[Asterisk-Users] Sip call question

[Skuse, Phil]

There's something that I want to set up in our lab for testing purposes, but
I'm not sure how to do it.

I would like to be able to call an asterisk extension, and then enter a SIP
address using DTMF, and then have asterisk make a SIP transfer to that
address.

For example:

If I dial <extn> followed by 4444*192*168*0*10*5060 I would like to be
transferred to sip://4444@192.168.0.10:5060. But I don't want to have to
register the IP address beforehand in any config files.

Any idea how I would do this? I'm guessing that I either need to collect the
DTMF, format it into a sip address and then somehow get asterisk to dial
that address, or perhaps I can take it from $EXTEN somehow.

Is this possible with the existing apps/scripts/macros, or do I need to
write some new ones?

Phil Skuse <phil.skuse at vicorp.com>
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 UNIX System Administrator, Vicorp Group Limited.   
 Tel  +44 (0)1753 660523  Fax +44 (0)1753 660501
 The Telephony Engine Company http://www.vicorp.com
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