similar to: Asttapi - what's wrong?

Hello all, Just to update, Instruction's can be found at www.omniis.com/asttapi, including where to download it from. This is update 0.02, this now includes a little feedback from Asterisk so that when click to dial has occurred then it is indicated at the start and the end of the call. Now working on inbound calls. Any question, please send to me. Regards Nick

We're encountering a problem with AstTAPI crashing on Windows 2000 Workstations. The program we're using is called Amicus Attorney, it uses a standard TAPI interface to be able to dial our clients, but on the 2 Windows 2000 workstations we've tried it on it has crashed, no errors or anything. When we select the Asterisk TAPI driver, the whole windows just closes/crashes w/ no apparent

Hi, I evaluated asttapi, it fits our needs to dial with asterisk out of windows applications. But we are working with rdesktop on a windows 2000 terminal-server, 3 Users on 1 Machine Whenever a user Sets up asttapi via outlook the configuration changes for all users. For example when I enter sip/340 as my extension, the other users also have sip/340 in their configuration. Do you know a trick

I've finished a patched version of asttapi that will work with asterisk 1.2. There were fundamental changes to the Asterisk Management interface between 1.0 and 1.2 that broke asttapi. I think my patched version will work on 1.0 and 1.2 branches, but I have no way of testing since I don't have a 1.0 install nor do I want one :). I'm looking for testers, if anyone's willing to

Hi list! I'm trying to setup Digium TE205P E1 card to work with my provider (and Ericsson BP 250 - this comes later). When I execute "pri show span 1" this is what I get. pbxzg1*CLI> zap show status Description Alarms IRQ bpviol CRC4 T2XXP (PCI) Card 0 Span 1 OK 0 0 0 T2XXP (PCI) Card 0 Span 2 UNCONFIGUR 0 0 0 pbxzg1*CLI> pri show span 1 Primary D-channel: 16 Status:

Hi list! When I'm doing transfer, to what context/priority does that call goes? Can it be changed? Is it the same for blind_tr/att_tr/and for transfer that appears when phone replies with - 302 "Moved Temporarily"? The thing is that I'm trying to transfer incoming call from E1 interface back to E1 interface. Transfers will occur when user is going out and sets up all call

I have it running on my Windows 2000 mahcine using STI products and don't have much of a problem. I would guess that it might be something on the workstations instead of the AstTAPI. Also, might be a little faster, easier, cheaper to just upgrade your existing workstations? Just a thought. Feel free to contact me off list if you would like. Thanks, Dustin -----Original Message-----

Had I have been smart originally I would have done this to start. Some rudimentary documentation above and beyond Asttapi 0.10's poor documentation is available along with the download at http://www.kirkhamsystems.com/asttapi. Clint -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kerry Garrison Sent:

Hi, I am 'playing' with asttapi which looks great on a first installation but I must be missing something regarding the source code because I haven't been able to work with it without problems. If you have played with this, you already know that the code to talk to Asterisk is placed in a file called asttapi.tsp which is in your windows/system32 folder. The thing here is, the

Yes, I have the exact same problem. :( -----Original Message----- From: Tomislav Vojvodic [mailto:tomislav@vox-mundi.net] Sent: Thursday, May 11, 2006 5:48 AM To: xytek@hotmail.com; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk TAPI - Outlook click2dial Hey, thanks for your reply.. ;) I'm also using asttapi from website you posted

Hello, We are using AstTAPI to make call?s from Outlook and it woks perfect. Our issue is that it doesn?t work so well with our CRM. Although the TAPI line initialization is successful, AstTAPI doesn?t report the Phone Number and TAPI doesn?t know about our extension. Anybody knows how to solve this? Ours log: 2005/08/02 15:42:32:476 Information 0x50002 in Process 2040 Thread 1784 Location:

Hi, I have been playing around with the latest release of asttapi and I have found the 'hangup' problem already reported to the list here <http://lists.digium.com/pipermail/asterisk-users/2006-May/151260.html> Apparently hangup should be done by making use of UserEvent commands. So I have configured this context for being used when making calls from outlook: [outlook] exten =>

Hi, Did any one have configured Asterisk+Asttapi+Collect Software? Collect is software for Debt collection agencies with PD I am having hard time configuring it with asterisk. I need help. Regards -------------------------------------------- Faisal A. Ashraf -------------- next part -------------- An HTML attachment was scrubbed... URL:

Good day, does anyone have AstTAPI running for incoming calls, and would like to show some examples. My setting right now looks like this: sip.conf -------- [22] type=friend dtmfmode=info username=22 mailbox=22 secret=privat host=dynamic context=privat canreinvite=yes callgroup=1 incominglimit=2 extension.conf -------------- exten => 123,1,noop ;Hint(SIP/22) exten =>

Hi Guys! After 1 week or looking for answers about CTI and Asterisk, I havent been able to find the necessary applications to do what I want to do. Maybe you guys have more insight on this. I tried installing asttapi, and works great! Can make outbound calls from outlook, etc.. Nice work! For incoming calls.. Ive been trying software like ascendis callerid, call alert and identapop pro, but

Does anyone know of a way to make an originate request coming over the management interface (e.g. AstTapi click-to-dial) include the relevant Alert-Info SIP headers to enable the originating phone to auto-answer? I've tried setting up a custom context (see below), but the dial plan is only entered AFTER the originating call is answered, so the SIP header is added to the terminating call leg,

I've been testing the debug version of AstTAPI, which worked for a few calls, then a bit later in the day (and ever since), when the call is hung up, the TAPI client doesn't get notified. Looking at the server logs, The TAPI message that is sent upon hangup, isn't being sent. exten => h,1,UserEvent(TAPI|TAPIEVENT: LINE_CALLSTATE LINECALLSTATE_IDLE) This is in the same context as

Does the SugarCRM included with AAH 1.3 not meet this criteria for you? ------------------------------ Message: 3 Date: Tue, 26 Jul 2005 17:48:08 +0100 From: "JunkMail" <junkmail@segurajuda.dyndns.org> Subject: [Asterisk-Users] Dial using URI(web) or using FORM(web) To: <asterisk-users@lists.digium.com> Message-ID: <009101c59201$cd5cc9c0$0a00a8c0@segurajuda.local>

Hey Nick, what a nice project. This will help us in database and outlook aided dialing. Great One Problem: If you initiate a call via Outlook, * dials the extension, afterwards dials the tapi-requested number and proceeds bridging the two calls together. Thats fine. When you end the call via outlook hangup-button, *-manager gets a hangup-command (seen in verbose mode), but * doesn't really

I am trying to use a the SIP tapi from www.enum.at <http://www.enum.at/> . This works fine from all kinds of applications which support TAPI, like outlook and Dialer Pro. However when making tapi controlled calls, the signaling to and from PSTN seems to fail. I have used the digium hardware ISDN PRI boards, but also a SIP gateway. Both result in a audio message from asterisk