similar to: Beginner couple of questions

> 3. Supposed I have 2 fxo cards (right now I have one already) and 3 > fxs, and one of the fxo will have two phone (running pararell), is > there any way for * to: > a. It always dial the first fxo, if the fxo is busy or is being used > (have other people conversation), will * be able to switch it to > other fxo? Here's the approximiate the conditions of the phone.

Dear All, I''m using the kernel 2.6.6, iproute2-2.4.7.20020116, iptables v1.2.9, and gentoo. I have a leased-line 64 kbps. I can see the counter works in iptables, but in the htb, it doesn''t go to the right class (it always go to the default class). Any help will be appreciated here''s my htb conf #!/bin/bash tc qdisc del dev eth1 root tc qdisc add dev eth1 root

you can do that. But are u installing qmail and * on same box. i wont recommend that. i use qmail and *. qmail is strictly for internet email. * is on separate server not exposed to Internet. * box also has sendmail. i hv configured sendmail to use smart host (qmail server). This way its safe and secure. HTH, -B ----- Original Message ----- From: "Ing Isianto Istiadi"

Dear all, I have a question regarding the configuration of *. I have 3 port FXS, and 2 port FXO. I have 4 users that use analog phone connected to FXS (I have 3 phones). I need to limit the user's capability (user A can call International, user B can call long distance, etc). I want to implement the password say to call , he/she needs to puch 9(for outgoing call)2-4 digits password,then

Hy guys, I cannot understand How the asterisk works. I would like know how the h323.conf, sip.conf and extension.conf works. I don't understand the parameters and the [sections]. What I need to the asterisk get a SIP call and forward them to a H323 terminal. I working at the h323.conf and extension.conf but I cannot understand!!! Please someone can help me. I your can send me a example (with

This newbie has been trying out Asterisk. It has been both a) surprisingly painful and b) impressive in terms of helpful support from other users. Having got two phones to communicate and then got voicemail MWI going (neither painlessly) I decided the next step was to implement call transfer as per nearly all commercial PBX systems i.e. hold call consult another extension either exit and let

-- I get the following error message when I try to turn the RPM file into a tgz file. root@wizard:/home/vince# rpm2tgz winetools-0.9-3joi386.rpm /usr/bin/rpm2tgz: line 98: winetools-0.9-3joi386.rpm: No such file or directory /usr/bin/rpm2tgz: line 99: winetools-0.9-3joi386.rpm: No such file or directory Ultima Linux rocks: http://www.ultimalinux.com/ultima-central.html

No, DON'T use mpg321, it will not work. Make sure you're using mpg123. See the wiki. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Rich Adamson > Sent: Friday, October 01, 2004 12:13 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re:

I can't seem to get musiconhold to work. I'm running asterisk on a RH9 box, I have the mpg123 package installed. In my zapata.conf file I have the line MusicOnHold=default . In my musiconhold.conf file, in the classes section I uncommented default and loud. In my extensions.conf file I have a set musiconhold line. However if I get a call and I either put it on hold or hit flash

I've followed the docs on how to configure musiconhold. I then added an extension to test it like this: exten => 6601,1,WaitMusicOnHold(30) And when I dial that extension I do hear something for 30 seconds - but it is not what I'd call music... it sounds more like the sound effects from a scifi flick - now and then it sounds like jail cell doors closing and echoing. Kinda cool but

I have the following in extensions.conf: [global] MP3ROOT=/var/lib/asterisk/mohmp3 [default] exten => 1111,1,Answer ; Answer the line exten => 1111,2,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten => 1111,3,MP3Player(${MP3ROOT}/sample-hold.mp3) The command that runs is: 14030 pts/0 S 0:00 /usr/bin/mpg123 -q -s -b 1024 --mono -r 8000

Hi, I have P4 2.8 HT machine with Asterisk. I spot 3 processes of mpg321 taking 20% of CPU each even when no call is made to Asterisk or any other PBX activity is on ? Is this behaviour normal ? Regards, Robert.

Does anyone have use for a broadcast/annouce app? I wrote SystemAnnounce which will play a specified file to all active channels (in an UP or bridged state). This was originally to tell users to get off the system, but there are several other uses... I also wrote a new CallPickup and CallPark app, both of which work more as expected (supply actual extension numbers, etc). Let me know if there

I am trying to music on hold but I am having all sorts of problems with it. I am running RH9 and the latest version of Asterisk as of yesterday. Here is what I did to test it: 1. I manually deleted the mpg123 softlink to mpg321. 2. I downloaded mpg123-0.59r-1.n0i.src.rpm, compiled and installed the the archive and loaded ztdummy.o module. 3. I threw a couple extra files in the mohmp3 directory

I'm pretty much a newbie to this but still think I've been around the various help pages, voip-info.org etc to be fairly sure I'm not missing something here so your help is appreciated! I have a box running RedHat9 at home with the latest CVS of Asterisk and all works fine. At the office, we installed Gentoo linux on a machine, downloaded the latest CVS of Asterisk, set it up. All

Has anyone managed to set up a moh server for Asterisk? Reason would be to offload processing off the asterisk box, onto another system. The wiki is a bit light on details. If anyone managed to get it up and working, what software did you use on the server side, and what client app did you use? Mpg123? Mpg321? Madplayer? Something else? Also, putting legal ramifications aside, it'd be nifty

This was just a recent personal experience.... Maybe I missed a thread on this: We recently installed asterisk (CVS-HEAD) on a debian system using 2.6 kernel and the enhanced RTC for all timing. Also a custom compiled kernel for the CPU on the box (P4). We had a strange thing happen in that with Debian's MPG123 package: Sound files played in asterisk/mpg123 were heard at literally 1/10th

Hi, I'm using CentOS 5.1 on both server and desktops. I have a series of packages that I either rebuilt from CentOS SRPMS to modify them according to my needs (PHP for example), or that I built from Fedora Core SRPMS because they're not available on CentOS nor in the RPMForge repos (Seamonkey, MPD with libshout support, xmms-wma, ...). I want to create my own remote Yum repository

Hi, I'm banging my head over this. Usually, I'm using a SIP hardphone feature called "Call Pickup Starcode" to enhance BLF with Directed Call Pickup : basically, SIP hardphone (here a Thomson ST2030S) is configured to send an INVITE message whenever a BLF is pressed while blinking. The INVITE is build with the extension number (attached to the BLF that was blinking and pressed)

I have read every thread, I have Redhat 9, Asterisk 1.0.6, 2 T1 lines connected via TE405P. Everything works great, except MOH. I added an exten with MusicOnHold(30), and it plays just fine. Conferences have music when no one is in. I have SIP phones. When I place a call on hold, the CLI give no indication the call is on hold. I have set musiconhold(default) everywhere, removed it from everywhere,