similar to: Help with call scenario

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

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

Hi, I am looking into advising a client on the pro's and cons of using Installing asterisk on a server vs appliance(e.g digium mypbx). the appliance seems cheaper initially.

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

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

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

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, 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",

Hello; I need two scenarios: 1) If someone sent SMS message, then we need to query information from the database based on information sent by the SMS (like the name or the mobile number), after querying from the database, we need to reply by the SMS. Can asterisk do this? To which level? 2) I have vtiger CRM, and it is possible to send SMS (there is a button: send SMS), the question is: what is

Hello,

Hi, I noticed that asterisk manager interface will only accept the originate commands in sequential order. For example, if I want to ring two extensions through the AMI, and while first extension is ringing, AMI won't execute and ring second extension until first extension has answered the call. Anybody has any ideas as I had the same results even tested with telnet commands to AMI interface.

Hello(s), I know this might be test book question or one best suited for google but I will take the risk of asking. Here I go. What common routine maintenance tasks do you run on your asterisk box? Thanks James. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090422/8cfeb434/attachment.htm

Fix compilation of Pacemaker glue for ocfs2_controld when the underlying Pacemaker installation supports both the Heartbeat and OpenAIS stack Signed-off-by: Andrew Beekhof <abeekhof at suse.de> --- upstream/ocfs2_controld/pacemaker.c 2008-09-11 16:51:11.000000000 +0200 +++ dev/ocfs2_controld/pacemaker.c 2008-10-23 13:14:56.000000000 +0200 @@ -20,8 +20,16 @@ #include

Hopefully someone can help with a bit of lighttpd configuration for my production rails app in Mac OS X. I have an application which will be accessed by many different domains and IP addresses. Examples: http://crm.local/ http://localhost/ http://192.168.0.49/ http://65.256.120.90/ (fake, but you get the idea, an external Internet facing IP address) http://crm.mydomain.net/