similar to: FWD and IPCall

Hello Asterisc-Users List, I am new to the list. I joined with a question in mind: How would you set up an asterisc box so that: (A) Someone dials a number (B) They are presented with a menu (C) Entering a number, like 1, connects a call to me. (D) I am on a mixing board, running an internet radio show. I want to run asterisc into the board, and run an output from the board to asterisc. Is that

Okay, here's the dal. Me and my friend both have asterisk boxes. I want to be able to type extension 27 on my end and get his extension 27, and he wants to be able to type 277 on his end and get my extension 9999. We both have FQDN's, and would like to see about doing this either over sip or IAX... Any help much papreciated, Shane -- -Shane Blog: http://blind-geek.com/blog/ CoOwner:

Hello List, I am getting Asterisk set up. I am going to be installing Debian Linux on a laptop later. I would appreciate some detailed instructions on: (A) What to type into the shell to download and install Asterisk. (B) How to open the configuration files (*.conf) (C) If there is a way that I can change the configuration files remotely (SSH?). Thanks in advance. -- -Shane Blog:

TODAY, Friday January 4th at 12 Noon EST, 11 AM Central, 9AM Pacific, Mountain figure it out, 17:00 UTC Mark joins us to talk about IAX, the appliance, what's new in the asterisk worldwide communities and answer any questions you may have. Why not take this opportunity to ask questions or make comments? This conference is the largest *live* online meeting of asterisk users in the world. Each

Hi All, kind of need help on the conference module, i'm using freepbx and enabled conferencing, i created a conference number, 6000. when i dial to it, my phone says it is connected but i'm hearing nothing, maybe logs below can help you. also, when i hang up the phone, the conference did not disconnect me. how can i end a conference? thank you -- Executing

Ok, I have spent all night trying to figure this out, and hopefully somebody has a similar experience. I have a very basic asterisk config. Sample configs, with the only addition being by SIP phone, and my incoming voip. Last week I got everything setup, calls were working, etc.,. I cam across a tutorial for voicemail, followed it, and it worked. When I call my phone and dont answer, it goes

I tried this http://lists.digium.com/pipermail/asterisk-users/2008-January/203615.html But I am NOT getting call in asterisk. SIP.conf file : _________________ [general] port = 5060 bindaddr = 0.0.0.0 context = default externhost=59.160.44.21 localnet=192.168.0.2/255.255.255.0 ; register SIP account on remote machine if using SIP trunks ; register => testSIPtrunk:test at 10.10.10.16:5060 ;