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Hello - I have a TDM400P with 2 FXO and 2 FXS modules. Feeding the FXS modules are two VOIP lines which are terminated by VOIP adapters and have regular RJ11 wires connecting to the FXS ports. Since the two different VOIP lines have different phone numbers, and I know and can tell asterisk which VOIP line is connected to which FXS port, can I cause a distinctive ring on the extensions if a call

I hope the subject isn't too buzzword compliant :) I'm just curious: What have people done with Asterisk? I'm particularly interested in DIY projects and things that can be done on a small/home office (or even hobbiest's) budget. If you have clever hacks or creative functionality you've implemented, I'd love to hear what a few people have come up with. Thanks! -Brian

...he Net5501 has a small 4-pin molex header on the board, I wonder if a small to regular sized molex power cable would do the job to supply this card. If the Soekris isn't expected to work well, are there any mainstream small form factor/low-power solutions for a SoHo asterisk server? -- Brian McEntire Photographer & Owner B Scott Photography (240) 358-6655 studio www.bscottphoto.com

Hello - I've read Asterisk should be able to activate a do not disturb feature to turn off the ringers on extensions. I checked the wiki and can't find documentation for how to do it. Here's my attempt, added to extensions.conf: [dnd-on] exten => _#78,1,Answer exten => _#78,n,Wait(1) exten => _#78,n,Macro(user-callerid,) exten =>

Hello, Sorry, I know I read this somewhere but now I can't find it when I need it. I'd like to force a call to go out one line if we dial '9' first and then the number. Same for '8' only I will force it out a different line. There is a parameter or a method to allow the dialtone to come back after pressing the first 9... but I can't remember how to do it. Anyone know?

Hello everyone. I'm new to Asterisk but got some basic functionality going last night and I'm just giddy to have my own PBX ;-) Sorry if these are silly questions: My Asterisk server has the TDM22B (2 FXO, 2 FXS) interface. I have a very basic PSTN line coming in from the phone company, I tried to get the most no-frills line possible (didn't pay for caller ID, voice mail, etc.). I

Can anyone tell me what I missed? I'm trying to setup a simple extension (400) that reports the time when it is dialed. I searched the threads and it seems like this should work... Here's what's in my extensions.conf: exten => 400,1,Answer() exten => 400,n,Wait,1 exten => 400,n,SayUnixTime(,EST5EDT,) exten => 400,n,Playback(tt-weasels) [BTW, tt-weasels is hillarious!

I have Asterisk voice mail setup locally. It works great, I'm impressed! Some details about my system: I'm using a TDM22B card to interface with both the PSTN and a VOIP provider. I'm running 1.2-beta from CVS. I have a regular VTech phone plugged into one of the FXS ports. Asterisk is able to indicate when a local voicemail message is waiting via the LCD display of my analog phone.

Hi all - Been using Asterisk installed on Debian and love it. But it's time to rearrange some lines and looking for a few features I didn't enable or have in the dial plan the first time around and wondering if you would recommend doing it through configs again or if one of the prepackaged solutions would more easily support these needs. One that caught my eye was TrixBox but I'd be

For those that have experienced low VM recording volumes when using a Digium TDM04b (or similar analog pstn card), a work around has been committed to cvs-head. Need some folks to test it; it doesn't seem to work for me, but need some feedback from others to ensure the work around is actually functioning. (The work around relates back to bug #2023, and was committed around Thursday or Friday

I looked on the voip-info wiki and found sparse and conflicting information on how to do this with Asterisk... My incoming lines are all on Zaptel. Is there a simple why to implement a '*363" (do not disturb) toggle via the dialplan? It would be nice to be able to pick up an extension, dial *363, and have all calls sent to voicemail without ringing the extensions. Doing it again would

Greetings rsync gurus! I'm trying to run rsync from machine A to backup files on machine B. I want the files to be copied into a directory -- /backups/B -- on machine A and mimic the directory structure on machine B. Machine B has the following file systems (and some NFS mounts I don't want): / /boot /usr /usr/local /usr/local/apache The problem comes when I run rsync with

Hello, hoping someone out there has some ideas - I have a VOIP line that has call waiting. It is terminated at a Sipura 3000 and the POTS side of that device connects to an FXO port in my * box. I also have a POTS/PSTN line that terminates in another FXO port on my * box. There are two FXS ports which feed cordless phones. I'm using the Zaptel TDM400 card. This gives 2 extensions + 2 lines

I use Voicepulse as a VIOP provider, the line comes in via a Sipura 3002 box. That's connected to the Asterisk box via a TDM422B POTS card. I'd like to add a virtual phone number to my VOIP service so that I can direct calls to a home business to a different voice mail than calls to the home phone number. Is there a way for Asterisk to know which virtual number was dialed under this