill
Phoenix, AZ
 
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Bill Wesson wrote:
> My question: How do I configure AAH via AMP to make a connection 
> through our legacy PBX to the PSTN?
>  
> Details:
> We're trying out Asterisk through Asterisk @ Home.
>  
> Our legacy PBX has a modem type dial tone port that we hooked a Digium 
> FXO to.
>  
> Now I can dial from the XTEN client on my computer to any legacy PBX 
> extension.
>  
> If I connect a regular phone to the modem dial tone port, I can dial 9 
> to get an outside line.
>  
> Replacing the phone line back to the Digium FXO port, I cannot dial 9 
> and the phone number to route my call to the PSTN through the legacy PBX.
>  
> Looking at the AMP (Asterisk Management Portal)=>Outbound Route, I 
> have two routes created:
> PBX=>to several legacy PBX extensions: 250, 270, 280 (these are the 
> dial patterns)
> 9_outside=>to the default dial pattern included with AAH: 9|. (that is 
> the sole dial pattern)
>  
> I wonder if the digits get dialed too fast to connect to the PSTN? Can 
> I put a pause in somehow?
>  
> I can see that Asterisk does grab the outside line.
> When I dial, I get the following message, which I think is coming from 
> the PSTN:
> "We're sorry your call did not go through, Will you please hang up
and
> try your call again. This is a recording."
> Any ideas anyone?
>  
> Thanks,
> --Bill
> Phoenix, AZ
>  
Are you dialing 99?
First 9 gets you to the FXO, second 9 gets you through the PBX

Keep in mind that Asterisk doesn't bother with such small details as 
"listen for dial tone" before dialing so it is also possible that
digits
are getting lost.
Are the digits Asterisk is dialing too short in duration for the PBX to 
understand?
Is the interdigital time wrong for the destination PBX?
Do you have any sort of digit grabber you can place on the output to 
determine what is and is not getting through?
AAH can be tricky when it comes to modifying the dial string.
A wait before dialing can be inserted, but DTMF duration and 
interdigital time will need the source modified.

John Novack

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

  

My question: How do I configure AAH via AMP to make a connection through our
legacy PBX to the PSTN?
<snip> 

--I cannot help with the AMP way but what you need to do is something like
this in extensions.conf

Exten => _9NXXXXXX,1,Dial(zap/X/w9w${EXTEN:1}

That should give a pause before sending your PBX the 9, then add the 9, then
send the NXXXXXX to the PSTN after your PBX has seized the line.

Jason Kawakami
www.optellabs.com
Salt Lake City, UT