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Hi, Does OCFS require NAS hardware to run or does normal PC hard disk work? Thanks, Pete -------------- next part -------------- An HTML attachment was scrubbed... URL: http://oss.oracle.com/pipermail/ocfs2-users/attachments/20081219/1bc9110b/attachment.html

...cstorage=asterisk odbctable=voicemessages [default] ; Syntax for new entries looks like this: ; MailboxNumber => password,name,e-mail,pager,options ; (usually, the MailboxNumber is the same as the Extension) 2000 => 1234,Dave Robinson,outraspace at hotmail.com 2001 => 1234,Colleen Robinson,petedao at gmail.com 2002 => 1234,Matthew Robinson,outraspace at hotmail.com 2003 => 1234,Lisa Robinson,petedao at gmail.com,,delete=yes Here is my res_odbc.conf [ENV] INFORMIXSERVER => my_special_database INFORMIXDIR => /opt/informix ; All other sections are arbitrary names for database conn...

...s. I'm remote at the moment so I can't send you the code but google for mobile remote receiver and you'll find what you are looking for. Lots of people do it so they don't have calls to cell phones picked up by voicemail. Cheers dean -----Original Message----- From: Pete Kay <petedao at gmail.com> Sent: Monday, March 31, 2008 2:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: [asterisk-users] How to give user a prompt before connecting thecall Hello, Is it possible to request for the premission from the c...

Hi, I am havind deep trouble with a bunch of our newly arrived Optiplex 330 as it can't run Centos 5.2 property. The installation works fine, but when it boots up, it can't be connected to the network. I am getting error saying " link is not ready" when doing system-config-network. I check lspci and it can detect the network controller no problem. The light next to the cable

Hi, I am running a VOIP application on Centos 5.3. It is a 16 core box with 12 G of mem and all what it does is passing packets. What happens is that at around 2K channels running g711 ( 64k) codec, all eth0 is used up and no more traffic can go through. I have checked google and it talked about interrupt scheduler. does anyone know how to configure the kernel to allow it to use all CPSs for

Hi, I am looking to buy 2 used 1 or 4 ports E1 Cards. If you have one, would you please contact me? Thanks, Pete -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081217/48afd68d/attachment.htm

Hi, I am having problem with my Asterisk installation and find out it has to do with ztdummy. if the ztdummy module is loaded, the asterisk playback() command will not play files. DTMF is still properly received. If the ztdummy module is unloaded, sound playback works again. Here is my version zaptel-1.4.9.2 linux-source-2.6.18 asterisk-1.4.18 Can anyone tell me how to fix it? Or should I

Hi, I am sorry my questinos are too fundamental. I am new to Asterisk, and hope to catch up as fast as I can. Problem 1: I have my SIP client ( in one PC .102) and SIP server ( in another PC .101) within the same land. They can make SIP connection, but when the SIP client makes call to play an audio file, I can only hear a "beat" sounds, and then nothing else. In the console, I can

Hi, I have to install Broadcom driver because the Dell Optiplex 330 running Centos 5.2 is not able to connect to the network. I am trying to install a Broadcom driver, but I get the followng error: [root at localhost tg3-3.85l]# [root at localhost tg3-3.85l]# make make -C /lib/modules/2.6.18-92.el5/build SUBDIRS=/usr/src/Server/Linux/Driver/tg3-3.85l modules make[1]: Entering directory

Hi, I am working on setting up the voice mail. I can get message recored and can find the .wav file created. However, when I tried to play back, I can't hear anything. In the CLI, it does say: -- <SIP/2000-081ed640> Playing '/var/spool/asterisk/voicemail/default/2000/Old/msg0000' (language 'en') I am wondering if I did anything wrong in my setup that causes this

Hi, I want to try detecting if a call is a fax from Zap/1 channel and if it is, forward it to a fax number. How to do it? I have iaxmodem and Hylax working, but it can only receive, but not redirect the fax call. Also, I have read that Asteris has a tool call rxFax. Could someone help me to understanding the difference in terms of functionality between rxfax and hyfax? Which one is better?

Hi I got my hylafax running to receive and send fax. Since I only have one number, I want to know if there is anyway to detect if a call signal is fax then redirect it to fax-to-email, otherwise route the call to my analog phone? Is it something that can be done? Thank you in advance for your help. Pete -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I am having difficulty running Avantfax on Debian. When I try to launch the web UI, I get a whole page of PHP codes. It looks like my apache is not recognizing the PHP file. However, I am able to run phpmyadmin no problem which proves that apache2 is working with PHP. Any idea? Thanks, Pete -------------- next part -------------- An HTML attachment was scrubbed... URL:

My problem with Avantfax on Debian is resolved. It is just a simple dumb permission problem. Sorry to bother everyone. Pete -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080325/0b50b467/attachment.htm

Hi all, I want to allow different users to have their own unique MOH. Is there anyway to do it? Asterisk does not have a realtime MOH feature but I am wondering if there is anyway to get around it? Thank you for your suggestion. Thanks, Pete -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I am hoping someone can help me out on this. I want to be able to interrupt MOH every X seconds after the DIAL command is executed. The interrupt greeting is something like "please wait while we transfer your call". How can I do that? Within the DIAL options, I can't see any announce frequency or options that can help. Could anyone please tell me how that function can

Hi all, Is there anyway to have Asterisk to play greetings of different language based on the local/time zone of the user? I want to have the email body message to be sent using different languages based on the user's country. Also, users from different country should be hearing country-specific greeting in VoicemailMain. Is there anyway to do that with Asterisk? Thank you very much for all

Hi, Does anyone know how to turn on TOE ( TCP offload engine ) and RSS ( Receive-Side Scaling)? Thanks, pete

Hi, I would like to be able to log call details in Asterisk. The kind of logs that I like to generate is like this: From To Forward Time Incoming Call 604-343-3334 503-233-4454 13:33:32 Extension Routing 503-233-4454 Extension 403

Hi, I am new to Asterisk and I am having a setup problem that I am trying to resolved for the last couple days without any success. I am pretty much desperated on this issue and I don't know why. Can someone please kindly help me to troubleshoot this? I can't hear any audio from Asterisk when running Playback or VoiceMail tests. I have my Asterisk server ( running on Debian,