Java sip software mobile phone jobs

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    8,606 java sip software mobile phone jobs found, pricing in USD
    WebRTC / SIP expert 6 days left
    VERIFIED

    The client is using astrix server. They will give us a vpn connection to their network And we need to setup WebRTC client to talk to astrix server. We need to use open source functionality working which it can't currently get a function to work within a type. [login to view URL] So that they can make a 1 way audio call which is having issues with local and remote media streams to work. Need t...

    $20 / hr (Avg Bid)
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    16 bids

    Want to start a VOIP/calling card solution ,need to install the necessary software (asterix,freepbx,a2billing etc).Advice the best solution

    $299 (Avg Bid)
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    ...upgrades to SSL, Predictive dialing, IVR, Incoming calls, and potentially GSM hardware upgrade (need to evaluate solution) with Asterisk expert. - Server health - IP phone connection - SIP line - Bandwidth and latency of system to network - Connectivity Other task related to asterisk server that will be needed in future: We need to have a setup, where

    $13 / hr (Avg Bid)
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    Looking for freelancer to configure FusionPBX with PowerPBX hosting. [login to view URL], [login to view URL], Debian ...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 4 - 6 lines. PHP or LUA scripting, VOIP, interconnect

    $462 (Avg Bid)
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    Trophy icon Logo design for "Livingston Depot" 4 days left

    "Livingston Depot" is the Name we need a logo for.. - "Sip & Shop Station" is the Tag Line This is a boutique cafe that is located in Livingston, Tennessee. This shop offers local artists, hand crafted items for sale, with a small cafe offering beverages, and lunch items. The keyword inspirations are elegance, down home, boutique, station, depot

    $40 (Avg Bid)
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    117 entries
    Asterisk Expert 4 days left

    ...upgrades to SSL, Predictive dialing, IVR, Incoming calls, and potentially GSM hardware upgrade (need to evaluate solution) with Asterisk expert. - Server health - IP phone connection - SIP line - Bandwidth and latency of system to network - Connectivity Other task related to asterisk server that will be needed in future: We need to have a setup

    $17 / hr (Avg Bid)
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    I need a call with multi skype user via SIP. and i saw 3cx gateway for skype program. this product is no longer available for download nor maintained. [login to view URL] [login to view URL] The 3cx gateway for skype is running only windows 7. and needed skype old version

    $29 (Avg Bid)
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    3 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

    $30 (Avg Bid)
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    1 bids

    We need an implementation of the services mentioned in the subject. Thanks.

    $1495 (Avg Bid)
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    7 bids
    ejabberd configuration 1 day left
    VERIFIED

    I need help you configure 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.

    $22 / hr (Avg Bid)
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    2 bids

    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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    5 bids

    Setup a VoIP server for termination the requirements are: 1. SIP connections to gsm gateways through VPN is required 2. Billing software 3. Load balance traffic to gateways 4. setup connections with 3 clients

    $238 (Avg Bid)
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    7 bids

    ...HTML, PHP, Java etc. Additionally should have a vast knowledge of website platforms, front-end frameworks, plugins, and databases. • Advanced to expert coding skills • Strong understanding of functionality • Can build the website from scratch • Some familiarity with design, marketing, and SEO • Keeping up with the latest design software and technologies

    $177 (Avg Bid)
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    I am looking for an experienced FreeSwitch engineer who can work with latest freeswitch version with SIP Trunks to do following tasks I will be passing destination number , CLI , ring duration , max call duration , audio file , unique call id to Freeswitch (JSON API) , the dial plan should start dialing numbers and plays a voice file when the call

    $1272 (Avg Bid)
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    I have already built app on play store called "Loan Emi Calculator". Please review the app befo...Calculator". Please review the app before you bid, here is a store link.. [login to view URL] New features need to add 1. SIP Calculator 2. Update Emi Calculator for variable interest rates

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    I have an existing Shoretel system with a PRI switch. Using the SETU VTEP device I want to replace my existing PRIs with SIP trunking by using the SETU VTEP as a gateway. I need help programming the device to integrate with my existing Shoretel system.

    $200 (Avg Bid)
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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

    $318 (Avg Bid)
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    convert cisco ip phone to sip for working in yeastar s100

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    Need to Configure 2TATA Sip Trunk line on my freepbx via remote desktop like TeamViewer. FreePBX hardware is assembled and installed freepbx.

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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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    Hi, I'm looking for someone to connect a twilio number that we currently have to a sip endpoint thats already configured for voice. We are using the X-lite Voip client and would like to get SMS functionality enabled on it.

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    SIP CLIENT FOR RASPBERRY PI3 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

    $373 (Avg Bid)
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    Looking for someone knowledgeable on freepbX and mobile application development to create a VOIP application for customers service . The reference app is named as "smart line" You can check the app's features . I need the exactly same features with some new designs .

    $6 - $11 / hr
    Sealed
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    13 bids

    Early decay in babies teeth is a big problem and is called early childhood caries...Brush the child's teeth morning and night 4) Bacteria from your mouth can be transmitted to your baby's mouth so don't share spoons etc 5) After a feed clear the mouth with a sip of water 6) don't feed after that 3) don't let the baby breast feed all night or on demand

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    1 entries

    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.

    $21 / hr (Avg Bid)
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    8 bids

    Add the ability to enter sip trunk credentials to asterisk

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    I am looking for a senior architect/developer to design and develop gsm telecom solutions based on ss7 or diameter or sip communication protocols. The scope will be discussed later, the pay will be very good

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    15 bids

    ...writer to write an original story featuring a seductive buxom woman who's big unwieldy breasts contain a milk that will wickedly shrink the size of any man who drinks it. One sip is enough to make him a midget and too much will send him down to bug size or smaller. Find a way to work this weird concept into a sexy short story with a full plot and I'll

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    23 bids

    Hello, I need a logo for a beverage company. The name is "Soufrière Hills" The tag line is “every sip makes you family and comforts your soul.” Please use the following hex codes only: #549127 #10375d #edc857 Use all three or two colors in any combination. Please be as creative as possible. This logo is for a beverage that is from the island of

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    55 entries

    ...need some kind of bandwidth compression system ( up to 60-80% than usual SIP 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

    $1742 (Avg Bid)
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    6 bids

    ...simple sip to tapi tsp app with simple features like sip registration like an sip client no voice or codec are needed the incoming call must only transmitted to tapi tsp driver with caller in number the out bound calls will be managed with action url so from any application how supports tapi functionality can dial out by using the tapi via sip to his

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    Hello, we are selling fasteners in germany. We ...from 9 am GMT time to 3pm GMT. Also part time is possible. You need to speak very good and clear english. Please bid for 1000 Calls/contacts. I will provide a phone account to use. Skype or a SIP Account, so that you dont have costs. When you are good, this can be be a project for much more contacts.

    $590 (Avg Bid)
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    SIP intercom broadcast software View online status View the status of the call Setting up a voice recording Call management Single-key talks, multi-party meetings Surveillance, monitoring, shouting and deterrence Arbitrary Paging and Interphone Recording Talk queuing, strong

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    ...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) and SIP integration using a combi of sylk server and opensips. IMO a real challenge for experienced engineers. I can be reached by mail at max@axeos.com. We can start with some small

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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 Dear, We have dial plan for inblond calls . Mean ...NoOp(${ipaddress}) exten=>s,n,Set(ODBC_ADDLIVE(${DID},${CALLERID(num)},${CHANNEL},${UNIQUEID})=1) exten=>s,n,Set(ODBC_CALL_DID_MAPPING(${DID},${UNIQUEID})=1) exten=>s,n,Dial(SIP/${DID}@${ipaddress}, 60, o) exten=>s,n,Hangup() exten=>h,1,Set(ODBC_DELETELIVE(${CHANNEL})=1) exten=>h,2,Hangup(42)

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    you have to install asterisk 1.8 on Puppy Linux make sure asterisk work with sip & iax both also wireguard & TeamViewer you can download OS from here [login to view URL] also if you want you can use Ubuntu or Centos you need to make iso and iso size must be under 500MB i can pay up to 100USD do not ask for more

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    ...will reside on Tizen the function of the app is a bridge between sip client and Phone Native dialer The application working in APK format Full source code Simple manual for compiling and generating the application from source Features : -Route call from SIP to GSM -Convert audio from/to SIP and GSM networks -Able to run on Tizen. Must run on background

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    We are new small VOIP Services provider and We need a SIP Voip Softphone App for mobiles mainly, eventually for windows too. The App would be similar to Zoiper, Bria, SIPGo, SessionTalk, etc. but customized with our Logo, our proxy domain and most important the capability to use all video and audio codecs including the now free licensed G729.

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    ...things in right context, it is desirable, but not necessary, with experience and knowledge within • On premises deliveries of system and installation script • Telephony and SIP (FreeSwitch, Swyx (a German PABX) and eventually Asterisk) • Record keeping, eventually through a PHP framework • Access rights for different users, we work with an access token

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    Seeking somebody that can setup correctly Jitsi videobridge with Need closed ...be able to handle room authentication + concurrent usage per room previously setup Jigasi dialin number (At the moment SIp registers and enters room but no voice never) Online translation and online transciption Customize mobile app In other words, set it up correctly.

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    ...Manager, who will work ongoing as an authorized negotiator on our Schedule 36 GSA Schedule contract to ensure it is up-to-date and in compliance. More specifically... -Email and phone support to answer GSA Schedule contract related questions -The development, submission, negotiation and e-signing of all GSA Schedule contract modifications -Assistance with

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    Required for own numbers, SIP Trunk Server Setup.

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    We need to develop a sip dialer for Windows and Mac.

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    Implement security and failover SIP

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    VOIP SYSTEMS AG and [login to view URL] need a basic SIP dialer for Android, keeping in mind these features: * PREFERED someone that already worked with Dellmont dialers or similar. * Dialer itself with all the basic functionality of a softphone. * Call history (calls made by the app itself) * Address book * Login/signup activities * Source code aviability

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    We are installing a Cisco 3800 Series Integrated Services Router for a E1 PRI We need someone to configure for the...a E1 PRI We need someone to configure for the following 1. Set Cisco to DHCP 2. Setup Dial peer and Plans 3. Configure the PRI Setting to register on Carrier 4. Register to SIP server , for Inbound and Out bound calling to from the PRI

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    Configure our VOIP setup. We are using a TG100 Yeastar GSM Gateway and Groundwire on iOS. This task is to set them up so they work reliably. Calls from the iOS app will terminate using the SIM. Calls from the SIM will terminate to the iOS app using a push notification. The TG100 Yeastar GSM Gateway is behind a router with a web based interface and port forwarding has been setup.

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    ...for my small call center project . Install PBX and Asterisk on VPS server. First we want to create 10-20 Ext. and all my executive login or online with there SIP phone on there mobile . For this setup my calls come from another network and we need divert that calls on ext. We can use two methods . 1st : ring all ext. or we can divert calls according

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