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    12,540 voip sip web phone 找到的工作,价格在 USD

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

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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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    ...allready implemented A2Z voip calling system and can show me a demo. I want to start as soon as possible I want to start an international voip calling company with operations based in india. I am looking for someone who can walk me through the process and systems of a voip business and various integrations (hardware and software) The voip business will offer

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    i want to develop VoIP system. Details will be discussed over chat. urgent!!! This project is long term over 6 months.

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    Hello Our current development website is https://didicar.ca. We need to add Civic plugin to the website. [login to view URL] We tried to install several times, but cannot install and found some php errors. If you have experience with civic plugin, that will be plus. If this project going well, there will be more

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    Trophy icon PAINTnSIP | DESIGN A LOGO 1 日 left

    Design a logo "PaintnSip" We are an art studio teaching art classes for beginners how to paint. Our students are adults and we provide a..."PaintnSip" We are an art studio teaching art classes for beginners how to paint. Our students are adults and we provide a glass of wine to each student, so we call it paint and sip. We are looking for a fun logo.

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    157 作品

    ...Resultcodes - I can add here a resultcode to any projects what i have selected. -Search Records - I can find here any record in the database per project. -Search CDR - with the phone number i can see a record when its called and how many time its called by the dialer. -Download Recordings - I can download here records from the server. -Recordings - i can

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    Please set-up Twilio phone number with SIP so I can use on following site: [login to view URL]

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    VOIP - CALL DROPS 2 日 left
    已验证

    We use a XORCOM ELASTIX VOIP and have 2 office locations with a VPN tunnel. The second office cant make or receive calls which seems to be a SIP ALG or NAT issue. But need this fixed.

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    OBD CALLING 2 日 left

    Hello We have SIP,PRI and want to build script where we can make OBD Calling (asterisk),

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    ...colored box is the correct one that is coming from another VoIP provider. I gave the green color for your easier to understand. -- Executing [s@macro-user-callerid:37] Set("SIP/8902050098-00000024", "CALLERID(number)=8902050098") in new stack -- Executing [s@macro-user-callerid:38] Set("SIP/8902050098-00000024", "CALLERID(name)=+91882...

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    ...colored box is the correct one that is coming from another VoIP provider. I gave the green color for your easier to understand. -- Executing [s@macro-user-callerid:37] Set("SIP/8902050098-00000024", "CALLERID(number)=8902050098") in new stack -- Executing [s@macro-user-callerid:38] Set("SIP/8902050098-00000024", "CALLERID(name)=+9...

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    Looking for an experienced contractor to update configuration on our VOIP environment (a dozen or so phones). Currently using PiaF / FreePBX but happy to change to another distro if needed. Most phone are Mitel/Aastra 6739i and plus a couple cheaper aastra and a polycom conf phone. Key Issues / Targets: 1. Hot-desking - Configuration to allow for

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    ...Bhavin B., We are looking for a few hours of assistance with a voip/ip pabx setup issue. We are a small IT support company in Sydney. We have 3CX pabx that has been working well. As we are in an NBN enabled area we are preparing to retire our incoming ISDN lines and are setting up to test sip lines. We have an unusual router (peplink) and multiple redundant

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    Dialer and voip software 已经结束 left

    I need you to develop some software for me. I would like a dialer and voip to be developed for my company.

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    We need a Android app that take GSM Call and SMS to Send to Server so other side phone can see SMS and take calls instantly to avoid Roaming. Also maintain call logs, SMS logs, contacts on server and sync to APP that installed on other phone. We will provide you Linux server to setup.

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    Hi Kristen H.,We would like to hire you to prepare a new catalog of @ 170 products currently on a GSA contract to mirror and add to GSA Advantage through the SIP database program. All files will be supplied to you. You will need to organize in the correct format and upload both product descriptions and photos.

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    Build a full-featured conference web app. Global Reach & Best Voice Quality. Make calls to all countries on Tier-1 carrier networks with minimum latency and clear voice quality. something like group call on Facebook or Hangout. A chat room with specific members NOT anyone has the link can enter the room!!

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    We need an HTML based SIP client that can be designed to look and act like a in home intercom. For example, there should be buttons for rooms, that will let you page the rooms, and select either video or audio. This will have to be set up that each "station" can be configured which rooms it can page etc... There are a lot more features and customization

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    App to register with my Asterisk Server as a SIP extension. My Server will send VoIP calls to the App and the App will make a local call on the GSM network and path both calls together. In other words, the Android Phone will act as a VoIP / GSM gateway. Thamk you.

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    Objetivo: Provisionar teléfonos Cisco 7911G para plataforma SIP abierta (Voipswitch). Requerimiento: 1 - Selección de firmware compatible con SIP no propietario. 2 - Creación de "[login to view URL]" para provisionamiento remoto.

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    need to add voip integration to zoho crm..recommendation and configuration

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    Существующая сеть передачи данных и телефонии по SIP. Необходимо осуществлять мониторинг сети, программировать коммутаторы, медиашлюзы, поддерживать SBC и т.п.

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    I have setup a Trixbox CE server, now i want receive call that r forward thru ip to my server Just a small project, i have more work on the server later : *Custom scripts, for example, call from a list (excel or txt) automatic *Maintain server, Daily, weekly, add new trunks & routes *Security

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    Delphi sip client 已经结束 left

    I want to get Delphi sip client. delphi version is 7

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    Project ICT VoIP – Real-Time VoIP Billing for Current WHMCS version ICT VoIP Billing Panel addition (existing) VoIP  Extended Rates  Package Rates Extended New* - Enable/Disable toggle for Real-time Billing (just below VoIP Product/Package) (update table) New CDR table to collect all CDRs from [login to view URL] CRON All fields are the same as

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    Hey everyone, I'm working on a project to develop an interface for a VoIP server to allow users to add their own extensions and modify their call routing. I need a developer that is an expert in Node.js as well as PHP because this project will be developed using both languages. If you have strong experience in both languages please contact me with

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    Hello Daer we are looking for an account managers who did already work in voip field , which has the ability to discuss prices and negociate offers with sellers ,

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    Hello, i want script to Test sip accounts with Back SIP response codes Example : HOST = '[login to view URL]' SIP_PORT = 5060 LOCAL_IP = '[login to view URL]' PROTOCOL = 'UDP' USER = '509' PASS = '509123' and it will return me with [login to view URL] (200 OK or 301 Moved Permanently OR 401 Unauthorized etc...) +save

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    Hello, i want script to Test sip accounts with Back SIP response codes [login to view URL] I will provide : Server ip : Port : Tcp/Udb : Username : Password : and it will return me with 200 OK or 301 Moved Permanently OR 401 Unauthorized etc... +save output into text file +Be able to run in multi-thread Job urgent

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    Looking for VOIP app to be created for IDD calling, SMS, free call and video call apps to apps, balance transfer from app to app, top-up recharge The Virtual number will be the subscriber’s MSISDN Someone who has done VOIP app only contact. Don't waste time if haven't done it. I dont want proposals saying we can do it we have team...etc

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    hello, i have this package : [login to view URL] need to...package : [login to view URL] need to install on my server windows then build api requests to manage users and add sip accounts SO i will be able later to use on my custom cms

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    I need a python script to: 1. answer a SIP call using pjsip 2. listen & send the audio to google speech api (file or stream) 3. get the recognized text back Silence should be detected to stop the file recording or the stream to google Websockets might be used as well

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    VLC and FreePBX on Ubuntu 已经结束 left

    VLC server with the ability to stream locally, installed and tested FreePBX with the ability to connect 2 princess phones locally using either MGCP or SIP, installed and tested I have intermediate Linux knowledge and can assist

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    We need a project manager who can describe the needs of building the app, and make the app work. The design have to be connected to a Kolmisoft Mor switch

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    hi ,i am looking for a android devloper who can help me with opensource sdk for sip client like linphone , csipsimple [login to view URL] bid if you have experience with sip app , i do not use microsoft products so bid if you are experience in working with linux [login to view URL] will be long term project if satisfied with the [login to view URL] budget is 100$ for this project

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    VOIP server asterisk 已经结束 left

    Voglio un server voip che giri con asterisk

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    Customize microsip: Currently microsip allows parameters to be passed via command line Eg: [login to view URL] /hangupall [login to view URL] /answer [login to view URL] 3892014 (...) You must - change the format of the arguments, that will be passed in the format below: [login to view URL] msip:hangupall [login to view URL] msip:answer [login to view URL] msip:38192014 - add new methods that can...

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    hello, please check this doc [login to view URL] i need code to test my back Udp + Tcp connection to my sip server with show error Error = { TRYING = 100, RING = 180, TIMEOUT = 408, BUSY = 486, DECLINE = 603, OK = 200, UNAUTHORIZED = 401, FORBIDDEN = 403, NOTFOUND = 404, PROXY_AUTH_REQUIRED =

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    VOIP/SIP Project 已经结束 left

    ~50 people across 3 offices (Sydney, Hong Kong - new at WeWork, China) Looking at...(Sydney, Hong Kong - new at WeWork, China) Looking at setup cloud solution to connect existing cisco/ gransdstream sip phones. Currently using Faktortel from Australia - looking at Freeswitch/ asterisk + Twilio SIP trunk (HK phase 1) + Faktortel sip trunk (AU phase 2).

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    Project for Kristen H. 已经结束 left

    Hi kristenhutchiso8, I noticed your profile and would like to offer you my project. We would like to hire you to set up a new catalog upload to SIP with existing iProd iPrice and iPhoto database files to include @ 160 products on GSA Advantage with our new GSA contract. This is the first of 4 projects with GSA Advantage and FEDMALL we would like to

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    We are looking for a skype for business office365 consultant to assist in setting up a direct routing environment from the office365 environment to a SIP SBC.

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    Build script to record VOIP calls from system

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    ...like for you to do is: 1. Configure a Linux Instance (Please provide the build and we can organise it with AWS) to: a. Connects to a Client SIP Trunk (We will use a Hosted PABX to test) b. Connects to a Carrier SIP Trunk (We will use a Hosted PABX to test) c. Passes calls between the Client and the Carrier d. Will hide the IP of both Media and Signalling

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    I want to start voip sales business and need reseller account. Looking for experienced person who can guide me about USA and Europe zones VoIP.

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