Voice over Internet Protocol or VoIP refers to a host of transmission technologies meant to deliver voice communications over IP networks that include packet switched networks and Internet. VoIP systems include communication services such as facsimile, voice, and voice messaging applications. A number of firms are shifting to this system to promote a cost effective medium of communication. You can hire an expert who can help your firm to leverage the potential of this new system. Just post a job today! Hire VoIP Developers

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    I will give access to a linux box, you can install asterisk along with any other services, I want a web interface, where i will copy paste 200 phone numbers, also there should be option to upload a voice record (audio) , once i click play button. Each of the 200 numbers should get a call from their own numbers (spoof) simultaneously and all of them should hear the audio file played. Please ...

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    Project for Joel L. 9 days left
    VERIFIED

    Hi Joel L., I noticed your profile and would like to offer you my project. We can discuss any details over chat.

    $25 / hr (Avg Bid)
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    I have a CISCO 887 VA-M K9 VDSL Modem and of course internet connection, but i would like to find somebody who know how to make correct settings via Putty program. That modem/router can establish connection.... I have a teamviewer connection..

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    Hello i have Centos VPN Servers and i have Cisco VPN Routers i need to link this routers to the vps as vpn clients

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    We are in search for CMTS engineer with experience in DOCSIS, CMTS. The candidate must be permitted to work at European Union and able to relocate.

    $1835 - $3670
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    work needed for inbound outbound Call center Tele marketing/Sales

    $7 / hr (Avg Bid)
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    Set up an auto Fail-over trunk on asterisk, trunk will be given. Looking only for expert in asterisk to work with.

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    Skill Pre-Requisites: Knowledge of WebRTC Gateway and not WebRTC web-client. Have developed SIP Servers applications (SBC or PBX) Good knowledge in WebRTC, Web Sockets, HTTP Knowledge of Asterisk server and Jitsi Media. Experience in product development and system engineering for WebRTC. Knowledge on JAVA will be good to have (Not mandatory). Min 1 year of experience in WebRTC projects ...

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    kannel sms 6 days left

    we are looking for application of kannel ( that can send between 10 000 and 100 000 bulk sms per day ). And also it installation and configuration.

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    I am using Vtiger 7.0.1 and FreePBX(3cx) as phone dialer. In order to prevent data leaks I need to integrate Vtiger and FreePBX and also modify Vtiger to hide our customer's numbers. The idea is to start a call directly from VtigerCRM with a click on the phone icon(something like that) without being able to see the phone number being called. Actually, we use free 3CX as soft-phone. Our goal i...

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    As I said ago, I'm looking for someone who can assist me now for getting opportunity in future.I am looking for a assistant to create and manage my social media sites on a day to day basis. I will also require you to do basic data base work, emails on my behalf ect . Must have great English , written and spoken . Previous exp in real estate would be an advantage and also a strong marketing ba...

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    hi, i need to configure trunk sip for outbound campaign and inbound call. im stuck in dialplan setting. in asterisk debug i can show all circuits are busy. i wait your reply tanks

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    ASTPP Behind NAT 5 days left
    VERIFIED

    I need help to resolve problems with ASTPP behing NAT. The call complete but don't has audio or DTMF.

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    Call functions like mute, conference, hold, transfer, and call recording should be available. These options should be visible during a call. When you transfer a call, it will ask for your permission to move the call to another person first. Custom ringtones Echo cancellation and noise suppression Pop-up notifications Presence/online status Call history Voicemail messages and missed calls not...

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    Dear all, I used to have Elastix with CISCO 2800 as FXO gateway, now I have issabel instead of Elastix and moved all the trunk parameters along with the outbound route yet calls are failed to connect , seeking professional help thanks

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    i simply want to convert my android phone to behave as sip voip gsm gateway like GOIP, DINSTAR. android phone will register with my softswitch via 3g internet and when my softswitch will send call to android it will forward to gsm network. so it will work like DIALER(DIALS NUMBER)--> MY SOFTSWITCH --> ANDROID PHONE --> TO NUMBER DIALED BY DIALER.... This will be main future.. and...

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    Setup and Secure the Freepbx 4 days left
    VERIFIED

    Setup and Secure the Freepbx for one server.

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    Dear all, I used to have Elastix with CISCO 2800 as FXO gateway, now I have issabel instead of Elastix and moved all the trunk parameters along with the outbound route yet calls are failed to connect , seeking professional help thanks

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    Need Personalized IVR 4 days left
    VERIFIED

    We need IVR system hosted on cloud with custom IVR options based on the Called ID. Below is the basic flow: 1. Customer calls to our Customer Care 2. Our Cloud Hosted IVR will receive the inbound call and start to play a Greeting Message 3. In the mean time while playing the Greeting message, the system will fetch the Caller Details from our CRM based on Caller ID. 4. The caller will be p...

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    Must be experience in Open Source IP-PBX development product. Setting up the infrastructure, build key features and testing. Works on Debian® 2.8 distribution, some of the key FOSS components that support the unified communication and management functionality are Asterisk®, FreePBX®, Chan_Dongle, and RaspAP Web GUI. Build a IP-PBX product that allows users to make VoIP calls to...

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