[Asterisk-Users] SIP TAPI

[Clint Sharp]

FYI, I've got a working version of asttapi that will work with Asterisk
1.2 up on my site at http://www.kirkhamsystems.com/asttapi .  It's the
debug build, so it contains some extra code, but that's merely to help
me out if anyone sends in a bug report (which so far out of apparently
80 something downloads, no bug reports yet, I guess it's working well).

Only reason I mention it is that I can't imagine trying to drop down to
SIP level support in asterisk when the asterisk management interface
works so well with asttapi.

Clint

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Brent
Torrenga
Sent: Wednesday, May 24, 2006 10:32 AM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] SIP TAPI

Hello,

Anyone try to use SIP TAPI (http://www.enum.at/index.php?id=479) with
Asterisk?

Pretty nice, pretty simple. I am hung up on something, though, and
google
doesn't specifically address my issue.

The program seems to go to the s extension in the default context of the
sip
user it is configured for. Is there a way to set it to go to an
extension of
the default context? I couldn't figure out how...

Assuming you can't specify an extension within the default context, then
that leads me to believe that a SIP user needs to be created
specifically
for each instance of SIP TAPI. So I tried that. I made a context just
for
this, with the s extension set to immediately dial my real sip phone.
This
works, and Asterisk bridges the call, but that's it. The docs are
sparse,
and I don't think this program was specifically written for Asterisk -
what
am I missing?


Sincerely,

Brent A. Torrenga
brent.torrenga at torrenga.com

Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana 46321-1771

tel:+1 219 836 8918 x325
fax:+1 219 836 1138
www.torrenga.com

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