[EMAIL PROTECTED] 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


_______________________________________________
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

Reply via email to