similar to: TE405P w/ Intel SE7210TP1_E Motherboard

http://host-a.starnetworks.us/Members/kpfleming/spring_von/photoalbum_view Enjoy!

Hello, I have a client that has a fairly small installation (20 SIP Phones) that is running Stable. Asterisk appears to be consuming large quantities of memory, and growing uncontrollably to the point where after about 6 weeks the box starts to swap itself to death. I've been keeping my eye on it today, and in the last 12 hours, it has grown by about 8 megabytes, and there has been

Hello, Straight from the floor of Astricon 2004, I am happy to release my updated Asterisk 1.0 RPMS for RedHat 7.3 and RedHat 9.0 platform. Current Release --------------- asterisk-1.0-0 libpri-1.0-0 zaptel-1.0-0 kernel-module-zaptel-1.0-0 RedHat 7.3 ---------- ftp://ftp.nacs.net/asterisk/rh73/RPMS/ ftp://ftp.nacs.net/asterisk/rh73/SRPMS/ RedHat 9.0 ----------

I'm relatively new to Asterisk and I'm hoping attending Cluecon will be a good way to get up to speed on the project and hear about what others are doing with it. We currently use a Cisco IP phone system at my office, although I just added an asterisk box to provide soft phones to our travelling users. (IAX2 is a lot easier to get through firewalls than cisco's protocols). Terry

Hello, Anyone using the 1.0.4.55 firmware release with any success? I have had my Budgetone running 1.0.4.50 for about a month and a half now with no problems whatsoever, and I am a little leary about upgrading. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place!

hello I am fitteling with the astwind-installer-0.1.1.exe asterisk for windows and am having trouble getting the thing to connect to the meers to download the updates and stuff. I looked at the wiki and set up networking and stuff with no success, has any one got this thing to work successfully? my windows box is the faster of the 2 machines and my main linux box is down at the moment. I am

Is there a user guide for Comedian Mail? I need to give some training materials to my end users. So far, I have been unable to find anything through google or the Digium site. THanks, Wiley The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other

Hello, Please be conscious of Digium's bandwidth and use a Mirror when downloading 1.0. I have mirrored the tarballs at: ftp://ftp.nacs.net/asterisk/ Direct links: ftp://ftp.nacs.net/asterisk/asterisk-1.0.0.tar.gz ftp://ftp.nacs.net/asterisk/asterisk-sounds-1.0.0.tar.gz ftp://ftp.nacs.net/asterisk/libpri-1.0.0.tar.gz -- Vice President of N2Net, a New Age Consulting Service, Inc.

Hi, This might not be a very popular question, but I was just wondering if anyone have ever tried to run Asterisk on a Windows computer using Microsoft Virtual Server (http://www.microsoft.com/windowsserversystem/virtualserver/default.mspx). I am told that you can run Linux on a virtual server using this software, so in theory it should be possible. I run a windows based domain, but am also

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

I looked through this forum and other lists trying to get a definitive answer to the VMware question... I have a Windows 2003 Server box that is my development machine. It is fully loaded with many s/w packages, dev tools, etc. It's not very feasible to convert this box to Linux without losing an awful lot of stuff. On it, I'm running VMware GSX 3.1.0. Under VMware, I have a

Hello, I've got a mixture of SPIP 300 and 500 phones in production for various clients. I've got the XML settings configured for local conferencing, but I'm not seeing the expected behavior from the phone when I attempt to conference two calls together. According to the manual, while talking to the first party, you simply hit Conference, dial the second party and then Conference

Hello, I have a mixture of Polycom SP IP 500 and 300 phones. I have been reading through the administration manual to try and solve this problem, but I do not seem to be able to find the answers to my question. I figured I would ask here and see if anyone has some suggestions. The problem is kind of annoying. After dialing 4 digits, the phone seems to pause and miss the 5th digit, often

Howdy, I just did an "apt-get dist-upgrade" on my Debian unstable box, and noticed that the Asterisk version appears to be 1.0-1 in the unstable tree. I KNOW that 1.0 hasn't been released yet, so I am wondering who is responsible for the Debian packages? This will be VERY VERY confusing for people and it should be corrected ASAP. -rw-rw-r-- 1 1176 1176 1189696 Jun 7

I don't think I've ever seen so many commits to the CVS, I'm not complaining. To save lots of checkout/test cycles can a message be posted when the current surge is over? -- Dave Cotton Directeur Linux Autrement 193 rue Marcel Cerdan 84270 Vedene 04 90 23 30 81 <http://www.linuxautrement.com> IAX 17004902330 FWD 42651

Hi ! I need a solution to park incoming calls to an extension of my choice where a special announcement is played, park subsequent calls to specific pools so that they listen to announcements of my choice. any ideas ? Shah.

Has anyone does this with *? Ie, ask for the caller's name and provide that to the callee before bridging? For calls to an extension, it should be doable via the dialplan. For calls to queues, some changes would be required to app_queue.c to allow an addional file to be played after the announce message. -JimC

I have a * serving 15 sip clients. I use the digium 4 port t1 card. We have an autodialer that calls and reminds clients of there appointment. it uses a pri t1. I would like to plug its t1 output into asterisk to use voip. I am very new to * and am confused. Any help would be appreciated. _________________________________________________________________ Express yourself instantly with

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I have made my first run at building FC1 and FC2 rpms for Asterisk 1.0.0. These are based off of Greg's spec files which a few minor changes. Source and binary RPMs are available at: ftp://asterisk.purplehat.net/pub/asterisk/ Please report any problems with these files to myself and not Greg as he has nothing to do with these builds. :) Sorry these took so long; it was a wild week at