Asterisk voicexmlcông việc

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    12,767 asterisk voicexml công việc được tìm thấy, giá theo USD

    Tôi cần phat triển phần quản trị cho giai pháp ai smart dialer va cần nhân sự biet ve asterisk python va php mysql

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    Customize Asterisk Đã kết thúc left

    Customize IVR for Asterisk: 1; Play announcements 2; Record 3; Transfer

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    EducaCallcenter Đã kết thúc left

    Tích hợp plivo với freebpx/asterisk để xây dựng tổng đài support simbox và hotline

    $500 (Avg Bid)
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    Tôi đã có sẵn tổng đài Asterisk, đã có sẵn SuiteCRM. Cần lấy API từ tổng đài Asterisk để kết nối với Suitecrm.

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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 ( @). 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 làm một chức năng theo dõi cuộc gọi video của người dùng. Anh có làm được không ạ? Nếu được anh liên hệ trực tiếp với em qua email @ để trao đổi chi tiết hơn ạ.

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    i need a php page with a button that make a call trough an asterisk pbx in the same local network I'd like to have a click2call function embedded in our web pages when you click on the button you'll make a call trough our PBX

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    Hi, need to develop an application that analyzes presence of tone in early media and makes the decision to drop the channel. It must support multi channel dialing (eg: when dial application is called and it has multiple dialing destinations)

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    Need Expert person for asterisk , VOIP 4 ngày left
    ĐÃ XÁC THỰC

    Hi i have installed asterisk on server , want to set sip dialer (soft phone for voice calling) my admin portal is on laravel platform, i want to set phone on that , which will be connected to asterisk on backend, right now m using zoiper , but i want to have web app. SIP VOIP Asterisk VPN

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    Hello We need to build auto dialer + IVR Survey call on asterisk After this we need to integrate this PBX with ERPNext CRM Module

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    Setup asterisk server with VOXBEAM

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    Hello Guys, We are looking for someone who can help us to setup a multi tenant freePBX to start a basic Toll Free Number providing services to local businesses.

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    Hello, We're currently seeking a service provider with previous experience in Flutter or React Native to build an application for multiple platforms like IOS & Android. The candidate should have previous experience in building communication applications and also have a strong understanding of how VoIP, PBX, Asterisk work & interact behind the scenes. The app will be integrated with the open-source system, Asterix. The goal of the project is to replace the existing 3rd party solution we're currently using called Zoiper. If you have experienced in building communication apps and understand Asterix, VoIP, FreePBX & Flutter, please reach out for more information. We have AdobeXD designs & technical specifications document with use cases which I c...

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    FreePBX Firewall Configuration Đã kết thúc left

    My FreePBX Server is having issues with call deployment due to the firewall and fail2ban. I need someone who has hands on knowledge of configuring it all way to make it work. I am myself a Full Stack Developer and have good knowledge, so people who really knows what Asterisk SIP Settings and FreePBX Firewall is actually apply.

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    ...for development of a Zimbra Collaboration Suite add-in (called a "zimlet" (java)) which can talk to Asterisk (soft-PBX) using Asterisk AMI to allow calls to be initiated (the call is started and then your internal deskphone starts ringing and is connected to the outgoing call. Both Zimbra and Asterisk are open source Linux based products. There have been open source projects to do this but based on older versions of ZCS and none support ZCS 8/9 and also Synacor even include click-to-dial for specific phone systems but not Asterisk. Here is a youtube video that explains how to create zimlets for ZCS 9 Here is a youtube video that explains how the Asterisk AMI can be used: Here is more documentation about zimlets

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    Need to improve certain area of the server as following. Server details: Current VoIP system: FreePBX (Asterisk 16.22.0) OS: Ubuntu 18.04 Database: MariaDB 10.4.22 Tasks: - Enable and configure Ubuntu firewall UFW - Install and configure Fail2Ban. When BruteForce comes on SIP registration try, register bad IP addresses in the list for 3 days to block any access. - Enable MariaDB to listen on TCP 3306. Currently working via ODBC with Asterisk - Install and configure NFS server - Setup MariaDB “Master-Slave” replication - Monitor Asterisk service running. If stops, send email alert and try to restart asterisk once. If you read the requirement, please include word "peace" in your proposal. Thank you.

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    suiteCRM and Freepbx Đã kết thúc left

    We would like to setup a Asterisk callcentre with suitecrm. . The asterisk needs to be integrated with suitecrm with callpopup , call logs and other standard features. For the CRM it needs to work with messenger ,email , whatsapp We won't like to use any 3rd party addons that we need to pay yearly or monthly. Only offer if you can build the connections yourself.

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    Hello We have the latest FreePBX server set with asterisk in it and Node Red to control the asterisk dial plan. Requirement is when we get an incoming call through a channel then Node Red identifies and takes over the call control. While Call forward and pre-recorded messages works fine however we are not able to get TTS based MP3 audio playback done via node red in asterisk. As per asterisk MPG123 will work however as using by nodered thus the file as to be injected via a function instead of node. Looking forward for solution to make following happen 1. Play mp3 file in asterisk(freePBX) via node red over an extension in incoming call. In a way node red will become dynamic IVR for bridge connections 2. Based on caller ID Node Red will HTTP request our...

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    Hey i am trying to build an app in asterisk. I am facing some problems with configuring file. Need guidance in configuring that file and need some answers of asterisks questions.

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    I am looking for someone who understands asterisk systems or VoIP systems. I need someone who can help support current clients.

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    Create an api in asterisks softswitch to handle calling on android and IOS linphone apps The call will work via Local number that is instead of the app calling via internet VoIP. , it will initiate the call via the app and the app fetches the Local number from the astericks softswitch. So when caller hits the call button the api fetches the DID local number from asteriks softswitch then places the call to the destination this works silently and there will be no tap tap sound . And if there is no internet in the phone it will send the call via thesame Local DID number in the astercks softswitch like this Country code,, destination number,,# that is in a pinless calling manner. But this way will make a tap tap sound . There are many apps that do calling using this type of l...

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    Setup asterisk server with VOXBEAM

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    I need a Asterisk PBX developer Đã kết thúc left

    I need to develop a cloud call center by using Asterisk in backend and need to create extensions , soft phones and configure from browser, if any one have an idea about this ping me ASAP

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    Asterisk and freepbx Đã kết thúc left

    must be set on a raspberry pi 4 asterisk and freepbx with 4 dongle channels. To create dialplan to send automatic calls from cli

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    ...management/monitoring tools, operating systems services (DHCP, DNS, NTP, SNMP, SMTP, BGP, OSPF, VPN , etc ...) - Cloud technologies knowledge(Amazon AWS, SoftLayer anIBM Company, Oracle Cloud, OVH, SoYouStart, DigitalOcean, etc ...); - web applications technologies (Tomcat, Apache, NGINX, Redis, PHP - fpm, Memcache, etc...); - basic telecommunications concepts and protocols knowledge(SIP/RTP) ( Asterisk ) - version control systems (GIT, SVN, CVS) - database engines administration and optimization ( MySQL, MongoDB, PostGreSql, etc ...) - database administration application / installation / troubleshooting / support (Oracle 9i - 12g) - good knowledge of Windows Server (Hyper-V, Active Directory, Group Policy DNS, DHCP, File Server) - experience providing support in medium to large ...

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    -SIP-based call to Softphone (web app) - Video & Audio SIP Integration We want users to be able to join active conference by dialing a specific URI and be able to actively participate using audio and videos and be able to share screen with the sip phone or terminal emulator respectively - Dial Out We want our users to be able to call a user ...meeting using our application (internal extentation or any mobile number) - User to User Call We want our users to be able to make user-user call on our existing mobile app using internet which will support video and audio with clear voice and lower bandwidth. - User to Phone Call We want our users to be users which are not on our APP Features : Recording, Call Forwarding *Have experience in Asterisk Issabel json API If have any solution...

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    Soporte de Asterisk Đã kết thúc left

    Necesito realizar el siguiente trabajo 1.- Configurar una ruta entrante desde asterisk (puro) para que pueda ser enrrutado a una cola y contexto 2.- El agente (a discresión) cuando entre (login) le caigan las llamadas de esa cola o contexto

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    Need a inbound call centre. You need to set up asterisk server for below features, but then another coder I work with will create a custom panel, and need to integrate both together. So it is a joint project. You will need to create API to integrate with custom panel. Ability to transfer calls to a closer Ability to create agents, 2 types of accounts, closer/opener Ability to add scripts, script will show according to Campaign/Agent type. Example, Car Campaign, show Opener script to Opener agent for Car campaign. Ability to have custom hold music, customise it per campaign or have a default one for all. Agent Alert sound for inbound calls. Ability to have a queue, many people call, all agents busy, play hold music and custom speech, saying 'You are now number 4 in the queue&#...

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    Linux System Administrator Đã kết thúc left

    ...management/monitoring tools, operating systems services (DHCP, DNS, NTP, SNMP, SMTP, BGP, OSPF, VPN , etc ...) - Cloud technologies knowledge(Amazon AWS, SoftLayer anIBM Company, Oracle Cloud, OVH, SoYouStart, DigitalOcean, etc ...); - web applications technologies (Tomcat, Apache, NGINX, Redis, PHP - fpm, Memcache, etc...); - basic telecommunications concepts and protocols knowledge(SIP/RTP) ( Asterisk ) - version control systems (GIT, SVN, CVS) - database engines administration and optimization ( MySQL, MongoDB, PostGreSql, etc ...) - database administration application / installation / troubleshooting / support (Oracle 9i - 12g) - good knowledge of Windows Server (Hyper-V, Active Directory, Group Policy DNS, DHCP, File Server) - experience providing support in medium to large ...

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    hi i want someone to set sip dialer(client - web phone) in front end made in laravel for making audio calls from asterisk server there are so many projects on github for web sip dialer u can set this with laravel. so on clicking on dialer button in front end dialer will open get registered on asterisk server and we can call. he has do set up on server also if required regarding socket for web phone settings are authenticated in database, everything is dynamic voip

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    Each files duration is no longer than 1 minute. The assignment requires: 1. Indicate the gender of all speakers on the recording; 2. Indicate all the languages ​​represented on the record (for Chinese, you must also indicate the dialect)...following rules: a) if the language is not familiar, then indicate “unknown. language.”; b) if the language is only Chinese, then it is necessary to indicate “Chinese (dialect name)”; c) if in addition to Tajik there is some other language (1-2 words are not a full phrase), then you must specify the name of the second language with an asterisk. Example: “Chinese (dialect), japanese*”; d) if there are full-fledged phrases in another language, then an asterisk next to the second language does not need to be...

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    Each files duration is no longer than 1 minute. The assignment requires: 1. Indicate the gender of all speakers on the recording; 2. Indicate all the languages ​​represented on the record (for Tajik, you must also indicate the dialect) a...to the following rules: a) if the language is not familiar, then indicate “unknown. language.”; b) if the language is only Tajik, then it is necessary to indicate “Tajik (dialect name)”; c) if in addition to Tajik there is some other language (1-2 words are not a full phrase), then you must specify the name of the second language with an asterisk. Example: “Tajik (dialect), turkish*”; d) if there are full-fledged phrases in another language, then an asterisk next to the second language does not need to be...

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    Each files duration is no longer than 1 minute. The assignment requires: 1. Indicate the gender of all speakers on the recording; 2. Indicate all the languages ​​represented on the record (for Azerbaijani, you must also indicate t...rules: a) if the language is not familiar, then indicate “unknown. language.”; b) if the language is only Azerbaijani, then it is necessary to indicate “Azerbaijani (dialect name)”; c) if in addition to Azerbaijani there is some other language (1-2 words are not a full phrase), then you must specify the name of the second language with an asterisk. Example: “Azerbaijani (dialect), turkish*”; d) if there are full-fledged phrases in another language, then an asterisk next to the second language does not need to be...

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    configuration of kamailio Đã kết thúc left

    we have installed Kamailio with some asterisk server behind and we need some help to configure kamailio to be a loadbalancer and acting as SIP proxy to just pass all sip messages including register to asterisk servers behind we need this to be done asap

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    We have Cisco phone CP-8861 and we want to connect it with Asterisk

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    We need vTiger Asterisk connector expert to connect asterisk and vTiger CRM. Asterisk and FreePBX are on AWS cloud. vTiger is on another server and the edition of vTiger is opensource You must have experience in connecting asterisk and vTiger , please give us your previous work demo link . We have 25 -75$ budget. Keep this in your mind and bid please accordingly Thank you

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    asterisk webrtc configuration Đã kết thúc left

    i need to configure asterisk with webrtc

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    ...using VoIP. You're a full stack developer. Requirements: - Backend: FastAPI / Python + Asterisk - Frontend: Svelte / Bootstrap 5 / HTML5 / CSS3 You will: 1. Develop the front-end by taking the mockups into account. 2. Develop the backend & the API routes & all mechanisms to make the front-end dynamic (except the VoIP related features). 3. Install & configure Asterisk with a Telnyx account to be able to do spoofing and 100 simultaneous calls. To test the numbers, automatic calls will have to be done. You must answer the some questions. 1. What is your expertise level in VoIP web apps ? An example ? 2. How will you interface the front-end receiving & sending the audio to the python backend and Asterisk ? 3. Could you provide an example of a Fa...

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    Asterisk Docker image Đã kết thúc left

    Need someone to build Asterisk Docker image

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    we are required a Asterisk developer to customized the asterisk software and develop the Asterisk based pbx

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    Hi I need help with various settings on the subject of voip

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    Hi I need help with various settings on the subject of voip - required skills - diligence, flexibility, understanding, desire to progress and learn new things, consistency, availability

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    Develop a high-quality backend voip call mobile application on Dart. The frontend is written in flutter. Sip server asterisk install on Linux servers in AWS. We need to write the backend logic: users buy a monthly unlimited conversational tariff. Payment via api acquiring. Develop SMS to confirm mobile number. The backend code is written in Dart, read this information.

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    I want to integrate FreePBX and in the future maybe other PBXs like a clean asterisk to a Billing system so my clients will be able to get their recordings, balance and be able to pay through the same platform. At the moment I have an MagnusBilling system that is not integrated yet, Maybe we need to create a new custom system or use the MagnusBilling system. I'm open to suggestions. One thing which is a must- English conversational level. I've had a bad experience because of that. At the end of this project, I want my clients to be able to gather all the information I've mentioned above by going to the link of the billing system which might or might not be the connection of the extensions. The budget of the project will be discussed as we discuss further details a...

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    Asterisk SIp server Đã kết thúc left

    Hello, Came across your profile during the search and I am looing for a professional who can create a sip server with APIs which will be use to make incoming and outgoing calls. Please let me know if you have time the we can discuss in details. Best Regards, Rohit

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    i need someone to install for me the best for 500-1000 cps , i need to put a blacklist of 1 millio number i tried with mysql it goes down any suggestion

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