similar to: a couple of new tutorials

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, 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, AstPligg is a new Digg-like website devoted to * and VoIP news. At the moment, it's in beta stage and very basic - no fancy custom templates. It allows posting new stories, comments on stories, RSS feeds and tags. Still, it can be very useful, as the number of * sites and blogs grows every day, and keeping track of what is hot in the * world is increasingly difficult. Yes,

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 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

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 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 put together a quick note about how to "see" oh323 calls while they are handled by your * box. http://www.oinko.net/astrecipes/index.php?n=89 The article is just a draft with usage examples; I'd love to hear your comments and updates if there is something I got wrong. Thanks l. -- Assum est, versa et manduca.

Regarding this article (1) I have one question to make. What can I do to record the call if the agent makes a transfer using the "flash" button instead of "transfer button" or using blindxfer or atxfer defined in features. conf If the agent makes the transfer with "flash", the comunication between the person who is calling and is already in the queue and the target

Hi, ALL: I install my oh323 channel driver following steps of http://www.oinko.net/astrecipes/index.php?action=artikel&cat=270174&id=10&artlang=en I works my asterisk well before install the chan_oh323.so. But after I do "make install" the oh_323, my asterisk crash and show me the following message (asterisk -vvvvvvc). Does anyone have any idea about it? What's wrong

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

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 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 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 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

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