Install elastix server asterisk server jobs

We are looking for a technical person who has good telecom experience and can help us developing ringless voicemail broadcasting system. There are very few companies are doing this and we have no idea how do they do that. Here is detail about companies that provide ringless VM : [login to view URL] [login to view URL] Http://[login to view URL] You can read about it and tell me if you can do it....

...OVERVIEW I have to shift my existing asterisk vicidial server files ( my dial plans, modules , database ,IVR , custom code on the server, user details, ALL CONFIGURATIONS. etc ) to a better hardware /server with Xeon processor 3.2 GHz.& 16GB RAM NEW VICIDIAL server details & its configuration ● Centos7, Asterisk, FreePBX & VTiger is already ...

This current project shall be awarded to individual who can install IP based PBX alongwith open source billing module ([login to view URL]). The IP PbX shall be asterisk based on linux platform. VPS for installation will be provided. The payment terms include 50% payout at the time of 50% work completion and rest at

After login, receive message "cannot connect to asterisk"

Dear all, I am seeking someone who can write a script that will install LAMP, asterisk, a2billing 2.1.1, and FreePBX automatically on CENTOS 7 by firing one or 2 command line(s).

connect vtiger 7 to elastix 4 can see recording when call popup that call incoming

Hi, I look for Freelancer who can implement Asterisk with my MGCP provider. Not SIP. Can somebody do? Thanks

I have requirement of asterisk PBX developer

...(Cacti, mrtg, Nagios, PRTG, OpenNMS) b. Server Management Tools (Hyperic HQ, Zabbix) c. Backup Management Tools (FreeNAS) d. Inventory/Asset Management Tools ([login to view URL] with GLPI) e. Visitor Management Tools ([login to view URL]) f. Printer Management Tools g. ID Management Tools (Open LDAP) h. VoIP Tools (Asterisk, FreePBX, FreeSwitch) i. Messenger Tools

Hi -I want to install freepbx and configure termination gateways on my server using vpn. I already installed asterisk and a2Billing -configure a connection with clients

...MT7628N CPU Cores: 1 CPU MHz: 580 Flash MB: 8 RAM MB: 64 server send call to router >>router send to voip gateway we install our program on router and our router OS is openwrt like server to router resisted and router to Voip gateway all connection By SIP asterisk server to creytiv/re router then creytiv/re to VOIP gateway some one alre...

im looking for asterisk pjsip developer for support our system

In need of software or asterisk/freepbx hosted solution for "Voice Broadcast" or "Robo Dialer". - Connects to SIP Trunk - Alternate Caller ID; Area Code Caller ID campaigns with alternating numbers - Ability to play and add audio files messages that are randomized - Multichannel lines (100) to make hundred or thousands calls per minute - Choose number

We have 2 platforms: 1. Freepbx 2.11 and Asterisk 11 2. FreeBpx 14 and Asterisk 16 CallBack makes callbacks simple. When your agent receives a callback request, they will see all of the caller’s information in the screen. They can click one button to accept the callback request and [login to view URL] will automatically connect them to the caller. This process

need someone to fix the extension file on the asterisk script

i need to VOIP SIP to SIP call G729 optimization bandwidth SERVER A =asterisk server Server B =Openwrt ROuter server A&B connected to VPN VOIP device COnnected to Openwrt ROuter we send calls Server A to VOIP device SIP to SIP codec g729 its use per calls 30KBPS we want to conpress it if you can make it under 15KB we are happy i saw too many peppole

We have installed OpenAnswer, FreePBX, and Asterisk on a AWS. We are looking for someone familiar with the software to finish fully setting it up for us. This includes changing all the default passwords. After setting it up we want to upgrade to PHP 7.2, and NodeJS 8. We are also looking to remove all test files, and default configuration files that

I have a vps server centos7-64 I would like to install the latest version of the following -asterisk/freepbx/a2billing/fail2ban/vpn -a2billing load balance traffic across termination gateways -IVR filter to drop test call generator (press DTMF to continue call)

we have a java script which is connected to asterisk we need someone to help change the settings

I have a vps server centos7-64 I would like to install the latest version of the following -asterisk/freepbx/a2billing/fail2ban/vpn -a2billing load balance traffic across termination gateways -IVR filter to drop test call generator (press DTMF to continue call)

I have setup an Asterisk Cloud deployment and need some help tweaking the final steps of the setup

...generate a voucher of the customers choice, enter the customers number, then click buy. The consumer will get a SMS containing Voucher number. (SMS sent out via local Asterisk Server) Major features for the initial design include: ● A minimal UI than allowed vendors to generate a unique token for client use. (UI needs to be made modern) ● App will

i want SIP Proxy for my voip Dialer and voip device i have asterisk server i need to register my soft phone to my SIP server but sometime we have blockage issue so i need a SIP Proxy if you can provision then good for me check the Attach FIie if you can make it let me know

...ISSABEL install on two server and they are configured as high availability active /passive mode. The problem is the call center module. When we install the module, the agent login facility is not working as per configuration. We have created agent, campaign, list and form but the agent still cannot log in. If you have experience in Elastix call center

i had asterisk/freepbx installed on centos recently. One DID on the SIP trunk is working fine but other DID has two issues. Both DIDs are on same network and similar SIP phones 1. there is a delay of 50 sec after caller presses a digit to reach staff before phones ring 2. phones ring only once....so calls are lost freelancer who installed this for

Need Asterisk Web Developer, Need Application like VOIP Service, Miss Call Alert service, You can check demo like myoperator , knowlarity etc

I have a very serious problem with my hosted PBX: It is available online but I already change the default 5060 to 10... Firewall and fail2ban are already setup. Sip phones can call using dialcheap and megavoip. The problem is that hackers make calls using dialcheap and megavoip and I cannot stop it. ##, *2, Tr are already disabled

...and connects both parties 2) Ability to add an audio file message (in wav or mp3) to the placed call, greeting both parties upon pick up 3) To store voice messages on my server (in wav or Mp3), email notification when message is received and ability to ftp download message Some of the existing systems: [login to view URL] , [login to view URL]

We are looking for a company who has 24/7 server administration. We have between 4-6 servers running at the same time, and we are looking for a new company, who can be in charge of everything concerning the servers. It is important that there is 24/7 support, and that there is system admins attached, if it is not only system admins. If every case goes

I would like create agent survey on Issabel/Elastix. Need modification in UI as well like adding menuitem under call center reports section to show Agent survey results page within elastix/issabel UI.

we have 2 extension files we want someone to copy the dial out routine from one to the other

after cloud server down we have: [2018-Dec-04 10:22:27] [CRITICAL] (admin/[login to view URL]) - Connection attmempt to AMI failed when we try: [root@pbx]# fwconsole restart Asterisk not currently running Running FreePBX shutdown... UCP Node Server is not running Running FreePBX startup... Starting Asterisk... [--------------->------------] 13

...Push-To-Talk button + Mute button + Speaker button. - Group can be created by users added as contact (or connected into one group in back-end). - Developer can choose between Asterisk / FreePBX -or- XMPP / OpenFire as user management. - Communication must use Over-The-Air must be always 256-AES encrypted. (All live Push-To-Talk recordings must be 256-AES

...Push-To-Talk button + Mute button + Speaker button. - Group can be created by users added as contact (or connected into one group in back-end). - Developer can choose between Asterisk / FreePBX -or- XMPP / OpenFire as user management. - Communication must use Over-The-Air must be always 256-AES encrypted. (All live Push-To-Talk recordings must be 256-AES

We are looking for Java developer who can develop CTI & IM desktop software for our asterisk based PBX. for instant messaging we are currently using Openfire server but other XMPP servers are also welcome. We already have design and initial code to start with. Details of project will be shared on request.

...notification on my TV when incoming call, i want to see the caller id on my TV) 4 - [login to view URL] (i already have media plex server at home, i want to manage it from my dashboard and control player and choose from library and stream it on one of my chromecast) 5 - [login to view URL]

...business). The business name is "Asterixx Entertainment". I have attached my current logo, but keep in mind this is not what I am after. If you are able to incorporate an 'asterisk' into the logo that is a bonus but not necessarily going to be the winning design. This logo will be featured on business cards, future website and facebook pages/promotional

now i have one like when a calls come to server [login to view URL] [macro-answer] exten => s,1,AGI([login to view URL],${ARG1}) and php agi write all incoming calls to database and web php read from DB but i dont know where is prob i saw some people they read active call every second same like console CLI real-time i need something realtime like it have

PLEASE DO NOT OFFER TO MY PROJECT BEFORE READING. IF YOU READ THE PROJECT PLEASE TYPE "YES I READ YOUR PROJECT" ON YOUR OFFER, AND D...will not ask Twilio for the same number if it is already asked before and already the database. P.S.; definitely we do not want to use the asterisk. We would like to run it on VoipSwitch. please do not offer asterisk.

Asterisx PBX application - Receive call, check filesystem (configured UNC path) to determine if audio files needs to be played. - Transfer (bridge) and record call. -...played. - Transfer (bridge) and record call. - When call ends, send recorded audio file to a http endpoint (HTTP). - Configure all into a Hyper-V virtual machine (Windows 2016 server)

I want to customize my A2billing with a more current theme, and configure an asterisk live, to have the activated of the graphic CLI.

Hello, You are supposed to configure the FreePBX and Asterisk with FXS and GSM Gateway. Freepbx and Asterisk is already installed on the server. Need immediate start on this.

I want to develop a desktop app for asterisk to integrate with outlook. There is a open source software called outcall, but does not work well. I need the same thing. I would possibly want to develop a mobile app to for my clients

1 - We need to configure Asterisk FREPBX in a centos machine with the newer version from centos. 2- The developer needs to provide a full documentation from the work with detailed description with step by step of how to configurate it. 3 - The developer will register some of our SIP channels into the freepbx. We currently have both sip clients: -

I want to make Asterisk server for Dynamic Black list on Destination numbers As I have calls running on my softswitch I want to add this ASTERISK server to block the calls under the following criteria: 1- if the Destination number hit 5 times and total duration less than or = 10 Sec 2- if Destination number make 5 hits in less than 10 minutes 3-

hi , i am running two asterisk connected through a iax trunk over openvpn, everything is working fine just the voice quality is choppy and some time robotic voice ,need to troubleshoot , please bid if you have done troubleshooting of iax trunks before , my budget is 50$ for this project.

I hope make to enable zrtp in motif channel for asterisk. Please bid if you have experience in this area.

Hi. We need to configure asterisk on our ubuntu machine. We need someone high skilled to do the job. We will also need to make it visible over internet with public IP. Also we need to configure some IP phones on it. After all, we need a manual with step by step about how to configure it.

I have lan calls working fine, but require someone with Cisco/Asterisk SIP knowledge to help resolve this issue with NAT, no audio not working.

I search a developer for create my project for Asterisk. This is project for automatic call and deliver alarm message from alarm station. See file for more information.