similar to: Inbound call center - reliability \ scalabil ity with queues

We are wanting to move off of our legacy inter-tel phoneswitch and move to VoIP and asterisk. We are looking for a new PBX because the inter-tel switch is too difficult to integrate our existing (and new) software into. We are a technical support center. All our calls currently come in on toll free numbers via T1's, and there are 3 of them. I want to use a media gateway to convert the

Hello, I have finished my basic polishing of the Asterisk GUI client I have been writing in Perl/TK and have released a first beta version on sourceforge: http://sourceforge.net/projects/astguiclient/ I am still working on a user manual for the application, but the code works and we have been using the same basic client for the last month here at my company and it is working just fine. I'm

Hello, We just had some new T1s turned up today to replace others that our contract has run out on and we are now getting more echo on the new T1 lines than we had on the old ones. The T1 type is 24-channel, D4/AMI SF Robbed-bit(the same as the T1s they replaced) The problem is that we are getting echo on about 10% of the calls in and out placed on these new T1s compared to less than 1% with

Hello Everybody, I am working on Fastagi and I am making use of Asterisk-java. But I don't find the class file for SIPPeersAction. Hence I am getting the error message when compiling my java code. ---------------------------------------------------------------------------- ------------------------------------------------ [root@localhost asterisk-java-0.1]#

Hello, I'm trying to sum up current options for doing small (up to 20 agents) inbound-only CC. I've found: astguiclient, maybe there are some other CC solutions? And on the other side: witch is better to have 20 PC w/ softphones or one T1 channel bank and normal phones with hands-free sets. (whitch set whould you recomend) ? kd, -- Krzysztof Drewicz Affordable 2/4 span E1/T1

I have 12 T1's going into 3 servers, 4 in each into "Digium, Inc. Wildcard TE410P Quad-Span togglable E1/T1/J1 card 3.3v (rev 02)" cards. Each "group" of T1's have the primary D on 24 and the secondary D on 96. The first server (ts20) and the last server (ts22) can playback "demo-congrats" fine. The "middle" server (ts21) cannot -- just dead air.

Hi, suppose I have agents waiting on a queue and I configure asterisk to dial out and to forward the call to the first agent enqueued. Asterisk will do it even if the answer to the call is "busy". Is it possible to configure asterisk to detect the busy signal and, in that case, dial another number, without wasting agent's time? Thanks

Hi, Please forgive my terminology, still a bit new to T1s and such. I'm looking for a way to have 5 T1s from a carrier terminate into some type of box (multiplexer?), then be able to plug 7 asterisk servers into that box (each with single port T1 card) and be able to have 2 * servers go down at any given time and not actually have the carrier see that anything has happened. Obviously if a *

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

Hi, we are currently considering different options for rolling out a large scale IP PBX to handle around 3,000 + concurrent calls. Can this be done with Asterisk? Has it been done before? I really would like an input on this. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL:

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-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

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 would like to create an application where a file is played repeatedly in the background while two parties are having a conversation. Does anyone know of a way to achieve this? I have been looking into the ManagerAPI to redirect the call to a meetme room. Then I try to introduce a third party that will just play the file continuously, but with no luck. Is there a different/simpler

Pardon my ignorance, but I was hoping someone could clear up something for me. - For a few POTS lines, digium has a single port card for that, or a T1 card to a channel bank. - For 10 or more lines, digium has a T1 or E1 card for that too based on PRI channels - For 100's to 1000's of lines, I suspect a soft-switch is in order??? A traditional phone company will sell: - POTS lines for

i cannot see caller ID of the call originated from outside zap channel. i hv configured both zapata.conf and extensions.conf. i m right now in india i think asterisk only supports Bellcore enable caller ID. so is it the same bug of BT caller ID problem in UK? or it is the bug of my asterisk configuration? i hv enabled callerID from my TELCO. -------------- next part -------------- An HTML

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

I tried to install astguiclient and it gives me for each follow page: Object not found! Looking into the apache log file I find: [Sun Feb 27 16:18:30 2005] [error] [client 192.168.250.108] File does not exist: /srv/www/htdocs/astguiclient/method=POST, referer: http://vpbx.elmit.com/astguiclient/phone_stats.php?extension=gs102&server_ip=192.168.250.20 I have uncommented the ### If you have

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 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