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    11,347 video sip symbian pekerjaan ditemukan, seharga USD
    Softphone android Berakhir left

    Un softphone que maneje cuentas SIP para ANDROID

    $524 (Avg Bid)
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    Saya Pernah Direkrut Ngentry Dikantor Badan Stastistik Itu Terjadi Tahun 2015 . Jam Ngentrynya Dimulai Jam 12 Siang Sampai Jam 4 Pagi , Kebetulan Saya Sip Jam 9 Malam Sampai Jam 4 Pagi .

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    setting IP PBX Berakhir left

    jasa setting untuk keperluan kami yaitu...yaitu: Billing System / Budgeting per ID / User Recording All Call IVR / Auto Attendance Call Fax to Mail & Mail to Fax Bisa Cabang IP PBX Yealink BLF [Busy Lamp Field] Sip Messenger connect Android APPS VOIP saat ini kami memakai program elastix dengan hardware openvox a814hj0582xk dengan chip 8fxo

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    ...Care untuk bekerjasama dengan PT Kronus Indonesia - Perawat, Fisioterapi, Okupasi Terapi, Terapi Wicara & Bidan - Laki-laki dan Perempuan - Usia 21th – 45th - Memiliki STR/SIP yang masih berlaku - Mampu bekerjasama dengan Team / Independent - Penempatan wilayah Jakarta dan Luar Jakarta (Bandung, Semarang, Yogyakarta, Surabaya, Medan, Makassar atau

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    this program for information tcooperation in Goverment "Pemerentah kabupaten WONOGIRI"(Dinas Koperasi dan UMKM) INDONESIA, Can use for ALL Dinas koperasi and UMKM ind Indonesia

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    HARJONO file Berakhir left

    ...Blackberry, Nokia, dan merek lainnya :) , Apa itu Android Android adalah sistem operasi yang digunakan di smartphone dan juga tablet PC. Fungsinya sama seperti sistem operasi Symbian di Nokia, iOS di Apple dan BlackBerry OS. Android tidak terikat ke satu merek Handphone saja, beberapa vendor terkenal yang sudah memakai Android antara lain Samsung , Sony

    $182 - $548
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    We are looking for someone who has good experience in video chatting app and H.323, SIP. It is very urgent project, so if you don't have good experience in this field, please don't waste our time. Thank you.

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    sip VoIP dialer with rcs futures voip call sms chat video ex...

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    i need someone that can login to my computer and setup my fusion PBX should accept calls from vitelity SIP Provider, and also to make calls thru my outgoing provider.

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    Robocall server 5 Hari left
    TERVERIFIKASI

    A center of family planning is looking for a Robocall solution to remind patient to visit doctors, etc. Informa...reminder them the date of the visit. at the end of each day, a full report should be given. The voices will be prerecorded by our team here. We will use goip gsm gateway (sip) for the calling. We have servers to host the application.

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    ...– simple dial surface, numbers, #, * etc. all via PC/laptop windows 10. NO landline or mobile phone for dialing only the PC and PC microphone + PC headset. Each user has own SIP account and should use the program separately. 6. Dial #2 – open phone book. Select a city/name/phone number from phonebook. Given list each item (each lien) is one-click call

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    I need a simple softphone that allow to SIP register on a freepbx server. With just the dialer, not accessing phone contacts and all the features that most softphones comes. It must run in the background when closed so it can receive calls. Guess is just that, hope your offers! Thanks

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    Goautodial specific functions 2 Hari left
    TERVERIFIKASI

    ...any number in any list: show a different calling number on customer phone for each attempt. Will give you 10 calling numbers that we have 3) Set up load balancing on 3 trunks (sip provider). When dialing on lists, send one call to one trunk, next to the second trunk and next to third trunk and so on... 4) Failover : after dialing a number on a trunk, if

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    Digital logo 2 Hari left

    I'd like to create a logo for my website and work as an author and speaker on personal development. My initial idea is a play on my initials (SIP) and I've attached a sketch of versions that I have used. Would like someone to play around with that and come up with something professional/digital.

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    We have setup bitrix24 onpremise and we have asterisk-vici installed. We want to use SIP connector. Any one with experience in setting up onpremise Bitrix24 and telephony with SIP connector can respond.

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    Hi All I have a SIP client on my network, its a Intercom. It registers to my FreePBX/Asterisk solution and makes calls to my mobile phone fine. I want to intergrate Skype because Amazon Echo's can answer/make calls using Skype. The intention is to receive the Intercom call on my home network to an Echo Dot, which I believe will work when the FreePBX

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    ...kind of logs. [2019-02-14 08:56:15] WARNING[7089] chan_sip.c: Autodestruct on dialog '223205511-3759123182-18707@TcomNextoneSBC4-DRC.mydomain.com-onnet_1-b2b_1' with owner SIP/telsam_ip-00003486 in place (Method: BYE). Rescheduling destruction for 10000 ms. if you interested in I can share you more information. and budget is negotiable fixed or hourly

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    We need an expert in OpenSIPS that can change the SIP response of 200 ok to something else or delay it full details will be given later.

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    ...allow only Checked In guests to sign in using surname and Room number. once signed in also allow guest to connect to the Hotel WIFI also within the APP we will need a build SIP/VOIP on the App which will interface with hotel PABX and the app will become the guest extension as long as his signed in and for the duration of his stay. This app will need

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    WebRTC / SIP expert 23 jam left
    TERVERIFIKASI

    The client is using astrix server. They will give us a vpn connection to their network And we need to setup WebRTC client to talk to astrix server. We need to use open source functionality working which it can't currently get a function to work within a type. [login untuk melihat URL] So that they can make a 1 way audio call which is having issues with local and remote media streams to work. ...

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    Find me some Leads for a VoIP Service. The Company offers cheap rates for VoIP calls via Mobile App (Andorid, Symbian, IPhone). Please, I do not need Email Addresses to send bulk Emails. I need some traffic twoards my site. And acctual leads that generating paying customer. Any suggestions appreciated.

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    Want to start a VOIP/calling card solution ,need to install the necessary software (asterix,freepbx,a2billing etc).Advice the best solution

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    ...SSL, Predictive dialing, IVR, Incoming calls, and potentially GSM hardware upgrade (need to evaluate solution) with Asterisk expert. - Server health - IP phone connection - SIP line - Bandwidth and latency of system to network - Connectivity Other task related to asterisk server that will be needed in future: We need to have a setup, where we are more

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    Looking for freelancer to configure FusionPBX with PowerPBX hosting. [login untuk melihat URL], [login untuk melihat URL], Debian ...Plans, PBX Extensions, I/O Trunks, Voicemail, DISA, IVR & local number display on the go, Call Forwarding, Call waiting, Linphone setup and instructions, call recording, setup sip trunks. for 4 - 6 lines. PHP or LUA scripting, VOIP, interconnect

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    Trophy icon Logo design for "Livingston Depot" Berakhir left

    "Livingston Depot" is the Name we need a logo for.. - "Sip & Shop Station" is the Tag Line This is a boutique cafe that is located in Livingston, Tennessee. This shop offers local artists, hand crafted items for sale, with a small cafe offering beverages, and lunch items. The keyword inspirations are elegance, down home, boutique, station, depot

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    Asterisk Expert Berakhir left

    ... Predictive dialing, IVR, Incoming calls, and potentially GSM hardware upgrade (need to evaluate solution) with Asterisk expert. - Server health - IP phone connection - SIP line - Bandwidth and latency of system to network - Connectivity Other task related to asterisk server that will be needed in future: We need to have a setup, where we

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    I need a call with multi skype user via SIP. and i saw 3cx gateway for skype program. this product is no longer available for download nor maintained. [login untuk melihat URL] [login untuk melihat URL] The 3cx gateway for skype is running only windows 7. and needed skype old version

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    must be connect multi Skype user to a SIP-PBX. and call via skype to pstn or mobile number. multi skype user can running on asterisk machine. extensions are calling via each skype-account

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    We need an implementation of the services mentioned in the subject. Thanks.

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    I need help you configure my ejabberd server with sylkserver for translate jingle/sip between xmpp client and SIP proxy and/or PBX. Goal is that XMPP client can make a call through the PBX to the outside work (as a regular UA) and that the XMPP client can be called via the PBX.

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    We have a multi-tenant PBX system and need a simple SIP phone provisioning tool written in Javascript Node.JS, using Angular or React, running on a MEAN/MERN stack. This will not be integrated with the PBX, however our B/OSS system will add, suspend, and delete tenants and extensions to this new system you will create via a REST API. A commercial

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    Setup a VoIP server for termination the requirements are: 1. SIP connections to gsm gateways through VPN is required 2. Billing software 3. Load balance traffic to gateways 4. setup connections with 3 clients

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    ...Financial Planning ;Invest online;Testimonials ;Blogs; Careers ;Contact us With A live Chat options & watsapp messenger Few Algorithms need to be used –like calculator for – sip;retirement;marriage;child education etc...

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    I am looking for an experienced FreeSwitch engineer who can work with latest freeswitch version with SIP Trunks to do following tasks I will be passing destination number , CLI , ring duration , max call duration , audio file , unique call id to Freeswitch (JSON API) , the dial plan should start dialing numbers and plays a voice file when the call

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    I have already built app on play store called "Loan Emi Calculator". Please review the app befo...Calculator". Please review the app before you bid, here is a store link.. [login untuk melihat URL] New features need to add 1. SIP Calculator 2. Update Emi Calculator for variable interest rates

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    I have an existing Shoretel system with a PRI switch. Using the SETU VTEP device I want to replace my existing PRIs with SIP trunking by using the SETU VTEP as a gateway. I need help programming the device to integrate with my existing Shoretel system.

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    1) Set up Asterisk. You will get a machine with Linux OS of your choice. 2) Install UniMRCP and UniMRCP Kaldi Plugin. Configure SIP line for inbound calls. We will provide the URI to configure in the plugin. 3) Configure an Inbound call plan that does the following for inbound calls: (a) Prompt the caller to punch in a 4 digit code. (b) Post

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    convert cisco ip phone to sip for working in yeastar s100

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    Need to Configure 2TATA Sip Trunk line on my freepbx via remote desktop like TeamViewer. FreePBX hardware is assembled and installed freepbx.

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    Sip client for Raspberry pi3 We need to make sip client for raspberry pi3 on parking system machine, Software must have source code in python and auto start with raspberry linux SIP client must call just one number that we insert in code, over GPIO inputs trigger sip client start call. (Gpio inputs must configurable in code) Sip client must have

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    Hi, I'm looking for someone to connect a twilio number that we currently have to a sip endpoint thats already configured for voice. We are using the X-lite Voip client and would like to get SMS functionality enabled on it.

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    SIP CLIENT FOR RASPBERRY PI3 this software to be developed for Linux using Python. SIP client for Raspberry pi3 We need to make sip client for raspberry pi3 on parking system machine, Software must have source code in python and auto start with raspberry linux SIP client must call just one number that we insert in the code, over GPIO inputs we

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    Looking for someone knowledgeable on freepbX and mobile application development to create a VOIP application for customers service . The reference app is named as "smart line" You can check the app's features . I need the exactly same features with some new designs .

    $6 - $11 / hr
    Bersegel
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    I need an android library who permits to have analog modem call (CSD Data Call, like a fax) to 9600bps, as well as symbian os and windows mobile

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    Set up SylkServer Berakhir left

    Set up sylkserver to translate jingle/sip between xmpp client and SIP proxy and/or PBX. Goal is that XMPP client can make a call through the PBX to the outside work (as a regular UA) and that the XMPP client can be called via the PBX.

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    Add the ability to enter sip trunk credentials to asterisk

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    I am looking for a senior architect/developer to design and develop gsm telecom solutions based on ss7 or diameter or sip communication protocols. The scope will be discussed later, the pay will be very good

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    ...writer to write an original story featuring a seductive buxom woman who's big unwieldy breasts contain a milk that will wickedly shrink the size of any man who drinks it. One sip is enough to make him a midget and too much will send him down to bug size or smaller. Find a way to work this weird concept into a sexy short story with a full plot and I'll

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    Trophy icon Need a logo for a company beverage Berakhir left

    Hello, I need a logo for a beverage company. The name is "Soufrière Hills" The tag line is “every sip makes you family and comforts your soul.” Please use the following hex codes only: #549127 #10375d #edc857 Use all three or two colors in any combination. Please be as creative as possible. This logo is for a beverage that is from the island of

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    ...need some kind of bandwidth compression system ( up to 60-80% than usual SIP calls )from Server A to Server B. Server A = Asterisk server Server B = Asterisk Client server Explanation of the scenario: 1. server A ( asterisk server, with static IP) receiving VoIP calls , with sip protocol, using G711,G729 and/or G723.1 codec and sending calls to Server

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