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    10,589 need someone setup asterisk jobb har hittats, med prissättning USD
    BACKGROUND SERVICES ANDROID 6 dagar left
    VERIFIERAD

    Hello, We have an issue with our VoIP (Asterisk / PBX) based application that it receives no calls when running on the background. The issue is currently that; 1) APP is in the Android recent-tabs it sometimes receives calls / broadcasts. 2) APP when not (closed) from the recent-tabs it never receives calls / broadcasts. 3) APP crashes when not directly

    $181 (Avg Bid)
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    ...Vladimir U., I noticed your profile and would like to offer you my project. We can discuss any details over chat. I need to move our Asterisk system to FreePBX on Microsoft Azure cloud. I'd prefer to install it on Centos 7. I also will need to create couple extensions and connect them to a test account. There is also some basic ACD function: Dial 1,2 or

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    anyone who has worked with asterisk or a2 billing before please bid and i will give details of project

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    I need you to configure asterisk to receive calls and play an audio. Also to make calls. I have already setup asterisk and have a configuration just need to be sure

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    I need a web application make in php, mysql, ajax with nice css style. A form that allows the inclusion in a table (which is updated with ajax) of the various practices made by a user of a team. USER LOG IN - Drop-down menu from which to choose the customer - Drop-down menu in which to indicate the name of the practice performed - Time taken - Small

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    Hi Arshad, we're looking for someone skilled in FreeSwitch or Asterisk to build a ringless voicemail system. We operate [Logga in för att visa URL] which is a popular ringless voicemail platform in the US. However, we use other provider's APIs to do the delivery. This isn't cost effective for us. We need to operate our own delivery infrastructure. Is t...

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    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 : [Logga in för att visa URL] [Logga in för att visa URL] Http://[Logga in för att visa URL] You can read...

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    ...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 Installed

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    ...shall be awarded to individual who can install IP based PBX alongwith open source billing module ([Logga in för att visa 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 the project closure. Please

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    Troubleshoot FreePBX 21 timmar left

    After login, receive message "cannot connect to asterisk"

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    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).

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

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    I have requirement of asterisk PBX developer

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    I need an article about 1000 words and graphs and pictures to cover following points -Few open source and completely free tools to manage IT Infrastructure a. Network Management Tools (Cacti, mrtg, Nagios, PRTG, OpenNMS) b. Server Management Tools (Hyperic HQ, Zabbix) c. Backup Management Tools (FreeNAS) d. Inventory/Asset Management Tools ([Logga in för att visa URL] with

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    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

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    ...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 already made it they can run more then 64calls but if you can run 32calls i am Happy need a easy way make a pakage we just install package on openwrt router and input sip server IP

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    im looking for asterisk pjsip developer for support our system

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    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

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    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 [Logga in för att visa URL] will automatically connect them to the caller. This process

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    need someone to fix the extension file on the asterisk script

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    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

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    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)

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

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    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)

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

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    ...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 confirm transactions through SMS, as well as provide link/access token to enable service activation (Need to confirm functionality) ● App

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    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

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    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

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

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    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

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    ...[Logga in för att visa URL] , [Logga in för att visa URL] etc You don’t even need a fax machine…,Receive Fax from customer as PDF in your Email. The system will support setting up/administrate the call center SIP accounts (with hunt groups). Of course system needs configuring Asterisk for custom DID:s, call forwarding, internal calling, voicemail, ability

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    ...level 1 or level 2 support first, then they must be very quick in sending them up the chain, if they can not solve the problem, and system admins must be available 24/7. We need an SLA where error-correction is initiated within maximally 15 minutes from we have created a case/ticket if it is urgent – We sometimes use KVM and thus we can see if you actually

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    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.

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

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    ... [root@pbx]# fwconsole restart Asterisk not currently running Running FreePBX shutdown... UCP Node Server is not running Running FreePBX startup... Starting Asterisk... [--------------->------------] 13 secs In [Logga in för att visa URL] line 187: Unable to connect to Asterisk. Did it start?

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    ...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

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    ...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

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    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.

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    Hi, i need to have someone who can integrate these components on my Home Assistant ([Logga in för att visa URL]) 1 - [Logga in för att visa URL] . i want to receive on my dashboard 1 minute snapshot retrieve from the onvif hub of arlo pro 2 camera (its custom development) 2 - [Logga in för att visa URL] (i want to

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    ...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 to show all my active dial and all calls on my web panel real time my current one i can do but its take too much time to load have a look on

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    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.

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    Asterisx PBX application - Receive call, check filesystem (configured UNC path) to determine if audio files needs to be 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)

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

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    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.

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    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

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    Förseglat Sekretessavtal
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    28 bud

    Proyecto: Implementacion de una mesa para practicas de laboratorio Voip con servidor Asterisk

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    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-

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    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.

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

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    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.

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