Design sip client using java source codeproiecte

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    8,611 design sip client using java source code proiecte găsite, la prețul de USD
    Scrie un software S-a încheiat left

    Am nevoie de cineva sa imi configureze un telefon cisco 7960 pentru serviciul de sip.

    $37 (Avg Bid)
    $37 Oferta medie
    1 oferte

    Salut, Am un VPS cu FreePBX functional. Doresc sa adaug HylaFax + AvantFax si sa pot sa trimit faxuri prin iax sau sip. Am vazut ca ai facut ceva similar. 40$ fixed rate Poti sa ma ajuti?

    $23 - $194
    $23 - $194
    0 oferte

    Salut, Am un VPS cu FreePBX functional. Doresc sa adaug HylaFax + AvantFax si sa pot sa trimit faxuri prin iax sau sip. Am vazut ca ai facut ceva similar. 40$ fixed rate Poti sa ma ajuti?

    $31 (Avg Bid)
    $31 Oferta medie
    1 oferte

    Sunt interesat sa discutam pentru un proiect android, de voip (sip) asa cum este si viber sau skype... in primul rand ma intereseaza aplicatia pentru client dar si partea de server ca aplicatia sa fie functionala

    $30 - $250
    $30 - $250
    0 oferte
    Aplicatie android voip S-a încheiat left

    Sunt interesat sa discutam pentru un proiect android, de voip (sip) asa cum este si viber sau skype... in primul rand ma intereseaza aplicatia pentru client dar si partea de server ca aplicatia sa fie functionala

    $150 (Avg Bid)
    $150 Oferta medie
    1 oferte

    Ma intereseaza sa discutam mai multe legate de o aplicatie de voip pe android ca si viber , caut pe cineva care sa faca partea de client sau sa customizeze un client open source de sip existent

    $231 (Avg Bid)
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    Ma intereseaza sa discutam mai multe legate de o aplicatie de voip pe android ca si viber , caut pe cineva care sa faca partea de client sau sa customizeze un client open source de sip existent

    $150 (Avg Bid)
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    1 oferte
    Aplicatie de voip (sip) S-a încheiat left

    Buna seara, as vrea sa discutam cu privire la o aplica...de voip in care utilizatorul sa se logheze pe baza de nr de tel si sa primeasca sms pt confirmare precum si prin facebook, sa aibe o parte din functiile viber, configurarea SIP sa se faca printr-un api fara ca sa trebuiasca sa faca ceva utilizatorul final pentru a configura la inceput aplicatia.

    $150 (Avg Bid)
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    Soft Phone S-a încheiat left

    ...inregistreaza prin 3 parametrii: Server Username Password (campurile username si password devin hash-uri si nu pot fi vazute in clar) Telefonul functioneaza atat pe protocolul SIP cat si pe IAX2. Telefonul poate deschide oricate linii in paralel. Eventual, fiecare linie noua deschisa, le trece pe celelalte pe Hold, in sensul ca se aude numai linia

    $250 - $750
    $250 - $750
    0 oferte
    Goautodial specific functions 6 zile left
    VERIFICAT

    ...function: when agent click on button, the client can enter credit card details by DTMF and get back to agent to continue discussion 2) Regarding any number in any list: show a different calling number on customer phone for each attempt. Will give you 10 calling numbers that we have 3) Set up load balancing on 3 trunks (sip provider). When dialing on lists,

    $238 (Avg Bid)
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    ...to launch our new product to the world Naturally flavoured Coffee beans. Imagine its early morning your somewhere between being asleep and being awake and you have that first sip of the freshest, best quality, real Arabica coffee with a hint of rich Caribbean vanilla rum and tropical spices or smooth almond marzipan. Ahhh absolute haven, aromatherapy

    $250 (Avg Bid)
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    $250
    4 intrări
    Digital logo 6 zile left

    I'd like to create a logo for my website and work as an author and speaker on personal development. My initial idea is a play on my initials (SIP) and I've attached a sketch of versions that I have used. Would like someone to play around with that and come up with something professional/digital.

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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.

    $578 (Avg Bid)
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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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    ...kind of logs. [2019-02-14 08:56:15] WARNING[7089] chan_sip.c: Autodestruct on dialog '223205511-3759123182-18707@TcomNextoneSBC4-DRC.mydomain.com-onnet_1-b2b_1' with owner SIP/telsam_ip-00003486 in place (Method: BYE). Rescheduling destruction for 10000 ms. if you interested in I can share you more information. and budget is negotiable fixed or hourly

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    We need an expert in OpenSIPS that can change the SIP response of 200 ok to something else or delay it full details will be given later.

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    ...guest within the hotel. App needs to interface with PMS which will allow only Checked In guests to sign in using surname and Room number. once signed in also allow guest to connect to the Hotel WIFI also within the APP we will need a build SIP/VOIP on the App which will interface with hotel PABX and the app will become the guest extension as long as

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    WebRTC / SIP expert 4 zile left
    VERIFICAT

    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. [conectează-te pentru a consulta linkul URL]

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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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    ...required, and also who can help us to support any client Asterisk instances that we manage. Currently system needs changes and 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

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    Looking for freelancer to configure FusionPBX with PowerPBX hosting. [conectează-te pentru a consulta linkul URL], [conectează-te pentru a consulta linkul 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 sc...

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    "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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    ...and also who can help us to support any client Asterisk instances that we manage. Currently system needs changes and 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

    $17 / hr (Avg Bid)
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    5 oferte

    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. [conectează-te pentru a consulta linkul URL] [conectează-te pentru a consulta linkul URL] The 3cx gateway for skype is running only windows 7. and needed skype old version

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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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    1 oferte

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

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    ejabberd configuration S-a încheiat left

    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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    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

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    5 oferte

    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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    ...Solid artistic and design skills • Basic to advanced coding knowledge • Very proficient in graphic design • Familiarity with SEO and marketing As a Web Developer A developer should have a quality to take a mock-up of a website and bring it to life through coding. Should be proficient several different coding languages like HTML, PHP, Java etc. Additionally

    $178 (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

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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.. [conectează-te pentru a consulta linkul 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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    Asterisk Voice IVR setup S-a încheiat left

    ...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 the 4 digit code to a web URI we provide. (c) Prompt user to speak

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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 ...

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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 th...

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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
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    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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    Set up SylkServer S-a încheiat left

    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)
    $21 / hr Oferta medie
    8 oferte

    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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    Trophy icon Need a logo for a company beverage S-a încheiat left

    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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    ...BW in client 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

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    sip to tapi tsp file S-a încheiat left

    ...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 give you a list of 1...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 S-a încheiat left

    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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