Sipml5 jobs

Se requiere una persona con experiencia configurando la libreria de Javascript SIPML5 y asterisk. Tenemos todo instalado y configurado. El softphone web se registra al asterisk, emite y recibe llamadas, pero cuando se atiende la llamada no transmite el audio. El servidor tiene instalado una VPN. Cuando el usuario se conecta a la VPN, entonces funciona el audio de la comunicación pero cuando no se conecta a la VPN entonces vuelve el problema del audio. Se requiere que el softphone funcione sin VPN, solo por internet. Version asterisk 18 OS: Ubuntu Server 14 +++++++++++ A person with experience configuring the SIPML5 and Asterisk Javascript library is required. We have everything installed and configured. The web softphone registers to Asterisk, sends and receives call...

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'm using FreePBX 14 and asterisk 18. I want config FreePBX to work with sipml5 or jssip. I'm trying to config webrtc, websocket to work with sipml5 but I'm not success. Please help me!

I have installed magnusbilling which come with Asterisk 13 on Debian Need someone to configure Webrtc clients to connect to it useing web phones And provide a " how to " guide file Thanks

I have instslled ASTPP which comes with freeswitch I need someone to configure WebRTC clients to connect web phones For example And provide " how to " guide

I'm looking for experienced web developer with WebRTC experience bu...dialing systems from scratch. We have a cluster of multi-tenant PBXs (VitalPBX) the new dialer must be integrated with. We expect this dialer to allow agents to place/receive calls, hold calls, transfer calls and any other normal call centre operation features to be implemented. We are open for the technologies you would like to use, e.g. it can easily be React.js with JSSIP library, or Vue with Sipml5 etc. It's would be also nice to stay in touch with developer for further development and some support in case we need something to know or to get be able to consultate with developer per hour/project basis. Minimum experience required: 2+ years web dev expertise with solid Javascript knowledge, WebRTC...

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 de...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.

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.

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.

Looking for someone with skills in: ReactJS Redux PostGresSQL (Sequelize) Pro in Asterisk and WebRTC Excellent Web responsive developer COMMUNICATION IS IMPORTANT TO ME, GOOD, RELIABLE COMMUNICATION - !IMPORTANT WEBRTC WITH SIP via SIPJS, JANUS, or SIPML5 - !IMPORTANT SKILLED AT MAKING DIALPLANS IN ASTERISK AMI AND ASTERISK AGI - !IMPORTANT NEED SOMEONE WITH VERY GOOD ENGLISH - !IMPORTANT The application is in the process of being built. I need someone to take care of the additional tasks. I need someone who can dedicate their time and provide clean and organized code. 1. "Build in call routing into the groups. Inside the SMS Group settings, when a call comes in we need to be able to decide if the call will: - Be forwarded to ring the last agent that spoke to the customer w...

...the styles that GrapesJs creates right in the <style> element of the created html snippet. I welcome advise and consultancy on how things should look like visually, as well as on how we can add additional GrapesJs modules (blocks, components with their own js) to make it comfortable for a user to create functional pages easily. We're going to have components like 'bpmn editor', 'google maps', 'sipml5 phone' that the user will be able to place onto their pages. Creating these types of blocks/components will be our next tasks. In your proposal, please answer or comment the following: Describe your experience with GrapesJs. How is it compared to its analogs? Ask questions regarding the task. Is the scope clear or you need any clari...

Hello - Ahmed referred me to you. I have an issue with WebRTC working with a FreePBX/Asterisk PBX server. I use sipML5 and everything connects fine. The problem is there is no audio when OUTSIDE the network. If I am connected to the VPN, everything works how it's supposed to. As soon as the VPN gets disconnected, the audio goes away on both ends. If I connect my PC without a router and give it a static public ip with firewall turned off, I am also able to get the audio to work even without VPN. As soon as I put my PC back behind router with a private IP it then stops working again. RTP trace shows packets getting routed the same as when I use a PC softphone. The softphone works just fine with audio whereas the webrtc client has no audio besides the two cases I described above. ...

Language: Javascript (NO others) Use I need a simple proof of concept for a sip client on a html5 mobile device with android. We are using an asterisk server, and can provide a pjsip or sip extension. I want run the webrtc code as a function on a bigger web page based application, so you can create a simple , and contain the phone display, logic and functionality inside of a div. I want it to pop up a div when an inbound call comes in, with an "accept", and "deny" button, just as you might expect if you were receiving a phone call. There should be a way to hide/show the div. There should be a little div, with the most current status displayed This div should also have a "call" button, which triggers a dialpad appear on he screen, with a dial

Looking for some one who can connect JSSip or SIPml5 to asterisk with incoming and outgoing calls,

Hi, I have fully installed Asterisk 16 and working webphone based on sipjs. Audio on that setup are working, what I need is a someone who reconfigure it and make for me working videochat on this. It could be sipjs or sipml5 i dont care. It have to works audio call _AND_ video call

We need experienced users who have already done similar projects We need integration of embedded SIP cilent woth WEBRTC to work with FREEPBX with all functions supported : hold transfer - attended , unatennded dial etc .. link : regards

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I have a running asterisk 11.24.1 I also have webrtc installed but my sipml5(dialer) cannot called out keep having "Not acceptable here".

Experianced developer needed for open source scripts, 1- to install on as a STANDALONE working script (no depended url's exepted) 2- to modifie and make smaller fully customised other packages clik2dial, => will transformed to (and also SSL dependency will be solved from) webrtc2sip, webrtc-everywhere, Boghe, iDoubs, IMSDroid, libSigComp, Server-side, ...needed for open source scripts, 1- to install on as a STANDALONE working script (no depended url's exepted) 2- to modifie and make smaller fully customised other packages clik2dial, => will transformed to (and also SSL dependency will be solved from) webrtc2sip, webrtc-everywhere, Boghe, iDoubs, IMSDroid, libSigComp, Server-side, telepresence, Flash2IMS, sipML5

Create embeddable javascript widget and remote script so that the javacript can be copy and paste to any website or blog. The idea is to allow embedded of SIPML5 phone() to any website/blog. Requirements: 1) Work in all browsers 2) Short codes 3) Shouldn’t interfere with any other events on the target page 4) In case of service down, should not block target page from loading (eg. async) 5) Can't rely on any external Javascript Libraries and not pollute namespace of target page

Hello we would like to someone to build a web phone that would work in the chrome browser. This web phone would need to connect to a asterisk server to make and receive phone calls. We would like to use webrtc / sipml5 or any other technology that will work successfully. Please respond with "What up Dingo " at the beginning of your message so that i know you have read completely

I need to integrate my asterisk server to my website with Webphone with Multiple call Support . Asterisk server is ready.

We need to install the latest stable release of OPENSIPS and ASTPP in the server and then establish calls between users using SIPML5. Calls should be established between - Chrome to Chrome browsers - Chrome to Android devices having the app installed and vice-versa - Chrome to IOS devices having the app installed and vice-versa - also includes a Chat function and the ability to call the PSTN For video calling purpose we need the help of webrtc. The video call, voice call, voice chat should work seamlessly in both 2G and 3G networks.

On our Centos 6 server, with asterisk 13 installed, we need to implement all the necessary settings in order to make sipml5 work. The job will be considered finished after testing that all the inbound and outbound calls work. We provide access to the machine just and only though teamviewer and/or anydesk.

Need little bit changes in Its open source , download the codes and check whether you can modify or not.

We're building a PREDICTIVE DIALER system written in PHP utilizing webRTC/SIPML5. We're looking for an EXPERIENCED, qualified individual who has much knowledge with predictive dialers. Must be WELL VERSED with the following: php, mysql, asterisk pbx, webrtc, javascript, html5, css. Looking for someone in the UNITED STATES who is available for consistent work.

would like to take and just show two image buttons to "accept" and "hangup" and pre-poulate form objects instead of asking for user for any input. see additional design concept in attached file - basically looking for 1 page that I can embed in an iframe. I built the whole backend VoiP stuff so only need the web stuff

I have 2 teams at the end of a project and we are running into an issue with securing the web sockets for outbound calls through our CRM. Our CRM is Microsoft SQL and our dialer is Asterisk based. I need a consultant to help us find our issue and fi...web sockets for outbound calls through our CRM. Our CRM is Microsoft SQL and our dialer is Asterisk based. I need a consultant to help us find our issue and find a solution. PLEASE ONLY EXPERIENCED RESPONSES ONLY!!!!! Below I have listed the 1. Must be familiar with Asterisk Server 11 2. Must be familiar with WebSockets and Secure WebSockets 3. Must know how to use Secure WebSockets on SIPML5 and be able to receive inbound calls 4. Must be familiar with MeetMe conference Asterisk App 5. Must know how to detect RTP and SR...

We need a basic webpage integrated with SIPML5 () You just have to develop the Front-end w already took care of the back-end. What you have to do is: - Login page (index) based in a simple mysql table. - Two buttons, 1 for login and one for logout - Third scroll down button for audio and video call - Status of the connection (connected - disconnected) - Text field (alpha numeric) tp enter the number to call All above features are already developed for this library an you can find an example and also you can use as start point at () We don't want more than the features we listed above. You can start with the sipml5 example and get rid of the things we don't want. - If you click on the button 'expert mode'

Hello, we are looking for someone to develop a sip enabled web phone using WebRTC + Javascript SIP/SDP stack + Asterisk. We would like a webpage created that will do the following: Works on Chrome, Firefox, IE, Safari, Opera and Bowser Audio / Video call Screen/Desktop sharing...Firefox, IE, Safari, Opera and Bowser Audio / Video call Screen/Desktop sharing from Chrome to any SIP client Instant messaging Presence Call Hold / Resume Explicit Call transfer Multi-line and multi-account Dual-tone multi-frequency signaling (DTMF) using SIP INFO Click-to-Call SIP TelePresence (Video Group chat) 3GPP IMS standards Links: Please Respond with "What usp Dingo" at the beggaining of your message so that I know you have read.

I need some changes to an existing website. Currently seeking expert in new web development technologies. Experience with the following required: design, sipml5, webrtc, and sip. I need buttons to call pre-configured associated sip user initiate audio full duplex via existing hosted IP PBX.

Hi TeqSystems, I noticed your profile and would like to offer you my project. We can discuss any details over chat. We need an expert review of an integration of a 2-way and 3-way conferencing application that uses Freeswitch, WebRTC, SIPml5, Verto, C# or C++ and some custom integration.

We need to configure Asterisk PBX 13 on FreeBSD to work with sipML5 client.

Hello, We need to customize sipML5 opensource soft web phone to work and be installed on our servers. Please check their website and ask us any questions you might have. We are looking for a fast turnaround. Thank you

The server informed bellow has an Asterisk 13.5.0 installed and we need to configure it to work with SIPML5 free demo at Fell free to install anything in this server (will send the server info for the ones who apply) and change configurations, but please document everything that was needed to make it done. It need to work with webbrowsers like Chrome and Firefox.

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We have installed test Kazoo cluster with two nodes. We needed t...have installed test Kazoo cluster with two nodes. We needed that WebRTC properly work in Kazoo include client with IPv6 addresses so need to apply patch . Now we have some errors with jssip: SIP/2.0 488 Not Acceptable Here mod_dptools.c:3277 Originate Failed. Cause: INCOMPATIBLE_DESTINATION and sipml5: SIP/2.0 603 Failed to get local SDP - when I trying to answer on incoming call. Client has ringing but can't answer. Additional task - need to enable incoming call from external carrier. (Now I can't select option Peer in Kazoo-ui carrier settings). After fix this issues we need simple report how to solve it in feature.

I’m looking for someone to help me develop a SIP phone using the sipML5 API located at Please research this site and see how easy it is to use this API. (Please state you are not a robot in your Bid) Your job is to create a web based phone that appears like a modern cell phone with the ability to make and receive phone call via an asterisk server to local or off site server. I’ll provide you server and credentials for testing. Your created phone should be able to function on a webpage or desktop with http service running. We'll need an installer that simple to use. Basic Phone Features: 1. Make phone calls to any phone number. 2. Receive phone call from any phone number 3. Ability to redial last number dialed. 4. Able to place a call on-hold wit...

We are building a web phone based on sipML5 project. Project is to customize the User Interface for our brand. demo can be found here : programmer's guid: Please only bid if you will do this project yourself, also new freelancers are welcome. freelancers with sipML5 experience in their past project are prefered. We need to start immediately.

Need a devleopr to debug and fix our SIPML5 website. We installed Vtiger on a Ubuntu server and we need help finishing up the configuration. We have it currently working in a test enviroment but it does not have sound and will not work with Asterisk on another server.

Install and configure FREESWITCH on CentOS so that it successfully integrates with sipml5 based web client to handle outbound calls via multiple trunks (wholesale VOIP providers)

Android, iOS and HTML Sip phone Development.

We are wanting to setup and use SIPml5 as a viable application for users to log into their voice accounts with a web page. This is what SIPml5 does but the problem we are seeing is that the REGISTER comes from some unknown ip to us 188.165.231.30. This is probably becuase we are not programmers and do not have a full understanding of how to get this package to work properly with our system. Already Done Work - setup and installed SIPML5 from on a server - configured a SIP Proxy to work with SIPML5 - tested Registrations What we need you to do - setup and configure SIPml5 for ease of use with our system SipXecs / FreeSWICH - setup a landing page for SIPml5 configurations - change the code so we do not use for registrations - wo...

We have FreeSWTICH installed on our Ubuntu server. Now we need to configure it with webRTC client "". Work ground would be to configure freeSWITCH for webRTC with Sipml5, users of freeSWTICH can log in on this website and will be able to have Audio communication with each other.

I need someone to send me a working config file for setting up kamailio to act as an outbound proxy for webrtc sip clients. An example javascript client to be done in jssip or sipml5. You will provide me a kamailio config setup for websocket outbound proxy.

I want customization of Sipml5 with click2call button. I have a freeswitch/Kamailio server that I need a click2call button . We can work with other platforms like jsip, but we want most supported on browser Task 1. Customize a webrtc software eg sipml5, - 2. Provide interface for creating click2call button, we should be able to link to image 3. Click to call should anonymously initiate call to our desired extension on the SIP Server eg 1001. Website visitor should not require to login. This should also work when visiting link from mobile phone

I'd like to deploy a simple Webrtc2SIP implementation ( - open source BSD license) using the 4x modules i.e. 1) SIP Proxy 2) RTCWeb Breaker 3) Media Coder 4) Click-to-Call I run an Asterisk 10.12.1 PBX and the aim is to build something similar to the demo at This is a simple task for the right freelancer. Thanks

We need to install the latest stable release of OPENSIPS and ASTPP in the server and then establish calls between users using SIPML5. Calls should be established between - Chrome to Chrome browsers - Chrome to Android devices having the app installed and vice-versa - Chrome to IOS devices having the app installed and vice-versa For video calling purpose we need the help of webrtc. The video call, voice call, voice chat should work seamlessly in both 2G and 3G networks.

We would like to offer our students a free multiplatform PRECONFIGURED SIP softphone preferebly using PJSIP sipstack and should use OPUS as...be made). Examples of softphones using PJSIP: Android -> we currently use this and are quite satisfied and it would need no modifications, just preconfigure a sip account. iPhone/iPad -> WebRTC - open source SIP safari -> The integration will be with an asterisk pbx and I will need to test each platform for compatibility. The use of OPUS is a VERY HIGH PRIORITY so we can offer exceptional voice quality.

Project Description: I need someone who has experience with the open source HTML5 SIP client software () to create a SIP client in HTML5 to run on a smartphone that can work with a 3CX VOIP server. NOTE TO ALL BIDDERS: 1) I am NOT looking for a native iPhone or Android app. Read the project description carefully and dont waste time by posting generic responses. I want an HTML5 app ... NOT a native iOS or Android app. 2) Only respond if you have previous experience working with the sipml5 library and are very familiar with the SIP protocol.. 3) I dont want to deal with a company - only bid if YOU intend to do the work yourself.. Thank you