similar to: monitor outgoing calls in queue / campaings

Hello, I''m getting an error when running functional tests as follows, using Mac OSX 10.4.9, Rails 1.2.3 . When I run: ruby test/functional/campaign_controller_test.rb I get this error: Started E Finished in 0.027402 seconds. 1) Error: test_should_get_icon_data(CampaignControllerTest): Errno::ENOENT: No such file or directory - /usr/src/projects/myproject/

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, 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 =>

I have made 3 tables +------------+ +----------------+ +------------+ |people | |people_campaigns| |campaigns | +------------+ +----------------+ +------------+ |id :int | <----> |people_id :int | /-> |id :int | |name :string| |campaign_id :int| <-/ |name :string| |etc ... | |exported :int |

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, 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

In order solve my incoming caller ID problem, I upgrade the dahdi to version 2.6.1 from version 2.4.x. After upgrade, I found the echo cancellation doesn't working (I'm using Digium AEX800B PCI Express card). I can hear my self talking on the phone. How to solve this? I think I need to recompile dahdi 2.6.1 with OSLEC support? how? [root at callcenter ~]# dahdi_cfg -vvv DAHDI Tools

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

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.

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 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:

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, 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 a small call center with 4 Zap lines and 4 agents. Agents login using sip phones with AgentCallbackLogin. I occasionally gets a complaint that when customers call the call center, after the initial greeting is over the call gets cut after playing the thank you message. I started investigating and found that that happens when the call gets transferred to an agent who is making an

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

Hi i'm planning to migrate a callcenter to asterisk and VOIP, the call center can have up to 25 cuncurrents agents logged in. I'll have some simplty IVR business logic and the some queues. Can a normal server with 1 GB ram 100 GB HDD Pentium 4 3.6 Ghz CPU Ethernet 10/100/1000 Support this? Would you suggest me a particular products? The server and the agents will be in the same LAN,

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

Hello List, our setup: Callcenter IBM Hardware, 1x TE420, 1x xircom analog switch, 4x different cellular providers on the xircom analog port, ~60 agents Debian 5.0.1 (Lenny) Asterisk 1.4.21.2 Debian Package recompiled with additional app_queue segfault fix Zaptel 1.4.11 Debian Package My Problem is I have two channels (Zap/9-1 and Zap/6-1) which have a duration of over 4 hours. I am

I have 4 sipuras 3000 in a small callcenter connected to the PSTN receiving calls and forwarding them to Asterisk and viceversa. In addtiion I have some x100s, linksys FXSs, etc Strange things are happening with the Sipura and Asterisk which I cannot seem to figure out. During off hours at the callcenter, when no one is placing calls, if I place or receive a call with any of the Sipura,