Viber sip asterisk trabalhos
I need a little customization on Issabel Call Center Community Edition. I just need to send the calls with some delay to the SIP Trunk, lets say, half second of delay on every call originated by the dialer. The goal its to reduce the CPS load on the trunk. I need someone with skill to get the job done! Its urgent!
Hi, I have a phone API to built. You must gather all the information that the API could offer, such has retreive client phone call. We need an angular pho...check our DB is the client exist, so we need a small square that will show us the client information on the call we receive and once we press the button we will be redirected to the client file. We have a CRM that we use, so you will have to create a module for that. So you need to have experience, your code will be double check for the quality of it. You need experience with CI, PHP, Angular, SIP protocol and you need to be able to understand the API has the company does not offer any supprt. I will provide informations, picture and link for you to complete the work. Come to chat I will anwser all your questions. Good l...
delay the buy signal on asterisk or opensips, when the call is hangup delay between 6 or 7 seconds to receive the bye
Hi, I need a windows pc softphone with following features. 1. SIP server information will be configurable through an API. 2. balance display through api. 3. Auto answer the incoming call 4. Play a sound file to other party after answering the call. 5. Option to add delete sound files. 6. Branding Thanks Vidhya Sagar Dixit
Hello, I am using Issabel 4 with asterisk 16. At the moment I can only upload audio file at 8000 hz . I would like to know if it's possible to use .wav files at an higher rates like 16000 hz or 48000 hz in order to increase audio quality. Audio files should be used for all the audio parts: IVR, Announcements etc...
Looking for someone experienced in mainly Mirta PBX or Asterisk by itself is less preferred but might be an option.. you would likely be a voice Engineer and will need to train people in use of and maintenance the PBX.
Hi, I need a windows pc softphone with following features. 1. SIP server information will be configurable through an API. 2. balance display through api. 3. Auto answer the incoming call 4. Play a sound file to other party after answering the call. 5. Branding Thanks Vidhya Sagar Dixit
The project involves creating an outbound dialer software with IVR functionality.: • Develop a system that is capable of making bulk calls to recipients. • The system must be able to be hosted online and be able to connect to operator E1/SIP Trunk. • The system must be able to schedule phone calls. The system must have a phone book, that includes name, number and sex (male or female) fields, the most important field is the number. The ability to create a group should also be available. • The system should be able to create different Campaigns (Campaign creation). • The system should be able to create IVR Menu’s for specific campaigns. Example Campaign: Training people about financial literacy in rural areas, the course has 6 lessons which will be de...
Dear all, I have CISCO router act as a FXO gateway for Asterisk server, it was working find upto the point where the incoming calls are not being ended when hangup , it keep active which make the line always busy,
I need to build an Android app that will receive a list of residents from and API. when pressed on one, a phone call using FreePBX will be initiated. The resident will be able to speak to the user. If the residents presses #0 on his phone, a string will be sent through socket communication to the door opener, and the door will open. Next phase will be an app that will allow Sip Video Calls.
We are looking for someone to run pen tests using sipvicous pentest tool running through python making use of the full tool kit to the best of their ability. The contract will be ongoing and paid at a healthy rate. If you feel you are qualified for this task please get in touch. Thankyou
Need to create a super scaleable conference bridge using an open source PBX. The user management and auth will be handled separately. Knowledge of Asterisk or FreeSwitch is relevant.
I have an ISO image (operating system for Boom Infotainment system for Harley Davidson). I'm interested in unpacking the ISO so that it has intelligible documents that I can edit graphically. Perhaps changing the logo is all I will do. I don't know if this is possible and if not please let me know. I don't know what language it's coded in. The IS...image (operating system for Boom Infotainment system for Harley Davidson). I'm interested in unpacking the ISO so that it has intelligible documents that I can edit graphically. Perhaps changing the logo is all I will do. I don't know if this is possible and if not please let me know. I don't know what language it's coded in. The ISO image is here:
I need you to develope multi tenant pbx, USA for me. must be experienced person
I need you to build Sip trunk pbx for multi tenants call in USA. Person must be experienced in management of trunk and server.
Hi Ameer Ullah A., I noticed your profile and would like to offer you my project. We can discuss any details over chat. I want you setup my sip trunk to fusion pbx and outbount and a inbound rule. I already have extensions
Hi, We have an application which is React JS front end and Django backend. We are looking to incorporate webrtc within this application: We would like 5 features if possible: 1. webrtc - audio to other users on the application 2. webrtc - video to other users on the application 3. webrtc - chat to other users on the application 4. webrtc - softphone that connects to an Asterisk PBX - to click to dial from data on the user panel - basic softphone functionality 5. An autodialer - basically - it loops through the users contacts - dials the number, if it answers, it opens up the screen of the client. The two repos are on github, we have a development server also. Certain user information is on the django backend which the reactJS front end will need up access/update etc.
SIP mobile dialer With LOGIN and history and information
...request. Multitenant. User login with e-mailaddress Admin login, reseller login, tenant admin login, user login. BLF and speeddial buttons, button templates, server templates and tenant_policy Call history Orders and licenses to tenants. Based on MySQL database. To be more clear: we don't use a third party API like Twilio, but the websocket connection is between the clients browser and i.e. an Asterisk server. We have both a prototype of the management interface and the Web dialer interface. Please ask....
Hi guy's, I have a sip provider, but I don't want to install a softphone on my computer, instead I want to use a HTML SIP / JS so I can make a module to use with my CRM. I have find those two references page and it seem it can be acheive. and For this projet, I want to be able to connect with my phone provider and receive a call I can anwser or make a call. Basic HTML5 will do it. Ping me if you have question, we talk about price in the chat.
...and everything else related to keeping the business running. A few months ago I purchased a GoIP4 SIP GSM Gateway, but been having a hard time configuring it, so it hasn't worked as expected up to now. I've got 4 local sim cards in the machine already but I never got them to work properly. I directed the calls to an SIP address on one of my virtual PBX, I want those sim cards to receive and make outgoing calls and I want all 4 people to be able to do so right from their stations at home even though we are not in the same network. At this point, I can receive calls but calls always get disconnected only after 10 to 15 seconds for unknown reasons. When I make calls, the calls are charged to my SIP provider instead of being charged to the SIM card, so I�...
when incoming and outgoing call going in asterisk some time audio not coming
GOIP brand 8 port gsm gateway. Need to configure the equipment to achieve the following: using an app of SIP soft phone, to be able to receive calls from the sim cards in the GOIP GW physically present. One app that can have all the connection for all the phone numbers of the sims in GOIP. Able to make outgoing calls as well, selecting which line/sim for outgoing call. Receive and send SMS texts for each sim as well. Essentially using the GOIP GW, but have access to all sim cards in an app on one mobile device.
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...need to build a proof of concept python multithreaded application that will connect to one or multiple Asterisk servers using AMI protocol (preferably using a library) and trigger a callback function when receiving events. At the same time, one or multiple clients can connect to this python application using a predefined port (like 5039) and using the AMI protocol receive the events from the asterisk servers and send commands to one or multiple clients. When an event is received from the asterisk server, the application will send the same event to all or some of the clients connected. When a command is received from any of the clients connected, the same command will be sent to the asterisk server. In other words, I need a "wrapper". This is just a ...
...JavaScript 4) Kotlin 5) Spring boot 6) Docker Внердрение эквайринга Тинькофф: 1) Kotlin. 2) Springboot 3) Javascript. 4) Flutter Для андроид и эппл приложений навыки важны для специалиста: #Flutter Бекенд и фронтенд: Бэк: Kotlin, Springboot Фронт crm: Flutterweb, JavaScript Фронт открытого сайта:. Nuxtjs База данных MySql. Впрыгивай в новаторский проект. Ехуу Пиши сразу сюда свой whatsap/viber. Предварительно опыт, созвон по скайпу. Прикрепи кейсы, если уверен пиши так. Обязательные требования Наш технологический стек – Python (Django) на бэкенде, REST API, Vue.js на фронтенде. Реализована интеграция с движком платформы Stepik для хранения и проверки задач. Опыт frontend разработки от 2 лет. Опыт работы c JavaScript (ES6). Опыт работы со SPA фреймворками (например: V...
I sell and install Mypbx S-series phone system (asterisk based) and i would like to integration Cliniko CRM into the Phone system. Features - when there is an inbound call on the PBX the caller id will be sent to cliniko and Cliniko will display the customers details automatically in the cliniko software. If the caller id is not in the cliniko software as a known caller id display a customer card with the caller id ready to add new customer. (PBX system) (softphone) They used our open API which is available here: Telecommunications company that has already enabled this
I'd like to hire a professional with some experience in Asterisk. I would like to create 60 multiple choice questions for Asterisk based in my book Complete Asterisk Training. I will hire this project at least three times from different professionals. Your are going to receive a copy from the Complete Asterisk Training book to extract questions from there. The questions in the end of the chapter will not count. Questions may be: Multiple Choice (Radiobutton) Fill in the Blank True/False Multiple Choice (Radiobutton) Multiple Choice (Dropdown) Multiple Correct Essay (Evaluated by Admin) DragandMatch Matching DragandMatch A maximum of 10% of true/false questions will be accepted. Multiple choice questions should have at least 4 options (None of the prev...
I have a 3CX system running locally and I have a CM sip trunk. I also have a Yealink W60B basestation with 1 phone. I can't get it working, no basic receiving or calling. This project is about helping me fix it.
My Requirement is like this: We have rooted android device (TV Box), installed open source Linphone Voip Sip client on it. We required: 1) access & modify the settings of the linphone app through web browser within LAN using device IP address. 2) Android app to Discovery of all these Linphone devices in a single page & change there settings from there only.
At the top of the menu center - Sip Boba Teas should be the title. Create a Boba menu 38.75" L x 39"W. I will supply 2 licensed photos from Shutter Stock. The menu should be on a white background. I have attached a spread sheet with the pricing, the names of drinks and Add Ons. Boba Tea is broken into two groups: Milk Teas and Fresh Teas. I would like this highlighted on the menu. One Photo shows Milk Teas and the other photo shows Fresh Teas. I would like top of the menu to show Fresh Teas with the photo and the bottom of the menu to show Milk Teas. A would like a color graduated bar to go across the bottom to labeling the zero sweetness, 50% sweetness and 100% sweetness. The bar does not need to be more than 8" inches. Email questions.
...(from) clients using existing Asterisk PBX, actually the client had Salesforce Enterprise Lightning (consider for development related Salesforce API calls enabled). Scenario: The calls need to be run using integrated softphone on Salesforce UI (not an external softphone like X-Lite or Zoiper using callbacks methods), that should be possible using Open CTI API integrated with Asterisk PBX, each call need to be logging to Salesforce activities with all call metadata (source, destination, total time, timestamp started, timestamp ended, user who made or get the call and the others fields given by Asterisk PBX) furthermore, it´s needed to be stored a filed with record link to Asterisk PBX record store (for this point it´s necessary some kind of web se...
Доброго времени суток. Пару лет назад вы откликались вот на этот заказ: https://www.freelancer.com.ru/projects/mobile-phone/whatsapp-sip-gateway/?ngsw-bypass=&w=f Такая же задача сейчас для меня актуальна. Вы его реализовывали? Можете такую задачу сделать?
PHP VOIP inbound and outbound call web app integration with Asterisk. This project require a website interface not only integration with asterisk. Need and Admin, Login, Authentication, Client Management and Call Tracking capabilities.
Customize Vicidial: 1. Automate Regular Task so even a person with Zero Knowledge of Asterisk, Vicidial can do things like Make new campaigns, increase call flow, pause, load leads, add lists/ ad users etc. 2. Create customize Reports - generate reports with single click or button. 3. Change Login web console... make it more secure and reduce agents logging into somebody else;s account.
I'd like to hire a professional with some experience in Asterisk. I would like to create 60 multiple choice questions for Asterisk based in my book Complete Asterisk Training. I will hire this project at least three times from different professionals. Your are going to receive a copy from the Complete Asterisk Training book to extract questions from there. The questions in the end of the chapter will not count. Questions may be: Multiple Choice (Radiobutton) Fill in the Blank True/False Multiple Choice (Radiobutton) Multiple Choice (Dropdown) Multiple Correct Essay (Evaluated by Admin) DragandMatch Matching DragandMatch A maximum of 10% of true/false questions will be accepted. Multiple choice questions should have at least 4 options (None of the prev...
Recently converted CUCM from T1 to SIP. Currently the Sonicwall is the border element and is having consistency issues. Cisco router is licensed as a CUBE and has enough processing power to handle the call load, so I want to set it up as the CUBE for CUCM using then validate the removal of the original T1 configuration from the router. Will need to be familiar with Sonicwall.
I want an calling application people can call via internet to mobile(sim) with this method :- > METHOD + VOIP Server like Asterisk / FreePBX + Protocol like SIP/PJSIP/WebRTC + Library like liblinphone for Android or iOS. + Making outgoing call using this above library functions like , () + A little programming to change number every time. ONLY BID IF YOU KNOW THIS METHOD AND TYPE "READ" IN YOUR PROPOSAL IN TOP. OTHERWISE I WILL IGNORE YOU
I want an calling application people can call via internet to mobile(sim) with this method :- > METHOD + VOIP Server like Asterisk / FreePBX + Protocol like SIP/PJSIP/WebRTC + Library like liblinphone for Android or iOS. + Making outgoing call using this above library functions like , () + A little programming to change number every time. ONLY BID IF YOU KNOW THIS METHOD AND TYPE "READ" IN YOUR PROPOSAL IN TOP. OTHERWISE I WILL IGNORE YOU
I have an occasionally issue with audio between Freeswitch and our PBX with PBX (Sip Wise) . CAll enter Freeswitch and then continue to PBX. This issue is that we cannot hear a greetings from our PBX and it looks more random since in most of the cases it is working fine. We are looking for expert who can analyze and fix the issue within the Freeswitch.
I want an calling application people can call via internet to mobile(sim) with this method :- > METHOD + VOIP Server like Asterisk / FreePBX + Protocol like SIP/PJSIP/WebRTC + Library like liblinphone for Android or iOS. + Making outgoing call using this above library functions like , () + A little programming to change number every time. ONLY BID IF YOU KNOW THIS METHOD AND TYPE "READ" IN YOUR PROPOSAL IN TOP. OTHERWISE I WILL IGNORE YOU
Freelance opportunity for an Avaya Certified Implementation Specialist (ACIS) or Avaya Certified Support Specialist (ACSS) to configure & upgrade an Avaya Session Border Controller Enterprise (ASBCE) for Avaya Equinox OTT Service Provider solution 9.1.11. Inc. SSL certificate installation, reverse proxy configuration and call flow rules for WebRTC client, SIP audio trunks and SIP Video endpoints. The ASBCE is deployed on version 8.1.2 and will need to updates applied. Remote access is available via VPN.
Looking for someone with experience in AI Voice Bots and FreeSwitch or Asterisk. Must have experience with STT or Transcribe.
Job Description Should have worked on: 1. IP based Call Center. 2. Dial Plan in Asterisk. 3. Good understanding of VXML and IVR development. 4. Good understanding of VoIP, SIP, SS7. 5. Prior working on voice cards, media gateway, WebRTC etc. 6. Expert in Freepbx 7. Understanding of Telephony, CCXML,VXML, json - Scripting skills (Perl/Shell etc.) - Understanding of Linux/Unix commands/distributions (RHEL,CentOS,Fedora) 8. Expert in Cisco voip along will get priority
...bills so we can provide them with a comparison quote. Product: We can typically save our prospective customers between 35-60% on their current call spend. We also offer the following features as part of our package: Hosted PBX Call Center Contact Center Call Recording Call Monitoring Real Time reporting (call costs, current spend, agent dials etc...) Voice-mail to email SIP Trunk Support (Remote and customer service) Number porting. Skills: Leads, Marketing, Research, Sales, Telemarketing See more: VOIP lead generation, we need sales agent, we are looking for sales agent in usa, voip phone for business, voip for business, voip business phone system, voice mail recording service, time in miami, telesales only, telesales call, telephone mar...
Need a simple solution to import +3.000 contacts into Skype so I can call them using Skype OUT. Therefore, the solution cannot be import of CSV file since import of CSV contacts does not app...using Skype OUT. Therefore, the solution cannot be import of CSV file since import of CSV contacts does not appear in Skype Out list. Ideally, the solution should be made in .Net. for Windows 8.2 or later (If interested, other related projects/upgrades will be likely be possible if interested). For interested bidders, I will provide a dummy file for you to test with. I am available on Skype, Viber, Whatsappl, Line for quick communication after bidder has been chosen. IMPORTANT Please write why you are the right programmer for this project and what you have done previously that is relate...
Having a paint party for about 20 people in my backyard. Would like someone to decorate the yard (rent tables, chairs) and provide decorations for an intimate private venue. Also need help with finding a caterer for the event July 31st. Would like a caribbean food served or soul food.
Дорогие друзья, мы ищем в нашу команду или для работы удаленно Талантливого копирайтера (не того, кто хочет сменить сферу деятельности) с минимальным опытом-год. Функциональные обязанности: 1. Написание текстов в области SEO и диджитал - маркетинга 2. Написание и редактирование текстов сайта, email рассылкам, sms – рассылки, viber - рассылки 3. Написание текстов для новостей и пресс-релизов 4. Написание статей для блога 5. Написание текстов для промо, акций 6. Поиск интересных фишек в копирайтинге 7. Изучение работ конкурентов Работа на румынском языке. Будет большим плюсом, если будет знание дополнительного языка: русского или английского.
