[Asterisk-Users] Call did not go through

2004-02-21 Thread Jim Sneeringer
Title: Call did not go through






Whenever an outside number is dialed, Asterisk says Were sorry. Your call did can not be completed as dialed. Please check the number and dial again or call your attendant to help you. I have tried many configurations, but let me give the simplest: It fails when a local number is dialed and the context contains only the following rule:

[default]

exten = _9NXX,1,Dial(${TRUNK}/${EXTEN:1})

where

TRUNK = Zap/1Zap/2

which are Digium FXO cards.

It works with

 exten =9,1,Dial(${TRUNK}/${EXTEN})

Furthermore, it was working before. To my knowledge, the only thing I changed to make it fail was to shut down the working test system, move it to the actual environment, and make it live. I had been testing with only one of the two CO lines. Maybe I changed something in extension.conf, but if so I dont know what it was.

Incoming and intercom calls work fine.

Can anyone tell me what is wrong? Thanks.

Jim




Re: [Asterisk-Users] Call did not go through

2004-02-21 Thread info-lists
Jim Sneeringer said:
 Whenever an outside number is dialed, Asterisk says We're sorry. Your
 call
 did can not be completed as dialed. Please check the number and dial again
 or call your attendant to help you.  I have tried many configurations,
 but
 let me give the simplest:  It fails when a local number is dialed and the
 context contains only the following rule:

 [default]
 exten = _9NXX,1,Dial(${TRUNK}/${EXTEN:1})

 where

 TRUNK = Zap/1Zap/2

 which are Digium FXO cards.

 It works with

   exten =9,1,Dial(${TRUNK}/${EXTEN})

 Furthermore, it was working before.  To my knowledge, the only thing I
 changed to make it fail was to shut down the working test system, move it
 to
 the actual environment, and make it live.  I had been testing with only
 one
 of the two CO lines.  Maybe I changed something in extension.conf, but if
 so
 I don't know what it was.

 Incoming and intercom calls work fine.

 Can anyone tell me what is wrong?  Thanks.

 Jim


In:  exten = _9NXX,1,Dial(${TRUNK}/${EXTEN:1}) you strip off the
leading 9

in  exten =9,1,Dial(${TRUNK}/${EXTEN})  you do NOT strip the 9 off. 
Could it be that your external line connected to the Digium card is
actually connected to some other system and needs the 9?   (and the
message is actually coming from that system) Connect an analog phone to it
and see what dial string it needs.

Robert
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] Call did not go through

2004-02-21 Thread James Golovich


On Sat, 21 Feb 2004, Jim Sneeringer wrote:

 [default]
 exten = _9NXX,1,Dial(${TRUNK}/${EXTEN:1})
 
 where
 
 TRUNK = Zap/1Zap/2
 
 which are Digium FXO cards.
 
 It works with
 
   exten =9,1,Dial(${TRUNK}/${EXTEN})
 
 Furthermore, it was working before.  To my knowledge, the only thing I
 changed to make it fail was to shut down the working test system, move it to
 the actual environment, and make it live.  I had been testing with only one
 of the two CO lines.  Maybe I changed something in extension.conf, but if so
 I don't know what it was.
 

The message you are getting is not from asterisk, its from you telco.  To
simulate what * is doing in this case, plug a phone into your POTS line
and just pick it up without dialing any digits.

You should be using a zaptel group for this, as Dial'ng the way you
are now won't work properly.

Assuming you are dialing 95551234 this is what would be dialed
Dial(Zap/1Zap/2/5551234)

So it will only actually dial a number when the first Zap channel isn't in
use.

Make sure you have a group set in /etc/asterisk/zapata.conf before both of
your channel definitions.  Like this:

group = 2
channel = 1
channel = 2

Then change your TRUNK to:
TRUNK=Zap/g2

James

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users