similar to: Asterisk/FreePBX/Alcatel2400

Hello, I have 2 analogue ports on an Alcatel PBX patched to 2 FXO ports on my <mailto:*@home> *@home 2.8 running on top of CentOS. Both FXO Ports are on ONE Digium card. What I would like is: If someone calls extn 281 on my Alcatel PBX it is routed through to Extn 233 on my * thruogh FXO port/module 4 If someone calls extn 282 on my Alcatel PBX it is routed through to Extn 234 on my *

Are there any Snom 200 users that have two extns defined on their phone? I've been trying to get two (or more) extns defined in such a way that when extn #1 rings, LED #1 flashs; extn #2 rings, LED #2 flashes, etc. (Answer greating will be different depending upon which extension is called.) I can get multiple extns to register and work with * just fine, but regardless of which extn is

i am using the freebsd port, which seems to not yet have WaitExten(), which i kinda want to use thusly [ext-666] exten => _.,1,SetVar(areacode=666) exten => _.,2,Background(zz-in-who) ; give them list of extns exten => _.,3,WaitExten(10) ; let them enter extn to call include => extensions include => applications include => speeddials

Hello all, Just got hold of a CISCO 7960G. Updated the Firmware to the latest 8.2. As you all know it has 6 lines which is why i bought it. Just would like to know from you experts if this piece will connect to 6 different providers over the internet or will it only work as 6 extns with 1 provider. Im not able to get connected to more than one provider. Also i have found that the time is not

Hello, I'm trying to create a source package (on Win2k system). I followed the instructions from R-extns.pdf, installed ActivePerl and RTools with setting the DOS path to it. Then I did "Rcmd check" and had an error: ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ C:\R\rw2001\bin>Rcmd check D:/prime7/SpnInstall/spectrino * checking for working latex ...latex: not found NO * using log directory

Hi , I;ve Asterisk 1.6.0 with static agents (sip softphones with extns 100 & 101 ) in a queue..When a caller arrives in queue , it lands on first 100 , 100 then does a blind transfer to 101 .. so that the caller can converse with 101 .. strangely enough the queue_log shows : 1253814090|1253814090.12|55365|NONE|ENTERQUEUE||98221232123

Please forgive me if this is a silly question. I've been following this thread in the hope that I could put my * server and snom 200 into full-time service very soon. I need to find out how to have the lines configured so that it does not return a busy reply when only one call instances is engaged. Am I supposed to create multiple extensions on my asterisk dialplan to reflect the 5 call

Hy, Does anybody knows how to show the digium addons in the freepbx GUI. The module is available in the GUI but sadly empty! Everything seems to be correctly installed bute the tables in the database are totally empty. Is there any script anywhere to fill those digium tables? Working with latest freepbx framework 2.9. regards Florent THOMAS -------------- next part -------------- An HTML

Hi there, Any of you know how can i configure two PBX servers with FreePBX with load balancing? Do I need a SIP server like OpenSER? Any of you have some configuration reference? I have installed AsteriskNow in both FreePBX servers. Thank you! Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, I am trying to install Asterisk with FreePBX while running install_amp following error is coming can any one help in this regards Thanks in advance.. Linga Reddy Connecting to database..OK Connecting to Asterisk manager interface..OK DB Error: no such tableGenerating AMP configs..OK Restarting Flash Operator Panel..OK

Hi, All Here is the install script that I promised. Let me know if there are any problems. I did not include nv_faxdetect in this yet, I am still working on that. The hylafax-iaxmodem script I posted earlier should work on CentOS 5. http://jonnt.users.taylortelephone.com/asterisk/centos5-asterisk1.4-install.sh Jonn

Hi, i use asterisk with freePbx for all configurations. Now i want use a openser with asterisk but also with freepbx. I pretend use the asterisk whit freepbx, but autentications for users in openSer.. it's possible?? thanks -- Carlos Jer?nimo

Hi all, I've just installed again my Asterisk using Xorcom repositories. I can make extensions, but when using any extension i want to dial anything, I got "404 not found" using Xlite. Any ideas of what can be happening? Regards, -- Diego Quintana a.k.a. RouterMaN Ingeniero de las Telecomunicaciones Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676 Ext.

Hi Everyone, I am looking into AMI (using PHP) to record every instance of HOLD that is generated by putting a caller on HOLD (press hold button on the phone set). There is no HOLD in Asterisk but the event Music on Hold is generated when HOLD is pressed. The complexity is that all of the the calls are handled by FreePBX so I don't have the channel IDs etc... Can someone please point out how

All my experience on asterisk was gained through configuring Trixbox, so a ready to go Asterisk system. Now i?m trying to install a server from scratch, so this question arose. When installing freepbx, the .conf files are written, when installing asterisk nothing is written unless I run the Make Samples. 1. Basic features like parking calls, transfering calls, contexts, and so on have to

Hello, This is what what I see after a Yum install asterisk16 asterisk16-config freepbx: Use of uninitialized value in string ne at /var/www/html/panel/op_server.plline 4997. Use of uninitialized value in substitution (s///) at /var/www/html/panel/ op_server.pl line 5439. Use of uninitialized value in substitution (s///) at /var/www/html/panel/ op_server.pl line 5440. Use of uninitialized value

I am experimenting with Asterisk having downloaded and installed the FreePBX i386 CentOS-6.3 based distro and updated it. The current package level on this system is: asterisk11-11.3.0-49_centos6 freepbx-2.11.0beta2-112 I am using twinkle-1.4.2-7.el6 as a softphone testing tool. There is no firewall on the asterisk host and SELinux is disabled on it. Fail2Ban is installed but I have made no

Please does anyone have Freepbx or Trixbox Powerpoint Presentation? Thanks _________________________________________________________________ Connect to the next generation of MSN Messenger? http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=wlmailtagline -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, Looking for Asterisk-FreePBX in a Flash Technical Support in Tampa Please contact me by phone: 813-842-6941 Thank You, [cid:image001.gif at 01CC76D3.D2584900] Secure2ware Inc. 813-425-5900 Keith A. Ware, ext. 211 Keith at secure2ware.com<mailto:Keith at secure2ware.com> www.secure2ware.com<http://www.secure2ware.com> -------------- next part -------------- An HTML

Hello everyone, I was wondering if anyone could share a video on how to install OPUS from scratch to server, I´m new to Linux but since we have now freepbx (Asterisk 1.8.32.2 and 11.23.0) I´m force to look for alternatives to improve voice quality on an extreme budget, and more than that to learn new and better codecs for free PBX. So far I have downloaded opus-1.1.3.tar, installed on on my