similar to: dialplan game

The solution you choose should be based on many factors which should include your business requirements, team's experience, your budget, growth expectations and more. You can choose Asterisk or Freeswitch as a platform and start building on that - but it is not simple and being new to VoIP you are likely to make mistakes. The "do-it-yourself" approach will some money initially, but

http://www.boingboing.net/2006/06/05/play_zork_by_phone.html Let me preface this idea with one comment: I don't have the time to do this - I don't even have time to eat these days. But someone out there has the cycles to do this... and it would be very cool. OK, so now Zork is attached to Asterisk, but using the less-than-clear Festival engine. There are beta tests of the LumenVox

Hi all, I'm new to VoIP, now we have a project that needs a PBX with client APPs. In our team we have argument for choosing PBX. By so far, we have following candidates: A: Open source 1) Asterisk PBX (http://www.asterisk.org) (with longest history that almost every one knows it, now the last version using the PJSIP stack) 2) FreeSwitch (http://www.freeswitch.org) (A lot people

I was tinkering with Asterisk and the Festival text-to-speech engine, and wrote some short Asterisk::AGI scripts to read back live weather reports. After that, I thought I needed something more interactive to work with... Then I had a flashback to 1996, first year university, standing in the C & O club at the University of Waterloo, where someone had just pulled out their US Robotics Palm

Dear Sir Can't find /etc/apt/sources.list file please help. When I tried the command on terminal some errors has occurred as follows- manjulata@manjulata-HP-Pavilion-dv4-Notebook-PC:~/Desktop/ndnSIM$ deb http://ftp.iitm.ac.in/cran/bin/linux/ubuntu raring/ No command 'deb' found, did you mean: Command 'dab' from package 'bsdgames' (universe) Command 'dwb'

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

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

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

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

Hi. I use NUT on embedded system, and after update distro to fresh one I've got a non-working usbhid APC monitoring. There's 2 troubles: 1) I/O error for root on usbhid device 2) Permissions error for nut user Root log: # /usr/bin/usbhid-ups -u root -DDDDDD -a infocom Network UPS Tools - Generic HID driver 0.38 (5.1.2-beta1-28-g04cd334) USB communication driver 0.32 0.000000

Wow! this got long on me. The problem seems similar to the one expressed here: http://www.mail-archive.com/samba at lists.samba.org/msg111029.html but I already had server signing = no in my config, and the 4 relevant options all seem correct. client schannel = Auto server schannel = Auto client signing = auto server signing = No On one of my PC's I can see 'If possible =

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

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

Hi. I use NUT on embedded system, and after update distro to fresh one I've got a non-working usbhid APC monitoring. There's 2 troubles: 1) I/O error for root on usbhid device 2) Permissions error for nut user Root log: # /usr/bin/usbhid-ups -u root -DDDDDD -a infocom Network UPS Tools - Generic HID driver 0.37 (5.1.2-beta1-28-g04cd334) USB communication driver 0.31 0.000000

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:

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

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

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