similar to: How To Transfer Asterisk Installation to a Different Machine

I am trying to get called back with a DISA dial tone when I call a trigger number. I got it to work almost the way I want, this is the callback context: [callback] exten=> 501,1,Congestion() exten=> 501,2,Hangup() exten =>h,1,System(cp /etc/asterisk/callback.info /var/spool/asterisk/outgoing) exten =>h,2,Hangup() With the above, the call comes into the trigger number, then the call

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 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

I want to essentially transplant my existing Asterisk server to a new machine, and take the old sever out of service. Assuming I install Asterisk on the new machine, does anyone know what files I would have to copy over? What comes to mind are the *.conf files in /etc/asterisk, as well as the voicemail audio files. Anything else? -------------- next part -------------- An HTML attachment was

When using a SIP phone with Asterisk, hitting the # key (pound or hash depending on where in the world you happen to be) tells Asterisk that there are no more digits coming, and to put the call through immediately based on the digits already entered. This is the same functionality as the PSTN (at least in North America). However, DISA just sees the # as another digit, and therefore pressing #

Hi, I need some help on how to manage the "full" log file. It's getting quite large now and I'd like to clear it. Is there any simple command for this or should I just delete the file (need to be sure this won't affect the system). Also - how do I keep the log file from growing so large? Thanks! Regards, Jan

Hi people! I am looking for a h.323 implementation guide for asterisk. I looked in the doc folder of the latest asterisk source distribution and I didn't fund anything acording to this subject. If you guys could give me any advise, I would thank you. Tamer

Hello List. I have a question regarding connecting two asterisk servers. I'm trying to learn how asterisk comunicates from server to server. I already have a server running smoothly now, I'm installing another one to test it along side the actual one. I would like to run different scenarios: 1. Have one of the boxes at a different location outside the LAN and have them communicate. 2.

Hi... I have what is, I am sure, a relatively common & straightforward problem (no, NOT that kind of problem!)... I'm trying to hook two asterisk servers together so I can make a "distributed" PBX. Here's the scenario: [MASTER] is in the office. It has unrestricted access to the internet, and a fixed IP address. It has 3 SIP hardphones configured & working, plus a

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

Thanks for the quick response! I read about logrotate at voip-info.org but I didn't quite understand it. I'm no asterisk/linux expert unfortunately. First of all. What exactly does happen when I run: /usr/sbin/asterisk -rx 'logger rotate' Does it clear the file and create a new one? Can I run this manually without any interruption in the system? And what does the script do? I

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

Greetings, all. I would like to record calls that are entered into queues and I'm not quite sure how to do it. Here's how I'm currently set up: - Call comes in and is placed into Queue #1 (which rings all phones for 15 sec). - If call drops out of this queue, it is placed into Queue #2 (which plays MoH until the call is picked up). I've tinkered with MixMonitor and I have my

What's the difference between user "123" and "devries"? Based on the output here, they seem the same..? tleilax*CLI> tleilax*CLI> sip show users Username Secret Accountcode Def.Context ACL Forcerport 201 password 201 default No Yes 123

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 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:

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

I'm trying to use the Digium suplied G.729 Codec, I have ran the register utility, and got my licenses written to /var/lib/asterisk/licenses, but when a start Asterisk I got the following errors: [Mar 24 12:33:08] NOTICE[4068] codec_g729a.c: G.729 transcoding module version 34, Copyright (C) 1999-2007 Digium, Inc. [Mar 24 12:33:08] NOTICE[4068] codec_g729a.c: This module is supplied under a

Just wondering what web GUI people like for asterisk. I installed asterisk from source and I was looking at possibly installing web GUI for system management. So far freepbx.org looks promising anybody else have any suggestions. Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 roy

I upgraded my server from Fedora Core 4 to Fedora Core 5. I was wondering if anybody else has run into the problem and know's the fix? I recompiled asterisk and if I don't have the /usr/lib/asterisk/modules/codec_g729a.so file in place it works. I use or used to use the licensed G729 Codec from Digium. This is the error message from asterisk -vvg: [app_playback.so] => (Sound File