similar to: 1and1 dedicated servers have been down for a few hours .

You can listen or join the Asterisk Users Conference Fridays at 12:30 PM EDT Today's subject suggestions: FAX capabilities, what's your solution? Multiple asterisk server implimentation: ENUM, DUNDI or even two servers connected Your subjects? Share your ideas, ask your questions! See http://x2z.eu for instructions on how to join or listen

Hello Fellow Asterisk Mailing ListMembers, When I tried to download the latest version of Asterisk this is what I get: http://ftp.digium.com/pub/asterisk/releases/asterisk-1.4.0.tar.gz Opening fileinfo database failed http://ftp.digium.com/pub/asterisk/releases/asterisk-1.4.0.tar.gz Opening fileinfo database failed Where are all the latest Asterisk 1.4.x source files? Thanks in advance, -E

Dear all I have asterisk 1.2 configuration and it is working fine but thing is that i have alread Avaya setup and i have intergrate my Linuxbox asterik with Avaya system avaya already use 4 digit dialplan (1644 example ) and in asterisk i have configure 2 digit dialplan ( 44 example ) now i want to configure 4 digit dialplan in asterik without any change in avaya or asterisk so

As usual, we'll be jawing about any and all asterisk-related subjects with the usual gang and any new people are always welcome, regardless of your level of expertise. You can even come and ask questions, it's guaranteed to be a more pleasant experience than it will be on IRC ;) http://VoipUsersConference.org/topics.php IRC; Freenode.net #voip-users-conference

how to adjust voice volume for sipura 2000 and cisco ata186?

I have read and followed as much as I can find but I am missing something. What I want to do is get as much as I can running from mysql and keep the *.conf files for static things. So I have setup a SIP users/peers table in a mysql database and I have populated it with a few peers. I have configured asterisk addons and from the asterisk CLI I am able to search the sip users / peers tables using

On 7/27/07, Ivo Emanuel Gon?alves <justivo@gmail.com> wrote: > On 7/24/07, Arseny Krasutsky <dtiger@mail.ru> wrote: > > So I can send this binaries to you if you want to place it online. > > I'm sure people need it :-) > > I do not know Jean-Marc's position on this -- it's his project, so I > don't want to meddle -- but if he does not oppose, I

This isn''t Rails or Ruby related, but I just need some advice and thought I''d ask the bright individuals here. I want to start my own blog but I''m not sure where to begin. I don''t want to get a blogger account because I want to have my own domain. Blogger also doesn''t have many features. My hosting service is 1AND1 so it supports php, cgi and mysql.

I have a CentOS virtual private server from 1and1.com If I checkout or pull something from a repository, it will contain an autogen.sh file Running this and then configure seems to work. However, when I run make I get a lot of error messages from libtool saying that such and such a command could not be found on some line or that there is a syntax error. This happens even with the latest

are there any open source sip softphone (Window OS version )?

Dear All The setup is te110p with an 8 channels PRI to make and receive all calls. SIP phones throughout the company. TDM400p with 4 FXS modules to send/receive faxes and make credit card transactions. I have an analogue phone on the tdm400p for testing. I can receive calls to the exten. There is a dialing tone. However, when I try to make a call I get a busy signal. Asterisk stated busy then

I recently had a failing disk replaced in my colo server. The server is running Centos 7.3. After the replacement, I was following these instructions to rebuild the RAID array that the disk was part of. https://help.1and1.co.uk/servers-c40665/dedicated-server-linux-c40577/rescue-and-recovery-c40581/rebuild-the-software-raid-array-after-a-drive-replacement-a729756.html Everything went fine until

This is an email to all New York based Asterisk users. For some time it's been bugging me that we don't have a local contact point/user community. If you are involved in Asterisk and in NY/NJ shoot me an email, I'm going to try and revitalize either meetup.com or some other shared environment for Asterisk users in NY. Shoot me an email and once I get an idea of how many

Does anyone know what options you need to use with "sox" to output the audio in the WAV49 format that Asterisk uses.

Hi everyone, How can I add the equivalent of: exten => s,n,SIPAddHeader(Alert-Info: Ring Answer) in a .call file? This is to support paging to Polycom phones... Thanks for all info! Steve

Hi all I sold new TDM800P card with 8 FXO ports, someone know if can be use this card on AsteriskNOW or trixbox? What can i do for use this card? Thanks. ---------- RafaelCanchola Product Development Engineer, FonetGlobal Inc. rcm at fonetglobal.com http://www.fonetglobal.com Ph. + 52 800 022 10 21 ext. 214 + 52 442 167 08 00 VoIP 523663899 d00d! cyberalph -------------- next part

Hello,

Hi All; Any one can advise for a good stable open source video conference or video server? Regards Bilal ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

Hi all, I will a TDM card with FXO modules on it. Below is the dial plan. When someone can 9123456, CLI will show dialing to 123456 and answered. After answered, the call hangup. I would like to know what will cause the case to happen. Anyone can give me some advice to solve it? exten => _9X.,n,Dial(Zap/g0/${EXTEN:1}|${RINGTIMEOUT}) exten => _9X.,n,Hangup zapata.conf

hi, all I creat new application app_myapp.c for asterisk 1.4.15. I add this in asterisk/apps dir. to load. after compiling asterisk app_myapp.o and app_myapp.so has been created but when i run " show applications" at cli> . my application not displayed. what's wrong??? any suggestion!!! thanks Bhrugu Mehta