similar to: Merge master.csv files

Hi there, Does asterisk keeps the master.csv open between writes? Right now I have 2 asterisk nodes sharing every configuration file (by using a distributed filesystem) except the master.csv files. If asterisk does not keep master.csv file open between writes, then I can share the master.csv file between both nodes right?If not, then any suggestions to merge both master.csv files? Thanks in

Hi, Is it possible to configure Asterisk to fetch for files from the spool directory in different directories? For example, fetch voicemail files in /abc/voicemail and call files in /cde/outgoing ?. And is it possible to configure the filename that Asterisk gives to files, like voicemail files? Thanks, Ricardo

Hi, I have set up asterisk to run as non root, and allow admin users to log in to the server as asterisk, which gives them privileges to edit configs in the asterisk home directory. As for connecting to the console with 'asterisk -r' - this by default does not work as asterisk is owned stored in /usr/sbin/asterisk I am reading that the best way to solve this is to use 'visudo' -

How can we set the CDR Master file to rollover at say 30 Meg and create a new one -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100827/2e98385f/attachment.htm

Hi all I got this solved. Turns out the script WAS executing, but I forgot that apparently you need to follow "cron rules" in any BASH scripts executed via System() from an Asterisk dialplan. E. g. all paths must be fully and absolutely specified, there are no relative path references available. So I changed the the file wireless.sh which was: #!/bin/bash touch wireless-executed to

Hello List, I am getting Asterisk set up. I am going to be installing Debian Linux on a laptop later. I would appreciate some detailed instructions on: (A) What to type into the shell to download and install Asterisk. (B) How to open the configuration files (*.conf) (C) If there is a way that I can change the configuration files remotely (SSH?). Thanks in advance. -- -Shane Blog:

Now that the g729 patents have expired, how do we use g729 in Asterisk? Will Digium be releasing a g729 codec for 'free' use or do we download the 'free' codec off the Internet now that we can use it without moral or legal restrictions? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com

Does anybody have (or know of) a command line application that would: ) Eliminate pops and other random loud noises. ) Trim leading and trailing silence. ) Trim pauses exceeding x milliseconds to y milliseconds. ) Normalize what's left. I know about normalize and have figured out how to trim leading and trailing silence in sox, but I'm looking for more :) Thanks in advance,

Hi all, My name is Ricardo and unfortunately I'm just crawling in this telecomm/asterisk world. So, after reading all day long i still don't understand a few things. :D I'm trying to "develop" a call recorder for a costumer. He has a small call center ( 10 agents ) and want to record all calls. Since he already has everything (ACD only) working perfectly in the PBX and

Here I am expecting to be configured following scenario: User calls : it will play a sound will ask for input DTMF, then call will be given to particular extension for any DTMF entered. But its not working as expected. I have attached the dial plan file. -------------- next part -------------- An HTML attachment was scrubbed... URL:

"Request For Comments on a Feature Suggestion" -- just wondering if others would find this useful. Frequently, when something really doesn't make sense, I like to bump up the console logging by editing logger.conf and changing console = error to console = debug,dtmf,error,event,info,notice,verbose,warning and entering logger reload When done, I

hi all, can anyone give me a guide on any asterisk admin solution / interface for config management, and monitoring? No database use is intended and I prefer open source. Thanks for support. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160516/98f6e448/attachment.html>

I'm trying to find a way to setup a SIP server that will mainly echo back a request from one agent only, my question is do i need to setup any of the other conf files besides extensions.conf and sip.conf? the book says "If you plan on a pure VoIP network, the only real requirement is the asterisk package," , but i'm doubtful about zaptel. can anyone help me out?

Hi Guys, Is it possible to harvest the output of system into a SetVar(variable)? exten => s,n,SetVar(var=system(*asterisk -rx "sip show channels" | grep -c "(ulaw)")* * * *??? any problem with the syntax? * * * * * *Thanks,* * * -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi List I am having trouble running the command siptest:~# asterisk -rx 'dialplan reload' most times it does what I expect and I get a response as below siptest:~# asterisk -rx 'dialplan reload' Dialplan reloaded. every now and then I get no response i.e. siptest:~# asterisk -rx 'dialplan reload' siptest:~# and a "verbose 10" setting shows [Mar

Hi... Im problem is this, i have a asterisk server (FC8 - kernel 2.6.24) a the asterisk version is 1.4.18. If in the machine is all ok, i can stop start the asterisk service no prob, my problem is when in another server (in my case, debian etch 4) using the ssh the stop service is ok, but the start service dosend finalise. Like this: ssh etx9 'sudo /etc/init.d/asterisk stop' operacao at

Hi. I'd like to get access to the CDR's generated by Asterisk (1.4) in real-time from a remote connection coming in on TCP. Basically what I have is a Windows application that is used to process incoming, outgoing and missed call records putting them into a database for some analysing etc. This app can connect to a TCP server and read from this connection the CDR's as they are

Our small company is replacing Cisco CallManager with Asterisk (because we are tired of sending them money) and I am documenting the process as I go on my blog. I am trying to make the notes as easy as possible in hopes that I can ease someone else's pain. Here is the link: http://qvlweb.blogspot.com/2009/03/asterisk-pbx-system-install-01-what-i.html Please feel free to comment on the

Hello(s), I know this might be test book question or one best suited for google but I will take the risk of asking. Here I go. What common routine maintenance tasks do you run on your asterisk box? Thanks James. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090422/8cfeb434/attachment.htm

Does anybody think it would make sense for /etc/init.d/asterisk to run /etc/asterisk/startup.d/*.sh on start like safe_asterisk did? Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP