[Asterisk-Users] AT&T access code entry by Asterisk

Philipp von Klitzing klitzing at pool.informatik.rwth-aachen.de
Wed Dec 17 06:15:39 MST 2003


> exten => _91NXXXXXXXXX,1,Dial(ZAP/g1/${EXTEN}www5555555,70)
> [...]
> I need to implement this functionality in the dialplan.  Does anyone
> have any suggestions? 

See if you can get this done with the help of a .call file. In that call 
file you call an extension first dials out and then does something like


Recently there was a user in this list that claimed that this was the 
only way for him to get things working - didn't test it myself. Old 
posting below.

Cheers, Philipp

From: Chee Foong 
Subject: [Asterisk-Users] WipeOut - gateway access with pin solution 
Date: Thu, 14 Aug 2003 21:53:04 -0700 

Helo WipeOut,
I have found a solution for sending dtmf after dial.
I use spooling. Take a look at the sample.call file inside asterisk dir. 
You need to edit this file and dump it in /var/spool/asterisk/outgoing. 
Asterisk will precess this file automaticlly
I create the sample.call do something like this:
Channel: OH323/4324324324 #dial the access way
MaxRetries: 3
RetryTime: 60
WaitTime: 30
Context: test-context #after connected to access gateway, proceed to 
context 'test-contet' in extension.conf
Extension: 1
Priority: 1
# set var to be used in extension.conf
SetVar: PINNO=1234
SerVar: NUMTOCALL=123123213123 # actual dest number
My extension(test-extension) is: (in extension.conf)
exten =>  1,1,SendDTMF(${PINNO})
exten =>  1,2, Wait, 3
exten =>  1,3,SendDTMF(${NUMTOCALL})
However, this might not suitable for you, if your user dial in manually. 
My situation works fine cause everyting is automated where calling number 
and called number is inserted into db in advanced.
also, chan_h323 has proplem sending DTMF, chan_oh323 works but sound 
quality is bad.

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