similar to: Installing asterisk on a server vs appliance(e.g digium mypbx)

Hello, First let me apologize for posting about a GUI topic on here. There's a reason why I did that, and it's because the underlying concept of this is connected to Asterisk.Here's my situation: Twenty wifi clients connecting to our wireless router (Cisco Linksys E4200 loaded with Tomato). All these WiFi clients are running eyeBeam (in case you're wondering where the calls come

Hellos I am using freepbx and asterisk. I am writing an AGI script to edit the values in findmefollow table. The script will enable users to delete and add follow me numbers from their phones. I want it to enable users enable/disable follow me. I can't seem to find a value in the database that deals with enabling/disabling followme. Please help -- Best Regards, James Mutuku Ndeti Agile

Hellos, I am looking for the sounds used in this ivr example http://www.voip-info.org/wiki/view/Asterisk+Tips+follow+me. The one with 6900. Any assistance is welcome. -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php

I am running asterisk and I want to achieve the following scenario My goal in the end is to achieve the scenario (example using extension A and Extension B) 1. Extension A has a line apperance of 4(4 calls can ring on it). 2. Extension B calls extension A(which is busy on one of the lines). 3. Extension A sees the second light blinking and hears the beeps (currently working). 4. Extension B is

Hello, I have 2 sites. One(Site 1) has an asterisk PBx and the Other(site 2) has 2 remote soft phones. The latency btw both sites is btw 500ms-700ms. I know this is a shot in the dark...but are there ways of improving the voice quality for the remote calls(btw site 1 and site 2), Other than increasing bandwidth? -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994

This is currently still at a proof of concept stage. After being mis-sold a Alcatel phone system, that does None of the things we asked for.... (Ok it takes calls but that's about it) We are looking at alternatives to try and bring some of the features we previously had on our old Analogue STS phone system. Looking at all the docs I can find Asterisks looks like it should be able to do the

Hellos, My voip carrier has assigned me a IP based account...where they only give me the IP to call through. I have setup the dial plan exten => _7XXX.,1,Answer() exten => _7XXX.,2,vmauthenticate(${CALLERID(number)}) exten => _7XXX.,3,Dial(SIP/${EXTEN:1}@Y.Y.Y.Y) exten => _7XXX.,4,Hungup() Where Y.Y.Y.Y is the assigned IP. After Dialing I asterisk logs the error SIP/Y.Y.Y.Y-35dc

Hellos, I want to setup Asterisk+a2billing for over 10,000 extensions for voip resale. Has anyone done this before. What are the hardware requirements and challenges? James -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090513/186b2bde/attachment.htm

Hellos, I have using asterisk 1.2 and freepbx 2.3. I need users to disable and enable followme from there phones. I can't see any support for it. Is this possible/available.? I have googled and I can't get information on it -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer

Hellos, I need to send personal directory from asterisk to the ersonal directory of the linksys spa 942. Is this possible? -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve

I am learning agi scripting using php. I m using phpagi 2.x on asterisk 1.2. I hve written a simple script that reads out the callerid using flite. My problem is that I seems the script is not getting the callerID. Bellow is the script _________________ #!/usr/bin/php -q <?php /** * @package phpAGI_examples * @version 2.0 */ set_time_limit(30);

Hello, I am writing an AGI script to achieve the following - Users can Disable/Enable followme from their extension. They can also change the followme details from their extensions. I have looked at the follow me table for freepbx. I can't see the field for the values enabling/disable followme. Is this value stored in the database? -- Best Regards, James Mutuku Ndeti Agile Systems

Hellos, I have asterisk 1.2 and freepbx 2.3. I have edited the agi script(dialparties.agi). Everytime I restart asterisk, the file gets overwritten. How do I make sure my changes are not overwritten? What generates dialparties.agi? Thanks -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer

Hellos, I am having issues with my meetme conferencing. When I dial the conferencing number, It hangs after a few seconds.I have read somewhere that I need to enable ztdummy, which I have done but still no changes. Here is my log ~=~=~=~=~=~=~=~=~=~= PuTTY log 2009.08.12 18:44:43 =~=~=~=~=~=~=~=~=~=~=~= -- Executing [1;36;40mMacro [0;37;40m(" [1;35;40mSIP/1215-fc5b [0;37;40m",

Hellos, I am having issues with my meetme conferencing. When I dial the conferencing number, It hangs after a few seconds.I have read somewhere that I need to enable ztdummy, which I have done but still no changes. Here is my log ~=~=~=~=~=~=~=~=~=~= PuTTY log 2009.08.12 18:44:43 =~=~=~=~=~=~=~=~=~=~=~= -- Executing [1;36;40mMacro[0;37;40m("[1;35;40mSIP/1215-fc5b[0;37;40m",

Hi, I have two asterisk servers with the same version of 1.4.29.1. The first server named it as MYE1. MYE1 is an incoming server that can accept incoming calls from PSTN(ZAP E1). The second server is a pbx functions server and named it as MYPBX(SIP). The sip.conf of MYE1 likes below: [MYPBX] type=peer host=mypbx.abc.com nat=no disallow=all allow=g729 canreinvite=yes qualify=no context=default

In particular, I'm comparing these two models: Yeastar NeoGate TG100 vs. Dinstar DWG2000-1G http://www.yeastar.com/products/NeoGate-TG100.asp http://www.dinstar.com/Product/Product_25.aspx?typeid=6 Wich model do you recommend me, Yeastar or Dinstar? Thanks in advance. -- Usuario Linux Registrado # 342019 --> http://linuxcounter.net/ <-- skype --> luedcortes gtalk -->

I am in the process of buying a TDM800 card from Yeastar ( http://www.yeastar.com/products.asp?TypeName=TDM800%20PCI%20Card&cTypeName=1 ) Any one has tested this cards? How reliable are them? I am specially interested in the FXO/FXS module. -- Gustavo Felisberto (HumpBack) Web: http://dev.gentoo.org/~humpback Blog: http://blog.felisberto.net/ ------------ It's most certainly GNU/Linux,

Hello, I have been trying to get a TrippLite Internet Office 750 UPS to talk to my Linux PBX for a couple of evenings now and I'm getting nowhere... I tried searching the list archives before posting here (I'm sure I'm not the first one to try to get this going) but they seem to be offline... The UPS is unfortunately USB based but I thought I'd give it a try anyway. Here is

Hello my friends, I'm very worry about a problem i'm having...my asterisk got freez some times, every 5 or 6 days with NO trace in /var/log/asterisk/messages What i want to know is if safe_asterisk has something to be with this? This is what i have on my server: [root at mypbx ~]# ps -A | grep asterisk 9118 ? 00:01:30 asterisk [root at dreampbx ~]# ps aux | grep asterisk root