similar to: OT: cisco voip vulnerability

http://blogs.computerworld.com/15082/omg_microsoft_patents_sudo_linux_and_macos_dead?source=CTWNLE_nlt_dailyam_2009-11-12 -- thanks ./francis -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.centos.org/pipermail/centos/attachments/20091112/1061174b/attachment-0003.html>

I am looking for a voip provider that provides good rates (below 5 cents/min or unlimited) to UK NCFA numbers. Braodvoice advertises they do unlimited to NCFA but does not have the ability to actually termiate those calls as per the CTO Nathan Stratton, and last he said they dont even have contracts in place to get service provisioned for that. As such I am looking for another provider to take

Anyone have any idea how soon RHEL and CentOS will be releasing the patch package? Excerpt: Computerworld - Developers of the Apache open-source project today warned users of the popular Web server software that a denial-of-service (DoS) tool is circulating that exploits a bug in the program. The tool, called "Apache Killer," showed up last Friday in a post to the "Full

has anyone else noticed what appears to be a threading issue in asterisk 1.2.9 (it broke sometime between 1.2.4 and 1.2.9) where if you have about > 50 calls and do asterisk -rx show channels it will display the header but nothing about channels, total calls, active calls, etc. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461 DE +49 801 777 555 3402

I think they are being vague to give people a time to upload to the latest version. Cheers, Dean > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Brian West > Sent: Thursday, 23 June 2005 11:45 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re:

Or you can use authenticate() and have it take its 'passwords' form a text file on your machine. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > trixter aka Bret McDanel > Sent: Sunday, January 29, 2006 5:37 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion

http://blogs.computerworld.com/16997/oracle_rips_red_hat_and_sort_of_launches_a_new_linux oracle: just another centos wanna-be? rday -- ======================================================================== Robert P. J. Day Waterloo, Ontario, CANADA Top-notch, inexpensive online Linux/OSS/kernel courses http://crashcourse.ca

I have been asked if asterisk can change codecs dynamically based on the calling party's caller id. I couldnt find anything, and dont know that this is something that asterisk can do, but it occurs to me that possibly with a reinvite it can be done, however I dont think you can issue those from the dialplan or agi. The only solution I can think of on this is to use something like ser

I am considering using iax.cc (sixtel) and wondering if anyone had opinions, good or bad. Are there outages with any regularity? How responsive are tech support? How is packet loss? I am particularly interested in termination to the UK, but will accept any comments people have. Thanks -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US

I am wondering how the g729 licenses are done during calls. If I have N licenses for g729, and N are in use and an additional call comes in that requests N+1 to be in use, how does asterisk handle that call? Does it dump it? Does it negotiate another codec automagically? Basically what happens to that call, obviously it wont (shouldnt) let you use more licenses than you have available, but

Is there any way within asterisk to limit the scope of contexts, basically to make one context totally unaware of another. The application I had in mind involved allowing users to create their own dial plans. To that end I wanted to make it so that a given user could not call a different users dialplan. I could filter everything and prepend a customer id to every context they specify, but

Has anyone built a game with the dialplan? I would think this would most easily be managed by an AGI, but its possible with realtime extensions. The game would be like 'adventure' that I first played on a prime in 1979. Or any of the infocom games (ie zork). Infact since the infocom spec is known it might be possible to plug in the data files directly from an AGI. If anyone has done

Has anyone successfully used a standard bluetooth enabled system to connect to a standard bluetooth enabled mobile phone (not the bluetooth to FXS converters) to create an audio path for phone calls with asterisk, if so is there a writeup on what was done so that others can replicate this. What I am thinking is that via alsa/oss/whatever you should be able to use the bluetooth audio channel as a

I am curious if anyone has had problems trunking iax2 with 100+ concurrent calls. I am planning on testing this out tomorrow, however I wanted to know if anyone else has had a problem with this prior to my test so I know what to look for if anything is known and what resolutions have been found if there are any known problems. Specifically I am doing this on fbsd 6 with asterisk 1.2.4 using

I was wondering if anyone has put together a comprehensive list (that is reasonably maintained) that lists country codes, landline numbers, mobile numbers, etc. The particular requirement is for a dialplan to know what is going to be charged to whom. For example, mobile and landline rates are the same in the US the US has a unified numbering plan of 1NXXNXXXXXX, while the UK has: 441xxx

Hello All, I have a question about glm in R. I would like to fit a model with glm function, I have a vector y (size n) which is my response variable and I have matrix X which is by size (n*f) where f is the number of features or columns. I have about 80 features, and when I fit a model using the following formula,? glmfit = glm(y ~ x1 + x2 + x3 + x4 + x5 + x6 + x7 + x8 + x9 + x10 + x11 + x12 + x13

SJVN's take on it: http://blogs.computerworld.com/16596/the_most_popular_web_server_linux_is

... here''s hoping they release it with Solaris drivers http://www.computerworld.com/action/article.do?command=viewArticleBasic&articleId=9121698&intsrc=hm_list -- This message posted from opensolaris.org

aside from festival are there are other TTS engines out there that are free? I have written a simple script to snarf files from a foreign site with a really good TTS engine, but there is a lot of latency so I was looking to use something on my system, however festival is hard for me to understand (far too mechanical). Basically the script I have uses sitepal.com's TTS engine (after a

has anyone experienced a problem where RTP audio cuts out when doing 30-40 concurrent channels via sip? The box is freebsd 6, asterisk 1.2.4, voip only (no libpri, no zaptel - not even a timing source) The box has plenty of bandwidth, when a call to the same box is iax2 it works, but when its sip a call gets connected a few frames of audio are passed and then silence. When the box is completly