similar to: hardware and network requirements

Was reading the long list of things being deprecated in the RHEL 8 release notes t'other day, and saw my old friend sendmail on the list. I've stuck by her because I've learned how she likes to be stroked to get the best out of her, and quail at the thought of taking on a new MTA and having to learn it all all over again. But I guess I'm going to have to, soon or late, probably

Hi all, i have the following setup PSTN -> routing server (asterisk 1.6.2.11) -> IAX -> callcenter asterisk 1.6.2.9 -> SIP -> agent Does work quit fine - then agent does have the abibility to transfer a call to a third party - the agent can initiate the transfer over a web interface - it does generate a asterisk manager atxfer request... So agent does initiate transfer - call

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

Hi all! I usually keep the panels on my Centos 6.6 (Gnome) so they auto-hide, but t'other day I wanted to keep the top panel visible for a while to make the clock visible for doing some rough timings. so I open the top panel, click properties, and UNcheck the "autohide" checkbox, close the properties menu, and to my surprise, the panel still autohides. Went thru that exercise

Hi, I have a server with 2 TE110P cards. 1 card is plugged to telco line, another card is plugged with a Hicom PBX. I want to send some call to VoIP phones and all other to my PBX. I don't known how to make my dialplan : ===========Extensions.conf========== [incoming_call] exten => 090200000,1,Goto(callcenter,100,1) exten => 022956353,1,Goto(callcenter,100,1) exten =>

Guys. How do those callcenter apps work with Asterisk where a call comes in and * send a URL and some screen popup up based on callerid or something or username or id and shows all the customers info? Anybody done that? What do you need to do that? If you are using ATAs or IP Phones, how do those integrate with the PC so the screen would popup?

Hello We are developing an application to be deployed on E1 lines (inbound and outbound calls) What is the best way to fully test the application if we do not have E1 lines in the development environment? Is there some kind of software tester to test IVR/Callcenter applications virtually?? thanks and best regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I have configured asterisk and a2billing.for inbound i have also configured did and its forwarded to sip extensions. But i want to enable queues with inbound numbers(DID).But i could not find a way to do this in a2billing. I want enable that if some did comes to asterisk/a2billing it should be forwarded to queues not sip extensions and their i want to enable hunting so if one

Hello list! I have mongrel service 0.1.0 working on my current production machine. Upgrading to a new server and also moving to mongrel service 0.3.1 has not worked yet. I am hoping someone will have an idea as to why. I have mongrel installed properly (I think): C:\rails\igacc>gem list --local *** LOCAL GEMS *** ... mongrel (1.0.1) A small fast HTTP library and server that runs

Anyone know a good replacement for phpagi? Unfortunately development stalled long ago and it has not been updated. What is the best solution for AMI and AGI on PHP? Thanks. -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez +52 (55)9116-91161

I am the network administrator at a small (20-30 employee) financial company. We are in the process of moving offices and will be obtaining a VoIP phone system when we do. Right now, it's down to the 3com nbx100 series and *. Having lurked on *-user for a few weeks and having seen the nifty features of asterisk, I'm convinced. The price difference has pretty much sold my superiors.

hi i need an open source callcenter manager system like queuemetrics but opensource any one know any? i prefer to search before start a new one thanks David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I am running a 5 seater inbound call center on 1.0.9-BRIstuffed-0.2.0-RC8h and it's running well. I am now trying to upgrade it to 1.2.4. So I installed 1.2.4 from source and copied all config files from original to the new server. But when a caller lands inside the queue no queue message is getting played. The gsm files are present in proper locations, whcih I am able to play using

hello, I've callcenter and our queue members want to see on their IP phone's display queue's name , from which incoming call was originated, for example "<client's_number> -> Sales". This problem appears when one member can belong to couple queues. Work around would be setting calling name with such information. Maybe there is another way (setting SIP

Hi, I have callcenter running with v 1.2 with AgentCallbackLogin and now trying to move to 1.4 using the example doc, doc/queues-with-callback-members.txt. From what I understand the basic idea in the example is to 1. Authenticate a caller with VMAuthenticate 2. Get his SIP Channel number 3. Use

I was spamassassin update spamassassin.i386 0:3.2.0-1.el4.rf Complete! and then,Require Perl-Package installed [30998] dbg: diag: module not installed: Mail::SPF ('require' failed) [30998] dbg: diag: module installed: Mail::SPF::Query, version 1.999001 [30998] dbg: diag: module installed: IP::Country::Fast, version 604.001 [30998] dbg: diag: module installed: Razor2::Client::Agent,

We're going to be setting up Asterisk at our data center, as well as our call center locations via an optical fiber point to point connection. Is it best to have the servers communicate to eachother via SIP using SER, or just use the Asterisk functions? Thanks, David -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have a small callcenter with 3 agents who login using AgentCallbackLogin. They normally receive calls, but needs to call outside also. When they call outside, though they are busy the "show agents" shows them as available, and calls gets routed to them. How can I make them busy when they call outside. Also they also need to move out for couple of minutes or to send a mails

I'm not really looking for working configurations as much as I am looking for people who can say "This is a solid product and I trust my business to a solution running Asterisk." As far as pre-sales work... Well, tell that to my consultant. I'm quite excited about *. I've got my company sold on it, they just want some reassurance that it's ready for prime-time

Hi! I'm trying to transfer a incomming call from a Queue to another extension. I'm receiving a call from a queue with the AgentCallbackLogin. The queu is as following: Queue(sales|t) Which should allow transfers. So as soon as the call is answered I would like to be able to transfer it When the agent presses the # I get the dialtone but as soon as I press any digit Asterisk tells me