similar to: TO: M.G. Ref: Dial using URI(web) or using FORM(web)

Hello! I have an Asterisk@home instalation with 7 users working OK, and I'ld like to implement either a -- Web dial feature, where the user would fill one form field with a phone number and a connection would be created between his extention and the entered number. OR -- Dial using an URI (callto:xxxxx link in a web page), having AstTapi installed and configured in all workstations.

Same problem here: if call come over ISDN PRI and it is for a SIP phone that equals to strong echo situation, at the SIP end. Interestingly this doesn't happen on all calls but it does on 95% of them. Asterisk load at that moment is insignificant - 1 to 2 calls. I have tried with all possible echo cancellers in zconfig.h, with and without MMX, and with and without CFLAGS+=-march=i686 in

Hi all! After much struggling I got my *@home working fine AND making use of two AVMFritz!PCI cards. Really nice ! (kernel 2.4.2x) There's however a silly glitch that's getting on my nerves, and, kind of a newbie that I am to linux, it should be easy to get help : -- "capiinit start" MUST BE run before Asterisk. (any other way makes * not to start because chan_capi

I have this silly problem : When I place a call, being either to an extention or to an outside line, DTMF signals are ignored by Asterisk. This is serious because I can't even transfer calls (#) or park them (#70). When I receive a call there's no such problem. When I recover a call from parking (71) all goes OK too, and so goes call capturing with *8... I already tested dtmfmode=inband,

I would like to hear stories from people using TAPI, CTI or CallerID software with Asterisk. What are you guys using, setup examples, etc. Has anybody sucessfully integrated SugarCRM with Asterisk and how did you do it. Are you running callerid software? Did you stumble into problems like using tapi and callerid software returned both the callerid and calledid? Hope you can help me out with

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

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-----

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

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

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:

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'

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

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

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 =>

We're looking at SugarCRM 5, but they strongly recommend PHP 5.2.4. Looking around I see that a few individuals are maintaining repos for CentOS, but I'd prefer to get it from one of the larger repositories. Anyone out there using http://www.jasonlitka.com/yum-repository/ ? Any idea when PHP 5.2x will be available? --Chris

Hi folks, I was wondering if there's a guide on how to configure sugarCRM Integration with Asterisk. I was looking in google and all i found was about Trixbox, which has sugarcrm integrated by default. Regards, -- Diego Quintana a.k.a. RouterMaN Ingeniero de las Telecomunicaciones Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676 Ext. 1011 FWD # 764839 Ext. 1011

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 =>