Joomla free call module sip工作

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    8,369 joomla free call module sip 找到的工作,价格在 HKD

    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 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

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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.

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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。

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    ...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),以及可交付的中转服务器; 9、并不需要完整的语音通讯软件,只需要能够实现语音传送、播放的功能模块即可。 10、演示内容:A用户点击“通话”按钮,中转服务器可通过设置好的IP地址,找到B用户,并发送通话请求。B用户接受请求后,双方可以进行通话。 11、你可以采用第三方组件,例如SIP,但是需要告诉我们是哪家的,效果如何。因为考虑到我国网络问题,所以建议是自己搭建服务器。 12、我们已经开发的软件可以在安卓、苹果商店找到,叫IEMAKER,通过安卓盒子,可以实现直播,但是目前还不能实现声音直播,因此,需要开发这个模块,用于该软件。

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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.

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    Freepbx and Asterisk settings 6 日 left
    已验证

    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

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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

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    ...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 that the Set nr2 of numbers show the right DIDs when they call external numbers... I need to check that Fail2Ban that

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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.

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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

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    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

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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

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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

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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.

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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

    ...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 Deadlin...

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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

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    Create SNOM Phone Dial Plan 3 日 left
    已验证

    ...digit long distance calling Toll Free 800/855/844/866/877/888 International Calling with 011 Emergency calling with 911 Directory Assistance with 411 We use "9" to get signify an outside call. So when dialing a 10 digit local number we want 9XXXXXXXXXX sent to the PBX as it will strip the '"9". A long distance call would be 91XXXXXXXXXX. We do no...

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    ...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] up to you as you like i just need pass a call on g729 codec under 15KBPS now as you

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    add sip trunk to elastix i have sip trunk from STC

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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

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    ...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 • Cal...

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    we wish to have someone connect and configure our [login to view URL] to our SIP and Trunk

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    Rebranding of Linphone 已经结束 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

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    Ringless Voicemail 已经结束 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 [login to view URL] [login to view URL] [login to view URL] https://www

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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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    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

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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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    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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    ...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 [login to view URL] and sip.us. The application should

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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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    I need SMS Gateway 已经结束 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

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    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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    ...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

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    Need content on mutual fund - SIP

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    ...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 2nd IP on my voip device i need a solutions to pass call server to local VOIP device i cant use any PC

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    chan_dongle web interface 已经结束 left

    ...dongles. 9- set a call timout to avoid voicemail and its charges, also those timed out calls should be considered as not completed, not answered with 00:00 duration which affects acd ( i have this problem). 10-just click beside the dongle active call to spy on it(quality control) ( i have it done just need it in GUi) 11-set dongle call duration limit

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    VICIDIAL Expert Needed 已经结束 left

    ...from [login to view 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, he presses 1 and the call will transferred to an available agent. b- Vote/Feedback/Survey: a pre-recorded

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    iOS Chat Application 已经结束 left

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

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    ios sip client 已经结束 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

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    Need a browser extension or appli...understand when the agent is available, on do not disturb or on a call. Need the availability to move a call 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 已经结束 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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    Setting up AsteriskNow 已经结束 left

    Phase 1 Need to setup asteriskNow with ippbx (tata telecom). Setting up sip trunk. redirecting calls on specific numbers to specific sip no. At specific work hours or if specified through API, then to redirect calls to Mobile no. Setup recording of calls.

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    ...using Zoiper as softphone application to register extensions with IPPBX System. Our employees using Zoiper from inside and outside office. The protocols we are using are SIP and IAX We started a new office in Egypt, and would like to use Zoiper to register with same IPPBX System by employees working there. Unfortunately, its not working. We have

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    Linphone customization 已经结束 left

    I need the Linphone client rebranded for Android/IOS and Windows/OSX. Simply with our logo / app-icon, removing options for creating or using linphone account, so only sip account is configurable. Furthermore changed to our colors. Packages for IOS and Android should be ready for deployment to the public stores, so we need the sourcecode when

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