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    8,368 java sip client windows 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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    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)
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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)
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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)
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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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    ...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 --> ready in fo...

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    ...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 that the Set nr2 of numbers show the right DIDs when they call external

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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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    12 竞标

    ...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 --> ready in fo...

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    VOIP yate call pass 5 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 5 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
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    PHP SIP client 5 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 5 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

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    本地
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    Create SNOM Phone Dial Plan 4 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]

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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 SIP...

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    Looking...• Roaming Extensions • Route by Caller ID • SMS Messaging • Spell / Say • TAPI - Dialling from Windows applications • Text-to-Speech • VoIP Gateways • Voicemail • Visual Indicator for Message Waiting • Voicemail to email • Web Voicemail Interface Mobile data collection to push data into CRM Create reports on the fly.

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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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    ...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 digest, TLS client certificates Final

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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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    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 [login to view 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 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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    ...v2/v3/v5/ i want to run 32calls you can connect server to local a VPN and MASQUERADE connections or you can install any SIP PBX on local route like sofia-sip free-switch and register server to local if you want you can install a SIP Signal service on openwrt and pass call to voip device also you can bridge network and assin a server IP on local Voip

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    chan_dongle web interface 已经结束 left

    ...cards, prepaid recharge cards) 23- hide/show numbers 22- Module to change IMEI number of dongles. setup security rules to avoid automated bruteforce attacks on server to gain sip authentication....

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    VICIDIAL Expert Needed 已经结束 left

    ...expert to fulfill the following: 1- Write a script for VICIDIAL installation with a single command line 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

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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 application for both windows and mac. The soft phone will require an agent to log in and be able to show when an agent is logged in, we also need to 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

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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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    Android SIP development 已经结束 left

    Datalink International is seeking a skilled Android and iOS Application developer. Must have 5 + years of proven development skills. All A...developer. Must have 5 + years of proven development skills. All Apps developed will link over GSM Cellular networks to existing GPS and SMS Host Software. Must be familiar with SIP and serial data connections

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    ...that allow me to send calls to the mobile via SIP ( VoIP ) and this calls can me dialed ( Outbound ) Via the GSM ( the SIM card which is installed in the phone ) In more details Now we have a mobile device and Asterisk server, I want to link the mobile to the server and this server will send the call (SIP) and the mobile will make this call via the SIM

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