similar to: Asterisk GUI Client Released!!!

Hello, Just confirmed this on my end, because of the massive changes that have been made to callerID handling in asterisk 1.0.5 many of the features of the astGUIclient suite will not work on this new version. The latest stable version recommended is Asterisk 1.0.3. We will work on trying to find ways around the new callerID rules that the asterisk developers have put in place and hope to have

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.

I just got a Polycom soundpoint and I set it up using the phone and web based admin. I cant seem to figure out the config files and they are confusing me greatly and I dont have time for it :) Some things are odd, like on every reboot it seems the volume I set is reset? is there any way to fix that. And the ringer seems low. - Even all the way up Anyone willing to point out a good asterisk

For an inbound call center with 4 T1s and 30-50 agents on you would do just fine with a single, one-processor machine. We have handled more than this on a single P4 server although we use astGUIclient instead of Asterisk queues, but the load is very similar. I would recommend a Sangoma Quad T1 card because they are about 30% more efficient than Digium T1 cards. When you say that you need to

Hello, You can do either, you can send multiple Originate actions in a long line without waiting for a response back(although the responses do usually come back very fast) or you can open multiple connections using each one to Originate a new call. We use the multiple connection method in the astGUIclient suite because if you get a pause or lag in Manager output on a single connection(which does

I have benn having trouble with the Monitor Command. Basically any time that I send a call into a MeetMe room I am only able to monitor half of the conversation. File-in is recorded with the incoming voice but file-out does NOT record anything. I have tried this with both the b and m option as well as without any options to the MeetMe command. Also the Monitor correctly records both sides of the

I have looked at matt's asterisk gui client at sourceforge. I am not a programmer by trade. The documentation there seems to be a bit lacking. Has anyone have the experience in installing the gui client and may perhaps have a how-to document available for sharing. -- David Kwok Tel: 612 99292086 ext 1002 Iaxtel/FWD # 17001813482 ext 1002 -------------- next part -------------- A non-text

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

I just received my first polycom 501, tried my best to follow all of the documentation and configs on the wiki ( though some conflicted in which case i tried both ways ), and at the end of the day can not connect my phone to asterisk. My Questsions: does the wiki information apply to the 501's? The images and config files at freedomphones, are those supposed to work with the 501? Has anyone

Hi All, We are in search of the latest Polycom firmware SIP 1.6.5 and BootROM 3.1.3as per http://www.polycom.com/resource_center/1,,pw-492,00.html Can someone help? We have legitimately obtained these phones but even our official distributor can't get their hands on updated firmware. The only thing we have found is http://www.freedomphones.net/polycom/files/?M=A which has only old versions.

I am getting to my wits end with these phones (and so is my boss). I am getting an random echo on these phones and I have an issue opened with Polycom and its been in their research and development department for almost a month with no results. I have noticed that I get a message "RFC3389 support incomplete. Turn off on client if possible" in asterisk. I have researched this and made

Hi, I was just wondering if anyone knows how predictive dialers detect voicemail and answering machines, and if they could explain to me how that works. Thanks! Brian.

Hello, I have made some changes/improvements/bug fixes to the Asterisk GUI client I have written in Perl/TK and have released a second beta version on sourceforge: http://sourceforge.net/projects/astguiclient/ Here are the screen shots of the client application running on Linux and Windows: http://www.freedomphones.net/astguiclient_linux_0.8.gif

Hello, While I've been searching for SIP hardphones and trying to pick between all of the different features that are available for them I had the idea that we (The Asterisk community) should create a label that hard-phone manufacturers can put on their products that meet basic requirements to function with the Asterisk VOIP system. Or, at the very least, a listing on a website that would

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 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

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 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

The method we use for web popups on incoming calls in the astGUIclient client app that we are working on for release next week is to use AJAX(Javascript + XMLHTTPRequest) It works in Firefox and IE5+ and doesn't require any META refreshes. We've been using this internally for the last month and it works great. MATT--- -----Original Message----- From: Anton Krall

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 * * * *