Sip phone configuration jobs

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    11,047 sip phone configuration jobs fundet, i prisklassen EUR

    Project for Alexander Project for AlexanderProject for Alexander

    €22 (Avg Bid)
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    5 bud
    Web development 6 days left

    I need a new website. I need you to design and build my online store. Service provider company:- - Credit report, - Home loan, mortgage loan, personal loan, busines...Service provider company:- - Credit report, - Home loan, mortgage loan, personal loan, business loan, gold loan, car loan - investments, fixed deposits, insurance, mutual fund, SIP.

    €129 (Avg Bid)
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    The business is a provider of cloud based communications which includes: Skype For Business Microsoft Teams Cloud Contact Centre Office 365 Sip Trunking Hosted PBX We require assistance in developing and revamping our website content, feature guides, product bundles, resources for customers (How2Guides/FAQ etc) some of which we have but need updating

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    21 bud

    About Our Business: Outpost Central Ltd. develops Internet of Things products and provides solutions to various applications such as commercial, industrial, environmental including farm and irrigation monitoring with its “Smart Water Meter and Sensors”. We are one of the largest and fastest growing providers of IOT Smart Water Meter solutions in Australasia. We've been in busines...

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    I have all details .i need to configure sip and configure asterisk server . VIP experts

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    Need to configure OpenVPN on Windows server. Clients need to - Access internet throught vpn - Acces files and directories like local network - Need to ping the server - Access Server through name server (DNS) and IP. Support to configure clients and confirm the succefull connection and all this itens above.

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    Cisco Router Configuration 4 days left
    VERIFICERET

    we have to Configure two Cisco Routers

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    ... 1) To set up and maintain an onboarding system that allows a user to sign up online and create their own extension, lines and phone system configuration 2) To setup an infrastructure that supports security, quick SIP registration with a proxy, easy call routing management, multi-tenant PBX for each user sign up and billing system ready for online

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    5 bud
    Voip Sip Filter 3 days left

    Hello i need to bulid sip filter to filter Retail Voip traffic

    €175 (Avg Bid)
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    7 bud
    Mobile app 3 days left

    We are looking to develop a mobile a...integrate specific API from our middle ware platform. Initially the app will only need to be for android devices. The mobile will need to integrate with a telephony platform for sip communication and a messenger platform. The majority of the mobile devices will not have cellular service and will be strictly wifi.

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    73 bud

    Dear ios programmer, i am looking for an IOS programmer to work with me in Hong Kong on an app. My office is TST Starhouse. Could we meet. My mobile is 669 664 05 and my email is the theowit3@[login to view URL] Thanks Theo

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    Looking for assistance in configuring Cisco call manager for sip trunking.

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    Adding a New Field in JDBC Plugin, So that we can get that new field while events are generated.

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    ...Linux server. I have a detailed requirement document which I can provide. The contractor(you) would be required to write small simple bash scripts that query a server configuration parameter such as logging directory or if a service is running and report = echo PASS or FAIL. This is urgent need and so I would require the scripts as soon as possible

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    I need a very simple customisation of a Android based free / open source SIP client(sCIPsimple,Linphone or anything). I want the default values of my server to be populated in place of blanks in configuration forms. So, all you need to do is replace the values and compile and give me.

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    8 bud

    Hello. We are looking VOIP Tunnel software solutions to bypass voip calls in blockage area . It is encrypts/decrypt SIP and RTP packets and send voip traffic between soft switches without blockage . We are looking experienced guy who already developed or worked on this .

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    We'd like to create an asterisk-based calling system for internal communications, including multicast messages and intercom calls. Incorporating a basic effective web interface, similar in effect to solutions such as Freepbx, with some custom functions and inputs.

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    ...of our workflow (which requires internal file references) and may need to use Egnyte or local backup on a limited basis. Looking for a freelancer that can help us with configuration and act as an advisor on an ongoing basis for all of our cloud and fileserver requirements. Please answer the following questions in your response: 1. What is your experience

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    ...initially provide reporting capabilities on Twilio's SIP trunking product. Using Twilio's REST API, you'll build the following: 1. User auth 2. Dashboard with usage data (much like what twilio provides now, but for the subaccount) 3. Historical reporting - whatever twilio provides data-wise on the SIP trunk side of the house (call date/time, ANI, duration

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    my project to connect Atmel XMEGA A1U Micro-controller through SPI to two RF micro-controllers AX8052F100 that are connected to two transceivers, one acts as transmitter and the other acts as a receiver. I need someone who can write a code on AX8052F100 to do the following : the task is to write two code in code-block to access and configure the SPI, one time to be master for the first AX8052F1...

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    I need an application that runs on Android phones (4.0 or above) and can send/receive calls through SIP (Can use any sip stack like sipdroid) and forward it to the GSM network. The application should then forward the audio and convert from SIP to GSM and vice versa. Full description below. Please review and message me with considerations. If any

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    Basically converting BICC to SIP and SIP to BICC using OpenSS7 or any other proved open source library. Performance and stability is an important factor

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    We need experienced users who have already done similar projects We need integration of embedded SIP cilent woth WEBRTC to work with FREEPBX with all functions supported : hold transfer - attended , unatennded dial etc .. link : [login to view URL] regards

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    I want to build a customized SIP/VoIP Softphone with one new Skin design as per choice and my logo for unlimited user license that runs on PCs. Windows. The server side it's done and use Asterisk server with PJSIP. I would like that the developer use other solutions to do that, doesn't start by the zero. I hope at least these features in the 1st step

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    I want to build a customized SIP/VoIP Softphone with one new Skin design as per choice and my logo for unlimited user license that runs on Android and iOS app. I would like that the cost of project be showed in separeted Android and iOS and hope that the developer use other solutions to do that, doesn't start by the zero. I hope at least these features

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    I want to build a customized SIP/VoIP Softphone with one new Skin design as per choice and my logo for unlimited user license that runs on PCs. Windows and Mac. And in the future on mobiles.

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    ...experience in this field , this is simple problem. This is my result to run my code. ***********My code************ from pycall import CallFile, Call, Application call = Call('SIP/flowroute/18882223333') action = Application('Playback', 'hello-world') c = CallFile(call, action) [login to view URL]() *******************result******************** Traceback (mo...

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    hello, community I am looking for an engineer in networks for making a set up of an extension in a sip server remote.

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    dear all , i have pcap file which it capture from wire shark for my sip switch .. i want such script which will locate pcap file , and extract audio data from it .. it is RTP ... but i do not know the encode format (g711 or g722... ) the one who can have this project should first extract audio from sample i will send to him then we can start

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    I need build a sip app with vpn, sip and ssl, please contact for detail

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    ...если все правильно и звонок пошел, то ему вылетит окошко, что вам звонят. Скрипт формирует USSD команду, исходя из введенных данных, форму которой я вам дам, логиниться на SIP сервер исспользуя Логин,Пароль и адрес сервера и отправляет комманду на сервер(делает вызов), отправляя эту команду происходит звонок с подменой номера, как только трубку поднимают

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    Company is looking vibrant developer who can build a SIP stack on a currently developing application. This facility will run on an existing app that is only available on Andriod. SIP stack must be designed to perform on single task. 1) Press a button to dial a fixed number on a SIP network.

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    ...project named "VIBER WHATSAPP WECHAT FACETIME SIP GATEWAY", that you have finished it 90% can we talk about this and maybe you can develop this for me also? Hello, We need to develop a SIP to Viber, Whatsapp, WeChat and Facetime gateway. The gateway should be able to pass voice calls incoming over SIP and forward them through Viber, Whatsapp, WeChat

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    Basic router configuration (IP Address, hostname, access credentials)  SIP Trunk configuration from Cisco Voice Routers 1 and 2 to Genesys Virtual IP  ISDN PRI ports configuration  Dial Plan configuration  Inbound and Outbound Calls Testing

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    Hello, We need to develop a SIP to Viber, Whatsapp, WeChat and Facetime gateway. The gateway should be able to pass voice calls incoming over SIP and forward them through Viber, Whatsapp, WeChat and Facetime to complete the call to the called party number. The development platform/operating system is not important. The project should be completed

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    15 bud

    Se trata de crear la configuración necesaria para que con una Raspberri pi sea posible: - Recibir llamadas de un telefonillo (Door Phone) - Comunicar con el telefonillo. - Activar la apertura de puerta. Una vez terminado el proyecto será necesario entregar: - Archivo img o similar para montar en tarjeta SD - Diagrama de conexión entre la raspberri

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    I need Sip Dialer With Include OpenVPN. Sip Dialer connect by OpenVPN. OpenVPN Connect only Sip dialer..... you need use PJSIP OpenSource SDK for SIP Dialer. I need Very Simple Apps... i need support g729 voice code. I can you OpenVPN server side. if you have good experience please contact me.

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    need to install cpanel on vultr along with configuration and extensions to do full setup the job must be done through teamviewer if you are not happy with this please do NOT bid!

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    We have A2billing server and we are building an app for our resellers to topup customers through app. We need the working API to Create and Manage custo...are building an app for our resellers to topup customers through app. We need the working API to Create and Manage customers - mainly Create customer with Callerid (including sip account) and topup.

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    Hi there, I have a start up business with a website that needs some final touches and help to configure and develop. My business is called The Boxing HIIT and I have built the site on a wordpress template, which has been half set up. There is some functionality and content upload that is needed to be helped with and configured. The current site can be viewed at [login to view URL] and would like i...

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    I'm looking for someone with OpenVPN experience. Using openvpn 2.4.6-I602 and NetGear Nighthawk R8000. Error: Fri Aug 31 07:09:05 2018 WARNING: No server certificate verification method has been enabled. See [login to view URL] for more info. Fri Aug 31 07:09:05 2018 OpenSSL: error:140AB18E:SSL routines:SSL_CTX_use_certificate:ca md too weak Fri Aug 31 07:09:05 2018 Cannot load certificate ...

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    download elastix pbx and connect it with sip and Asternic Call Center Stats Lite

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    New server 's speed optimization set up (Centos 7.5 + Plesk onyx) for wordpress sites. Need to be done same configuration like source server + correct ftp access in 2 plesk servers Full written instractions requested.

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    I need a website where I could sell mutual funds and insurances to my clients, and post regular information about the best funds and insurances. Also site must have SIP calculator and loan calculator. I need it preferably on wordpress.

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    Mandatory Skills: 1) Hands on expertise in SIP, RTP and RTCP 2) Good knowledge on TUN,STUN, NAT 3) Free-switch working knowledge on audio conference using SIP and Web RTC 4) Good knowledge on RTP Proxy and routed audio conferences concept where media would flow via free switch RTP Proxy 5) Working experience of High Availability and Cluster 6) SDP

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    I have recently obtained a new NAS, and need some help to setup backups (from PCs and Server, over LAN and internet) and also setup TV (DS Video Station). Possibly also configure it as an FTP server as well (with access from the internet).

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    I'm looking for programming help with [login to view URL] and Kamailio. Must have experience with SIP and preferably experience with [login to view URL] library. Will need 1 - 4 hours per week for 3 - 5 weeks.

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    We would like you debug and fix our sip trunking /Ext to Ext calling setup to work with our providers so that we may use Elastix MT 3.0 to the fullest with multiple organizations and users per organization, Elastix MT.3.0 has been installed already just need to fix some bugs. We are currently able to have our handsets receive registration, and our

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