similar to: Couple of new tutorials on asterisk 12 and ARI

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

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

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:

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

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

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 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 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:

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 list, I know it's a bit OT, but for those who will be at the Astricon, we are organizing a very informal meeting (maybe in front of a pint or two) to talk about Asterisk for call-centers. No marketing or anything - just a way to exchange ideas and meet f2f. I created a facebook group to organize it - see https://www.facebook.com/groups/507826572618269/ See you in Atlanta! l. --

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, 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, 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

Hello list, I posted a couple of tutorials lately, maybe someone can benefit from them: The first tutorial explains how to transform your Asterisk call recordings (in WAV or GSM) to lo-fi MP3 to save a lot of space. It's actually pretty easy to implement using a makefile. http://astrecipes.net/index.php?n=294 The other tutorial lets you implement a way to monitor all outgoing traffic

Hello list, I have been researching a bit into the way the queue app works and how different timeouts play together, and have prepared a short tutorial on understanding queue timeouts - see http://www.oinko.net/astrecipes/index.php?n=118 - any suggestion, error found or correction is welcome. While I was at it, I came across a strange bug: imageine you have three callback agents

Hello list, after spending the best part of an afternoon trying to build Asterisk on an old EPIA VIA C3, I thought that writing a tutorial would make life easier for future compilers: http://astrecipes.net/index.php?n=356 I had never compiled Asterisk for a different architecture, and I'm pretty disappointed at how complex it is - building Zaptel, Libpri and Asterisk requires

Hello there, I just wasted some time setting up a Grandstream HT-488 to be used with Asterisk, so I thought I'd share the experience by writing a small tutorial at: http://astrecipes.net/index.php?n=338 Most tutorials I came across were for old versions of the firmware, and I spent too much time trying to figure out what the Forward to VoIP option wanted.... :-( I hope this helps,

Hello all, as everybody else here - I guess - I have been playing with the new Asterisk 1.8 release. So far everything went smoothly - the compilation phase was really straightforward, and I have a box ready for real testing now. I prepared a tutorial out of my experience on how to compile Asterisk 1.8 with iCal, GTalk, SNMP, MySQL, cURL and DAHDI - the usual stuff - so if anybody is interested

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

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]