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Hi List; Did any one tried the H.323 module? How much it is stable and work fine? Regards, ------------ ITS IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 00965 9849460 ____________________________________________________________________________________Ready for the edge of your seat? Check out tonight's top picks on Yahoo! TV. http://tv.yahoo.com/

Hi People, We use the Grandstream GXP-2000 phones, firmware 1.1.1.14, Asterisk PBX, Slackware Linux 10.2, loaded on a Intel(R) Pentium(R) 4 CPU 2.66GHz Box... The issues that we are experiencing involves our Telephone Operator's/Receptionist whom answer multiple incoming calls... As an example.., when they answer line 1 and Line 2 starts to ring they would ask the person on line 1 to

Hello list, Can someone please point me out why would a stream like the following only write ONE line (the first) on the given file? Action: login Username: test Secret: 123456 Action: UpdateConfig SrcFilename: voicemail2.conf DstFilename: voicemail2.conf Action-000000: Append Cat-000000: default Var-000000: 127 Value-000000: >5555, Jason Bourne97, jason25 at noCia.gov.do ActionID:

...us/img219/7207/linksys3102cid1jj7.jpg http://img219.imageshack.us/img219/4625/linksys3102cid2ld5.jpg Does Zaptel support those on Digium TDM400 clones like those from OpenVox? Thank you. ------------------------------ Message: 9 Date: Mon, 22 Oct 2007 15:35:40 -0400 From: Rurouni Alucard <rakh at dangerclan.net> Subject: Re: [asterisk-users] Authenticate by IP? To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Message-ID: <471CFB8C.7070200 at dangerclan.net> Content-Type: text/plain; charset="utf-8" Saludos Carlos,...

Hi Folks, I dont know how exactly to start... so im going to (what i think is) the point... In a dialplan, after i set an autohangup (with AGI) , how could i send a sound (stream a sound ) into an open channel at X seconds before the autohangup time get to 0 for that channel? (Like public phones, that gives u a 'beep!!!' before ur time runs out, just like that...) Thank you very

Hi list, I have been experiencing a strange behavior with asterisk and i would like to know if someone else has face it. This is my scenario, 3 extensions created on sip.conf: 121 | 123 | 123 Everything work just perfect except for the following issue: I have this block on my extensions.conf [record] ;-------Extensiones individuales exten =>

Hello list, I'm using a PHP script to communicate with asterisk via AMI, and edit configuration files. So far everything went ok, but I came up with a little problem editing extensions.conf using 'updateconfig'. Is it possible to edit an existing line in extensions.conf file?, e.g. Given a 'category' in my extensions.conf as follow: [macro-mytest] exten =>

Hello List, I have been working on a PHP application in order to build a BLF style script. Until now everything is going Ok but something a little (in my oppinion) strange is going on with the 'ExtensionState' command; The problem is that it does not returns the 'Status' as it's suposed to, mentioned in the A.T.F.O.T book - version 2., where it sais something like: