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    13,410 free sip applet travaux trouvés au tarif de EUR

    ...démarcher de nouveaux prospects. Nous proposons diverses offres : Serveurs téléphoniques, call-center avec webinterface, micro-paiement, services de mise en relation, lignes SIP, etc.. ...

    €11 / hr (Avg Bid)
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    6 offres

    Bonjour, Nous recherchons un intégrateur spécialisé VOIP capable d'installer la pile complète de logiciel Goautodial 3.3 et y installer un trunk SIP pour permettre la mise en place de campagne d'appels sortantes (sondages, prise de rendez vous). Nous fournirons un serveur dédié (idéalement OVH) qu'il faudra paramét...

    €545 (Avg Bid)
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    8 offres

    ...téléphone IP 1 numéro de téléphone SIP fourni par OVH 1 serveur dédié sur lequel est installé KAMAILIO. Mon téléphone IP était directement connecté à ma ligne SIP chez OVH (Je pense que ma ligne SIP chez OVH est installée sur un serveur ASTERISK chez OVH) Mais depuis p...

    €192 (Avg Bid)
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    2 offres

    Bonjour, je chercher un développeur pour coder une application sous android et IOS avec les spécifications suivantes : 1- Appelle sortant via un server(SIP/IAX). 2- Forward les appelles Voip vers GSM avec un limitation de duré max 10 minutes/24 heures( Call voip ---->GSM) 3-l'application doit être exécuter sur le smartphone en arriè...

    €1712 (Avg Bid)
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    15 offres

    ...sites web depuis 1999, recherche une personne expérimentée dans le cadre d'une mission ponctuelle. La mission consiste à - installer un trunk SIP sur un serveur Asterisk - installer minimum 5 comptes SIP pour appels sortants avec enregistrement de chaque appel - installer un panel de gestion (XIVO ou autre)...

    €176 (Avg Bid)
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    6 offres

    ...d'une mission ponctuelle. La mission consiste à terminer la configuration du logiciel VOIP Xivo sur un serveur VPS Linux : installation d'un trunk SIP installation de minimum 5 comptes SIP pour appels sortants avec enregistrement de chaque appel...

    €232 (Avg Bid)
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    3 offres
    sip account ippi.com S'est terminé left

    Cherche particulier ou centre d'appel qui utilise l'offre "call center" de [se connecter pour voir l'URL] ou ippi fr. Pour plus de Détail. veuillez me contacter

    €1287 (Avg Bid)
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    1 offres

    ...notre bon vieux PBX PSTN par une solution IP. Nous avons opté pour le model Grandstream IP PBX UCM 6104 Nous allons utiliser six internes Grandstream GS-GXP2160 et des 3 lignes SIP. Nous cherchons une personne intéressée et capable de mettre en service notre système. Nous devons pouvoir utiliser ces services. (indicatif) - Répondeur avec hor...

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    4 offres

    Nous avons un serveur asterisk configuré avec quelque sip trunk ovh. Nous avons des problemes dans la presentation des alias dans les appelles sortant. Nous avons besoins de quelqu'un qui connait bien se fournisseur Merci

    €26 / hr (Avg Bid)
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    Codez un logiciel S'est terminé left

    Se connecter sur l'équipement sip goip, lire les statistics d'appels, et déduire ceux qui ont fait au moins x nombre d'appel échoué depuis la derniere verification. Relancer les appels sur les sims d'une durée de y minutes, avec le numero à appeler en input

    €134 (Avg Bid)
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    2 offres
    Write an Android application S'est terminé left

    IAX /SIP Dialer , codec support G729,GSM,G711

    €5221 (Avg Bid)
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    5 offres

    Dans le but d'optimiser vos revenus 3D MULTIMÉDIA propose a ces free lancers de revendre ces solutions call center tels que ; interface , CRM , numéros surtaxés de tout destination ( compte sip ) , script de prospection , suivi technique , formation coaching aux centres d'appels intéressés par la réception d'appels et la prospe...

    €169 (Avg Bid)
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    3 offres
    GOAUTODIAL S'est terminé left

    ...GOAUTODIAL devra être totalement personnalise "crutomisation" par mon logo et celui du client. 7- Payement par PAYPAL uniquement. 8- Assistance au paramétrage des comptes trunck sip. 9- Parlant français. FAIRE PROPOSITION PAR MAIL. Cordialement,...

    €33 - €272
    €33 - €272
    0 offres

    salut Write an Android application appelle et partage fichier avec sip via wifi wifi+ appelle avec sip + partage fichier voir video qui ajouter je suis etudiant jai besoin bon prix

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    SolSun Entreprise S'est terminé left

    Création d'un site complet sans scrolling si possible. 4sections ou catégorie. Applet vidéo en ligne. espace publicitaire à vendre. Section Blog.

    €385 (Avg Bid)
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    VPN service + website S'est terminé left

    Hello, I would lik...need: - A comprehensive website (php + sql + + design + customer space, paypal etc.) - Either a Windows + Mac + Linux software to connect to VPN -> multiserver OR java applet -> multiserver The website should be in https Price: 600USD + bonus if good work and fast (exemple : [se connecter pour voir l'URL] or [se connecter pour voir l'URL])

    €617 (Avg Bid)
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    OpenSIPS Expert Needed 6 jours left
    VERIFIÉ

    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.

    €289 (Avg Bid)
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    ...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 his signed in and for the duration of his stay. This app will need

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    WebRTC / SIP expert 5 jours left
    VERIFIÉ

    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. [se connecter pour voir l'URL] So that they can make a 1 way audio call which is having issues with local and remote media streams...

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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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    ...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 we are more

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    Looking for freelancer to configure FusionPBX with PowerPBX hosting. [se connecter pour voir l'URL], [se connecter pour voir l'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, int...

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    Trophy icon Logo design for "Livingston Depot" 3 jours 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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    93 propositions
    Asterisk Expert 3 jours left

    ... 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 we

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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. [se connecter pour voir l'URL] [se connecter pour voir l'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 offres

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

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    ...SmartCard over USB. For this, the Android device should be detected as an USB CCID by any Windows and Linux system. For simulating a SmartCard on the Android system a Java Card Applet must be possible to use. Similiar project: [se connecter pour voir l'URL] The FunctionFS gadget should be used. If a kernel patch is necessary

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    ejabberd configuration 1 jour left
    VERIFIÉ

    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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    VoIP server for Termination S'est terminé left

    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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    ...Financial Planning ;Invest online;Testimonials ;Blogs; Careers ;Contact us With A live Chat options & watsapp messenger Few Algorithms need to be used –like calculator for – sip;retirement;marriage;child education etc...

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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.. [se connecter pour voir l'URL] New features need to add 1. SIP Calculator 2. Update Emi Calculator for variable interest rates

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    33 offres

    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 offres
    Asterisk Voice IVR setup S'est terminé left

    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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    5 offres

    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

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    5 offres

    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 .

    €5 - €10 / hr
    Scellé
    €5 - €10 / hr
    13 offres

    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

    €12 (Avg Bid)
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    1 propositions
    Set up SylkServer S'est terminé 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.

    €18 / hr (Avg Bid)
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    8 offres

    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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    Write An Erotic Short Story3 S'est terminé left

    ...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'est terminé 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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    55 propositions

    ...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

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    sip to tapi tsp file S'est terminé 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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