similar to: Astricon - let's talk call centers?

Hi All, I wonder if any of you has some suggestions on which WebRTC client/softphone to use for a click-to-dial, webpage hosted solution. Any suggestions? Thanks l. -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I have been playing with the AMI quite a bit lately - mostly debugging WombatDialer in production, but that's a different story - and I have been frustrated by the lack of a simple way to interact CLI-like with the AMI itself. So I have decided to write something myself to make my life easier, or at least a bit less miserable. The result is a little webapp that you can use as a sort

Hello all, I created a set of Docker images running Asterisk and exposing AMI / ARI ports that i found to be quite useful for ARI / AMI development and regression. As they are based on Docker with whaleware, adding new configuration files to roll your own dialplan / queues / voicemail etc is pretty easy. And you can run quite a lot on the same box to simulate clusters. There is no SIP / RTP

Hi all, I need some advice - I have been working on originating multiple calls using AMI and then joining them. What I want to do is: - dial call 1 (where the caller is in a "channel" format, like SIp/1234 or Local/1234 at ext) and "park" it somehow - dial call 2 (where again the caller is in channel format) and join it to the previous call. As a requirement, I cannot use the

Hi all, I thought I'd share I script I made (based on some of Leif's works) that lets you download, compile and install Asterisk all in one go; and then removed the dev tools used. We use it quite a bit to provision systems using Ansible, but it is easier than remembering everything every time even if you are using a shell. At the moment I have scripts for Centos 7 and Asterisk 13, but

Hi all, I just prepared a little tutorial on installing Asterisk 13 on CentOS 6.5 64-bit. See http://astrecipes.net/index.php?n=668 Hope you like. :) l. -- Loway - home of QueueMetrics - http://queuemetrics.com Try the WombatDialer auto-dialer @ http://wombatdialer.com

Hello all, next week it's Astricon 10 time, so we thought we'd create something that the community could like and use for free. It's a pretty effective tool if you run a call-center or plan to run one. QueueWiz is the first free web app for interactive, quick and accurate call center sizing, cost and revenue simulation. Insert your data with the intuitive interface, measure traffic

Hi all, I put together a couple of new tutorials on compiling Asterisk 12 with PJSIP on CentOS 6.5 and test-driving ARI on the same box. You can find them at: http://astrecipes.net/index.php?q=AstRecipes/Compiling%20Asterisk%2012%20on%20CentOS%206.5 and http://astrecipes.net/index.php?q=AstRecipes/Getting%20started%20with%20ARI Comments welcome and happy holidays! :) l. -- Loway

Hello list, I am trying to set a custom SIP header on all calls that are made by the app queue because I want to track a certain state at the SIP level. If I use the following code: exten => s,n,SIPAddHeader(X-Unique-ID: ${UNIQUEID}) exten => s,n,Queue(myQueue) this works fine for the FIRST call made from the queue to an agent; but if that call does not go through, it's not repeated

Hello all, I am trying to set up a dynamic channel to be used as an Agent dialer for a queue - you know, trying to replace AgentCallBackLogin for an Asterisk 1.6. I would like to do something like: [myagents] exten => XXX,1,Set(realchan=${DB(myagent/${EXTEN})}) exten => XXX,n,Dial(${realchan},tT,60) This basically fetches the actual channel to be used for dialling and dials it. What I

Hi everyone, after doing the same thing multiple times and struggling to remember how it was done, I have prepared a small tutorial that explains how to save monitored files in different folders per day. This is quite useful becausethe resultingfile system is way more manageable than having maybe 100,000 files all saved in the same folder. You can find the tutorial here:

I got this notion monitor-format = wav49 wav49 presents much louder than regular wav and gsm in my experience -- _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Lenz Emilitri Sent: Friday, January 22, 2010 8:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users]

I agree with these fella's, this is a piss poor way of fixing it. I only know of one call center that used static agents, mostly because they were sold a peice of crap and they had no idea how to use it the other way. I think you will find the majority of call centers are callback centers. This decision has taken Asterisk out of the realm of providing reasonable call center solutions. VIVA

Hi list, I have a client with a strange requirement: putting a noise gate on the Asterisk channel. For those who are not familiar with them, noise gates are used in musical instruments to avoid entering low-level noise into the amp system. What they basically do is, they measure the volume of the channel, and when it's too low they just let the channel close, i.e send perfect silence,

Hello list, I have prepared a small tutorial today that deals with how to avoid Asterisk rebuilding DTMF tones when using it to connect industial appliances that use DTMF. You can find it at: http://astrecipes.net/index.php?n=248 I know it isn't everybody's piece of cake, but I thought somebody could be interested as well :) l. -- Home of QueueMetrics -

Hello list, I have been working lately on a small CDR parsing utility, and would like to do some performance testing on it. I am looking for some - possibly large - real-life Asterisk CDR datasets to run some performance monitoring. Anybody's got some CDRs that can be shared? Thanks in advance, l. -- Loway Research - Home of QueueMetrics http://queuemetrics.com

Hi there, I support a large number of enterprise users who contractually must connect to our support center via a 4G VOIP connection. I simply want to be able to auto detect all poor quality calls in realtme (as they are being made), play a message and drop the call - without user intervention. All decent call quality calls will be allowed through - to be handled by support staff. Its a

Hello list, I have prepared a couple of new tutorials you may find interesting: - Installing an Asterisk 1.4 beta system - at http://astrecipes.net/?n=216 - Installing the Digium's Asterisk GUI for 1.4 - at http://astrecipes.net/?n=217 It's nothing too complex, but you may find them interesting, especially the new Asterisk GUI. Any comment is welcome - the site is a wiki, so feel

Hello list, I have prepared a small recipe on how to compile Asterisk 1.2 beta 1 with a TDM400 card and H.323. You can find it at http://www.oinko.net/astrecipes/index.php?n=102 Any comment / suggestion / modification /bugfix is welcome! I was wondering: is there any way to build a version of Bristuff for 1.2 beta 1? Bye for now, l. -- Loway Research - Home of QueueMetrics

Hello list, after the last time that mpg123 wen ballistic on our production system, we decided to skip mp3 playback altogether and to go for raw files. After half an hour playing with mpg123 and sox parameters in order to translate a mp3 file to a wav file that can be streamed back through * with no need for an mp3 decoder, I thought I'd post the result to the list to avoid wasting