similar to: OT: cisco ip phone security problem

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:

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

Thought those that use cisco in conjunction with asterisk may want to read this. I dont use cisco so I havent read it to see if its actually anything new. Vulnerabilities in Cisco's VOIP system http://www.computerworld.com/securitytopics/security/story/0,10801,103240,00.html?source=x73 -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402

I just tried to call a NCFA number that is known working in the UK (+44 0870 ..) and got a US message from broadvoice 'that number is not in service please check the number and try again'. I am wondering if it is my setup or if broadvoice no longer provides unlimited access to those numbers. Could someone else verify if this is the case? It is still listed on their rate page as

Does anyone know of a BYOD provider that terminates calls to NCFA numbers (UK 'national rate'). I enjoyed broadvoices unlimited to those numbers, but this is getting silly now, it doesnt work and no answer if after switching to a new provider it will ever work. Can anyone suggest an alternative provider that serves NCFA? -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605

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

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

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

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

does anyone have knopsterisk for download, I assume that because its GPL the creator of that iso cant restrict spreading it. A friend wanted it to play on a box and the only thing I can find with google is the knopsterisk.com site which wants $10 to get a copy and does not provide (as far as I can tell) any free distribution access which is his/hers/its/them/they/whatever right (being politically

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

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

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

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

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 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

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

I have searched for how to locate echo cancelation on SIP clients, but cant find anything and echocancel=y doesnt seem to have any effect. Configuration: CVS-HEAD from last month iPAQ h5500 with SJPhone (gsm/ulaw/alaw) Problem description: When I place or receive a call I hear a faint delayed echo of myself. The other party hears a really bad nonmuted echo that makes the call unusable. Aside

Sprint Nextel is sueing vonage, voiceglo and theglobe.com for infringing on VoIP patents. Sprint Nextel claims to have about 100 patents on VoIP technologies. Does anyone know which ones this article is talking about, and if so does asterisk have any of those features? The reason I am asking is that the article is vague, Vonage uses a fairly standard codec set, I dont know about the others.