similar to: Asttapi Collect

We i have settup it like this it giveme caller id agent id and date-time on gsm file but i want them to be in folder on every day basis datewise. exten => _1NXXNXXXXXX,2,Set(CALLFILENAME=${ACCOUNTCODE}-${EXTEN}-${TIMESTAMP}) exten => _1NXXNXXXXXX,3,Monitor(gsm,/rec/asterisk/apr07/${CALLFILENAME},mb) Any Idea ? Faisal > ------------------------------ > > Message: 16 >

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 =>

Hello All, I am running mirror of a website i update my mirror using rsync but when i run rsync -avvvvvvzC --timeout=600 --delete --delete-after rsync.voip-info.org::voipmirror voip-info/ I get this error opening tcp connection to rsync.voip-info.org port 873 rsync: failed to connect to rsync.voip-info.org: Connection timed out (110) _exit_cleanup(code=10, file=clientserver.c, line=94):

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, Well David it worked after stopping the iptables.. Now what would be the rule for iptabels to allow rsync. I am not very good with iptables help will be appreciated Thanks again for help. > -------------------------------------------- > Faisal A. Ashraf > Web www.voip.com.pk > Your VoIP Solution Partner > -------------- next part -------------- HTML attachment scrubbed

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, I just upgraded my asterisk to Asterisk 1.4.21.1 I am getting this Notice can any one tell me what i need to see in order to fix this problem. [Jul 18 18:27:08] NOTICE[9779]: rtp.c:1286 ast_rtp_read: Unknown RTP codec 126 received from '0.0.0.0' [Jul 18 18:27:09] NOTICE[9780]: rtp.c:1286 ast_rtp_read: Unknown RTP codec 126 received from '0.0.0.0' -- With Best Regards,

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

Hi, I like my recordings to go to date wise folder i mean to say that for example today is 20070409 so all recordings should go directly to that folder instead of one folder for whole month. and then next day's recordings should go to next date folder. so how can i do that my current monitor context is like... exten => _1NXXNXXXXXX,3,Monitor(gsm,/rec/asterisk/apr07/${CALLFILENAME},mb)

Fasiha Ashraf --- On Sat, 22/8/09, Fasiha Ashraf <feehapk@yahoo.co.in> wrote: From: Fasiha Ashraf <feehapk@yahoo.co.in> Subject: Re: [Xen-users] my bootlog To: "Boris Derzhavets" <bderzhavets@yahoo.com> Date: Saturday, 22 August, 2009, 11:12 AM Please check what wrong here grub.conf        title Fedora (2.6.30-rc6-tip)         root (hd0,6)         kernel

Hi Ashraf, In that case I think you may need to structure the code to first build the list and only at the end supply that to the SpatialLines function, something like test.func <- function(x) { tt <- list() for ( i in ... ) { ... tt[[i]] <- (whatever) } return(SpatialLines(tt)) } Eric On Tue, Sep 26, 2017 at 12:36 PM, Ashraf Afana <asafaneh at