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    8,462 build sip client web application 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...

    €547 (Avg Bid)
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    ...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...

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    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 arri...

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    La société LWS, spécialisée dans l'hébergement de 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 ...

    €177 (Avg Bid)
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    ...LWS, spécialisée dans l'hébergement de sites web depuis 1999, recherche une personne expérimentée dans le cadre 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 appel...

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    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

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    ...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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    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

    €27 / 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

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    2 offres
    Write an Android application S'est terminé left

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

    €5241 (Avg Bid)
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    ...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 prospection sur diff&e...

    €170 (Avg Bid)
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    3 offres
    GOAUTODIAL S'est terminé left

    ...simultané et compte admin et agent . 6- 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 - €273
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    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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    Skype SIP client in PHP 6 jours left
    VERIFIÉ

    We have a few Skype Connect profiles with inbound numbers at...script should (persistent) connect to Skype servers, login and answer the calls by returning a recording (depending on the inbound number). That's it folks:-) It's all standard SIP, pretty straight forward. --> $100 FIXED (it's max 3 hours work, MAX, 30 minutes if you've done it before)

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    Hi, I look for Freelancer who can implement Asterisk with my MGCP provider. Not SIP. Can somebody do? Thanks

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    hi i want to register sip to sip and pass call g729 problem is i dont want to use big pc and lots of HW so i found a program its allow to do that i want [se connecter pour voir l'URL] i want to run this program on a router Router Link [se connecter pour voir l'URL] CPU: MediaTek MT7628N CPU Cores: 1 CPU MHz: 580

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    Programming Bandwith.com 3 jours left
    VERIFIÉ

    Help programming sip endpoints, applications voice & sms in [se connecter pour voir l'URL] I need to connect my [se connecter pour voir l'URL] account to my PBX, be able to forward DID numbers to another number, and direct sms's messages. I would also like to complete the project described here [se connecter pour voir l'URL] Prior experience

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    Multichannel Phone Dialer 3 jours left
    VERIFIÉ

    In need of software or asterisk/freepbx hosted solution for "Voice Broadcast" or "Robo Dialer". - Connects to SIP Trunk - Alternate Caller ID; Area Code Caller ID campaigns with alternating numbers - Ability to play and add audio files messages that are randomized - Multichannel lines (100) to make hundred or thousands calls per minute - Choose number

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    I need help with setting u...setting up the Outbound Parameters of my 3CX-based SIP Trunk (in the cloud). I am able to make & receive calls using Nexmo. But with Anveo, I only succeeded one time to make a call by tweaking the Outbound Parameters in the 3CX SIP Trunk Setup and I was never able to reproduce that again. Knowledge of SIP trunking a MUST.

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    sip to Native dialer APP 2 jours left
    VERIFIÉ

    Sip to native dialer app Features 1. this app need to be working in the background 2. the app file will have with all configuration user password to register to our sip server so client do not need to use user and password 3. the app register it self with a remote sip server 4. the result for this

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    i need to VOIP SIP to SIP call G729 optimization bandwidth SERVER A =asterisk server Server B =Openwrt ROuter server A&B connected to VPN VOIP device COnnected to Openwrt ROuter we send calls Server A to VOIP device SIP to SIP codec g729 its use per calls 30KBPS we want to conpress it if you can make it under 15KB we are happy i saw too many peppole

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    5 offres
    VOIP, SIP Expert 2 jours left
    VERIFIÉ

    I’m looking for an expert of VOIP, SIP, Hack for educational purpose

    €173 (Avg Bid)
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    10 offres

    I want my IP PBX to be connected to Twilio WebRTC so that can be able make calls using Twilio SIP account. Calls from my PBX will come from an IP so I need someone who can make the set up for me to start using Twilio voice trunk.

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    SIP Proxy Server install 1 jour left
    VERIFIÉ

    i want SIP Proxy for my voip Dialer and voip device i have asterisk server i need to register my soft phone to my SIP server but sometime we have blockage issue so i need a SIP Proxy if you can provision then good for me check the Attach FIie if you can make it let me know

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    you have to install skype for business 2019 and set it up. we have multiple sip phone numbers which have to be also connected to skype for business 2019 for in and outgoing calls.

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    i had asterisk/freepbx installed on centos recently. One DID on the SIP trunk is working fine but other DID has two issues. Both DIDs are on same network and similar SIP phones 1. there is a delay of 50 sec after caller presses a digit to reach staff before phones ring 2. phones ring only once....so calls are lost freelancer who installed this for

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    I have a very serious problem with my hosted PBX: It is available online but I already change the default 5060 to 10... Firewall and fail2ban are already setup. Sip phones can call using dialcheap and megavoip. The problem is that hackers make calls using dialcheap and megavoip and I cannot stop it. ##, *2, Tr are already disabled

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    ...com/click-to-call/ etc You don’t even need a fax machine…,Receive Fax from customer as PDF in your Email. The system will support setting up/administrate the call center SIP accounts (with hunt groups). Of course system needs configuring Asterisk for custom DID:s, call forwarding, internal calling, voicemail, ability to activate/deactivate service

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    Hi, I deliver VoIP services to my customers and i use 90% of the time the Yealink SIP phones. The problem in 50% of the cases is that when the client is turning on their voicemail, they can't see it anyway if it is turned on yes or no. I know it is possible to use something with XML and a webserver to create a connection with the phone which can enable

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    required a AVAYA Call Exchange PHP scripts for SIP, call recording, call records and so on ... along with DB

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    3 offres

    I need someone to setup my softphone with vpn and connect to an existing hosted pbx. my sip and voip are blocked by my isp

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    3 offres

    Cisco voice gateway connection to SIP server and PSTN on E1 PRI

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    I need someone who is expert in hacking, also expert in networks, cyber security, VOIP, Telecommunication servers.

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    sipxcom sip system setup S'est terminé left

    ...3.64, but i cannot setup outside domain, i want to use sip outside network, i know there are instructions but i cannot fix the problem. I know i just miss very little thing. i will give you teamviewer to setp up my pc. You need to do: 1)help me to set up sip system using sipxcom, i need the sip can be use in internal and external network. 2) tell me

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    I have just registered new firm in financial services. Stock broking, mutual fund, SIP investment advising, Various types loans & insurance services will be our products. I need logo & letterhead for this.

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    WE need to connect a cisco 7206VXR with VIC adapters for voice connection to the PSTN and to a Mizu SIP Server for users authentication and billing using sip/h323 protocol. We already done a major part of the setup and configuration of both the cisco and the softswitch but calls are still failing to authenticate on the cisco gateway to the pstn network

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    VOIP ASSISTANCE S'est terminé left

    VOIP ASSISTANCE SIP DEVELOPPEMENT

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    Trophy icon JT&T Brand 1 jour left

    I’m looking for a logo for business cards and letterhead I’m looking for some flyers so that I can advertise VoIP and SIP services the Name of the company is JT&T Communications we offer best in class SIP and VoIP. And phone system analysis design and implementation

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    Garanti
    €195
    653 propositions

    1 - We need to configure Asterisk FREP...documentation from the work with detailed description with step by step of how to configurate it. 3 - The developer will register some of our SIP channels into the freepbx. We currently have both sip clients: - SIP Phone - SIP on PC 4 - We need someone who is an expert on it, because we need it asap. Thanks.

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    Project for Prostitov A. S'est terminé left

    ...discutere i dettagli tramite chat. A) TRUNKS INFOSTRADA Protocollo SIP username 390415541985 From Domain [se connecter pour voir l'URL] secret GZD397XU Host [se connecter pour voir l'URL] port 5060 Register String 390415541985@[se connecter pour voir l'URL]:GZD397XU:390415541985@[se connecter pour voir l'URL] B) TRUNKS MESSAGENET Protocollo SIP username 540...

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    IAX SIP VOIP Softphone S'est terminé left

    ...you have IAX SIP VOIP Softphone system and apps? We would like to do similar system like www . zoiper . com Zoiper is a phone solution perfectly fit for end users, service providers, call centers or any business willing to benefit from VoIP communications. Zoiper softphone is a standalone client-side software VOIP phone application and is not bundled

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    HSMWorks-sip panels/ Panels S'est terminé left

    ...Experience using HSMWorks is a must. We have ongoing work for a competent programmer who is able to build machining routines for our 5 axis router. We are looking for someone who preferably has experience in processing large timber panels or Sip panels. Our models are prepared in Solidworks and require converting using HSMWorks software. We are

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    Lightweight SIP Client with Dynamic CRM which would display the customer information. CRM Field and information would display dynamically and action should collect and send to server Database. Required Features: SIP Call with transfer/conference/hold/Mute facility SIP messenger. Dynamic CRM-Dynamic field header and value display for each call according

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    VOIP, SIP, Cyber Security S'est terminé left

    I need some one who is expert in cyber security & hacking and specially VOIP, Sip, you should be expert in this field.

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    Avaya VOIP Phone System Help S'est terminé left

    ...we talk together. Current System: • Avaya IP Office 500 V2. Build Version – 9.0.2.0 Build 860 • 18 Avaya Phones – Model 1416 • 23-24 extensions Needed • 4 Trunk Cards That I see • 5 Analog Phone Numbers Following Licenses currently on system: • SIP Trunk Channels • SIP Trunk Channels • SIP Trunk Channels &b...

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    as the title indicates, need sip installed on cisco 7960 to work with astersik. total 10 phones

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    Hi. We need to configure asterisk on our ubuntu machine. We need someone high skilled to do the job. We will also need to make it visible over internet with public IP. Also we need to configure some IP phones on it. After all, we need a manual with step by step about how to configure it.

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    I have lan calls working fine, but require someone with Cisco/Asterisk SIP knowledge to help resolve this issue with NAT, no audio not working.

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    ... The application will feature voicemail detection function where an outbound call is placed from a csv list and is directly router to the recipient's voicemail without ringing. Afterwards a pre recorded message is automatically left. For placing calls I would like to use SIP trunking or VoIP such as [se connecter pour voir l'URL] and sip.us. The application should

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