similar to: Polycom IP 500 Dial Issues

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 ----------

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, 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, 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.

Hello, I'm looking at building a couple new PRI Gateway boxes using TE405P cards, and was wondering if anyone has had any experiences (good or bad) with the Intel SE7210TP1_E motherboards from Intel. General Technics builds some really nice (and cost effective) 1U servers based on the board: Server: http://www.gtweb.net/gt637.html Specs:

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

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'm replacing a SoundPoint IP 600 with a SoundPoint IP 650. I attempted to simply reuse the existing config files for the old phone on the new phone, but the new phone would lock up on the 4th digit when attempted to dial out certain numbers. So, I downloaded the newest firmware and config templates from Polycom, and attempted to migrate the settings. Seems I'm missing something from

Any Polycom gurus out there? If so, I have a few config file questions. First off, does anyone have the daylight savings time rules written for this Sunday's big change? Secondly, if there any way in the config file to tell the phone not to display the number of missed calls? I don't mind it keeping the missed calls list, I just don't want that running count. Lastly, I am trying

Just installed Asterisk 1.2 on a brand new clean machine running RedHat 9.0. I have a TDM400 card inside. When I boot, the card seems dead. When I do: modprobe wctdm modprobe Zaptel the lights come on and all seems fine, until I reboot that is... After a reboot I have to repeat the modprobe. I shouldn't have to do a modprobe every re-boot should I? How do you get the drivers to load

Hello all, It's my birthday today, so as my present I would like everyone possible to download and test my updated set of RPMS for Asterisk 0.7.1. By popular request, I installed and built a set of RPMS for RedHat 9.0, and in the process fixed a bunch of issues from the initial build. I have also updated and will be maintaining a page on the Asterisk Wiki located at:

Hello all, Saturday night, after a couple of shots of bourbon, I realized that I had an old PC sitting in the garage that I could use as an Asterisk gateway if I just blew the dust off it and reloaded it with a modern Linux distribution. In my characteristically impulsive manner, I grabbed it and started cleaning it up so that I could put it in my office without my wife having a fit. The

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

I have four Polycom 330 phones connected to an asterisk system. There are other VoIP phones connected too. All of the extensions are four digits beginning with 11. From any of the phones, except the Polycom, picking up the handset to call extension 1103 for example works fine. With the Polycom 330, as I press the second 1 of 1103 it stops taking input and gives me an error. I tried

Hi all. I just installed bunch of IP450's and everything went well and my customer is happy.... except that they are unable to transfer calls to other extenstions. They can dial them directly just fine. However, when the user is in a call and presses the transfer soft key, they get dial tone, and start typing the extension, say 1008. But by the time they get 100 typed in, the phone tries

I'm running Asterisk 1.2.8 with Polycom ip501's xten softphones The only problem I'm experiencing is the following: I can't seem to get blind transfers to work with my Polycom 501 phones Either through the feature code or the soft keys. Feature code blind transfers: I set up a feature map in features.conf like this: blindxfer => # This works for all my

This is a dialpaln issue. I solved the same problem recently. For 4 digit extensions you need to append the dialplan statement in the sip.cfg configuration file as follows <digitmap dialplan.digitmap="[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2 -9]xxxT|1xxxT" dialplan.digitmap.timeOut="3"/> Michael > I am not sure what I did but blind transfers do not

Hello all, I've been invited to do a presentation on Asterisk for the Ohio Linuxfest in Columbus this weekend (http://www.ohiolinux.org). Rough estimates are that nearly 500 people will be attending. I've been working on an outline for a couple of weeks and I would like to have some peer review of the information presented. I am going to have to cut down the content to make it fit in

Hello, I have a friend that is asking if he can use his Ericsson 3413 H.323 IP phone with Asterisk. I can't seem to find any reference to this phone on the Wiki... -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST

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