Sip voip app mobile工作

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    12,756 sip voip app mobile 找到的工作,价格在 USD

    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。

    $52 (Avg Bid)
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    4 竞标
    voip语音网关模块 已经结束 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

    $1180 (Avg Bid)
    $1180 平均报价
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    直拨,回拨,点对点语音,视频电话,文字聊天,可见好友位置,语音对讲,文件传输,充值支付,一键语音客服,邀请好友。业务管理系统。(Removed by Freelancer.com admin for breaching Terms and Conditions)

    $2631 (Avg Bid)
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    Sip client with the pjsip or other sip stack. Need the cloudy Address Book accroding to the API. Need the presence status. The project should be finished in 1 month.

    $1532 (Avg Bid)
    $1532 平均报价
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    VoIP Video IM等即时通讯功能。 支持Android,IPhone,Window Phone等客户端应用。 熟识SIP Webrtc XMPP。 提供Demo。

    $5555 (Avg Bid)
    $5555 平均报价
    1 竞标

    1、开发一个用于移动APP软件的即时语音通讯模块,发送端是IOS或者android平台,接收端是android平台; 2、即时通讯模块功能主要实现单对单语音即时聊天,无须实现视频聊天; 3、要求操作简单,点击“通话”功能按钮即可实现通讯; 4、由于即时通讯双方可能位置很远,因此必须经过互联网传输,需要中转服务器进行语音媒体的中转。中转服务器可以采用成熟的第三方软件或者协议,但是在项目完成后可以交付给我们; 5、发送方网络的最低可能是ADSL 2M。但是延时不能超过1S; 6、语音质量要高,不能有杂音、回音等问题,需要可以用于商业行为的那种; 7、需要有可供调用的组件和调用方法,我公司可通过调用方法进行上层功能的开发; 8、需要提供所有源代码和演示版本(演示版本的运行平台包括android、IOS)

    $2109 (Avg Bid)
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    i need voip website wuth all tools to start here in kuwait jjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjj

    $155 (Avg Bid)
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    voip website 已经结束 left

    i need voip website wuth all tools to start here in kuwait jjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjj

    $150 (Avg Bid)
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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.

    $10 / hr (Avg Bid)
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    Freepbx and Asterisk settings 6 日 left
    已验证

    ...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 completed

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    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

    $1120 (Avg Bid)
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    ...I have a IP PBX with 2 sets of 10 numbers. Set 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

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    i have a Linux server its running on Apache tomcat i have some .jsp file on webapp forder its an api for vos3000 like i want to insert update delete but i dont know how to do you have to fixed it check the attach file here is all info i dont release any fund without test

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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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    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

    $956 (Avg Bid)
    $956 平均报价
    7 竞标
    VOIP yate call pass 4 日 left
    已验证

    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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    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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    VOIP phone system 4 日 left
    已验证

    we need a phone system based on twilio platform.

    $652 (Avg Bid)
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    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 4 日 left
    已验证

    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
    $30 - $250
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    PHP SIP client 4 日 left
    已验证

    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 4 日 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

    $30 / hr (Avg Bid)
    本地
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    Create SNOM Phone Dial Plan 3 日 left
    已验证

    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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    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 [login to view URL] or yate or besip [login to view URL]

    $216 (Avg Bid)
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    Step 1: Homepage with logins Step2: Login page with details step3: VOIP or Live Stream features

    $390 (Avg Bid)
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    Integrazione api voip 2 日 left
    已验证

    Vorrei integrare un sistema VoIP nel mio sito web così cliente possono vedere la loro fattura e possiamo vedere i loro menù.

    $365 (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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    Paid VOIP services app 2 日 left
    已验证

    Application that allows people to instantly phone a specialist, on a pay-per-minute basis, to receive recommendations

    $837 (Avg Bid)
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    ...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 registered! I need to get all Registered, and the call is go through SIP...

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    ...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 • Call Parking • Call Queuing • Call Recording • Call Transfer &bu...

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    I want to use raspi as gsm gateway. This is the basic task. Details will be explained.

    $540 (Avg Bid)
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    we wish to have someone connect and configure our [login to view URL] to our SIP and Trunk

    $1063 (Avg Bid)
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    I have an iOS app which uses Sinch api for the VoIP calls, i'd like this replaced with an opensource solution, such as FreeSWITCH. The app uses usernames and not phone number. It's in Obj-C, with php services, mysql DB, hosted on Amazon AWS. I expect excellent clear quality calls.

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    Voip Calling App -- 2 已经结束 left

    Voip Calling App with unlimited Bandwidth, It will Be used for calling East African Countries, it must be functional.

    $170 (Avg Bid)
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    Voip Calling App 已经结束 left

    Voip Calling App with unlimited Bandwidth, It will Be used for calling East African Countries, it must be functional.

    $289 (Avg Bid)
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    I need to create and confing server. i need to forward calls to it and to see the header and make tests can be done on linuxcpanelwinvps

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    Rebranding of Linphone 已经结束 left

    ...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, and we

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    Ringless Voicemail 已经结束 left

    ...must connect from carrier to carrier to allow VM to be placed on a number 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 [login to view URL] [login to view URL]

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    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

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    La RCSOft software specializzata da anni in sviluppo software gestionali, per potere integrare al suo interno un CRM richiede una interfaccia con il Centralino Voip 3CX v15 in poi, per potere effettuare delle telefonate in uscita e per potere recuperare il numero chiamante in entrata in modo da posizionarsi sul form del cliente. L'interfaccia deve

    $25 / 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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    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.

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    Design project 已经结束 left

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

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    webrtc asterisk 已经结束 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)
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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

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    Dinstar GSM gateway VOIP 已经结束 left

    I have installed Dinstar GSM gateway, but I have voice quality problem. Who can solve it, please contact with me

    $34 (Avg Bid)
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    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

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    I require a voicemail drop ...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 [login to view URL] and sip.us. The application should feature the capacity to accept multiple channels for simultaneous dialing.

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    android sip gsm gateway 已经结束 left

    we are looking for en expert to develop a app that will act as sip gsm gateway i am including here a [login to view URL] to a software that was develop for window [login to view 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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