search for: msampson

...ack to their first call, finish it up and then go back to the second call. I hope that made sense. I'm sure there is a way to get it done, but how flexible is the current queue system in Asterisk with stuff like this? -- Michael Sampson Information Systems Manager Customer Contact Services msampson@yourccsteam.com 952-936-4000

...ueueMetrics and it is a good piece of software, but does not handle the DIDs the way I need. Really any recommendations for software to go with asterisk that inbound call centers are using and find useful would be great. -- Michael Sampson Information Systems Manager Customer Contact Services msampson@yourccsteam.com 952-936-4000

...ller. With asterisk@home I can it # then the extension to transfer to and it will ring there. But is there a simple way to announce the call before you transfer it. If not, does anyone have any good work arounds for this. -- Michael Sampson Information Systems Manager Customer Contact Services msampson@yourccsteam.com 952-936-4000

...a recording control that plugs in to a 2.5 mm headset jack, but it takes batteries so thats not going to work Does anyone else do something similar? Does anyone have any ideas about what producs/setup would work for this. -- Michael Sampson Information Systems Manager Customer Contact Services msampson@yourccsteam.com 952-936-4000

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

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

...telco, its called a release link transfer, or a 2 b channel transfer. I also do this through an analog T-1 with a flash. Does anyone have know what you need to send asterisk to tell it to do this, or if its even possible? -- Michael Sampson Information Systems Manager Customer Contact Services msampson@yourccsteam.com 952-936-4000

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

...tell me phone company to turn on echo canceling. If the echo was on the other end than it would be my problem? Is this right? What exactly should I say to my phone company so they know exactly what I'm talking about? -- Michael Sampson Information Systems Manager Customer Contact Services msampson@yourccsteam.com 952-936-4000

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 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

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 one use on our PBX who has been experiencing seemingly random disconnects. The user is on the same LAN as everyone else, using the same type of phone (79XX loaded with SIP firmware) as everyone else. He had some disconnects a few weeks ago, I suspected the phone, so I swapped his with mine. I have since not had issues with his old phone, however, he has had issues using mine. So, the

hi, Can anybody pls tell me, how to increase the time gap between 2 digits when you transfer a call. ie, the operator answers the call, and presses hash key to transfer, and then enters the extension number, some times, it timeouts too quickly before the operator enters the whole extension number (may be bcos the operator is slow). I tried the following, but it doesn't seems to be helping

I want to set up a dial rule like this 9304752#w9#w+NXXXXXXXXX The point of this is. It will dial into a pbx with the account number 9304752, wait a second, dial 9 to get an outside line, wait a second for the outside line, and then dial the number to be called. When ever I save this in amp anything after the first # disappears. When I try to call out it doesn't do the adding of the