Webphonecông việc

I need an expert who can create a link for our desktop app via Mobile App (android and ios) Users will download our 1 page app and activate it by email address only. This email address will be saved for that app ID so that no one else can use the same email for other app account. The App page will have our name/logo and a Button "Call Now". when clicked on that button, it will open our desktop UI to make the calls and for other transactions. We want a simple one page App for now but hopefully we will make it a full Mobile app later on. We can discuss the language/platform to be used. Remember: Its a low budget simple connection App.

Fusion PBX installation integrated with PSTN SIP trunk, Webphone(SIPML5 or any other for Audio, Video and conference), Webchat and Screensharing configured with Webrtc enabled. on GCP ubuntu22VM Recording Enabled

I am looking for a solution for our Voip webphone telephony. We have our own Asterisk server and receive calls via our CRM and can also make calls over it. Now we want to add more features: - It shows history, who called and when. How long the phone call went on - Click to call functions. In our CRM we want to click on the number and it should connect directly to it - We would also like to see when someone calls, if it is displayed in our database, i.e. CRM Ideally, you may have call center functionality to build in. We serve 3 numbers at the moment. For CRM we use a bootstrapframework php 7.4 Can you make our offer for it.

Hi I'm not a Cloudflare expert, I have an account with spectrum enterprise enabled (trial) and an Asterisk pbx server with a websocket webphone and stun. I need help to configure Cloudflare in order to proxy the voip calls. Max 100 euros. Please bid only if you have experience with Cloudflare and custom applications (spectrum).

Hi, i am looking for someone that can assist me with some small setups on vicidial. I have already setup a vicibox solution on the server and have a sip trunk. We did use fusionpbx for a small call center with only outbound calls but looking to move to vicidial. I am struggling to setup the...struggling to setup the carrier and have issues with the time sync on the vicidial agent side. Our time is out with 10min for some odd reason. I have already checked the files in apache and cli which is correct. the settings in vicidial is also set to +2 for time zone. I am looking for someone that can assist just to setup the sip trunk and also make sure that the agents can dial out with the webphone that is built into vicidial for now. Will you be able to assist today and what will the cost ...

i have one did setup and working with my webphone on vicidial outbound dialer - i will also use twilio i need someone who can set up multiple DIDs for my campaign along with their routing (maybe about 5 more DIDs) - as well as this same person creating me a video on how they did the DID setup and routing for me must know how to work vicidial and twilio this is for OUTBOUND dialing use the phrase "setup master" so i know you read and understand what i need please give ma an accurate price and time frame on this project add the price and timeframe to you bid request that assures me you read and understand and can do the job lets not waste each others time if you have no idea how to work either do not bid i have different groups in my dialer so i will need numbers attached ...

On your ec2 instance I will have asterisk, coturn and the sipml webphone all working

Buenas tardes, en estos momentos tenemos un Webphone desarrollo con la tecnologia SIPML5. El webphone funciona de manera correcta y esta desarrollado en javascript con .net. El inconveniente se da que a veces se suspende la sesion del explorador y necesitamos evitar eso. Entiendo que necesitamos un experto en javascript que nos ayude a ver este punto y que haya desarrollado un webphone antes.

install webphone on vicidial (vicibox10) - SSL installed i need only installation & configuration for install webphone on vicidial (other than viciphone)

I need a Sip Client Webphone so I can integrate this into my CRM as click to call button for my clients so basically my agents will see the clients details and when they click on the call button a call will initiate to the client number via Sip Client Webphone and WebRTC and right in the CRM agent talks with the client. I don't want normal integration to other platforms and softphones like zoiper, 3cx or others I want something like: but for some reason these are not working for my sip they asking AOR or WSS which my PBX/VoIP doesn't provide so it should connect to sip only using username, password, and domain of PBX like zoiper connect.

...I would prefer in an IPSEC tunnel and with username and password... but we need to take in consideration that not all clients are able to make IPSEC or have an IT guy or what ever. So should be also possible to have it only by user name and password or what ever, here I need a good advice to have it simple and secure. In 10 percent of the cases we will connect the SIP Trunk (connection) to his webphone (zoiper) or IP desk phone. IP deskphones which we are using are Motorola, FANVIL, Yealink, Grandstream for the moment. Please contact me for more details. The clients should have access to there services, so they can administrate the sip trunk and conversations etc......

Projeto similar ao mizu voip para fazer chamadas através de um navegador

Webphone que funcione através de navegadores, ex Chrome, Firefox e etc... Usando qualquer conta voip Similar ao mizu-voip

Need webphone developer for web based calling

Hi I have two servers and an iframe which should load a resource (a webphone on the Asterisk engine) from one server and web pages from the other server. If I put the resources (the webphone and the web pages) on the same server, everything works fine but if I try to split them, the webphone is not working and I'm not getting any error. I guess it's some problem with CORS or whatever and I don't know what to do since i'm not getting any error in the browser, i don't know how to proceed from here. Server is debian 10 with apache and the web pages are in php.

Hello, We are developping a softphone/webphone like zoiper. We developed user interfaces and now we are looking for someone for developping functionnalities like : SIP account registration Receiving call Calling out Transferring call On hold call Recording call ... Best regards,

Hi I have a web application with a frameset and a webrtc webphone (using websocket) and I'm unable to figure out how to properly configure a good CSP. Apache version 2.4 Thank you.

Currently we have several PBXs in Asterisk and there is a problem of Flooding of port 8089 Webrtc, it is not an attack since the traffic is valid and comes from IPs of our clients, this problem happens when the person who is using the Webphone has intermittence of internet, generating multiple connections in closing state that are observed in the Kernel log. We think that to solve this is to use an Kamailio such as Webrtc Gateway and that the flooding is controlled from this point. We need a professional to help us install and configure an Opensips as a mid-registar WSS, which allows us to log the extensions found in Asterisk of the SIP type. Observation: The asterisk actualy is not Real Time, and for local devolpment with cannot change that.

Se cuenta con una plataforma comercial, para ello se requiere una integración webphone de asterisk. Con la cual se pueda realizar llamadas de un área a otra, de una persona a otra, de un anexo al exterior. Entre las funcionalidades que debería contar Asterisk es: * Grabación de llamadas * Límitador de llamadas * Ocultamiento de número celular destino * Habilitación de soporte web para el servicio Asterisk * Habilitación de soporte webrtc * Programación de dialplan para click2call * Configuración de Troncal SIP con operador * Programación de soporte para anexos dinámicos y temporales * Creación de servicio web para generación de anexos temporales y dinámicos * Creación de ...

Hello, I work as a volunteer in an emotional support and suicide prevention NGO. We get about 10,000 calls every day, in which people let off steam for about 30 to 90 minutes. We currently use softphones installed on the computer but I would like to place a SIP proxy and run a webphone in the browser. For that I will need an executable docker-compose file or detailed instructions for: - Frontend - inputs SIP parameters as ip, login and password - button to connect - disconnect button - button to answer the call - button to end the connection - no need to make calls, just receive calls - it doesn't have to look nice, I will adapt the frontend to our internal application. - you can use sipjs, jssip or any other open source SIP library. - backend a backend ...

Tenho um projeto de plataforma Saas que deve conter um discador/softphone/webphone que preferencialmente deve ser desenvolvido em JS e que se comunique ou seja faça ligações através de servidores PABX's. O principal ponto do projeto é que o webphone deve efetuar ligações através de vários servidores PABX do mercado, ex:. Asterisk, FreeSwitch, intelbras e etc... Caso seja necessário uso de WebRTC Servers, fica a escolha do desenvolvedor. "Podemos analisar outra linguagem de programação se for facilitar a entrega".

Softphone - webphone 3CX. (BSG) JSON Webservices, Webhook, programar una web para que el usuario final puede hacer una llamada unicamente hacia la empresa.

I need a sip based webrtc webphone with all basic features and audio video calling plus instant messaging is desirable but not must

Hi Malay P., We use fusionPBX and we want to offer customers a webphone (webRTC) and we would like this system to be developed both the webrtc client (customized with the company logo) and the server (wss). So that it has security and stability where we do not need to pass the password ("sip") to the client. It would also be important that in the future if we change platforms (fusionPBX) we can port this system to this other platform with a few adjustments.

Hi Ram Krishna D., We use fusionPBX and we want to offer customers a webphone (webRTC) and we would like this system to be developed both the webrtc client (customized with the company logo) and the server (wss). So that it has security and stability where we do not need to pass the password ("sip") to the client. It would also be important that in the future if we change platforms (fusionPBX) we can port this system to this other platform with a few adjustments.

Webphone webRTC e servidor wss

Olá Malay P., We use fusionPBX and we want to offer customers a webphone (webRTC) and we would like this system to be developed both the webrtc client (customized with the company logo) and the server (wss). So that it has security and stability where we do not need to pass the password ("sip") to the client. It would also be important that in the future if we change platforms (fusionPBX) we can port this system to this other platform with a few adjustments.

Hi Ram Krishna D., We use fusionPBX and we want to offer customers a webphone (webRTC) and we would like this system to be developed both the webrtc client (customized with the company logo) and the server (wss). So that it has security and stability where we do not need to pass the password ("sip") to the client. It would also be important that in the future if we change platforms (fusionPBX) we can port this system to this other platform with a few adjustments.

Hello, we need somone to install a webphone on an existing Debian 8 Server that has apache + SSL already installed. Use webphone found here: We have a freeswitch Server installed and ready with test extensions. We need this done today! We are willing to pay a bit more for the rush. So, Please respond to the task with "Dingo I'm Ready" at the beggining of your message so that I know you have read.

mettre en place un cahier des charge pour un sophtephone ou webphone j'aimerais développer un shotphone, j'ai déjà un serveur Asterisk en place qui fonction et il me manque que le softphone pour mon application

Need to upgrade existing Free PBX platform and insure redirection of its press options to connect to a sip server via webphone/softphone. Experience with call routing on selections from PBX options to sip server and interconnectivity thru voip(softphone) and pstn are required. Knowledge of Javascript and API implementations(Documentation is available) are required to hand-shake Mizutech webphone software with the FreePBX. Migration of files from external hosted server to the PBX server may be required.

Currently we have several PBXs in Asterisk and there is a problem of Flooding of port 8089 Webrtc, it is not an attack since the traffic is valid and comes from IPs of our clients, this problem happens when the person who is using the Webphone has intermittence of internet, generating multiple connections in closing state that are observed in the Kernel log. We think that to solve this is to use an Opensips such as Webrtc Gateway and that the flooding is controlled from this point. We need a professional to help us install and configure an Opensips as a mid-registar WSS, which allows us to log the extensions found in Asterisk of the SIP type. Observation: The asterisk actualy is not Real Time, and for local devolpment with cannot change that.

I'm looking for product professional and UX / UI to help me with prototype development and service / product requirements. With product design skills, who can develop application UX / UI and write functionality requirements with all the necessary stories. I will do the product narrative, telling the story, showing the desired features, the business applications and the professional should answer questions about their doubts, research market applications and show at each meeting to show the evolution of the product. Preferably if you are familiar with the Portuguese or Spanish language. We can also perform in English, but we need to be clear that the communication can have a little more noise. The service will be performed on milestones and deliveries. The product will be a simple...

Hi, I am having one dedicated server where I hosted the vicidial server. I am done the necessary configuration. When I register the carrier settings, I am facing the SIP account is not registered and carrier settings is new to me. I am using the Webphone and it is also not registering. It needs to be resolved.

Hello. I need to make a webphone with video support. I like a ctxsip but this webphone hasn't video calls support. I search for a talented js developer who can add a video to ctxsip.

I have installed GoAutoDial V4. I need some help getting it to connect to my freepbx. I want to configure a press 1 campain. I would like to broadcast a message to a list of prospects. If Somone listens to the message and presses 1 than i want it to forward to my FreePBX Extension. I would like to also set it up for regular agent outbound diling using the webPhone. Alos i would like to install the vicidial gui so that i can access it as well the way it was with Goautodial 3.3 I can give TeamViewer access to my computer where you will have access to both the GoAutodail V4, SSH, as Well as FreePBX as i watch and learn. Looking to spend $15 CAD

Webphone for integrate to our webbase sales management system. Functions : Answer, Hold, Transfer, Reject, Mute API Strong Installation Documantation (Word, Video)

Necesitamos integrar un webphone a nuestro CRM el cual esta programado en PHP7, los requerimientos serian los siguientes. 1-Posibilidad de loguearse en conjunto con el usuario del CRM 2-Posibilidad de conectarse a central telefonica por el protocolo de comunicaciones sip 3-Posibilidad de realizar Click to call 4-Integración al CRM para realizar y recibir llamadas

i have a vici server and i am trying to install the webphone but i need to upgrade from asterisk 11 to 13

SIP SoftPhone for Salesforce Salesforce integration with the Mizu WebPhone

We have installed vicidial with viciphone we follow the instructions when configuring the phone but cannot get it to work

We need to be able to 1. allow the user to call the lead phone number, call the contact in opportunities, call the account 2. call a campaign All this should be used via vicidial with a combination of a webphone/soft phone. Since vicidial is based on asterisk we could use direct connection to asterisk also, but logs etc must be available of course.

We are looking for an experienced developer to help us finish our Chrome Extension. Telzio is a business phone service provider, and this extension should work as a way for our users to easily click on any phone number identified on a website, and make a phone call with either our webphone (on our website) or by sending a request to our backend and have their mobile app or desk phone ring (this part we're taking care of). As for UI/CSS/design, we can assist with that inhouse - the important is that we find a developer with a lot of experience building Chrome Extensions.

Realizzazione di una WebApp, che permetta di comunicare con PBX Asterisk: Webphone WebRTC BLF Status

Webphone for integrate to our webbase sales management system. Functions : Answer, Hold, Transfer, Reject, Mute API Strong Installation Documantation (Word, Video)

Did you give up on the webphone project? I noticed I wasnt able to message you

I have the latest Vicicdial installed and need the webphone installed and configured for Viciphone

1) Integração do 3CX com Zendesk e outros CRM's Via Call Flow Desinger - Realizar na URA: Validação de usuário pela URA, pelo CPF, CNPJ, ID do Ticket, ... 2) Exibir Webphone 3CX na interface do Zendesk 3) Abrir ticket no evento - incomming call, se ele estiver na base, Identificar e exibir dados do chamador 4) se ele na URA digitou o número do ticket, abrir instantaneamente o ticket em questão; Se ele é usuário identificado, abrir novo ticket instantaneamente novo ticket 5 ) Se não for usuário identificado abrir novo ticket em branco Exibir na time line caminho percorrido na URA, número do chamador, 6) ao termino da chamada popular a timeline do ticket com os dados da chamada, Data, hora, Dur...

...salesforce developer with CTI lightning experience to build app to integrate with phone system from cebod telecom. Developer will develop the app available in salesforce exchange for customers to purchase or download. Here are functionality with are looking for. Link for APIs Webphone: Login screen to put sip credential and save them. Every time user login, it should automatically register with the phone system. Provide Webphone in salesforce account, where user can make and receive calls. Should be able to change the status to available/busy/not available Inbound Caller Identification: For all incoming calls, new pop should come up showing callers account information. If new caller, give option to create a new lead or new contact., or a new

I want to integrate a webphone into the web interface of a web app called SmarterTrack. You can use a webphone like this: I know it will be easy to integrate a webphone like this into the interface, but I want the webphone connected to SmarterTrack, so that it utilizes the built-in call log feature. Here are instructions on how to do this: See attached for a rough mockup of how I want the webphone integrated into the interface. I just want it like a drop down menu.