similar to: AstTAPI - Incoming Calls

I'm Japanese. Sorry,English is not so understood,Please let me question by items. In Asterisk-1.2.1 and 1.2.2,I cannnot understand the operation of Authenticate application's 'j' option. exten => 123,1,Answer() exten => 123,2,Authenticate(789,j) exten => 123,3,Playback(pin-number-accepted) exten => 123,4,SayDigits(111) exten => 123,103,SayDigits(999) In this

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 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

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 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

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

When I try to call from asttapi one number, I get message "No one is available to answer at this time (1:0/0/0)". Immediately after that I try to call the same number from SIP phone (the same one that is used with asttapi) and call goes true. What have I done wrong? This is how it looks on CLI. **** == Parsing '/etc/asterisk/manager.conf': Found == Manager 'tomo'

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:

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

Hello All! I have a strange problem with Cisco 7905. It is forwarding unanswered calls to VoiceMail even thought I have setup it not to. My ring timer on cisco 7905 is 60s, and my ForwardToVMDelay is 3000s. This means that call should never be forwarded to VM! This is true if I call from internal number then this happens on asterisk: -- SIP/104-6073 is ringing -- Nobody picked up in

I have installed asterisk in a 4.11 RELEASE FreeBSD, and we are using two Zoom X5v with SIP and works fine, we can call each other and this is OK ----------extensions.conf---------- [general] static=yes writeprotect=no [sip] exten => 123,1,Dial(SIP/123,20) exten => 123,2,Voicemail(u${EXTEN}) exten => 123,3,Hangup exten => 123,103,Voicemail(b${EXTEN}) exten => 123,104,Hangup

WARNING[<pid>]: CAPI[contr3/123456]/178 already has PBX structure?? WARNING[<pid>]: CAPI[contr3/123456]/178 already has a call record?? WARNING[<pid>]: CDR on channel 'CAPI[contr3/12345]/177' already started WARNING[<pid>]: Thread 1109916592 Blocking 'CAPI[contr3/123456]/178', already blocked by thread 1116277680 in procedure ast_waitfor_nandfds

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

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

Hullo :) Can someone help me untangle a bit of a mess? I'm trying to set up a demo * server to show off how useful it can be to our business (as an IVR system and VoIP backup if our ISDN30s fail). I've not been able to get NT mode working with our InterTel Axxess PBX, so I've resorted to using normal TE mode and working on the basis the people dial one of the ISDN BRI extension

- ActivaTSP can't work with Astmanproxy as Asmanproxy needs to be patched, - Asttapi wouldn't terminate a completed call. Which option would you pick ? Is there any other option (free or commercial) for Outlook click2call ? Best regards ------=_Part_283_12644120.1196417210166 Content-Type: text/html; charset=ISO-8859-1 Content-Transfer-Encoding: 7bit Content-Disposition: inline

Hello list, Can someone please point me out why would a stream like the following only write ONE line (the first) on the given file? Action: login Username: test Secret: 123456 Action: UpdateConfig SrcFilename: voicemail2.conf DstFilename: voicemail2.conf Action-000000: Append Cat-000000: default Var-000000: 127 Value-000000: >5555, Jason Bourne97, jason25 at noCia.gov.do ActionID: