search for: callcent

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...

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 flow is...

...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 => 022956388,1,Goto(callcenter,100,1) exten => 022956355,1,Goto(callcenter,101,1) exten => s,1,Goto(go_to_pbx,200,1) [callcenter] exten => 100,1,Answer exten => 100,2,SetMusicOnHold(default) exten => 100,3,DigitTimeout...

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: http://lists.digium.com/pipermail/asterisk-users/attachments/20070131/3a1d7dd4/attachment.htm

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 th...

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 -------------- A...

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...

Hi, Can some one tell me what is the meaning of all the fields of show queue callcenter? for example in my system it gives: callcenter has 0 calls (max unlimited) in 'roundrobin' strategy (33s holdtime), C:429, A:12, SL:0.0% within 0s How is the holdtime calculated? what is A and SL? Also how can I see which of my zap interfaces are busy currently? I did a zap show ch...

Hi, I run a small inbound callcenter with 3 agents doing techsupport. The agents are logged in via softphone, using agentcallback login. Some times the agents PC running softphone hangs, and they reboot the PC. But * is not aware of this and tries to send calls to the PC, which gets rejected. -- outgoing agentcall, to agen...

.... mongrel_service (0.3.1) Mongrel Native Win32 Service Plugin for Rails (debug build) rails (1.2.2, 1.2.1) Web-application framework with template engine, control-flow layer, and ORM. ... I install the service correctly (I think): C:\rails\igacc>mongrel_rails service::install -N callcenter -p 4001 -e production -c "C:\rails\igacc" -r "C:\rails\igacc" -l "log/mongrel.log" -P "log/mongrel.pid" -t 0 -n 1024 -a "http://localhost:4001" Mongrel service ''callcenter'' installed as ''callcenter''. I attempt...

...a: exten => 900,1,Playback(queue-youarenext) exten => 900,2,Playback(queue-thereare) exten => 900,3,Playback(digits/three) exten => 900,4,Playback(queue-callswaiting) exten => 900,5,Playback(vm-ivr) The queue is invoked by: exten => s,1,Background(Welcome) exten => s,2,Queue(callcenter|tT|||300) extern => s,3,Hangup When I tried exten => s,2,Queue(callcenter|tTr|||300) It was ringing with out music on hold, but again with out any announcement. Queue.conf is: [general] [default] [callcenter] music=default leavewhenempty = yes monitor-format = wav strategy=rrmemory ti...

Hi, As Freelance programmer/consultant I'm looking for project/task of IVR/CTI/CRM/IP-based, my skils are as following, 1 Dialogic-based CTI/IVR software programming 2 Intervoice IVR development 3 Siebel CRM integration and development 4 IBM DirectTalk and WebShpere Voice(VoiceXML,...) 5 IP-based development(VoIP,h323,sip,...) 6 Cisco

Hello List, I just have a query.... is it possible to have 2 or more telephone number mapped to the same E1 line and if so will the TE120P card pick up the last 4 digits of each number - as it is currently doing for the one? -- Kind Regards Etienne Pretorius -------------- next part -------------- An HTML attachment was scrubbed... URL:

...ade 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 Version - 2.6.1 DAHDI Version: 2.6.1 Echo Canceller(s): HWEC Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 02) Channel...

Hi, I am investigating possibility of using asterisk as an call center controller, i.e. Clients phone in, interact with IVR, if IVR is not enough get redirected to human consultant. There should be possibility for supervisors to connect to ongoing conversation. Expected traffic will not exceed 30 concurrent calls. Asterisk box should be connected to Siemens "communication platform"

I'm curious what asterisk solutions there are out there for inbound call centers with multiple DIDs. I'm looking for solutions for a setup where single system may have 1000 DIDs going to it, one for each account. Each account may not get that many calls. Solutions that will all reporting on calls coming into different accounts, call routing for queues based on distribution groups. Like

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

...ghtly wary of the whole open-source thing. They have no way of knowing, for certain, that asterisk is production-quality until they sign the check and find out. I've been asked by my CTO and CEO to get some testimonials and/or case studies of asterisk in production use in small office / small callcenter environments. We'll be having a contractor configure an IVR, a call center with queues, call detail reporting, and a dialplan for our two inbound groups (our callcenter and our normal office traffic). Does anyone have their own success stories and/or have some verifiable customer testimonia...

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

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 nam...