similar to: Gnudialer

hi we need a predictive dialer which can be used with asterisk software. Is it possible? Bye Owais Bin Zuber --------------------------------- Do you Yahoo!? Yahoo! Search - Find what you’re looking for faster. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040310/a49a4b5c/attachment.htm

It sounds like Naren and company has their own CRM application. They need a predictive dialer that allows third party app integration. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Matt Florell Sent: Monday, March 13, 2006 8:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

Hello everybody, I'm installing an Asterisk 1.2.7.1 with GNUDialer 0.98-puff18. It also has zaptel from CVS. My FXO is an X100P Clone. The agents from GNUDialer log ok, and everything is fine until the GNUDialer makes a call, as soon as it engages (the phone starts to ring) asterisk crashes with these messages: > Channel Zap/1-1 was answered. -- Executing

Hello! I was curious to know if anyone out there had used Asterisk for voice broadcasting. We have a proprietary dialer that can do it, and we are using Global Connect - www.gc1.com <http://www.gc1.com/> - for some voice broadcasting campaigns. However I'd like to see if I could use Asterisk, possibly with Vicidial or gnuDialer, to do voice broadcasting. Essentially I'd have

Hi Tilghman! &nbsp; &gt; Clearly, you missed the point.&nbsp; Since there is a FREE predictive dialer out &gt; there, and your product costs something, you are not the world's cheapest &gt; predictive dialer.&nbsp; I respect your wording and the way you or other people think on the list about difference between cheapest and free predictive dialer. &nbsp; Surely

Does anyone have any experience using asterisk with an external predicitve dialer, like MediaTel? Specifically: The predictive dialer dials out over T1 circuits. It connects to asterisk via amphenol cable from an fxs card in the dialer to asterisk with a tdm2406 fxo card. In the analog world, the dialer dials out through the t1 circuit, and the fxs card is plugged into a 66 block so the

I know this question has been asked a lot before, but please I would like to know from personal experience. I'm looking to use Asterisk in a call center environment, where most of the calls will be outbound calls. They will have at start 100 agents. I have looked at vicidial and looks promising, however I would like to hear from users what they use and how they like it compared to other

Hello, We've released another update to our VICIDIAL/astGUIclient call center suite: 2.0.5 http://astguiclient.sf.net/ The call center suite client applications run on most modern web browsers on almost any GUI-capable operating system, and it includes the VICIDIAL call center suite. This package is free and AGPLv2. This package is geared towards Asterisk installations with SIP,IAX or Zap

Hello, We've released another update to our astGUIclient-VICIDIAL suite: 2.0.4 http://astguiclient.sf.net/ The client suite runs on most modern web browsers on almost any GUI-capable operating system, and it includes the VICIDIAL call center suite and the astGUIclient client-side web app which extends your phone's functionality. This package is free and GPL. (the suite is not an

Hello, We've released another update to our Asterisk GUI Client suite: http://astguiclient.sf.net/ Screen shots: http://astguiclient.sourceforge.net/screenshots.html The client suite runs on both Windows and UNIX and includes a dialer (the suite is not an asterisk configuration tool) In addition to the usual bug fixes, this is mostly an update for the VICIDIAL dialer application.

Hello, We've released another update to our astGUIclient suite: 2.0.2 http://astguiclient.sf.net/ The client suite runs on most modern web browsers on almost any GUI-capable operating system, and it includes the astGUIclient client-side web app which extends your phone's functionality and the VICIDIAL call center suite. This package is free and GPL. (the suite is not an asterisk

Greetings all Grandstream released a new firmware and it seems like the speaker phone problem has been fixed. However we updated to firmware 1.0.1.12<http://1.0.1.12>to fix the echo problem but found other problems were now created. The worst of these new problems is that the whole phone starts degrading, the volume starts getting lower and lower. The ringing starts fading and the calls

Some at 5:34 pm EST DAILY, all my call get disconnect. I tried RE-INSTALLATION, I tried Reinstallation on a virgin HDD, but its same. I tried changing VOIP provider I tried changing Internet Provider..But no help.. What could be the reason ? Here are my enties of crontab : ### recording mixing/compressing/ftping scripts 0,3,6,9,12,15,18,21,24,27,30,33,36,39,42,45,48,51,54,57 * * * *

hello all, I am looking for an implementation of a 10 man call center. low cost license or GPL will be preferred. I will be glad for your help. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160622/8e2e5ab8/attachment.html>

Hello, We've released another update to our astGUIclient suite: 2.0.3 http://astguiclient.sf.net/ The client suite runs on most modern web browsers on almost any GUI-capable operating system, and it includes the astGUIclient client-side web app which extends your phone's functionality and the VICIDIAL call center suite. This package is free and GPL. (the suite is not an asterisk

HI all: Iam planning to use my asterisk box as callcenter?,any one can advice me with the best callcenter open source program based on asterisk . ? Any help will be apreciated. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100307/116f1b75/attachment.htm

I have upgraded the GXP-2000 to the newest firmware 1.0.1.12<http://1.0.1.12>and the phone is much more usable However, I still have two slight sound quality issues: 1) There is static on the line at all times. It is not that noticable to me, but when I make calls out the PSTN the person on the other end hears it. If I use a Cisco ATA with an analog phone and call the same person again

I'm looking into the dialplan specifics: tleilax:~ # tleilax:~ # cat /etc/asterisk/extensions.conf [general] static=yes writeprotect=no [globals] CONSOLE=Console/dsp ; Console interface for demo TRUNK=DAHDI/r1 ; Trunk interface TRUNKX=DAHDI/r2 ; 2nd trunk interface TRUNKIAX=IAX2/ASTtest1:test at 10.10.10.16:4569 ; IAX trunk

Hi, I have 8 E1 PRI Lines and i have 200 phone numbers and 200 channels (25 channels per PRI). is there a utility available in Asterisk to dial out 200 numbers and run a campaign for 200 numbers concurrently and play a mp3 file ? Please suggest/guide Regards Kaushal

Hi all, I am trying to install Vicidial in an existent FreePBX installation (I'm using Xorcom packages for Debian Etch), but I didn't find any documentation, I found only this guide [0], but is for trixbox only, do you think it will work on FreePBX on Etch? [0] http://iptn.org/vicidial/index.html Regards, Diego Quintana Cruz