[Asterisk-Users] two-stage dialing

Tony Wasson wasson at azxws.com
Mon Mar 29 10:30:09 MST 2004


albor at ipeya.com wrote:
> 
> I am trying implement two-stage dialing.
> Scenario is following:
> 1. * Dials SIP agent
> 2. SIP agent answer the phone and provide dial tone
> 3. * Sends DTMF string
> 4. "Bridge" channel with calling party
> I thought that something like:
> exten => _2XX,2,Dial_but_not_connect_(SIP/BYEXTENSION,10)
> exten => _2XX,3,Wait,1
> exten => _2XX,4,SendDTMF($DTMF_DIGITS)

You can do "2 stage dialing" with a call file. This may be a hack, but 
it currently works. I feel it is backwards from all the samples I saw 
since it rings the Agent's SIP phone first. From what I see, call files 
  connect the Channel first, then run the Context/Exten/Priority after 
it's picked up.


--------
#Two stage dial call file
#Rings the Channel first the connects the "Extension".
#agent's SIP phone, like SIP/agent007
Channel: SIP/xws108

#Use a context that can dialout.
#Probably whatever the agent's phone is set as
Context: internal

#Phone number to ring....You may need a '9' prefix
#To Asterisk, Extensions don't have to be internal only...
Extension: 5551212

#use '1' unless you know what you're doing
Priority: 1

#If your telephone interface sends CallerID, be sure to set it.
Callerid: 888-555-1212

---------

Make this file and then copy it into /var/spool/asterisk/outgoing. Don't 
try to make the files in that directory, because Asterisk may read it 
before your script is done writing the file. This may make Asterisk mad.

P.S. You can also do this with the Manager Interface, I lifted enough 
perl code to make this work for me using a "Click to Dial" script that 
sucks in the phone number from a web interface.

http://www.azxws.com/asterisk


Tony Wasson




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