[Asterisk-Users] RE: SIP TAPI

Brent Torrenga lists at torrenga.com
Wed May 24 10:43:32 MST 2006


Clint,

Crap. Wish I would have seen your setup first. I played with asttapi for a
few days, and gave up. My problems were manager related, and you cover those
points well enough on your page.

I was able to get SIP TAPI to work this way:
 - each install of SIP TAPI needs a SIP user in sip.conf.
 - each SIP user made for SIP TAPI needs a context in extensions.conf.
 - each context made for SIP TAPI looks like:

[blah-tapi]
exten => s,1,Dial(SIP/blah)
Include => blah-internal-context

It seems to work great this way. The software is taken out of the loop
immediately after connecting to SIP/blah, thus does not have call state like
ast tapi does. However, I think this also means that you can have an
unlimited number of simultaneous calls, unlike ast tapi. Also, this does not
provide for pop-ups on incoming calls or call progress, whereas ast tapi
does. What I really don't like about my setup is the lack of "outbound"
caller-id on your phone - no way to use the redial button. I guess a plus
for SIP TAPI here is that it doesn't require manager events to be put into
the dial plan - yay!

Clint, in your opinion, do I have the differences between the two programs
summarized correctly?

>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


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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