[Asterisk-Users] two-stage dialing

[John Paine]

-----Original Message-----
From:	Ron McMillin [SMTP:sipnow at sbcglobal.net]
Sent:	Saturday, April 03, 2004 6:33 PM
To:	asterisk-users at lists.digium.com
Subject:	Re: [Asterisk-Users] two-stage dialing

Hi,
  This is not gonna work, is it? Is there such thing as 
Dial_but_not_connect_?  I am trying to do the same thing but don't know how 
to accomplish this. If you've or anyone here figured out, please let me 
know.
  Thank you very much,
Ron

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)

Should do it.

Thank you,
Alex Fedorov


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