search for: aussievoip

Well, I wake up this morning, and aussievoip isn't up. I ring godaddy, who _were_ hosting it, and they say that the machine's been compromised, and you can't have your data. Nyah Nyah. I spent 1 hour and 38 minutes on the phone to them, trying to convince them to let me somehow get access to it, but to no avail. I've reported...

I've finally got a chance to play with 1.0.1.9, and the wiki has been updated. At the moment, I don't know of _any_ bugs with it. I'm yet to play with complex things like early dial, and will update the wiki as I find information. http://aussievoip.com.au/wiki-GXP-2000 I've just been given official permission to offer the firmware as a download, so if you visit http://aussievoip.com.au/wiki-GXP-2000-Firmware there are links and instructions there. Enjoy! --Rob

Hi ! :) Does someone have an idea of the reason why my MusicOnHold (with clean mp3 downloaded from http://aussievoip.com.au/wiki-MOH) is always having interruptions and micro-cuts ? No high load of the system, no swap, no hard disk r/w, mpg123 seems running well... nothing ! Except a little message at startup : "Warning, flexibel rate not heavily tested!" I'm getting crazzzzzyyyy... :-| Cheers,...

...> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users@lists.digium.com> > Message-ID: <000001c68ba0$3df82a60$0201a8c0@mark> > Content-Type: text/plain; charset="us-ascii" > > Have you followed the wiki entry on aussievoip? You are better to use capi > because I don't think misdn actually supports fax. > > > Regards > > > Mark Brooker > T: 02 4959 8670 > M: 0415 846 865 > F: 02 4950 5609 > E: mark@mbit.com.au > W: http://www.mbit.com.au

Hi I've got a dell 2550 with an Eicon Diva server PRI card plugged into it. I can call out using the acopy2 test utility. I'm having trouble with asterisk making calls however... my capi.conf and modules.conf looks correct by the wiki instructions - does anyone have any advice on where to look ? I can attach conf files etc. if needed. Asterisk says it has 30 capi channels available,

Guys. I just downloaded the recent firmware for GS GXP 2000 and I must say the phone works great but... How do you make the conf button work?????? Anybody done that? Also, with great dissapointment I must ask, where is ILBC support? GS web page mentions it and the manual says it supports it almost using bolds :) soooooo where is it???? Any light on this?

While using Blindtransfer #Extension everything works fine. But how do i activate announced transfer with an Grandstream GPX2000 ? Greets Markus

Hi, I've been having problems getting Asterisk to work with a fritz card (in ptp mode). We've managed to get everything configured, however, as soon as the call is connected we get "Unhandled message: prim 281 len -22 from addr 1000000, dinfo 0 on this port" I checked, and mISDN_dsp is loaded. Any help would be much appreciated! Regards, Chris Jones // Network Administrator

It took a little longer then expected, but here it finally is, a field test for the idefisk for linux iax2 softphone. Freely downloadable from http://www.asteriskguru.com/tools/ You will probably need to copy the iaxclient lib into your library directory and run ldconfig before starting the phone. Please note that this is the first copy in the wild of the linux version and is not as tested

Anyone with any comments on DSS buttons and general phone features? Thanks, Shane -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050525/e11c72b7/attachment.htm

First question, is there a forum for asterisk@home specific questions? I've asked what must have been questions about asterisk@home here and gotten some indication they weren't welcome. Second, does anyone know what files need to be backed up? I don't need to back up the entire system since I can reinstall from the CD in fairly quick order, however, other than the files in

We are moving our asterisk 1.0 system to a new Asterisk @Home system (2.8) and I am the one in charge of doing it. I have run into a snag, though, on meetme conferences and with the transfer key. Regarding the transfer, it appears that both directions of all calls can transfer by pressing the # key. I do not like that ability. I would like to change it by doing 2 things: 1. Make the transfer

It's a udev configuration. Read UDEV.txt, or, read the AMP wiki http://aussievoip.com for a step-by-step. --Rob -----Original Message----- From: asterisk-users-bounces@lists.digium.com on behalf of Jachin Rupe Sent: Wed 7/09/2005 9:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] /dev/zap* is not showing up (gentoo, portage,aste...

Has anyone attempted to use FreePBX for a business in production mode? Initial take is there are lots of things scripted but a lot of limitations in terms of supporting basic business functions. Inability (or lack of flexibility) is handling multiple incoming pstn lines, dialplan limitations, poor/no documentation, etc, to mention a few. Maybe its just me, but it appears its no where near

Hey folks, Using FreeBSD 5.2.1 and I've got the current zaptel driver installed from ports (0.8_1) and current ports asterisk (1.0.1). I've set options HZ=1000 in my kernel config, recompiled and rebooted and as far as I can tell, I've done everything right but when I try to use the conference, the audio is very delayed, choppy and segmented -- totally unusable. At the