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..._newstyle(319) Error domain join verification (reused connection): NT_STATUS_INVALID_COMPUTER_NAME Unable to join domain BLABLABLA. Any help would be great! P.S. Running Suse 9.1 _____________________________________________________________________ For super low premiums ,click here http://www.dialdirect.co.za/quote

I am setting up 2 nufone lines. I want to make them both availiable for dial-out. How do you syntax it in extensions.conf so that it figures out which one is avaliable and dials out on it. Also how do you setup the name part of callerid for the outgoing lines?

Hi, We have a case where we need of 16 x FXS, 12 x FXO and 1 x E1. To do this using Digium products I need of 8 PCI slots. This is not possible to be done in one computer and that's why I try to start using TDMoE. Unfortunately all my tries are without success. The network is crashed everytime. Can you give me some ideas/suggestions how to solve this case? Best Regards,

Hello There, I am trying to get Asterisk to work with Billion ISDN Adaptor, But i couldnt get isdn4linux to work. I am pretty new with isdn card but this is the only available option here right now , I've looked at this post and found that the author has been successful in installation of the same card. Can someone please help me out by giving the author's email address , so i can

Dear Asterisk Group. I have two Asterisk servers serving two data/help desk centers, both centers have a near identical setup. However, when connected to one of my data centers, I call a user, I can see on the CLI that the phone is ringing, but I hear no ringing on my SIP soft phone? Has anyone had a similar scenario? How as it resolved. Warm Regards Shad Mortazavi

Hello, Can someone explain me how to do caller based routing. Here is my example. I have an asterisk between a PBX and the PSTN. The second company get the same, and so, I can interconnect them by VoIP. Classic architecture. My problem is when I want to place fax. The calls between the 2 sites are in gsm codec. So the fax doesn?t work! Is there any possibilities to do caller based routing in

hello, i have a problem starting asterisk with one x100p digium and two hfc chipset isdn cards with bri-stuff.0.0.2. "ztcfg -vv" shows me a this info: Zaptel Configuration ====================== SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03:

Hello, Has anyone used Asterisk in conjunction with a billing system like Rodopi? Is the Rodopi VOIP module worth getting, or can radius be used? Thanks, - Darren

Hello, I would like to buy a SIP IP phone, but I don't know wich one to choose... Can you tell me wich IP Phone is known to work with Asterisk please. I've seen the Cisco 7940, but I don't know if it works, and how expensive is it ? I'm french, so if you know some french resellers, tell me. Thanks a lot, ---------------------- Fabrice Tereszkiewicz Sawadka.org

Hi, I have a couple of questions about MeetMe and call queues. I'm still pretty new to Asterisk, but already having to write a Service Center call manager for it (which I might add, our director has agreed to make open source!). MeetMe: How can I get MeetMe (does it even do this) to ask the user to speak their name first, and play that as the new member announcement. It seems