[Voiceglue] out dial in voiceglue

[Corey Coblentz]

If you mean to transfer from within a VXML application, you want the
<transfer> tag. Transfers have to be to destinations supportable by
Asterisk. Doug posted recently indicating that there were some issues with
the <transfer> tag implementation at present, but it is working for me -
make sure you're transferring to the right protocol (SIP/IAX, etc...). 

e.g., <transfer name="xfer"> method="refer" bridge="false"
dest="sip:12345 at 127.0.0.1">
W3C describes the transfer tag in full.

If you mean to originate a call, VXML doesn't support this - CCXML does, but
is not part of the VoiceGlue project. Asterisk can originate calls very
easily.

Good luck!

Corey Coblentz

_____________________________________________
From: voiceglue-bounces at voiceglue.org
[mailto:voiceglue-bounces at voiceglue.org] On Behalf Of Murali Krishna
Sent: December 1, 2008 6:57 AM
To: voiceglue at voiceglue.org
Subject: [Voiceglue] out dial in voiceglue

Hi,

      Can any one tell me how can I configure out dial  of  voice glue
application 
      From my sip phone. Please give the configuration of extensions.conf


Thanks in Advance
A.Murali
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