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    8,408 foip server sip робіт знайдено, ціни вказані в USD

    Що потрібно… «Розвернути» на сервері ASTERISK Налаштувати транк sip gsm +transfer Запис розмов та фіксацію номерів трансферних Адмін панель Панель доступу user Надаштування GSM шлюзів

    $576 (Avg Bid)
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    5 заявки
    Country Sales Manager UK 6 дні(-в) left
    ПІДТВЕРДЖЕНО

    ...end of the year, we are going to launch a SaaS in the UK. Target group are delivery restaurants without website that allows ordering. We provide leads for a start, dedicated SIP phone numbers, a CRM, and training. We offer high sales commissions and contract extension commissions. We expect impeccable UK accent - no exceptions. We are looking for a

    $4110 (Avg Bid)
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    6 заявки
    Sales Representatives for SaaS in the UK 6 дні(-в) left
    ПІДТВЕРДЖЕНО

    ...end of the year, we are going to launch a SaaS in the UK. Target group are delivery restaurants without website that allows ordering. We provide leads for a start, dedicated SIP phone numbers, a CRM, and training. We offer high sales commissions and contract extension commissions. We expect impeccable UK accent - no exceptions. When you apply make sure

    $4203 (Avg Bid)
    $4203 Сер. заявка
    6 заявки

    We are in a need of a customization of the opensource VoIP / SIP client LinPhone. The development has to be done in two phases: Phase 1: Customization of mobile + tablet apps (iOS + Android) // Objective C = iOS + Java = Android --> ready in two weeks. Phase 2: Customization of PC apps (Windows, Mac OSX and Linux) // Qt/QML = Windows, Mac OSX and Linux

    $2853 (Avg Bid)
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    15 заявки
    Low bandwidth voip solution by asterisk. 6 дні(-в) left
    ПІДТВЕРДЖЕНО

    ...we take a server which run in static IP . all client should send calls to server IP from his SIP server and we need to create separate account for each gateway . and calls should send to specific termination . call should pass with sip , g729 or g723. in client end we want to setup asterisk module which can convert calls from sip to sip and pass...

    $988 (Avg Bid)
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    I have an Asterisk server for private use already running, and would like to acidify a Trunk using an FXO VoIP Gateway, for this is necessary to create a sip trunk in Asterisk and I do not know how to do. In my attempts, I can even connect SIP between them, but I can not complete calls.

    $96 (Avg Bid)
    $96 Сер. заявка
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    I need a server administrator with experience on VoIP technology. FreeBPX, Asterisk, SIP Clients Cisco SIP phone provisioning and some other SIP phones. Developer most commute to office in Rupnaghar India. If you don't leave in India please don't apply.

    $11 / hr (Avg Bid)
    Місцевий
    $11 / hr Сер. заявка
    8 заявки
    Freepbx and Asterisk settings 5 дні(-в) left
    ПІДТВЕРДЖЕНО

    I am using for our office a pbx with regular sip phones and some softphones. The softphones work using Bria Mobile with Push notifications enabled. However, the phone doesn’t work properly on background meaning that push notifications don’t work as expected for receiving calls. Here is the page where Bria mobile explains the settings that needs to be

    $36 (Avg Bid)
    $36 Сер. заявка
    4 заявки
    Personalisation LinPhone Security & Design 4 дні(-в) left
    ПІДТВЕРДЖЕНО

    We are in a need of a customization of the opensource VoIP / SIP client LinPhone. The development has to be done in two phases: Phase 1: Customization of mobile + tablet apps (iOS + Android) // Objective C = iOS + Java = Android --> ready in two weeks. Phase 2: Customization of PC apps (Windows, Mac OSX and Linux) // Qt/QML = Windows, Mac OSX and Linux

    $1126 (Avg Bid)
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    22 заявки
    Project for Anton F. 6 дні(-в) left

    ...nr1: comes through normal ISDN to IP PBX Set nr2: comes through incoming Vitual Voip DID. All the numbers of set nr 2 work but i have 1 nr that does not work as i get error sip 503. For security reasons , you will be operating through Teamviewer on my computer which is connected to the remote site of my client where the IP PBX is. I also need to check

    $45 (Avg Bid)
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    Looking for someone to pass asterisk logs to, to determine why some calls are dropped or why calls are not routing properly.

    $52 / hr (Avg Bid)
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    16 заявки
    Customization of LinPhone SIP / VoIP Client 3 дні(-в) left
    ПІДТВЕРДЖЕНО

    We are in a need of a customization of the opensource VoIP / SIP client LinPhone. The development has to be done in two phases: Phase 1: Customization of mobile + tablet apps (iOS + Android) // Objective C = iOS + Java = Android --> ready in two weeks. Phase 2: Customization of PC apps (Windows, Mac OSX and Linux) // Qt/QML = Windows, Mac OSX and Linux

    $836 (Avg Bid)
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    10 заявки
    VOIP yate call pass 3 дні(-в) left
    ПІДТВЕРДЖЕНО

    i need to install yate on openwrt and pass calls server to my gateway we pass call useing sip to sip if you can make it please bid

    $200 (Avg Bid)
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    app development 3 дні(-в) left

    Android sip cellular gateway we are looking for an expert in developing mobile app to develop an app that will expect voice calls using usip server and dial out local number using the mobile network

    $602 (Avg Bid)
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    Wordpress Plugin for Phone Call 3 дні(-в) left
    ПІДТВЕРДЖЕНО

    Hi, I need a wordpress plugin which will allow the users to place a call on the website. I will be integrating a sip gateway for same. Users can place free calls with some restrictions like 1 minutes, or a 10 seconds ad after every 1 minute. Paying users can place unlimited calls until their credits are exhausted I wish to achieve a website like https://ievaphone

    $123 (Avg Bid)
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    Actionscript 3.0 SIP 3 дні(-в) left
    ПІДТВЕРДЖЕНО

    1. REGISTER + SUBSCRIBE to SIP server, with authentication 2. Accept INVITES 3. play inbound SIP packets and convert microphone to outbound sip That's it. If you've done it before, it's 1 hour work. Fixed payment $100. We need Actionscript in AIR.

    $30 - $250
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    0 заявки
    PHP SIP client 2 дні(-в) left
    ПІДТВЕРДЖЕНО

    PHP SIP client: 1. REGISTER + SUBSCRIBE to SIP server, WITH authentication 2. Accept INVITEs to send only, NO recv (so no traffic inbound) 3. Return WAV file to caller; depending on called number either file 1 or file 2 4. BYE to disconnect, that's it. - FIXED $40 for working code (do NOT ASK FOR MORE) - It's not even 30 minutes work if you've done

    PHP
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    VOIP Project 2 дні(-в) left

    Signalling work with CISCO CUCM Understanding VOIP - SIP (including Blind and Attended Transfer implementation) - VOIP Call analysis – Wire Shark or similar - Visual Basic Script language - Proprietary Cisco SIP protocol extensions - Cisco CUCM - Call flows of the Attended Call Transfer - Cisco Finesse handling of the Call Transfer Budget as outlined

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    Requirement two-way audio/video SIP based communications form the mobile phones to Asterisks on ARM based targets and web browser targets

    $34 / hr (Avg Bid)
    Місцевий
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    Create SNOM Phone Dial Plan 2 дні(-в) left
    ПІДТВЕРДЖЕНО

    I am looking for a SNOM phone guru that can create a dial plan for me with the following...strip the '"9". A long distance call would be 91XXXXXXXXXX. We do not want to use time outs for anything but the international calling. We require the complete string for SIP account 1, 2, & 3, Delivery in a text file. PBX: 3CX Professional Phone: SNOM D765

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    SIP To SIP register over Masquerade VPN 1 день left
    ПІДТВЕРДЖЕНО

    i have voip device and i have small router it have 32mb ram os Openwrt i want to run 32calls so you need to use SIP2SIP register USA SIP server >>Bangladesh Openwrt Router>>>VOIP Device we cant use asterisk cz its need too much space i think best libre [увійдіть, щоб побачити URL] or yate or besip [увійдіть, щоб побачити URL]

    $216 (Avg Bid)
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    add sip trunk to elastix i have sip trunk from STC

    $18 / hr (Avg Bid)
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    2 FreePBX Servers: Register SIP Trunk with TLS 17 годин(-и) left
    ПІДТВЕРДЖЕНО

    I have 2 FreePBX Servers [A & B] Server A: On Cloud And SIP Trunk has been created SSL is configured by "Let's Encrypt" Server B: On Local And SIP Trunk has been configured SSL is configured by "Self Signed" Result: Server A - SIP Trunk is appeared as not registered Server B - SIP Trunk is appeared as registere...

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    Looking to buy ready to use Call Centre CRM. Get back to me with demo. Complete documented installation and setup guide will also be required. I need you to develop some software for me. I would like this software to be developed for Linux using PHP, with Asterisk voip for call center. Solutions - Features - Call Features • Call Detail Records • Call Forward • Call Monitoring &bul...

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    Independent dialer system Закінчився left

    we wish to have someone connect and configure our [увійдіть, щоб побачити URL] to our SIP and Trunk

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    SIP面板中找到100家从事房屋建筑的公司 如果事情不明确,请写信给我,我会尽力解释。 在中文里,我通过谷歌翻译写作。 我的母语是俄语。

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    voip语音网关模块 Закінчився left

    具备sip协议,支持30路以上fxs,可以实现点对点通信,需要提供相关技术支持

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    Rebranding of Linphone Закінчився left

    I need Linphone rebranded with our Company info, logo, colors and only allow the setup of SIP accounts over TLS. Removal of options to create a linphone account. This must be done on: IOS, Android, Windows and OSX The result must be packages ready to deploy through app stores. and source code must be handed over to us, as Linphone is open source

    $572 (Avg Bid)
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    Ringless Voicemail Закінчився left

    ...without ringing the phone. App will need to have standard features such as dashboards to see send data, user data, cost from carrier data. Must be able to connect to any SIP provider. Ringless Voicemail examples [увійдіть, щоб побачити URL] [увійдіть, щоб побачити URL] [увійдіть, щоб побачити URL] https://www

    $614 (Avg Bid)
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    Tech Blogger -Telecommunications Закінчився left

    Looking for a Tech Blogger to cover all the topics around VoIP, SIP trunking, DIDs, different countries legislations and Etc. Candidate needs to be familiar with the industry and love tech-savvy content Please when apply, send your samples

    $27 / hr (Avg Bid)
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    the project is to build ionic app plugin out of linphone sip framework or linphone-cordova-plugin. Plugin should have following features. Audio calls Multiple calls management (pause & resume) Call transfer Audio conferencing (merge calls into a conference) Call History Echo Cancellation Call quality indicator Secure user authentication : md5 / SHA256

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    1 заявки
    Full Actionscript 3.0 SIP-RTMP bridge Закінчився left

    We need to bridge standard SIP calls to/from our iOs/Android app written in Adobe AIR Actionscript. In other words: handle the RTP media part of SIP to/from spk/mic. If you don't know by now what is required, PLEASE DO NOT RESPOND! Fixed payment $200.

    $200 (Avg Bid)
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    1 заявки
    Design project Закінчився left

    I’m throwing a sip and paint party. I’m looking for a artist to guide the class, to do a painting

    $83 (Avg Bid)
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    webrtc asterisk Закінчився left

    Necesitamos hacer un softphone basado en WEBrtc, que funciones desde todos los navegadores compatibles con esta tecnología, se conectan por sip a nuestro servidor asterisk

    $300 (Avg Bid)
    $300 Сер. заявка
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    I am unable to get calls from PSTN to freeswitch working. Calls from a SIP user into the switch (over the gateway) work. Calls between extensions and outbound calls work. I'm loading dialplan, configuration and directory with xml_curl (I had used a lua xml_handler script). I can confirm there is no problem on the carrier end...same carrier works with

    $180 (Avg Bid)
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    Actionscript 3.0 SIP client Закінчився left

    I need a SIP client COMPLETELY written in ActionScript, so NO external libraries or other dependencies. It should be able to connect with a SIP server, ACCEPT calls only (so don't worry about dialing and invites) and handle that 2-way phone call (mic/speaker). That's it, nothing specific! If you know what you're doing, you don't need anything else

    $64 (Avg Bid)
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    Voicemail Drop Windows Application Закінчився left

    I require a voicemail d...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 [увійдіть, щоб побачити URL] and sip.us. The application should feature the capacity to accept multiple channels for simultaneous dialing.

    $491 (Avg Bid)
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    android sip gsm gateway Закінчився left

    we are looking for en expert to develop a app that will act as sip gsm gateway i am including here a [увійдіть, щоб побачити URL] to a software that was develop for window [увійдіть, щоб побачити URL] 2, the actual software for window s i will drop it to you once we hire you 3. a suggestion

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    I need SMS Gateway Закінчився left

    Hi, I need SMS gateway, Without using any third party services like twilio etc. I need my own gateway. Freelancer must have experience with Voip/PBX/SIP etc

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    We are looking person with specific experties SIP protcol PHP DontNet Also goo din graphics

    $416 (Avg Bid)
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    Front-End for Asterisk - VoIP Portal Закінчився left

    Build a front-end with REST API for Asterisk (login/off, register customers, password reminder) in Laravel, CakePHP or similar ...with REST API for Asterisk (login/off, register customers, password reminder) in Laravel, CakePHP or similar PHP framework, manipulate the Asterisk PBX from the interface (ivr, Sip/iax, did, ami/agi, voicemail, routes etc.)

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    Build a VoIP/SMPP/PBX SMS Gateway Закінчився left

    ...SMS Gateway service that utilizes either PBX, SMPP, or VoIP - the platform must be able to run stand-alone meaning it can not use any 3rd party services such as a 3rd party sip provider, a 3rd party SMS service such as Twilio or Plivo. It must not need SIM Cards or modems. It must be able to create VoIP numbers (on it's own) ex; not using Google Voice

    $588 (Avg Bid)
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    I have A Server and server have 2 Real IP IS ON USA i have a openwrt Router its Bangladesh and router have connected a VOIP DEVICE i want to send call THIS VOIP DEVICE now i have 2options 1) Install VPN on server and openwrt ROUTER and bridge network and set a static IP same serial on VOIP DEVICE useing MASQUERADE 2) make remote connections and USE

    $563 (Avg Bid)
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    chan_dongle web interface Закінчився left

    i want to setup an gsm voip gateway GUI using asterisk+a2b+chan dongle, use huawei e173u-1 and e153u-1. i already have a running server, but i want to setup a new server with the following features in a GUI: 1-sim card suspension detection based consecutive unsuccessful number of calls and discard(dongle stop now) the suspended ones as well as not

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    VICIDIAL Expert Needed Закінчився left

    ...expert to fulfill the following: 1- Write a script for VICIDIAL installation with a single command line from [увійдіть, щоб побачити URL] 2- Tutor the proper basic setup for interconnecting with SIP trunk 3- Tutor on how to use VICIDIAL in the following concepts: a- Press-1 campaign: where we will send a pre-recorded message that has an offer, if the customer is interested

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    iOS Chat Application Закінчився left

    XMPP-based client application integrated with a SIP-based client application.

    $457 (Avg Bid)
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    ios sip client Закінчився left

    We want someone to create a very simple and basic sip client app , we have a open source sip application which you can look at for reference , we need as simple as this . add sip authentication / dial screen

    $240 (Avg Bid)
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    Need a browser extension or application for both windows and mac. The soft phone will require an agent to log in and be able to show when an ...from one agent to another agent. All call details will need to captured - Date & Time, Caller ID (Number) Start of Call and Finish, Agent ID Needs to be able to work across SIP and PBX providers using TAPI

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    Multitenant Asterisk PBX Закінчився left

    ...interface to create extensions, reports, online and offline ext, and PBX functions FUNCTIONS: - queues; - reports; (with export to .pdf and .csv file) - IVR; - group capture ext; - SIP TRUNK configuration; - DAC; Mandatory: - send documentation from development; - log from customers and admins about changes; - each user have your color/logo (size from picture

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