Freeswitch voicexml asteriskcông việc

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    13,965 freeswitch voicexml asterisk 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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    Project for hatvott Đã kết thúc left

    Chào bạn, Tôi tên là Tiến, ở tại Sài Gòn; cần tích hợp OpenSIPs và FusionPBX (FreeSWITCH). Regards,

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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

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    Project for Bao N. Đã kết thúc left

    Hi Bảo, mình đang làm project về fusionpbx core freeswitch Đang vướng phần add push lên ios và android để wakeup call, Bảo có thể support mình không

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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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    Setup asterisk server with VOXBEAM -- 2 5 ngày left
    ĐÃ XÁC THỰC

    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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    Class5 Soft switch based on FreeSwitch -- 2 4 ngày left
    ĐÃ XÁC THỰC

    Looking for a Class5 Soft Switch based on FreeSwitch with Android and IOS APP. *Account Management *Calling Cards *Rate Groups / Tariffs *Call Routing Strategies *Call Rates *DID Management *Product Management *Invoicing and Billing *Reports and Alerts *System Settings *Payment Gateways

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    FreePBX Firewall Configuration 3 ngày left
    ĐÃ XÁC THỰC

    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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    VOIP project 5 ngày left

    Hi Eremin P., I noticed your profile and would like to offer you my VOIP project. Cloud PBX based on Opensips and FreeSWITCH. This would be a long term project. We can discuss any details over chat.

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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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    Our current situation We are a call centre and have been on the market for many years. We operate a classic telephone service / telephone secretariat. We are currently work...whole thing. The briefing also includes ideas on how to implement the whole thing or which existing functions could possibly be used. The operator panel should be available as a stand-alone web application and work together with the telephone system / FusionPBX / Freeswitch. The WebApp will then certainly be able to be connected to the telephone system somehow in a config file. That would at least be good. The script language on which the WebApp / operator panel is based or developed is up to you. We can install Freeswitch and FusionPBX on a Server and set up a SIP Trunk. We will give you ...

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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 1 ngày left
    ĐÃ XÁC THỰC

    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 have recently taken this job off a developer that has been slow in completing the development. Theref...someone to make changes on the look and feel of the website, add DIDWW and DIDX APIs, Mobile Top UP API like TransferTo and Prepay Nation so that clients can add their user names and logins when integrated well, implement report to the existing SMS system as there is no report there. Test the system to make sure all components work well before we can release it. You must be knowledgeable in Freeswitch as the system is a combination of two popular programs merged together. We only want developers who are knowledgeable in FusionPBX and ASTBB and have worked on these systems before with results. We will use these systems as a yardstick to measuring how good the best candidat...

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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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    Looking for a Class5 Soft Switch based on FreeSwitch with Android and IOS APP.

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    Nổi bật
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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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    Analisar pacotes RTP/SIP do freeswitch, usando script Dejavu Python

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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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    Hi, We are a startup and need to hire a FreeSWITCH / OpenSIPs telecom engineer to help us with tasks from time to time. We would like to work long term with only 1 developer / engineer, you must also know how to install and setup FreeSWITCH/OpenSIPs on AWS. We will pay by the hour, please send your resume or experience and price per hour you charge. thank you.

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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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    Hi, We are looking for a developer with FreeSWITCH experience on AWS . I need several things done. 1 - The last developer setup FreeSWITCH where the call recordings save in BOTH the EC2 & S3 Bucket, we just want to save them in the S3 so they need to stop being saved in the EC2 instance. 2 - I need some sort of monitor so when a registration fails or calls fail it will send us a notification, I see there are some FreeSWITCH modules for this that. 3 - A FreeSWITCH dashboard to show real time the current state and calls coming in/out, I also see some open-source ones online but am open to recommendations as long as there is no recurring fee's for them.

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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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    Windows FreeSwitch Đã kết thúc left

    1. FreeSwitch in Windows platform/OS 2. FOIP in Windows OS FreeSwitch 3. SMS module 4. Conference module 5. Speech-to-Text module

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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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    Project for NetworkLab Đã kết thúc left

    Hi NetworkLab, you had done some work for me in the past, it was very high quality and good work :) Can you please send me a price quote in install one of the FreeSWITCH monitors so if a registration or phone number fails it will notify us, also, do you guys like any FreeSWITCH dashboards to monitor activity? If so which one and how much to also install that, thank you!

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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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    freeswitch dial plan Đã kết thúc left

    Hi I need a dial plan for an extension in fusion pbx where I will have an extension as follows. I call the did number that points to this extension - the system tells me to dial a phone number for identification (that is, what will be the callerid for the call - what identification will appear at the destination) - the system repeats the number I keyed and gives these options to confirm press 1 to press again press 2 - If you press 1 - please enter the phone number you want to call (destination). The system repeats the keystrokes and so on - if you press 1, the system dials using a trunk that you will have to set up (of my provider) - the call is automatically recorded. And I can receive the recording by email/ see and hear it in Fusion's interface. The budget for this is $80

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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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