similar to: context question

I'm trying to clarify contexts and their uses. I do have a good general understanding of them. My question is about "undeclared" and "non-existant" contexts. If I have a block somewhere (in sip.conf, for example), and it has no "context=thiscontext" field, does it just automatically use the "default" context? Or is this settable? (I see there is an

I have to design a dialplan for mulitple contexts (multiple companies) and I'm not sure how to go about it and I thought someone may offer help. Here is some background. There are three separate companies, let's say A, B and C. Each has their own context and each has their own set of numbers (these are just examples, not the actual config): [ContextA] exten =>

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

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

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

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 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 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

hi, how can I add or remove this line "include => context" by the command CLI ? regards Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030827/979ddd76/attachment.htm

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

Hello; we are looking to replace our current PBX with a *-box; it is connected to ONE ppp isdn connection that is terminated by the NC. We got on this box 4 msn's configured. currently we are working with pstn fxo's behind the PBX; it works but we can't use the CSID information behind it. We want to migrate and keep the MSN's to decide routing in combination with the CID.

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 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

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

Hello, all. I've got a problem where we set up call pickup for a customer. If the Bob's extension rings and Bob is in Jim's office, Bob can press the button on his Snom 320 that says "Bob" and pick up his line. It works great for calls coming in from the outside but does not work for internal calls. Internal calls generate a app_directed_pickup.c:204 pickup_exec: No

Hi, one short question: Is it possible for the zaptel driver to deal with multiple phone numbers on one single E1 PRI line? I could make my carrier route +49 xxx aaaaa-zzz and +49 xxx bbbbb-zzz and others down one single PRI trunk to our asterisk box terminating in a Digium TE410P. Does the driver handle this and can I put calls coming in all on the same physical interface put into