similar to: using sym-links with Samba

All, I run a bunch of (well 20+ actually) Asterisk boxes at home, work, friends and the lie with our own dialplan in the form 8EEXXXX where 'EE' is the exchange number and 'XXXX' is the extension number. This arrangement has been in for 2+ years and worked well with a central box (asterisk.thorcom.net) acting as the routing hub and SIP exchange point with various public

Mark, While these samples are pretty good they do not work "out of the box" - there are a couple of issues: 1. the samples are 44100 samples/second and Asterisk needs them to be at 8000 samples/second. This is what happens if you prune out all of the Amercian voicemail prompts and substitute yours: Asterisk 1.2.7, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark

Stephan, Ok, I'll re-state the problem... I have two devices that I want to talk to each other: 1. an Asterisk PBX 2. a Damm Cellular TETRAFLEX digital radio system (www.damm.dk) both devices are effectively "gateways" because they have many subscribers behind them. The Damm Cellular system controller is based on Windows-XP Embedded and its sub-systems used the OpenH323

Hello! My name is Tubby Frederickson, I'm a computer guru, and an independent associate with a NASDAQ corporation that offers very innovative products to broadband and non broadband telephone users. I saw your email address on a forum that was discussing broadband and telephone issues. Right now we are offering unlimited local and long distance calling throughout North America for only

The feedback we are getting so far has been excellent! As more is decided the list will be updated, if you'd like to be involved in helping, please join us on the IRC channel, #asterisk-uk on irc.freenode.net. If your company would like more involvement with the event, please email me directly. I would really like to hear from people/companies who would like to: - # Exhibit a product or

Gents, I've built an Asterisk system to replace our PBX at work and have Cisco 7960 phones (SIP 7.4) running with Asterisk 1.0.7. How to I get Asterisk to recognise the '#' being pressed during a call? In sip.conf I have entries likle this: [2001] type=friend context=local-phone auth=md5 username=2001 secret=xyzzy callerid=Jack Tubby <2001>

All, I have upgraded by home machine from Fedora Core 5 (FC5) to the recent FC6 and am struggling to build Zaptel-1.2.10 and Asterisk-1.2.13 on the box... which is an Intep P4 2.8GHz HT processor box with 845 chipset, hence the kernel installed is 2.6.18-1.2798.fc6-i686 so we hve this: [root@gate zaptel-1.2.10]# uname -a Linux gate.tubby.org 2.6.18-1.2798.fc6 #1 SMP Mon Oct 16 14:54:20

All, I'm haveing a bit of trouble getting my head around H.323 and call routing with Gatekeepers, Zones and intra-zone calls - hopefully someone who is more informed in things H.323 will be able to point me in the right direction...? I already have a mature network of Asterisk boxes dotted around the UK and overseas with hundreds of extensions and our own number-plan/dial-plan in the form