similar to: Call Center sofphone

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

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

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 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 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

Hi, I'm trying to write a small Visual Basic app to throw a popup with CallerIDNum when a call center agent answers a queue call. Does anyone know what is the right manager event to intercept? Thanks Mimmus

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, at my Asterisk box, I have a few of IAX2 phones (configured with alaw/ulaw/gsm codecs, in this order) and a PRI E1 line. In iax.conf I hav: disallow=all allow=alaw allow=ulaw allow=gsm During some incoming call, I read at console: -- Executing Dial("Zap/2-1", "IAX2/215|20|TtwW") in new stack -- Called 215 -- Call accepted by 10.97.1.7 (format ulaw) --

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

Hi, I have this setup: (PSTN E1 PRI) -- Asterisk -- (crosscable) -- Alcatel PBX --- analog phones and a few of VoIP phones directly connected to Asterisk. Calling a number (only one until now!) - an automatic responder (IVR) - from VoIP phones works, from analog phones doesn't work: NOANSWER after a few seconds. I'm using no 'r' in dial options (this caused a problem with an IVR

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

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]

Hi, how can I convert .wav files to .WAV: # file greet.* greet.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz greet.WAV: RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz using 'sox'? Thanks -- Domenico Viggiani

Hi, I'm trying to edit an AMP-derived dialplan: the macro "dial" uses the AGI script "dialparties.agi" to find the extension to call. I'd like to drop this script: does anyone can explain me what is its main job? Thanks -- Domenico Viggiani

I'm using a few GXP-2000 with firmware 1.0.2.13 and everything seems to be working fine. However, there are a couple of issues I'd like to know if are possible: 1) Even though the phone has 4 line appearances, if I am speaking on a line, the phone can no longer receive phone calls. I can manually select another line and make calls, but when Asterisk tries to send a call to it, I

Where can I investigate the origin of brief silences during calls from/to my SIP phone? Only rare pauses of 0.5-1.0 sec when I'm not able to hear anything. Thanks Mimmus

Hi, I'm seeing a lot of 'h' as destination numbers in my CDR logs. Some time ago I solved this problem but now I'm not able to remember anymore. Something related to match-all extension? Any help? Thanks Mimmus