similar to: Delay Before Join announcement

Hello, I have Elastix ISO install (FreePBX 2.7.0.3) My current Setup is as follows: Inbound Route > Queue > (Dynamic Agents) The queue distributes calls based on rrMemory. I have been asked to redesign the call distribution as follows: Calls will be delievered to Level-1 Agents (say 4 dynamic agents) in rrMemory format. When Level-1 Agents are busy, distribute calls to Level-2 Agents

Hi Pan & Dhaval, In the past 8 weeks, we have delivered a load-balanced asterisks (1.4) based call center with our flexqueue application for icson.com. It has the below features, 1. 2 x asterisk 1.4 boxes, 1 x mysql db box and 1 x flexqueue box(the two are failover configured with heartbeat and custom script, and mysql master-slave replication between two svr), 2 x kamailio boxes(failover

Hi, I would like to see the DNID in my MySQL CDR logs. I have read one big thread in the Asterisk Developer List, but I could not figure out how to implement it ? Is there a simple step-by-step. Thx in advance. Vai

Hello all, The Asterisk Management Portal (AMP) is now known as FreePBX. FreePBX 2.0.1 is now available for download. A **BIG** thank you goes out to the project developers for all their hard work, and to beta testers for running FreePBX through it's paces! This exciting new release boasts a better user experience, additional functionality, and a new module system. The module system is

Hi list! I'm using Asterisk 1.2.7.1. with FreePBX 2.0.1 on a CentOS 3.7 box. On the * box I also have a samba share where our CRM app can dump call files and a cron script is moving the call files every second to the asterisk directory. Everything goes really quickly, the call file is placed on the samba share and very quickly moved to the asterisk dir, so far so good. But then the call

OK, i am hoping that someone will be able to help me out. I am using FreePBX 2.8.1.4 I have two asterisk servers connected with a iax trunk. The trunk is working fine when used via the outbound route setting. meaning an extension on one server can call a specific extension on the other server. now what i want to do is set it up so that an incoming call (from a third server) is redirected to the

Hi, I am using Trixbox trixbox CE 2.6.2.3 (Stable) using Asterisk 1.4.22-4 I am looking for the following functionality: `````````````````````````````````````````````````````````````````````````````````````````````` I receive a call from Mr. A. I put Mr. A on hold. I dial Mr. B I connect Mr. A's call (which was on hold) to Mr. B and I get out of the call. Mr. A & Mr. B are in

Hi, I seem to be facing an intrusion issue, inspite of firewall (script attached). What am I missing ?? Any suggestions / recommendation are welcome pls. Best regards, Sans -------------- next part -------------- #!/bin/bash echo 0 > /proc/sys/net/ipv4/ip_forward # Clear any existing firewall stuff before we start /sbin/iptables --flush # As the default policies, drop all incoming

Hi, Does a parameter exist for a queue to delay ringing/sending a caller to all agent phones after the previous call is answered by an agent? My queue ring strategy is set to ringall. I am using Polycom KIRK wireless DECT SIP phones. And it looks like the KIRK wireless server may need a split send to realize all wireless phones are no longer ringing (busy) after 1 call rings & is unanswered,

Dear list, I've set up an RPM repository with several asterisk-related RPMs that I think contain some improvements upon what are already out there. The first goal is to be able to build an Asterisk + FreePBX system on CentOS 5 with the EPEL repo enabled; in our environment, where all our systems are under configuration management, either an appliance-type solution or a manual

I'm having a problem receiving incoming faxes and I'm hoping someone here can help me out. My system is a PBX in a Flash with one dahdi card for my incoming analog lines and another dahdi card for my analog devices (fax and modem). My dahdi-channels.conf file looks like: ; Autogenerated by /usr/sbin/dahdi_genconf on Tue Jun 23 14:56:24 2009 ; If you edit this file and execute

Hi, I have latest Elastix 64 bit setup and running fine (Asterisk 1.6.2.13) I would like to customize the file name of call recordings: /var/spool/asterisk/monitor/20110511-110858-1305126538.912.wav I would like to include the extension number in the file name. Did a lot of googling but not much help. Pls advice. Thx Sans -------------- next part -------------- An HTML attachment was

I am running Asterisk 1.6.1.6 and DAHDI/DAHDI tools 2.2.0.2 compiled from source on a Debian Lenny box, also running FreePBX 2.5.2. I also have an OpenVOX TDM400 card installed with an FXO port on port 1, and an FXS port on port 2. I have a POTS line installed and working on the FXO port. However I've encountered a weird problem that I can't seem to figure out. I have incoming POTS

Dear Mr. Beginning to work on Linux. I am trying to install R into Ubuntu Gustsy. I installed version 2.5.1 and worked fine. But then I tried to upgrade by adding these lines to sources.list: deb http://cran.fiocruz.br/bin/linux/debian etch-cran/ deb http://cran.fiocruz.br/bin/linux/ubuntu gutsy/ I introduced the authorization keys (as recommended) and the files were downloaded. But then I

Hi, I am taking my first steps towards virtualization. I have 3 Gateway Servers; all on CentOS 5.3 (non-graphical mode) + Squid. We have 3 separate teams each of 5-7 users who use 1 ISP each via their own Gateway server. I am keen to use virtualization and consolidate all 3 servers on a single server. I plan to install ZEN on a Pentium Core2Duo with 2GB RAM & 1 nic. Then create 3 virtual

Hi, On 64 bit centos 5.6 I have virtualbox 4 and 64 bit elastix latest. Since yesterday cpu utilization has been constantly peaking 65-75%. Hardly 1-2 concurrent calls. No other activity on server. Top shows asterisk on top. Its quad xeon server with 4 gb ram. Any suggestion where should I start and how should I go about with my investigation. Thank you and have a great weekend. Sans

I have a TDM22B (TDM400 PCI + 2xFXO + 2xFXS). One FXO is connected to the PSTN. When Asterisk places a call, it dials using DTMF. If I listen in on the line during the dialing, there is a roughly one second pause between the penultimate and the final digit -- proportionally much longer than the pause between the preceeding digits. This accounts for about 25% of the time taken to dial. Anyone

Greetings- First, my apologies for the OT post. Yes, I understand this is not the FreePBX-users mailing list. But, there are a large number of people that use FreePBX and I'm hoping they can be of assistance. I have a system running Asterisk 1.4.27 (see... relevance!!!) and FreePBX 2.6.0. There are a large number of inbound routes configured for the various DID's coming in via PRI, SIP,

Hi everybody, we use Asterisk to route all calls to a inbound phone number to a specific outbund mobile phone number, depending on time and date. I'd like to send a notification email to a specific email address, each time we receive a call. For this I used the tip of "dicko" here [1]. I'm a Asterisk newbie. Unfortunately it doesn't work. The System() command is not

I'm pleased to announce that Flash Operator Panel 0.26 has been released! FOP is a GPL'd switchboard type application for the Asterisk PBX. It runs on a web browser with the flash plugin. It is able to display information about your Asterisk box in real time. It is included in FreePBX, Asterisk@Home, DeStar, startShop, and several other projects both free and commercial. You can grab the