similar to: Inbound Callcenter with multiple DIDs

Say I have agents using a softphone like eyebeam that has 6 lines. They log in to the queue. Say there are 3 agents in my queue. 3 calls come in and all three agents are on a call. Now a fourth call comes in. Is it possible to have it setup so that the 4 call rings on line 2 of one of my agents, if they don't get it within the time limit it rings on line 2 of another agent and so on. An

With our current pbx system, a call comes in from the PSTN to the receptionist. She then hits flash, which puts the caller on hold, calls my extension, says "so and so is on the phone for you", I say "ok put him through", she hangs up and I am connected to the caller. With asterisk@home I can it # then the extension to transfer to and it will ring there. But is there a

Hi, we are trying to setup a prototype Asterisk machine for our call center (15-20 users). We are encountering some difficulties in finding a 'good' softphone (SIP/IAX). Suggestion/experience? Is there some product available for Windows with modifiable code? Is there some freelance developer potentially interested in creating custom version for us? Thanks in advance -- Mimmus

On a 55 station install onto a Cox PRI with a TE110P (Polycom 501 phones) a few users are complaiining about echo. According to the users, the echo seems to be phone number dependant. They claim that certain phone numbers have echo while others dont. Are there any tuning parametes like there is for a TDM400 card? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's

I work for a call center and we are looking at using asterisk to have our operators take calls. Our message taking software records all the calls on the operators computers. Right now we use these recording controls from radio shack that plug in between the wall jack and the phone and plug in via a 1/8 inch stereo connector to the mic input on the computer. If I buy an IP phone I can't do

Hi all, Has anyone implemented trunk to trunk forwarding with an asterisk PBX. For the purpose I have in mind its quite important that once the call has been sent onwards to the new desination the lines into the PBX are no longer held. If anyone has UK-specific experience of getting this up and running that would be incredibly useful! Nic

I have been looking at moving from SIP-based DID (Illinois) providers to one that uses the IAX protocol for DIDs. After a search, I've come up with the following: http://connect.voicepulse.com -- $8/month, many rate-centers http://www.iax.cc -- $1.50/month + 0.014/min, many rate-centers Can that be all that there is? I like the pricing plan at iax.cc, because it would allow me to set up

Maintainer: Express Line Date opened: January 17, 2006 Status: Open Value of bounty: $5000.00 Licensing for code: We retain intellectual rights to the underlying source code. We need Asterisk (stable version) to be able to perform a 2B channel transfer for a NI2 B8ZS PRI line. We can't use a channelized T1 at the time for our work. This feature is commonly called a call transfer on analog

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:

Hello all, Anyone expecialized with call center monitoring and reporting solution based on asterisk. A client of us, want to install a call center reporting solution for an asterisk server but I do not know which could be the best tool for that. I need a tool for reporting queue calls, agent calls, and disconnect cause. Any clue will be appreciated. Thanks in advance. VoipCrazy

I have running * without any Digium (or any other) hardware. Now I need to connect analog FAX machine to it. I think that cheapest and easiest way is to buy ATA. Please correct me if I'm wrong. Now, which ATA should I buy? Local dealer sells those four. I can buy something else (if there is any reason for it), but I prefer something of this. One more question, can I plug two lines in any of

Does anyone have any recommendations on how to connect 160 analogue phones to an asterisk PBX? Background information: A client wishes to replace their current PBX with a new VoIP system. Currently they have 2 PRIs. I intent to set up 2 asterisk PBXs with Debian GNU/Linux on raided drives. These drives will be mounted only read-only to recover gracefully from power-cycles. I am considering 2

Hi all, I am going to be building an Asterisk system to replace the current aging (aged) Nortel Meridian system in a travel agency. There is already a voice T-1 in place and currently there are about 20 extensions in use. I would want to move up to about 25 extensions immediately and about 30-35 within the year. I am going to want IVR and voicemail, plus the ability to ring a group of

I have a problem with my Asterisk system. When I use my phone to call my office mailbox I have to end my password with #. (The office do not use Asterisk) " # " is also used as a transfer button on my asterisk, so when I press it I hear my Asterisk trying to transfer the call. Is there any way to change the transfer button or remove it ? Fredrik

I want the ability to setup DIDs in a variety of different remote locations in Canada. There are various providers that have DIDs in major cities, but none that focus on the "smaller" cities. The question is how do I actually setup these DIDs? Thanks, Geoff

Hello All I am trying to answer incoming SIP calls, first, by dialing an extension, thence into voicemail, which works; and secondly by going straight into voice mail which does not. The extension.conf that works is like this; [incomingSIP] exten=>_.,1,Dial,Zap/2|1 exten=>_.,2,Voicemail,u5152 exten=>_.,3,Hangup the extension.conf which does not is like this; [incomingSIP]

I have several toll free numbers that get forwarded to a single local number assigned to a trunkgroup. I've asked the telco to not forward those toll free numbers but to assign them as DIDs to the trunkgroup, so that I can differentiate via DNID. They said that they can't do that. That toll free numbers must forward. I know that I could have them each forward to different local DIDs

I am a Linux sysadmin who has been tasked with developing the phone system for our nonprofit's new US headquarters building. We cannot bring our legacy phone system with us, so I am building this completely from scratch. I have already read "Asterisk: The Future of Telephony" and done a fair amount of googling. I am completely sold on Asterisk, and the new building's

Does anyone know where i can get DIDs in New Zealand. I am look for area code 06. -James --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 / Virus Database: 481 - Release Date: 7/22/2004

Asterisk 1.2.7 Redhat 9 I have DiDs from two different ITSP both set up as IAX2. Each one works when it's the only one in my iax.conf, but when I have them both defined in iax.conf at the same time, only one will work. My iax.conf is provided below. Any ideas how to fix? I'd like to use both DiDs! Thanks, H My iax.conf is below. When I dial the DiD provided by ITSP_B, the other