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. Clint -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brent Torrenga Sent: Wednesday, May 24, 2006 12:44 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] RE: SIP TAPI 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 [EMAIL PROTECTED] Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 tel:+1 219 836 8918 x325 fax:+1 219 836 1138 www.torrenga.com _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users