Asterisk sip tls srtpcông việc

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    18,918 asterisk sip tls srtp công việc được tìm thấy, giá USD

    Tôi đã có sẵn tổng đài Asterisk, đã có sẵn SuiteCRM. Cần lấy API từ tổng đài Asterisk để kết nối với Suitecrm.

    $30 - $250
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    Project for Vu L. Đã kết thúc left

    Hi vule83, mình đang có dự án Odoo CRM tích hợp tổng đài voIP sử dụng Asterisk 11. Bạn đã từng làm dự án nào tương tự chưa? Nội dung cụ thể thì cho mình số dt của bạn để trao đổi cho nhanh nhé

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    Chào các bạn, hiện tại tôi đang rất muốn kết nối vtiger opensource 7 với tổng đài assterisk, nhưng tôi không biết làm việc này, rất mong các bạn hỗ trợ tôi kết nối giữa 2 ứng dụng này, tôi sẽ trả phí kết nối cho các bạn. Trân trọng cảm ơn!

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    Project for nttranbao Đã kết thúc left

    Hi nttranbao, I'm a newbee asterisk. Now, my project are doing function monitor video call with asterisk. Do you do that function? If you can, you can send email for me ( [đăng nhập để xem URL]@[đăng nhập để xem URL]). Thank you :) Hy vọng anh biết tiếng Việt. vì tiếng anh của em không tốt lắm. Em mới nghiên cứu asterisk và hiện tại dự án của em đang muốn l...

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    Mình cần hỗ trợ config Elastix, inbound, outbound và sip trunk

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    Java meeting application 6 ngày left
    ĐÃ XÁC THỰC

    Need to fork apache open meeting server and customization of UI=> custom template and security implementation + Asterisk integration.

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    Implement IP PBX with Asterisk 6 ngày left
    ĐÃ XÁC THỰC

    We are a company with 15 sites in Portugal. with over 40 employees spread across the various centers. We intend to have a solution to talk between centers via IP, conference calls, IVR messages, automatic call forwarding, etc.

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    Hello, i am working on a project. I have ordered a VOIP number, i want to automatically accept each incoming call to the number and record the call. I also need to get the recorded file via an api. Currently i don't know how to do this. I like to have an application that can be run either be run on windows, raspberry, in the cloud (i prefer cloud) Details about the phone number, sip number...

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    Migrate FreePBX to Azure Cloud 6 ngày left
    ĐÃ XÁC THỰC

    The project involves migration of existing Asterisk PBX to a Azure cloud. There is existing Azure account. Install FreePBX on Azure cloud. Install Linux and FreePBX GUI. After the installation, migrate exiting asterisk configuration and voice prompts and confirm that the system works accordingly. Copies of the Asterisk files will be provided (have to be specified). I would also like consultation...

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    need someone for make backup of asterisk 6 ngày left
    ĐÃ XÁC THỰC

    Hi im looking for someone who will take asterisk config backup from existing centos 7 and then making installer for centos 7

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    Tendo un PBX Elastix 5 de 3CX en AWS con 7 extenciones moviles y un troncal SIP que me funciona actualmente con las llamadas entrantes, pero necesito que me configuren las llamadas salientes y ademas que me instalen 2 telefonos fijos GrandStrean GXP126 en 2 de las 7 extenciones.

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    Goautodial v4 Voip dialer crm experienced freelancer need for install and make SIP/Agent settings

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    Python script 5 ngày left
    ĐÃ XÁC THỰC

    Hello i need a python script to check my smtp servers in bulk (more than 200 smtp) so i want to create a txt file which contain smtp serversin this format " IP:port,username,pass,tls(if its required) " in checking process if an smtp works fine the script will send a test to my email from the same smtp and so on... for parameters User, pass and ssl sometimes they are not required so t...

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    Candidate should be familiar and comfortable with - Preference would be given to having experience on IVR and Call Center application development. - Hands on experience in Kamailio SIP Server - Integration with Database (PostgreS-Q-L) with knowledge of queries and stored procedures. - Experience in installing Hosted VoIP Candidate Profile The candidate must have: - 2 - 4 years of experience - Stro...

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    Hello. I'm look for someone to draw a set of plans for a log home with a day or two. 25' x 41' (center line) building. 8' basement. 8' first level with loft. 12" logs. 4/12 roof pitch. SIP 1st level floor and roof. East/West eve overhang - 4'. Gable end eves - 6'. I've attached The Trapper model home as an example to what I'm looking for....

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    Nồi Bật Gấp
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    We need to develop a SIP to Whatsapp gateway. The gateway should be able to pass recorded audio as voice calls over SIP and forward them through whatsapp to complete the call to the called party number. The development platform/operating system is not important. The project should be completed either by using the Linux/Windows whatsapp executables, or by using the Android / Windows Phone mobile ve...

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    Details will be shared winning bidder. please bid if you have the experience.

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    Call transfers on PBX 4 ngày left
    ĐÃ XÁC THỰC

    To make assisted transfers on FreePBX/Asterisk with one click. FreePBX / Asterisk with WebRTC ! This will be the first milestone to be discussed. For the whole projrct, see attachment.

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    VICIDIAL Customization -- 2 3 ngày left
    ĐÃ XÁC THỰC

    Need a call center solution using VICIDIAL, Need all the standard features of Vicidial and it should work with Asterisk 1.6 or 1.8 or both. (or higher) All Vicidial UI and branding has to be replaced with our custom brand name, and interface should be changes sufficiently to look different. Logo branding removed Marketing automation Lead analytics dashboard Analytics dashboards SMS +++ To Email ...

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    astreisk_auto call just one rules number 1 2 ngày left
    ĐÃ XÁC THỰC

    i want make rules in my server 1- i want port number 1 call number 2 to i mint 2 i want when i send call for port number one when it finsh it make stop this port 2 or 3 min exe exten => _[*#0-9]!,n,Dial(Mobile/A0/${EXTEN},60,tTm,K) exten => _[*#0-9]!,n,Dial(Mobile/A1/${EXTEN},60,tTm,K) --------------- A0 port number 1 A1 port number 2 If a user make a call from a sip account, asterisk w...

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    astreisk_rules 2 ngày left

    i want make rules in my server 1- i want port number 1 call number 2 to i mint 2 i want when i send call for port number one when it finsh it make stop this port 2 or 3 min exe exten => _[*#0-9]!,n,Dial(Mobile/A0/${EXTEN},60,tTm,K) exten => _[*#0-9]!,n,Dial(Mobile/A1/${EXTEN},60,tTm,K) --------------- A0 port number 1 A1 port number 2 If a user make a call from a sip account, asterisk w...

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    We need an app developed on kazoo monster-ui so that SIP Phones/Devices can be auto provisioned just like this: [đăng nhập để xem URL] The script in the video is discontinued/not working as it was for version kazoo 3.x, and the latest kazoo version is 4.3 (major update). I don't need every handset brand/model, just 1 model but the app should be flexible so that we can add more brands/models...

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    WebRTC-Asterisk-FreePBX 1 ngày left
    ĐÃ XÁC THỰC

    PLEASE READ THIS FIRST ! Asterisk with WebRTC for Inbound call center. This is only for freelancers with a very broad knowledge-base and who know how to get things done with the terminal. Some requirements: FreePBX/Asterisk, Java Script, pjSip, Bootstrap, CSS, HTML5, PHP, Web Dev., Stun / Turn Servers, Queues / Parking for calls, SSL, Ubuntu, etc. There are a few fixes to make (see attachment). ...

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    Asterisk agi script manged from php page 1 ngày left
    ĐÃ XÁC THỰC

    High level description: The IVR need to "collect" digits from the coller (personal ID number) via DTMF or Google voice to text service. The collected info needs to be stored on SQL table and passed as a parameter to dynamic (user defined) URL Task prerequisite: Asterisk 11, freepbx 2.11 Asterisk 16, freepbx 15 Full installation doc. All parameters will be via php page on /var/www...

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    Develop ionic application where SIP calls can be made

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    VoIP Cloud PBX migration 17 giờ left
    ĐÃ XÁC THỰC

    We are seeking a VoIP Engineer to move clients from Kerio Operator to Vital PBX You will need to configure Vital PBX with the same setup/settings the client current has in Kerio Operator. No client is to go live on the new PBX, this is just setup without activating the SIP Account. First test will be our office to go live, pending no issues one by one we will do the final step to turn other cli...

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    I have a project using SIM868 / SIM800 with ESP32 and Azure IoTHub, wich uses MQTT secured (TLS). SIM868 has hardware TLS but not work as expected so I want to use software. There are some ideas I looked arround: -WifiClientSecure It is a wrapper of WiFiClient using mbedTLS library to get software TLS support. -BearSSL -ESP32-PPPOS-EXAMPLE wich uses TLS in some part -[đăng nhập để xem URL] (I th...

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    Build phone system I am looking to set up sip trunking/pbx etc. I'm wondering if you are able to help me. I'm not very knowledgeable on the subject matter but i'll tell you what I have, and what I need. I have: - a VPS server with Centos 7 installed - a Twilio account with 2 numbers -number 1 is being forwarded directly from Twilio to my cell phone -number 2 is involving...

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    App para Android en Ionic 3 giờ left
    ĐÃ XÁC THỰC

    APP a desarrollar. Objetivo: La app va a tener los datos desde el servidor para autoprovisionarse de los datos de setup de las aplicaciones a instalarse, mantener en el servidor el status del teléfono, actualizar las nuevas entradas del registro de llamadas y la interfase de usuario. De forma funcional la app va a tener el siguiente comportamiento: Por webservice con el servidor manda: a....

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    I have developed BlockChain based TLS based notary service project. Wanted to hire a BlockChain Expert who can test and evaluate my project and compare the results with other BlockChain technologies

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    Hi, we need an experienced Xamarin developer to develop a cross-platform (Android/iOS) chatting app for internal use of a company. The project is split into phases. The first phase is a VoIP SIP softphone that supports Voice calling, Instant Messaging, File Transfer and Presence. Exactly the same as Whatsapp. The following features are required in the app for this phase of the project: - Voice ...

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    Nồi Bật
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    sip asterisk specialist Đã kết thúc left

    sip asterisk specialist to get in touch with our team in order to drive inner development of a core product. SIP webrtc, SIP API sip asterisk master configuration

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    We need an expert in jack audio connection tool box to create virtual cable between sip client and android speaker and mic

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    Asterisk - Help for Asterisk Đã kết thúc left

    PLEASE READ THIS FIRST ! Asterisk with WebRTC - A real Specialist is needed.! For Inbound call center with Asterisk and WebRTC. This is only for freelancers with a very broad knowledge-base and who know how to get things done with the terminal. Some requirements: FreePBX/Asterisk, Java Script, pjSip, Bootstrap, CSS, HTML5, PHP, Web Dev., Stun / Turn Servers, Queues / Parking for calls, SSL, Ubun...

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    i have a java project for you? i have developed a project on blockchain based TLS notary service and want to test it for man-in-middle attack. And also test it to compare the blockchain technology i have used with other blockchain technology would you be able to help me out ? i have built the project using nebulas blockchain i have used nebulas blockchain and have to do the performance testing and...

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    I have mostly setup the Twilio account already but need here/there help to understand the thing fully and be able to continue developing. I would need you for 1-2h per day and probably need 5-10 days in total. I will do homework between days. We'll be using Chrome remote desktop, TeamViewer, [đăng nhập để xem URL] or whatever so you see my screen and can instruct me. First tasks Get Twilio w...

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    Test the project Đã kết thúc left

    Hi I have implemented a nebulas blockchain based TLS notary service to tackle the Man-in-the-Middle attack on local host but i am unable to think on how to evaluate it and its required urgently can you help me out with it ?

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    Hi Ghanshyam P., I noticed your profile and would like to offer you my project. I am looking for sip dialer customization. Let me know if you are interested.

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    Hi Sunil C., I noticed your profile and would like to offer you my project. I am looking for SIP Android and IoS app developer. Please let me know if you are interested.

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    Asterisk Speech To Text Đã kết thúc left

    necesitamos desarrollar una pieza de software que trasmita el Audio desde Asterisk via Streaming para integrarla con un servicio de Speech to text

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    Asterisk expert Đã kết thúc left

    Asterisk with WebRTC Specialist needed.! PLEASE READ THIS FIRST ! For Inbound call center with Asterisk and WebRTC for agent access. This is only for freelancers with a very broad knowledge-base. Some requirements: FreePBX/Asterisk, Java Script, pjSip, Bootstrap, CSS, HTML5, PHP, Web Dev., Stun / Turn Servers, Queues / Parking for calls, SSL, Ubuntu, etc. The physical Servers are installed and mos...

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    Hello, I am looking for a data scraper who can get me leads from America from Instagram for the personal trainer niche. I need 1,000 leads for now. If you do a good job, I will hire you for a lot more business. I need the following from each personal trainer / health coach’s Instagram: First name*, last name, business name, phone number* email address*, City*, State*. Anything in the asteri...

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    Asterisk Help Đã kết thúc left

    Asterisk with WebRTC Specialist needed.! PLEASE READ THIS FIRST ! For Inbound call center with Asterisk and WebRTC for agent access. This is only for freelancers with a very broad knowledge-base. Some requirements: FreePBX/Asterisk, Java Script, pjSip, Bootstrap, CSS, HTML5, PHP, Web Dev., Stun / Turn Servers, Queues / Parking for calls, SSL, Ubuntu, etc. The physical Servers are installed and mo...

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    15 lượt đặt giá

    An application is required where mainly SIP calls can be made. It also requires that the development platform be hybridized, specifically in Ionic.

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    Hi Mohammad Abu S., I contacted you on fivvr. I never knew you also existed on Freelancer. You follow the rules well. I wonder if you would be open to helping me fix my freepbx installation. The only trouble I have is: (1) Receiving calls in from twilio sip trunk. It connects but every now and then it doesn't. Looking into the logs, it seams to be because twilio sends from 4 different ...

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    Build React Native VoIP App/SDK Đã kết thúc left

    Build React Native VoIP App/SDK with SIP protocol which able to integrate with current React Native solution. Feature/Function that require as below: • Push Notifications • Receive VoIP calls on the lock screen, active VoIP calls are no longer disrupted by incoming cellular calls and VoIP calls are now stored in the native applications call log. • Call Transfer • Call Record...

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    I want simple dialpad with all dial functionality in web application (html as ui) and call flow should be initiated from asterisk / pbx

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    I need someone to set up an authenticated SIP trunk on an audio codes SBC that is then connected to a Lync server. Lync servers do not allow authenticated trunks and as such the SBC is required to do the authentication with the external SIP server then pass it on to the Lync server. I am looking for someone who has already done this before. Not someone that just has experience in these technologie...

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    SIP Packet Analysis Program Đã kết thúc left

    We require a program which will change and manipulate SIP parameters in realtime to enable us to test end to end equipment response.

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    General Asterisk Đã kết thúc left

    Fixing and writing new asterisk codes for our servers.

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