similar to: FreePBX in production?

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I'm in charge of two Asterisk servers. One of these servers is running an Asterisk@Home version under CentOS, the other one is running Debian Sarge. The Debian server has been installed by me using this howto: http://www.astpp.org/index.php?n=ASTPP.DebianSarge-AMP On both servers, the AMP version is 1.10.010. However, the

Hi All: I used FreePBX to configure Asterisk, and tables are create in MySQL by using FreePBX install script. I created two x-lite softphone accounts by using FreePBX, they are stored in table sip as friend. I followed wiki doc to edit the extconfig.conf file. I can not get those two softphone to talk since I got the error message from Xlite as: Call failed: 503 service Unavailable I noticed

Hi, I have lookedaround and don't see this anywhere. Is there a way to tell the ip500 to not make the aural MWI blips?

Does anyone know if FreePBX can be installed on a Linux box that was built using Asterisk@Home. I would prefer to manage Asterisk with FreePBX over the AAH build. I have just not had good luck building an Asterisk system from scratch and the Centos based Amp ISO and prebuilt config files are a wonderful place to start. Nothing against Asterisk or Linux. My build from scratch issues are only

Hi I've got a dell 2550 with an Eicon Diva server PRI card plugged into it. I can call out using the acopy2 test utility. I'm having trouble with asterisk making calls however... my capi.conf and modules.conf looks correct by the wiki instructions - does anyone have any advice on where to look ? I can attach conf files etc. if needed. Asterisk says it has 30 capi channels available,

We are moving our asterisk 1.0 system to a new Asterisk @Home system (2.8) and I am the one in charge of doing it. I have run into a snag, though, on meetme conferences and with the transfer key. Regarding the transfer, it appears that both directions of all calls can transfer by pressing the # key. I do not like that ability. I would like to change it by doing 2 things: 1. Make the transfer

Hi, I'm trying to edit an AMP-derived dialplan: the macro "dial" uses the AGI script "dialparties.agi" to find the extension to call. I'd like to drop this script: does anyone can explain me what is its main job? Thanks -- Domenico Viggiani

Ok so I added to my Freepbx config running Asterisk 1.2.4 in extensions_custom.conf ; intercom exten => _7XXX,1,SIPAddHeader(Alert-Info: Auto Answer) exten => _7XXX,2,Dial(SIP/${EXTEN:1},12,Tt) and configured my Polycoms via this page http://www.voip-info.org/wiki/view/Polycom+auto-answer+config for auto answer and that works fine if I dial 7 then the 3 digit extension. No

I am intending to rebuild our asterisk server on a Dell 2800 with Dual CPUs. I am currently using Centos release 3.5 (Final) with the 2.4.21-32.0.1.ELsmp Kernel on a Dell 1750. I have a TE410P (Quad PRI) card and a TDM card. I remember that there were issues with the 2.6 Kernel, but Iam sure that it was with the older asterisk builds. Also, the new server has the X64 intel CPUs, so I would like

Figuring I knew what I was doing (I didn't - surprise) I added a totally unnecessary line in /etc/asterisk/extensions_additional.conf a couple of days ago. Troubleshooting a dialing rule issue, I'm now realizing that FreePBX is updating its database with the new settings but is not rewriting/updating extensions_additional.conf with the changes I'm making. I've tried renaming the

Does FreePBX support virtualization of its services? For example, can I use it to provide virtual PBX to different clients under the same instance of FreePBX? Or is it more geared to single office-type installation? Thanks, Daniel

Is anyone using FreePBX in production level systems because I'm just wondering if its stable enough to use. Currently I'm editing my own *.conf scripts but it sure would be nice if there were some sort of web interface for other people to use. The only thing holding me back is the stability of the FreePBX package... Any comments on this? Thanks in advance. Regards, Min Chang

Hello- I have a low traffic AAH setup, a few hardphones, a few softphones, 50 calls per day max. I used the AMP Digital Receptionist to make a simple voice menu: "Thank you for calling xxxx". I did this for both Normal times and After Hours times. It worked fine. I then went to the AMP Maintenance window, Config Edit, got the "phpconfig for Asterisk PBX" page, and selected

Hi, I'm actually using a slightly old version of AAH with Asterisk 1.2.1, because at first install it was perfect for my moderate knowledge of Asterisk. It is working well but I gradually introduced many changes to dialplan during normal use and now I'm feeling like in a straitjacket! Moreover I'd like to have the chance to upgrade Asterisk regularly. I have not the experience to

Hey guys, Is anyone aware of a single PRI (E1) card that has onboard hardware echo cancellation? I can only seem to find 4-port cards (Digium and Sangoma) and I really don't need the other three ports. Thanks, Avi -- National Manager - Special Projects < Sydney / Melbourne / Canberra / Hobart / London /> 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F:

Hi I have a TDM card with 4 lines on a hunt group coming in. I can answer the phones with exten => s,1,Answer() exten => s,n,Dial(Zap....&Zap....) ... The problem is I don't know how to find out what extension was originally dialed. And, trying to match on the extension always fails. E.g. exten => 1234567,1,Answer() # never gets here I thought I could get the extension on

Hi People, I'm writing to get some advice on where to start when learning asterisk? I was going to begin learning with AAH but I wanted to find out if there is a certain build to avoid or if there is a Gui/front end that is better then another. I have been working with dialogic cards for the past 5 years and with auto dialers but I want to get into providing voip service, support and

Had a Pent 4 server running fc3 crash (kernel panic) and am rebuilding from scratch. I installed FreePBX (CentOs) from scratch and asterisk was running, but had not yet been configured. It too crashed with a kernel panic. Ran memtest for 24 hours; no errors or issues uncovered. I then noticed that FreePBX installed using a SMP kernel (and grub indicated a non-SMP kernel was installed as

Hey guys This is probably for Armin, but I thought maybe someone else might have something I could try. I'm having a problem with one of my Eicon Diva V-4BRI cards and I'm trying to work out if its a driver configuration error, card failure or telco problem: I have an Asterisk box running -- the Eicon drivers see all four ports on my card, capi.conf is configured with all four ports

Hi I make a Digital Receptionist ,but how can I attach it to an extension Help regards ********************************************* No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic