Need develop simple sip api allow create product jobs

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    10,601 need develop simple sip api allow create product jobs found, pricing in USD
    WebRTC / SIP expert 6 days left
    VERIFIED

    ...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

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

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    ...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 we are more secured against breakdowns

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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

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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

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    Asterisk Expert 3 days left

    ...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 we are more secured against breakdowns

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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

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    Hello, for this project i need a freelancer to allow two users who are playing against each other to be in a chatroom with eachother. To connect to the chat i have a invite system, when user declines nothing happens. but when user accepts, i need both users to be able to message eachother in [login to view URL]

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

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    i need someone with at least 3 years of experience to change our POS the actual situation right now : we have some product that we manage from the point of sale, this products are not for sale to customers but we added them to POS to make a sale for a virtual customer one each day for an internal use those products are tracked by serial number what

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    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.

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    ...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 example of

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    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

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    Need an experience designer & developer for my the development & designing of my financial website. As a website designer & developer we look for the below requisites : A designer should be • Specialist in creating and laying out the visual aesthetics of a website. • Excellent with conceptualizing projects • Solid artistic and design skills • Basic

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    ...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 is connected

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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.

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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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    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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    Hello, for this project I am needing - To let users send game invites to other users when a user's status is "Request Game" If user sends invite, bring up modal for user who was invited with accept and decline options.

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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

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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 .

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    Sealed
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    13 bids

    Hi I have a website [login to view URL], I would like to allow user sign/log in with their facebook account on top of old model to sign up with registration. I need expert to help me immediately. Thanks and awaiting response. Kakada

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    ...night or even prolonged breast feeding. This simple fact still remains unknown to many mothers. Sugary drinks (milk is also a sugar ..lactose) and foods are broken down by Streptococcus mutans bacteria in the mouth to form acid. This is what causes the rot (decay or cavities). By age 2 a child may need several extractions and filling and metal

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    Preferably a VB.Net (maybe C#)piece of code that I can call as a DLL from another app, or include the source code within an app as a Class. The data file is "virtual dive" It is "mounted" inside my app and appears as a hard drive for example E: Only a specific User can start my app and mount that drive. However once it is mounted, anyone has access to it. I only want the U...

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    Design and implement a c++ program to catalogue 3 different items: magical spells, items, customers. The system should act as a database and allow user to load in multiple tables of data, run basic queries on the data, and then store the data off for later use.

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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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    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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    ...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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    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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    ...side with good quality of voice. We 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

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    we need an 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

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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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    He...provide a briefing. Working hourse should be 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.

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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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    hi..i want hire for image processing project using camera pass only ir ray and block visible light

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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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    ...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 i dont make payment before see you need to show me on team

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    we need a utility app that 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

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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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    WoWonder script modification to allow non-loggedin users (guest users) to view content such as blog, events, market, forum and pages. You must be familiar with the WoWonder script. See here for details before bidding: [login to view URL]

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    ...deliveries consist of adaptation and combination of our product portfolio. To make all this work, Teleoffice have to develop applications so that the total solution helps the customer to a higher level of service to their customers. Teleoffice has developed a structure with database, web server and REST api that can be accessed via apps and web interfaces

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