similar to: eBay Sip Phone Scam.

Hello guys, i have been on this mailing list for some weeks now, and i was wondering if someone here has installed linux on the XBOX and use it as a dedicated server. Its a 200 USD computer and could make it perfect to asterisk, its little and doesnt really take much space. My question is could this make it for a stable server??? here are some links i found for linux on XBOX

I am trying to get the following phones for testing. Is there a distributor in the US that is able to sell me these Sip phone and ATA adapters? I can not afford the Cisco phones there too hard to configure and too expensive! 1 - Sipura SPA-2000 2 - Grandstream Sip phone BT-102 1 - Grandstream HT-286 1 - Snom 105 Sip phone. I have called and emailed chagres but they have not reply. Nor

I am sure this has been asked before, but why not use Google newsgroup or at least some forum BBS software instead of this cumbersome mailing list process? -- Costas Menico Meezon Software Corp 201-224-8111 costas@meezon.com --

I would appreciate some help with this. I read the GPL license and basically it says you can do whatever you want with the software (sell, modify) as long as you include the source code, the License and make any changes you make available in the same manner to all others. My questions is this: If I develop an external application (say a Call Center application or a GUI management application)

Hi, I am an experienced developer with Windows and familiar with Linux. I am looking for a SIP solution. 1) How does Asterisk compare to VOCAL in terms of support. 2) Is Asterisk free? 3) Where are the docs? Or even better. Where do I start? 4) Will it run on RH9? Thanks in advance. Costas -- Costas Menico Meezon Software Corp 201-224-8111 costas@meezon.com --

Hi, Does anyone have sample * configuration on how I can get an incoming call to ring all SIP phones (small setup, say 4 phones) at the same time. 1) I would like to pickup up any phone and the ringing should stop (of course) 2) Put on hold and pick up on a different phone set. Do I need special phone features to achieve this? E.g. would the Grandstream 100 do it? Thanks -- Costas Menico

I want to be able to demo * on a notebook at a client's site. This means no FXO gateways; just 2 sip phones (like SNOM) and maybe a softphone (GnoPhone?). I already have RH9 running on my notebook. I would like to have one SIP phone dial and go through IVR before making a choice and ringing the other phone extensions. Of course the notebook would have to be running Asterisk. How can i setup

I just ordered the Asterisk Developers Lite kit. My environment will be the RH9 Linux server and a Windows workstation with Samba. I also of course have analog lines and DSL. I am interested in SIP development. I already downloaded the Asterisk software. What else should I download. Is there a doc that basically tells you the steps to install Asterisk and get it up and running? I would like a

This is troubling. Shouldn't your hubs/routers autosense the 10MBPS? ---------- Original Message ---------------------------------- From: WipeOut <wipe_out@lycos.co.uk> Reply-To: asterisk-users@lists.digium.com Date: Wed, 15 Oct 2003 07:53:13 +0100 >Steven J. Sobol wrote: > >>On Wed, 15 Oct 2003, Jon Pounder wrote: >> >> >>Nothing works. Call transfer

I installed Asterisk as per instructions in the FAQ on the digium.com site. Double checked it. I also think they have a bug in the zapata.conf where the context should be incoming and not internal. 1) I hear no dialtone when I pickup the phone on the S100U. Asterisk sees the event and displays the message on the screen. I tried dialing but nothing happens. I hangup and * shows the hangup event.

I am having serious issues with RH9 when it comes to speed. It may just be gnome or kde but it is slow launching apps. Does anyone know if Asterisk will compile under other distros. Many people are recommending http://www.gentoo.org/ and I am considering it. What is it about RH9 thats different when compiling Asterisk? Thanks -- Costas Menico Meezon Software Corp 201-224-8111 costas@meezon.com

A note to all those who are avoiding writing up an AGI becasue it looks two complicated.. I have just written up my first and its awesome.. It makes Asterisk open to all sorts of possibilities.. let your imagination run wild.. I put off writing an AGI script because a) I could not find any docs b) it looked like the only way to do it was perl and I know nothing about perl and c) I am not a

I am using a brand new RH9.0 installation. I installed Asterisk afterwards so I am not sure if Asterisk caused the problem below. The ps doesn't work. It could also be something else. I also tried installing a some video package. But I thought to ask here first if someone has seen this before. [root@localhost asterisk]# ps ps: error while loading shared libraries: libproc.so.2.0.6: cannot

Is there any kind of Configuration guide available? I've been trying to get a SIP Soft phone to work and all I get is: NOTICE[81926]: File chan_sip.c Line 4716 handle_request): Registration from '<sip:xxx@xxx.xxx.xxx.xxx> failed for 'yyy.yyy.yyy.yyy' I know I'm missing something in the configuration. But don't know much about * yet. Thanks, Kyle --------------

Hi! I am doing a research about the prices of SIP telephones. If someone can tell me which one are the cheapest and have an acceptable quality... it will be very kind. Best Regards, Mireia

>I went back to the example system direct from CVS with small >additions to sip.conf and extnsion.conf needed to make one >xten X-Lite phone work. I can dail in and hear the anouncements, >call the demo server at Digium. The audio quality I hear >comming from Asterisk back to X-Lite is good (9 on a 10 scale) >but the sound volume comming from the X-Lite extension is very low

I am using IAX to send a call to my cell phone. I want to be able to hit # and transfer it back into the office. I have added tTr to the dial command and hitting # prompts me for the transfer, but after I start dialing 103, it stops at 1 and tries to transfer it within nufone instead of my dialplan. This is the debug output: -- Called me@NuFone/1515480XXXX -- Call accepted by

I hate to bring this thread back to life, but... > it may be possible to get it supported, do you think the price >point is remotely competitive with Digium hardware? Also as I am not >about to divulge my information to them to look in the downloads >section, what is the licensing of their SDK? What is the licensing of >the driver? > >Steven >On Tue, 2002-11-26 at 14:52,

I have a DID with VoicePulse Connect, but the sound quality is horrible, it is often choppy and the caller's voice cuts out for 2-3 seconds at least once a minute, I have contacted VoicePulse many times, and they do not do anything about it! Does anyone have any similar problems? It isnt my Asterisk config because I have 0 problems using NuFone.

I have upgraded my grandstream phone from firmware 1.0.3.78 to 10.0.4.30 and now I am having problems with early dial. On the older firmware earlydial worked fine with my asterisk server, but now as soon as I have dialed the number I get a congested tone, and the number 4 flashes up on the LCD screen. Has anyone had this problem, and if so, how do I fix it? -------------- next part