Yeah, that sounds about right. I can see advantages and disadvantages
to both. The main advantage I see to AstTapi besides signaling incoming
calls (which I haven't tested on my modified code, I guess I should work
on that) is that once you've setup a user in the Asterisk Management
interface and modified your dial plan accordingly, you're done, you
don't have to add new entries for every instance of AstTapi. That would
be a burden I'd think in a larger installation of SIPTapi with Asterisk.
The nice advantage also to AstTapi is that signaling is ongoing while
the call is in progress, so you can end the call from the TAPI
application. This is a real boon in real CTI setups for callcenters
where the phones might be set to autoanswer incoming calls on a headset,
display information, and the user ends the call.
Seems like there should be a simpler way to do an TAPI interface with
the Asterisk management interface w/o a bunch of UserEvents though. I
think I'll look into that, because it'd be nice if all you had to do was
add the user to the manager.conf and be done. I know I could probably
do that on outbound calls, incoming calls might be a little more
difficult. It could probably be done with some assumptions about
extension length, etc. Sorry, just thinking aloud, but that's probably
where it should go from here.
[mailto:email@example.com] On Behalf Of Brent
Sent: Wednesday, May 24, 2006 12:44 PM
Subject: [Asterisk-Users] RE: SIP TAPI
Crap. Wish I would have seen your setup first. I played with asttapi for
few days, and gave up. My problems were manager related, and you cover
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:
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
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
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
for SIP TAPI here is that it doesn't require manager events to be put
the dial plan - yay!
Clint, in your opinion, do I have the differences between the two
Post by 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.
Brent A. Torrenga
Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana 46321-1771
tel:+1 219 836 8918 x325
fax:+1 219 836 1138
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