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    8,219 web based java sip application งานที่พบ, การเสนอราคา USD

    PBX System SIP Server on Cloud สำหรับธุรกิจขนาดกลางและเล็ก โดยไม่ต้องลงทุนอุปกรณ์ Server PABX PBX ต่างๆ ไม่ต้องเตรียม infra ให้ยุ่งอยาก เคลื่อนย้าย ไปตาม Site สะดวก โทรหา site to site ฟรี บริการค่าโทรแบบเหมา แบบรายนาที ทั่วโลก

    $540 (Avg Bid)
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    Project for Bhavin B. 8 วัน left

    ...retire our incoming ISDN lines and are setting up to test sip lines. We have an unusual router (peplink) and multiple redundant internet connections. We have spend many hours trying to setup our router to enable SIP connectivity however without success. We are looking for someone with 3CX, SIP and good networking /router skills. Hourly rate to be discussed

    $37 / hr (Avg Bid)
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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.

    $250 (Avg Bid)
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    Code an HTML based SIP client with certain customizations. 2 วัน left
    ยืนยันแล้ว

    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

    $1355 (Avg Bid)
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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.

    $631 (Avg Bid)
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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 "[เข้าสู่ระบบเพื่อดู URL]" para provisionamiento remoto.

    $130 (Avg Bid)
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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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    $269 \u0e01\u0e32\u0e23\u0e1b\u0e23\u0e30\u0e21\u0e39\u0e25\u0e40\u0e09\u0e25\u0e35\u0e48\u0e22
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    ...for call $duration = rounded up time to calculate cost based on Package Rate inbound or outbound as in [เข้าสู่ระบบเพื่อดู URL] $realtimecost = calculation of call to remove from credit with Tax associated with Client. i.e Client 10563101 (from Prevost, Quebec) Tax Rule rate = 14.98% 2min outbound call on SIP Canuk 200 plan 0.020 = 6sec increment 120sec 2min * 0

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    Back end developer for project using Node.js and PHP 7 ชม. left
    ยืนยันแล้ว

    Hey everyone, I'm working on a project to develop an interface for a VoIP server to allow users to add their own ...languages please contact me with a proposal and make sure to write "iluitech" in your message so I know you actually read the requirements and know that you can do it. If you have SIP experience and know PHPAGI it would be a huge plus!

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

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    Voip sip registering script 2 ชม. left
    ยืนยันแล้ว

    Hello, i want script to Test sip accounts with Back SIP response codes [เข้าสู่ระบบเพื่อดู 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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    hello, i have this package : [เข้าสู่ระบบเพื่อดู URL] need to...package : [เข้าสู่ระบบเพื่อดู 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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    Recognize audio & get the text by google speech หมดเขตแล้ว left

    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

    $523 (Avg Bid)
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    VLC and FreePBX on Ubuntu หมดเขตแล้ว 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

    $158 (Avg Bid)
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    Looking For Android sip app Devloper หมดเขตแล้ว left

    hi ,i am looking for a android devloper who can help me with opensource sdk for sip client like linphone , csipsimple [เข้าสู่ระบบเพื่อดู 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 [เข้าสู่ระบบเพื่อดู URL] will be long term project if satisfied with the [เข้าสู่ระบบเพื่อดู URL] budget is 100$ for t...

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    Customize microsip - open source sip dialer หมดเขตแล้ว left

    Customize microsip: Currently microsip allows parameters to be passed via command line Eg: [เข้าสู่ระบบเพื่อดู URL] /hangupall [เข้าสู่ระบบเพื่อดู URL] /answer [เข้าสู่ระบบเพื่อดู URL] 3892014 (...) You must - change the format of the arguments, that will be passed in the format below: [เข้าสู่ระบบเพื่อดู URL] msip:hangupall [เข้าสู่ระบบเพื่อดู URL] msip:answer [เข้าสู่ระบบเพื่อดู URL] msip:381920...

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    want build code using nmap หมดเขตแล้ว left

    hello, please check this doc [เข้าสู่ระบบเพื่อดู 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 หมดเขตแล้ว 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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    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

    $250 (Avg Bid)
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    Skype for business Direct Routing หมดเขตแล้ว left

    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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    ...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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    9 การประมูล

    I am looki...connections (Upstream Providers) and 1 x PABX / Opensips / downstream. Initial network configuration is completed. Configuration is required for the above + basic call routing and SIP headers. with the requirement for a basic configuration document (outlining works completed) Initial configuration could lead to additional future works.

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    Perl Script SIP Calling Bot (Prankscript) หมดเขตแล้ว left

    I'm searching for a multithreaded perl script that calls a phonennumber mutiple times with predefined sip-accounts and 4 additional functions.

    $233 (Avg Bid)
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    Sip dialer voice recorder coding หมดเขตแล้ว left

    Sip dialer voice recorder coding Must know about PHP and .net programming. Experienced person.

    $215 (Avg Bid)
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    Help setting up Freepbx หมดเขตแล้ว left

    i have set up a new freepbx server it is online but i cannot get the SIP registration to go through and actually connect to the provider this is not a big project i am missing a setting somewhere and i need someone to connect to my server via team viewer and fix it for me

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    SIP to Local call App หมดเขตแล้ว left

    Need to create a SIP app that can install on Android. Register to SIP server Receive call from SIP server and initiate call via telephone sim card Connect audio from telephone call to SIP call

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    Need to integegrate yeaster S100 with ZOHO CRM หมดเขตแล้ว left

    Need the Integration of Yeaster S100 with Zoho CRM for 32 Sip Accounts and 10 Lines

    $157 (Avg Bid)
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    Need the Integration of Yeaster S100 with Zoho CRM for 32 Sip Accounts and 10 Lines

    $235 (Avg Bid)
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    ...outbound SIP proxy server using Kamailio. There is no database or authentication required. The Kamailio server should perform the following functions: 1. Perform NAT traversal on any incoming or outgoing SIP connections using the "nathelper" module 2. Simply relay the SIP connection to an EXISTING PBX server (3CX). ***Kamailio should relay ALL SIP requests

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    install FreePBX v.13.0.121+; หมดเขตแล้ว left

    1. Install FreePBX v.13.0.121+ in to Linode server 2. configure SIP trunk engin (Australia) , (can be copy from existing PABX) 3. Connect PSTN to sip using ATA adaptor 4 Connect with Britex 24

    $78 (Avg Bid)
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    Looking to have a application that registers to a sip server based on sip 2.0 and or direct ip calling have the ability to make calls. open source can be used. up to 10 favorite contacts can be created in tile format. does not need to incorportate with native android phone contacts(for now) each contact(favorite should have a place to put in rtsp string

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    SIP Media Gateway Engine for Chatbot หมดเขตแล้ว left

    ...MessageGateway Engine will bring voice to text-based chatbots. The Gateway accepts SIP VoIP and then communicates with a chatbot over a JSON/REST interface. The Gateway also communicates with Google over gRPC for Speech to Text and Amazon Poly over JSON/REST for Text to Speech. Interfaces SIP Interface - SIP signalling interface. RTP Interface - G.711

    $26 / hr (Avg Bid)
    แนะนำ ด่วน NDA
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    linphone + uvc camera หมดเขตแล้ว left

    I would like to use the uvc camera on my linphone app to make video connections. If the ...would like to use the uvc camera on my linphone app to make video connections. If the linphone app gives you the source that is normally built, register it according to the sip information I provided and make the video connection possible through the uvc camera.

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    ...user is doing in realtime from admin Basicall its almost all P2P . This project would be developed using Technology such as NODE.js/ React native app and react native web When building the app like whatsapp we need to make sure that all in network calls stay on the app server , we a place on the admin to add all the millions of numbers we

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    Asterisk Black listing หมดเขตแล้ว left

    We need a solution in asterisk that help us block incoming calls with the same numbers (robot calls). We are star...if any new call is coming in, then look into the Database to see if the incoming number is in the database, and if it is, then just block the call and sent a "486 Busy Here" Sip response, otherwise asterisk can place the call as usual.

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    Android with SIP/VoIP caller program หมดเขตแล้ว left

    I need to develop an Android Studio application that allows to include the SIP / VOIP SDK to connect to ASTERISK The project must: 1.- Connect via SIP to an Asterisk telephone exchange 2.- Call from a number console 3.- Call from a number of clients (via Json) 4.- Configure (add / modify / delete) SIP accounts 5.- Management Module More details (ask)

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    I need to develop an application in APPInventor or APPBuilder that allows to include the SIP / VOIP SDK to connect to ASTERISK as an option and if it were not possible in ANDROID Studio The project must: 1.- Connect via SIP to an Asterisk telephone exchange 2.- Call from a number console 3.- Call from a number of clients (via Json) 4.- Configure (add

    $30 - $250
    $30 - $250
    0 การประมูล
    Trophy icon Content Writing หมดเขตแล้ว left

    ... Its all about Fun Art, not Fine Art! And remember, we are a BYO studio, so bring along your favourite bottle or two of liquid creativity to enjoy while you paint and sip with us on the day. You supply the wine and we’ll provide all the art materials, glassware and good vibes. Leave the rest to us! This is not your typical painting class

    $11 (Avg Bid)
    รับประกัน
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    Paint and sip for kids หมดเขตแล้ว left

    looking for an artist/painter/instructor to teach kids and art classes/parties for different ages. 3-5 hours a day, 2 times a week. Ideal candidate should be positive, patient, energetic. Looking to hire immediately.

    $27 (Avg Bid)
    พื้นที่
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    Vicidial webform หมดเขตแล้ว left

    ...gin=gs102&phone_pass=865748&fronter=321&closer=321&group=MORTCP&channel_group=MORTCP&SQLdate=2018-05-22+113857&epoch=1527014338&uniqueid=1527014305.751&customer_zap_channel=SIP/enterprise-000000ff&customer_server_ip=&server_ip=69.64.71.182&SIPexten=gs102&session_id=8600056&phone=7029973001&parked_by=55502&dispo=&dialed_n...

    $146 (Avg Bid)
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    Asterisk Management System with PHP หมดเขตแล้ว left

    ...not the best with Asterisk and I am unsure how the control panel would communicate with Asterisk. This is what we need the control to do: - Add a SIP Trunk Channel (For PSTN connectivity) - Remove a SIP Trunk Channel - Add Telephone Numbers to the database which can be used by extensions for outbound and inbound calls - Remove telephone numbers - Create

    $739 (Avg Bid)
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    Asterisk dialplan หมดเขตแล้ว left

    ...can be achieved with a call conference or call bridging. when dialing user (A) calls user (B) they are put into a call conference or bridge. if the dialer(user a) hangs up(sip 487) before user b answers, a disconnects but the conference / bridge channel stays open until b hangs up or for a max of 30 seconds cli needs to be passed from a to be etc

    $196 (Avg Bid)
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    Moile Sip App coding หมดเขตแล้ว left

    We need sip protocol writer. There is sip application for testing for voice also need recording of voice on app

    $540 (Avg Bid)
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    I need a python script to: 1. answer a SIP call using pjsip 2. listen & send the audio in chunks to google speech 3. get the recognized text back I would like to either detect the silence to send the audio to speech service or continuous send and get the text as it is ready

    $528 (Avg Bid)
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    Install IP PBX หมดเขตแล้ว left

    I am seeking a techie, who could install IP based PBX on my cloud setup. It will be a software based PBX either 3CX or any other asterisk based. Here are the precise requirements:- 1. Install and configure IP based PBX system.(Product shall be decided mutually). 2. Configure TLS/SRTP to traverse voice over ssl. 3. Ensure that traffic is passed through

    $419 (Avg Bid)
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    We need a Phone Sales Person (English Speaking Countries) Part time for at least 20h/week. PEQRFECT ENGLISH IS A REQUIREMENT. You will work with our CRM / SIP Phone.

    $5 / hr (Avg Bid)
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    Trophy icon Design a fun poster for our Paint Studio หมดเขตแล้ว left

    We would like a poster designed which will be hung on the wall of our Paint and Sip studio. The poster should be 300 dpi/high resolution as it will be printed out at about 45 inch wide to 90 inch high (so thats the ratio on size ) and hung on our wall. The poster shows a list of fun things that people might do at our studio while at our events. So

    $80 (Avg Bid)
    รับประกัน
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    35 ผลงาน
    i want create my own website หมดเขตแล้ว left

    i want to create own website like mutual fund , sip and financial and admin login client login and sip calculator also available in my web site and nse,mcx stock market api also required

    $187 (Avg Bid)
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    26 การประมูล

    I'm looking for someone to develop an Outlook plugin for windows that you login with a VoIP SIP extension credentials and whenever a call comes in it searches the number in Outlook and if it finds a contact it opens up that contact in a new window. If it doesn't find a contact it opens a new contact window with filed in the details from the caller ID

    $621 (Avg Bid)
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    5 การประมูล