Visual basic voip sip jobs

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    12,661 visual basic voip sip jobs fundet, i prisklassen EUR

    ...colored box is the correct one that is coming from another VoIP provider. I gave the green color for your easier to understand. -- Executing [s@macro-user-callerid:37] Set("SIP/8902050098-00000024", "CALLERID(number)=8902050098") in new stack -- Executing [s@macro-user-callerid:38] Set("SIP/8902050098-00000024", "CALLERID(name)=+9...

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    8 bud

    Looking for an experienced contractor to update configuration on our VOIP environment (a dozen or so phones). Currently using PiaF / FreePBX but happy to change to another distro if needed. Most phone are Mitel/Aastra 6739i and plus a couple cheaper aastra and a polycom conf phone. Key Issues / Targets: 1. Hot-desking - Configuration to allow for

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    9 bud

    ...Bhavin B., We are looking for a few hours of assistance with a voip/ip pabx setup issue. We are a small IT support company in Sydney. We have 3CX pabx that has been working well. As we are in an NBN enabled area we are preparing to retire our incoming ISDN lines and are setting up to test sip lines. We have an unusual router (peplink) and multiple redundant

    €32 / hr (Avg Bid)
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    1 bud

    I need you to develop some software for me. I would like a dialer and voip to be developed for my company.

    €116 (Avg Bid)
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    3 bud

    We need a Android app that take GSM Call and SMS to Send to Server so other side phone can see SMS and take calls instantly to avoid Roaming. Also maintain call logs, SMS logs, contacts on server and sync to APP that installed on other phone. We will provide you Linux server to setup.

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    6 bud

    Build a full-featured conference web app. Global Reach & Best Voice Quality. Make calls to all countries on Tier-1 carrier networks with minimum latency and clear voice quality. something like group call on Facebook or Hangout. A chat room with specific members NOT anyone has the link can enter the room!!

    €1097 (Avg Bid)
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    28 bud

    We need an HTML based SIP client that can be designed to look and act like a in home intercom. For example, there should be buttons for rooms, that will let you page the rooms, and select either video or audio. This will have to be set up that each "station" can be configured which rooms it can page etc... There are a lot more features and customization

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    29 bud

    App to register with my Asterisk Server as a SIP extension. My Server will send VoIP calls to the App and the App will make a local call on the GSM network and path both calls together. In other words, the Android Phone will act as a VoIP / GSM gateway. Thamk you.

    €550 (Avg Bid)
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    12 bud

    Objetivo: Provisionar teléfonos Cisco 7911G para plataforma SIP abierta (Voipswitch). Requerimiento: 1 - Selección de firmware compatible con SIP no propietario. 2 - Creación de "[login to view URL]" para provisionamiento remoto.

    €111 (Avg Bid)
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    4 bud

    need to add voip integration to zoho crm..recommendation and configuration

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    4 bud

    Существующая сеть передачи данных и телефонии по SIP. Необходимо осуществлять мониторинг сети, программировать коммутаторы, медиашлюзы, поддерживать SBC и т.п.

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    7 bud

    I have setup a Trixbox CE server, now i want receive call that r forward thru ip to my server Just a small project, i have more work on the server later : *Custom scripts, for example, call from a list (excel or txt) automatic *Maintain server, Daily, weekly, add new trunks & routes *Security

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    8 bud

    Project ICT VoIP – Real-Time VoIP Billing for Current WHMCS version ICT VoIP Billing Panel addition (existing) VoIP  Extended Rates  Package Rates Extended New* - Enable/Disable toggle for Real-time Billing (just below VoIP Product/Package) (update table) New CDR table to collect all CDRs from [login to view URL] CRON All fields are the same as

    €522 (Avg Bid)
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    9 bud

    Hey everyone, I'm working on a project to develop an interface for a VoIP server to allow users to add their own extensions and modify their call routing. I need a developer that is an expert in Node.js as well as PHP because this project will be developed using both languages. If you have strong experience in both languages please contact me with

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    50 bud

    Hello Daer we are looking for an account managers who did already work in voip field , which has the ability to discuss prices and negociate offers with sellers ,

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    Looking for VOIP app to be created for IDD calling, SMS, free call and video call apps to apps, balance transfer from app to app, top-up recharge The Virtual number will be the subscriber’s MSISDN Someone who has done VOIP app only contact. Don't waste time if haven't done it. I dont want proposals saying we can do it we have team...etc

    €927 (Avg Bid)
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    15 bud

    I need a python script to: 1. answer a SIP call using pjsip 2. listen & send the audio to google speech api (file or stream) 3. get the recognized text back Silence should be detected to stop the file recording or the stream to google Websockets might be used as well

    €448 (Avg Bid)
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    10 bud

    VLC server with the ability to stream locally, installed and tested FreePBX with the ability to connect 2 princess phones locally using either MGCP or SIP, installed and tested I have intermediate Linux knowledge and can assist

    €135 (Avg Bid)
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    11 bud

    We need a project manager who can describe the needs of building the app, and make the app work. The design have to be connected to a Kolmisoft Mor switch

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    23 bud

    hi ,i am looking for a android devloper who can help me with opensource sdk for sip client like linphone , csipsimple [login to view URL] bid if you have experience with sip app , i do not use microsoft products so bid if you are experience in working with linux [login to view URL] will be long term project if satisfied with the [login to view URL] budget is 100$ for this project

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    13 bud

    Voglio un server voip che giri con asterisk

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    Customize microsip: Currently microsip allows parameters to be passed via command line Eg: [login to view URL] /hangupall [login to view URL] /answer [login to view URL] 3892014 (...) You must - change the format of the arguments, that will be passed in the format below: [login to view URL] msip:hangupall [login to view URL] msip:answer [login to view URL] msip:38192014 - add new methods that can...

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    hello, please check this doc [login to view URL] i need code to test my back Udp + Tcp connection to my sip server with show error Error = { TRYING = 100, RING = 180, TIMEOUT = 408, BUSY = 486, DECLINE = 603, OK = 200, UNAUTHORIZED = 401, FORBIDDEN = 403, NOTFOUND = 404, PROXY_AUTH_REQUIRED =

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    ~50 people across 3 offices (Sydney, Hong Kong - new at WeWork, China) Looking at...(Sydney, Hong Kong - new at WeWork, China) Looking at setup cloud solution to connect existing cisco/ gransdstream sip phones. Currently using Faktortel from Australia - looking at Freeswitch/ asterisk + Twilio SIP trunk (HK phase 1) + Faktortel sip trunk (AU phase 2).

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    We are looking for a skype for business office365 consultant to assist in setting up a direct routing environment from the office365 environment to a SIP SBC.

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    Build script to record VOIP calls from system

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    1 bud

    ...with AWS) to: a. Connects to a Client SIP Trunk (We will use a Hosted PABX to test) b. Connects to a Carrier SIP Trunk (We will use a Hosted PABX to test) c. Passes calls between the Client and the Carrier d. Will hide the IP of both Media and Signalling e. Server will be built on AWS 2. Assist and provide basic training / demo to understand the configurations

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    I want a server voip

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    I am looking for someo...(Upstream Providers) and 1 x PABX / Opensips / downstream. Initial network configuration is completed. Configuration is required for the above + basic call routing and SIP headers. with the requirement for a basic configuration document (outlining works completed) Initial configuration could lead to additional future works.

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    I'm searching for a multithreaded perl script that calls a phonennumber mutiple times with predefined sip-accounts and 4 additional functions.

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    Sip dialer voice recorder coding Must know about PHP and .net programming. Experienced person.

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    SaaS Call Forwarding App Preferred Language: PHP Preferred Framework: Laravel Background: This project consist of development of back end and also implementation of product into website. The main task is the back end and admin panel. For this as a template we can use ADMIN LTE. The main function of this app is to allow customers to sign up for a call forwarding service which allows custome...

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    i have set up a new freepbx server it is online but i cannot get the SIP registration to go through and actually connect to the provider this is not a big project i am missing a setting somewhere and i need someone to connect to my server via team viewer and fix it for me

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    5 bud

    Need to create a SIP app that can install on Android. Register to SIP server Receive call from SIP server and initiate call via telephone sim card Connect audio from telephone call to SIP call

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    I have recently taken this job off a developer that has been slow in completing the development. Therefore I need someone to make changes on the look and feel of the website, add DIDWW and DIDX APIs, Mobile Top UP API like TransferTo and Prepay Nation so that clients can add their user names and logins when integrated well, implement report to the existing SMS system as there is no report there....

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    Need the Integration of Yeaster S100 with Zoho CRM for 32 Sip Accounts and 10 Lines

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    Need the Integration of Yeaster S100 with Zoho CRM for 32 Sip Accounts and 10 Lines

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    ...outbound SIP proxy server using Kamailio. There is no database or authentication required. The Kamailio server should perform the following functions: 1. Perform NAT traversal on any incoming or outgoing SIP connections using the "nathelper" module 2. Simply relay the SIP connection to an EXISTING PBX server (3CX). ***Kamailio should relay ALL SIP requests

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    4 bud

    I look for a voip dialer working that can send traffic voip and broadcast voice message

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    20 bud

    1. Install FreePBX v.13.0.121+ in to Linode server 2. configure SIP trunk engin (Australia) , (can be copy from existing PABX) 3. Connect PSTN to sip using ATA adaptor 4 Connect with Britex 24

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    Looking to have a application that registers to a sip server based on sip 2.0 and or direct ip calling have the ability to make calls. open source can be used. up to 10 favorite contacts can be created in tile format. does not need to incorportate with native android phone contacts(for now) each contact(favorite should have a place to put in rtsp string

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    ...will bring voice to text-based chatbots. The Gateway accepts SIP VoIP and then communicates with a chatbot over a JSON/REST interface. The Gateway also communicates with Google over gRPC for Speech to Text and Amazon Poly over JSON/REST for Text to Speech. Interfaces SIP Interface - SIP signalling interface. RTP Interface - G.711 and G.722 RTP media

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    Fremhævet Haster NDA
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    Hi, I need asterisk voip training and learning more about voip industry

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    19 bud

    I would like to use the uvc camera on my linphone app to make video connections. If the ...would like to use the uvc camera on my linphone app to make video connections. If the linphone app gives you the source that is normally built, register it according to the sip information I provided and make the video connection possible through the uvc camera.

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    6 bud

    ...configured to run on multiple servers for scalability and redundancy to handle over 500 million users P2p Network Basically all in network activity would be P2p All voip calls, video, file sharing, pic, everything All video calls woukd remain between the app users .with just the vcdr staying with us. We have to make sure the video quality

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    We need a solution in asterisk that help us block incoming calls with the same numbers (robot calls). We are starting a wholesale VoIP business, but we have some clients that if we have 8 or let say 24 ports open, some times they will send 8 or 24 calls depending on the ports available and all ports will be ringing at the same time with the same

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    18 bud

    Necesitamos virtualizar unas lineas móviles y fijas por medio de un gateway e implementar un pbx virtual 3cx.

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    ...develop an Android Studio application that allows to include the SIP / VOIP SDK to connect to ASTERISK The project must: 1.- Connect via SIP to an Asterisk telephone exchange 2.- Call from a number console 3.- Call from a number of clients (via Json) 4.- Configure (add / modify / delete) SIP accounts 5.- Management Module More details (ask) ////////

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    4 bud

    ...APPBuilder that allows to include the SIP / VOIP SDK to connect to ASTERISK as an option and if it were not possible in ANDROID Studio The project must: 1.- Connect via SIP to an Asterisk telephone exchange 2.- Call from a number console 3.- Call from a number of clients (via Json) 4.- Configure (add / modify / delete) SIP accounts 5.- Management Module

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    Trophy icon Content Writing Ended

    ... Its all about Fun Art, not Fine Art! And remember, we are a BYO studio, so bring along your favourite bottle or two of liquid creativity to enjoy while you paint and sip with us on the day. You supply the wine and we’ll provide all the art materials, glassware and good vibes. Leave the rest to us! This is not your typical painting class

    €9 (Avg Bid)
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