similar to: DeStar 0.2.2 released!

People, I have a Debian box with Asterisk and I've installed the Destar package in order to get web managing of my voip system. After I installed Destar, it runs on "localhost:8080", but my server does not have X-Window to access to it so I can engter the web interface.. So how can I change localhost:8080 to IP_ASTERISK:8080 in order to access Destar via web from another PC ???

Has anyone ever encountered an echo on the IP phone side of a call? It is an echo of the user's own voice. I believe that no one else in the office is experiencing this problem. The phone itself is a Snom 320. I've asked Snom for assistance since my source no longer carries Snom, but unlike previous times they've been slow to respond. ----- Mike Hammett Intelligent Computing

Is there a version of the pv_ops kernel that will do HVM and work with fc11? I have been failing miserably trying to get it to work. I have followed Boris''s guide at: http://bderzhavets.wordpress.com/2009/06/10/setup-fedora-11-pv-domu-at-xen-3-4-1-dom0-kernel-2-6-30-rc6-tip-on-top-of-fedora-11/ But have had no success so far. I am wanting to know what is the version of the pv_ops

I'm working with the final details of the Astricon agenda. I haven't got anything so far on Asterisk GUI's and there are plenty of projects out there. I would like to invite developer's of Asterisk GUI's, both open source and commercial, to participate. What I'm thinking of is giving each GUI a slot of 10-15 minutes for a presentation and then a panel discussion on the GUI

Good day Im look at http://www.voip-info.org/wiki-Asterisk+GUI And I see there are a few GUI for asterisk What do you guys prefer? What is the best and simplest? Id like something that give me access to backend for a little bit of customization Thanks for you help and time -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm pleased to announce that Flash Operator Panel 0.26 has been released! FOP is a GPL'd switchboard type application for the Asterisk PBX. It runs on a web browser with the flash plugin. It is able to display information about your Asterisk box in real time. It is included in FreePBX, Asterisk@Home, DeStar, startShop, and several other projects both free and commercial. You can grab the

Hello all I would like to know your opinions on free GUI used to manage Asterisk. Which is better? My setup is quite small, about 15-20 phones. I've seen the liste on voip-info. Thanks all. fred -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url :

hi, I'm a newbie in this. I'm part of little company with 20 users, we need a pbx/central with access to and from the PSTN. i know that it is possible with asterisk, but i want to know which kind of devices i need, (interfaces and phones) thanks, -- santiago jos? ruano rinc?n administraci?n servidores y servicios de internet red de datos universidad del cauca -----BEGIN PGP

Hi, Asterisk Version : 1.2.15 Card : TDM11B (1 x FXO , 1 x FXS) I have internal dialling working okay SIP->ZAP (analogue phone) and ZAP (analogue phone) -> SIP. The problem comes when I try and make a outbound call. Here is my extensions.conf :- Code: [incoming] exten => s,1,GoToIfTime(17:00-09:00\mon-fri\*\*?outofhours|s,1) exten => s,2,GoToIfTime(*\sat-sun\*\*?outofhours|s,1)

I was searching thru the internet and I found a wide variety of different web interfaces for asterisks I was curious which one is best suited for asterisks. Thanks

Hi, I finally got Asterisk installed using the German installation CD from http://www.asterisk.de.ms. I got two SIP phones working (SIPPS) asterisk*CLI> sip show inuse Username incoming Limit outgoing Limit 5678 0 N/A 0 N/A 1234 0 N/A 0 N/A

Hi ! I just wrote cdr_sqlite.c, see http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0001986 This little app creates automatically the sqlite database file in /var/log/asterisk/cdr.db, creates a table 'cdr' inside it and inserts all CDR records into this table. Please comment. I'll use this in my project DESTAR (http://www.holgerschurig.de/destar.html) to show these

hi, i have a external usrobotics modem, i want to use it with asterisk to interact with the pstn, what i have to do? thanks, -- santiago jos? ruano rinc?n administraci?n servidores y servicios de internet red de datos universidad del cauca -----BEGIN PGP MESSAGE----- Version: GnuPG v1.0.6 (GNU/Linux) Comment: For info see http://www.gnupg.org

hi list, when I run asterisk, appears the following: .... WARNING[1074459808]: File chan_oss.c, Line 346 (setformat): Requested 8000 Hz, got 8178 Hz -- sound may be choppy WARNING[1074459808]: File chan_oss.c, Line 974 (load_module): XXX I don't work right with non-full duplex sound cards XXX WARNING[1133735216]: File chan_oss.c, Line 232 (sound_thread): Read error on sound device: Resource

Hi, all, Can anyone tell where can I find the perl library for manipulating 'ini files'? Thanks, kaiduan ______________________________________________________________________ Post your free ad now! http://personals.yahoo.ca

Hello all, I'm looking for a solution to offer Virtual PBX, to my clients. I just saw software with multi-tenant support and I tested it, but no one likes me enought. Finally, I want to offer this service like a kind of hosting. Has you experience with multi-tenant software? Which has you tested? Has anyone experience about "vhost", "vserver", or something similar to run

Hello, I have setup XCP using the install .iso and have been trying to do an install over http, this has failed miserably each time. So now I want to just carve out an lvm from the disk and put my .iso files in there and install off of them. How is this done? I can create an lvm easy enough but I am not sure how to mount it and expose it to XCP. Is there some type of magic I am missing? Is there

Hi! I have an automated dialout via a call file to a mobile. Can I detect when the call is not answered but forwarded to the mobile operator voicebox? I would like to stop the dialout if this is the case. TIA, Mike

Source: xen Severity: normal User: debian-lts at lists.debian.org Usertags: upstream-trixie X-Debbugs-Cc: debian-lts at lists.debian.org Dear xen maintainers, Testing (trixie) currently ships xen 4.17, which, according to the upstream support matrix [x], will get security support until 2025-12-12. The latest upstream release (4.19) will get security support until 2027-07-29. I believe it would

Hello all, Just starting to setup asterisk v 1.4.11 and need to run three distinct phone systems for three different companies. So far, I have inbound lines going to the appropriate dial plan within the extensions.conf file. I'm using exten => _X.,1,NoOp(FROM NUMBER: ${SIP_HEADER(TO):5:10}) to determine which number is being dialed by the caller and then using a gotoif to get to