similar to: best gui

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

https://bugzilla.mindrot.org/show_bug.cgi?id=2473 Bug ID: 2473 Summary: sshd and -R port forwardings on 127.0.0.0/8 Product: Portable OpenSSH Version: 6.6p1 Hardware: Other OS: Linux Status: NEW Severity: enhancement Priority: P5 Component: sshd Assignee: unassigned-bugs at

Dear users, I am new to this mailing list. Can someone send me a guide or steps to configure Asterisk on Linux box? I will highly appreciate. Regards, Afzaal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050725/14265bb5/attachment.htm

Hi, I just got some interesting information from Kirk Telecom (www.kirktelecom.com). This company has been in the business of providing DECT solutions (IP gateway, base stations, repeaters and handsets) either to be used with Cisco CallManager (SCCP protocol) or with the Innovaphone IP PBX system (H.323). Two important elements: 1. It seems they foresee a SIP version of their product in Q1

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 ???

Hello, I'm glad to announce that DeStar 0.2.2 version has been released. This release contains a large number of bugfixes and new features, see CHANGELOG.txt for the full list. You can find it in the usual place: http://developer.berlios.de/project/showfiles.php?group_id=2112 Thanks for using DeStar, Santiago Ruano Rinc?n http://destar.berlios.de -------------- next part -------------- A

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

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 :

Can Asterisk light the message waiting light on a Grandstream BudgeTone phone? If so please reply with any related configlets :-) Thanks!

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, 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

Hi, I am looking for a very low cost way of receiving and sending T38 fax reliably. Is there any possible solution using Asterisk as the PSTN SIP gateay and Digium E1/T1 card? Is there other open source package that can help to accomplish this purpose? Regards, Mark -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi guys! We have a call centre that has been moved across from an old Ericsson MD110 PABX to an Asterisk server with those in the call centre using X-Lite as their softphone. I'm trying to get Agent presence configured so that X-Lite gives the operators a visual indicator of their status - logged on, off and on "pause". I'm using chan_agent for the agents, so agents are

I have a DID from IPKall.com which is forwarded to my asterisk box. Then this extension should call my ip phone using Dial application. Everything works fine, except when I pickup the phone, I can talk, the other party can hear me, but I cannot hear anything the person says on the ip phone. Then after a couple of seconds, the call hangs up. I don't know why. Here is the message I get:

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

I have looked at matt's asterisk gui client at sourceforge. I am not a programmer by trade. The documentation there seems to be a bit lacking. Has anyone have the experience in installing the gui client and may perhaps have a how-to document available for sharing. -- David Kwok Tel: 612 99292086 ext 1002 Iaxtel/FWD # 17001813482 ext 1002 -------------- next part -------------- A non-text