Asterisk pbx open source jobs

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    13,438 asterisk pbx open source jobs found, pricing in USD

    Hi Aqs Y., I noticed your profile and would like to offer you my project. We can discuss any details over chat. I am looking for some auto dialer based on asterisk

    $35 / hr (Avg Bid)
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    1 bids
    Click to Call For Free PBX 6 days left
    VERIFIED

    We are currently using FreePBX and looking to integrate some sort of Click to Call for our sales team. What we need it to do is everytime is recognizes and telephone number in the following formats (###) ###-### or ###_###_#### or 1-###-###-#### that it makes the telephone number clickable and it automatically places a call via our Bria softphones. I need this done as quickly as possible

    $187 (Avg Bid)
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    We have setup bitrix24 onpremise and we have asterisk-vici installed. We want to use SIP connector. Any one with experience in setting up onpremise Bitrix24 and telephony with SIP connector can respond.

    $654 (Avg Bid)
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    we need some one who can happily use fusion and setup and over come the challenges of our clients

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    Hi All I have a SIP client on my network, its a Intercom. It registers to my FreePBX/Asterisk solution and makes calls to my mobile phone fine. I want to intergrate Skype because Amazon Echo's can answer/make calls using Skype. The intention is to receive the Intercom call on my home network to an Echo Dot, which I believe will work when the FreePBX

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    Hi Kingasterisk T., We have a problem with our asterisk configuration. as a brief when call directed to extension queue after ivr. there is silence and no result. caller cant reach to extension. as example we get this kind of logs. [2019-02-14 08:56:15] WARNING[7089] chan_sip.c: Autodestruct on dialog '223205511-3759123182-18707@TcomNextoneSBC4-DRC

    $100 (Avg Bid)
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    I'm new to FusionPBX/FreeSwitch. Can't set up the same thing I've set up in minutes in Asterisk. Need remote assistance/general consulting or just a setup example in accordance with my infrastructure conditions. My FusionPBX is a VM that has only one LAN NIC. It uses pfSense VM as a gateway. pfSense VM has both LAN and WAN links with a public IP on

    $27 / hr (Avg Bid)
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    7 bids

    I Need help with a vicidial server that is not leaving vvoicemail message and is dropping calls in a press 1 campaign

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    6 bids

    We have developed a voice bot that understands vernacular languages of India. We are looking for a developer who can help us integrate a user call with our voice b...asks a question - e.g. what are the current offers? The bot responds: Buy goods worth Rs 5000 and get cash back worth Rs 1000. Please get in touch immediately if you know Asterisk well.

    $2779 (Avg Bid)
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    We have developed a voice bot that understands vernacular languages of India. We are looking for a developer who can help us integrate a user call with our voice b...asks a question - e.g. what are the current offers? The bot responds: Buy goods worth Rs 5000 and get cash back worth Rs 1000. Please get in touch immediately if you know Asterisk well.

    $2791 (Avg Bid)
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    7 bids

    I'm looking for a Twilio function ([login to view URL]) to be created that simulates a DISA ([login to view URL]). Basically it is a phone number that when dialed: 1. Automatically answers 2. Prompts the user for a phone number 3. Connects the user to the phone number they dialed I have

    $173 (Avg Bid)
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    Issabel is a PBX like FreePBX. In Call Recordings, I have limited ways to search for a recording. I would like to add more serac options in the interface. I believe that the required skills are: MySQL , Web and sterisk

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    AMD detection in Asterisk 4 days left
    VERIFIED

    I want someone expert in asterisk who can fix my AMD detection in Asterisk, I'm using Vicibox and running Press 1 Survey Campaign.

    $54 / hr (Avg Bid)
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    I need full voip solution, android app, website, admin control panel, reseller control panell , pbx, softphone setup, Anyway if you have experience and u have done this work before, only u can contact with me. Otherwise DON'T waste your time. also if you are from bangladesh you are most welcome...

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    im looking for someone to make landing page simple nice looking landing page have form and need image

    $29 / hr (Avg Bid)
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    I need you to configure our fusion pbx and configure our softphone so that we can add additional softphone without any issues along help with provisioning

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    Asterisk program is being hosted by our own server in a VM environment, we will provide access to VM and internal NAT. I'm looking for someone who is experienced with Asterisk PBX platform, who can assist with our internal PBX setup, provide some support when needed, make some adjustments to the setup when required, and also who can help us to support

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    Looking for freelancer to configure FusionPBX with PowerPBX hosting. [login to view URL], [login to view URL], Debian 8. FusionPBX system already installed. Configure Dial Plans, PBX Extensions, I/O Trunks, Voicemail, DISA, IVR & local number display on the go, Call Forwarding, Call waiting, Linphone setup and instructions, call recording, setup sip trunks. for

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    We have a call centre system using Asterisk. With Asterisk only supporting the following call distribution. We need to tweak the queue behaviour to reset agent idle time in two event, after agent has hung up an incoming call and after agent end pause. If you base on the ring stretagy of least recent, we can achieve the agent idle time reset on the incoming

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    Asterisk Expert 2 days left

    Asterisk program is being hosted by our own server in a VM environment, we will provide access to VM and internal NAT. I'm looking for someone who is experienced with Asterisk PBX platform, who can assist with our internal PBX setup, provide some support when needed, make some adjustments to the setup when required, and also who can help us to support

    $17 / hr (Avg Bid)
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    hello i need to stream live calls from my asterisk more or so like chansky but i need to stream in demand and to wav files so i can play it in brower i am useing freepbx

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    15 bids

    must be connect multi Skype user to a SIP-PBX. and call via skype to pstn or mobile number. multi skype user can running on asterisk machine. extensions are calling via each skype-account

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    We are looking for an Asterisk expert who has experience installing Kamailio or similar software to be used as an outbound proxy for NAT Traversal front ending multiple Asterisk/FreePBX servers. Some PBXs are having load spikes due to frequent device registration/authentication and NAT traversal functionality, so we're looking to install a solution

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    A CRM solution to help with sales and order management. We are choosing Suite CRM because its free and comes highly recommended but we also need to integrate this with a PBX so we can manage orders through a call center.

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    10 bids
    ejabberd configuration Ending left
    VERIFIED

    I need help you configure my ejabberd server with sylkserver for translate jingle/sip between xm...my ejabberd server with sylkserver for translate jingle/sip between xmpp client and SIP proxy and/or PBX. Goal is that XMPP client can make a call through the PBX to the outside work (as a regular UA) and that the XMPP client can be called via the PBX.

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    Add ADI input to Asterisk frontend. Thank you!

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    We have a multi-tenant PBX system and need a simple SIP phone provisioning tool written in Javascript Node.JS, using Angular or React, running on a MEAN/MERN stack. This will not be integrated with the PBX, however our B/OSS system will add, suspend, and delete tenants and extensions to this new system you will create via a REST API. A commercial

    $198 (Avg Bid)
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    Need to develop end to end solution of call center.

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    i need help fop2 its installed on a standalone (separate) server than Asterisk Freepbx i can get the recordings to work.

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    i need help fop2 its installed on a standalone (separate) server than Asterisk Freepbx i can get the recordings to work,

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    What I am looking for is a PHP application that works similiar to FreePBX for creating IVR menus and DISA and uses the Telnyx API to make changes to the call flow...that works similiar to FreePBX for creating IVR menus and DISA and uses the Telnyx API to make changes to the call flow on the Telnyx servers so that I don't have to run my own PBX server.

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    I want someone expert in ASTERISK and freeswitch and ios to send calls over bluetooth but to use whatsapp or viber instead of gsm dialing to be smart enough to tell asterisk use whatsapp when dialing (send calls direct to whatsapp) android or iphone whatever they can fix

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    Customer is looking for an Avaya consultant to provide design and configuration assistance with an Avaya consolidation / move project. The first ...but must have fluency in English and must be available to work local Eastern Time zone hours and weekends. There’s no BOM available but the equipment is likely older hybrid PBX / IP infrastructure.

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    Current software versions: FreePBX: [login to view URL] Asterisk: 13.11.2 Project Description : . After a call, our Call Center Agents will transfer a customer call to an extension where the survey should be executed. Survey will consist of 2 fixed questions. We need to be able to define the Survey number and Name, then record the questions and select valid

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    1) Set up Asterisk. You will get a machine with Linux OS of your choice. 2) Install UniMRCP and UniMRCP Kaldi Plugin. Configure SIP line for inbound calls. We will provide the URI to configure in the plugin. 3) Configure an Inbound call plan that does the following for inbound calls: (a) Prompt the caller to punch in a 4 digit code. (b) Post

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    Hi, The goal is to establish an asterisk server and freepbx for the purpose of voip termination in GSM network. The freelancer has to do the following: - Install asterisk server and configure it to operate with Huawei usb dongles (modems) having prepaid simcards to connect to the GSM network. - Be able to attach more huawei usb modems to the server

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    Sip client for Raspberry pi3 We need to make sip client for raspberry pi3 on parking system machine, Software must have source code in python and auto start with raspberry linux SIP client must call just one number that we insert in code, over GPIO inputs trigger sip client start call. (Gpio inputs must configurable in code) Sip client must have

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    Hello, We are looking for someone to deploy a FusionPBX (Freeswitch) for us from scratch we even need everything to be migrated to the new PBX from our Asterisk.

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    ...this software to be developed for Linux using Python. SIP client for Raspberry pi3 We need to make sip client for raspberry pi3 on parking system machine, Software must have source code in python and auto start with raspberry linux SIP client must call just one number that we insert in the code, over GPIO inputs we trigger sip client to start call

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    Set up sylkserver to translate jingle/sip between xmpp client and SIP proxy and/or PBX. Goal is that XMPP client can make a call through the PBX to the outside work (as a regular UA) and that the XMPP client can be called via the PBX.

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    Add the ability to enter sip trunk credentials to asterisk

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    ...calls )from Server A to Server B. Server A = Asterisk server Server B = Asterisk Client server Explanation of the scenario: 1. server A ( asterisk server, with static IP) receiving VoIP calls , with sip protocol, using G711,G729 and/or G723.1 codec and sending calls to Server B 2. Server B ( Asterisk server with PRIVATE NETWORK IP), receiving calls

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    Build Re-seller Front-end for Asterisk Phone system to allow total control of asterisk via Html5 for management and configuring. To be about to add sites exts phone numbers etc

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    Build Asterisk Phone system for re seller with html 5 front end for management

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    I am looking to add a few phone lines to my PBX system which is already up and running. To work on remote connection

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    Hi mula9211, I want to check if you are open to work at regular hours for us (much more hours than I propose now). We exist for over 10 years in NL and we have an asterisk based cloud PBX. We sell this as a service to companies and resellers. It is based using our own heavily customised linux distro. We also want to experiment with jingle (xmpp client)

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    fail2ban config and fix hacked phone.

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    i have installed 3cx pbx and every thing is good then i install sip trunk and configure the did/dod put when call my office number sometimes it ring and sometimes the number are busy need help

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    Hi Viktor S., I noticed your profile and would like to offer you my project. We need to install and configure asterisk/A2Billing for voip wholesale. Please let me know, when you can be available to discuss the details. Thanks, Nestor.

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    helping in Project to finish product.

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