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Hello, We need to develop a SIP to Whatsapp gateway. The gateway should be able to pass voice calls incoming over SIP and forward them through WhatsApp to complete the call to the called party number. The development platform/operating system is not important. The project should be completed either by using the Linux/Windows WhatsApp executables, or by using the Android / Windows Phone mobile versions of the application, no matters on the version number. The implementation should return the correct call error codes to the SIP backend, i.e. CALL SUCCESS, BUSY, UNAVAILABLE, etc. For a successful project, we'll select the one, triggering successfully continuous SIP/Viber calls. Functional flow 1) Calls originating will send to WhatsApp gateway 2) whatsapp gateway converts the sip/iax...
Need to develop ERP (Odoo) + EMR + Phone service VOIP + Zen Desk into one platform I want to have ERP features, live chat , voip, sms. Looking to hire several developers on a monthly basis 40 hours a week.
I need a BYOD platform for VoIP where we can make it easier, cheaper and simpler for Small businesses to connect any VoIP-enabled device and be up and running in few minutes. It must be a Zero touch platform. Clients must be able to easily signup and allocate their own calling plans, load airtime, create extensions and allocate, etc. Here are the most basic features this platform must cater for: Uncapped/Capped Calls / Easy Usage IOS & Android App Integration / Call Groups / Superb Voice Quality / Voicemail & Call Routing / Cloud Hosting / Basic or Advanced IVR / Live Chat / International Extensions / Quick & Easy Set-up / Complete Scalability & Flexibility / Video Conferencing
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)
Hi, We need assistance to set up Elastix last version 5 1. 7 internal extensions 2. 4 fxo ports with analog lines (sangoma a400 card) 3. Dedicated outgoing calls from extension through the correct fxo port 4. Incoming call to dedicated extension on chosen fxo. 5. Fax to email Reach me by whatsappp 198o322872two
We are currently building a Kamailio SBC which will be as Sip Proxy for our B2B, but we are facing a bug. Seems there to be a misconfiguration in Kamailio because when we are using a dispatch request, re-invite and ACK is not working to be sent in right way, because the re-Invite is sent back to dispatcher list which is making it lost and it is not forwarding to callee. This issue is mostly present at ACK and BYE. Hope some one can solve this and fix this SBC.
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
Communicate in English or other languages through inbound and/or outbound voice and/or non-voice communications to support customer service, telemarketing, credit and collections and/or provide remote technical support to customers. Verify and update customer information, respond to inquiries and resolve issues. Quickly resolve customer inquiries by providing appropriate and accurate information using the highest quality customer service standards. From data entry, updating records, invoices and claims to validating information and closing transactions. Follow up in a timely manner to ensure customer satisfaction and maintain optimal service levels and customer relationships.
Hello We need to build auto dialer + IVR Survey call on asterisk After this we need to integrate this PBX with ERPNext CRM Module
Answer calls, resolve/assist concerns; transfer calls if needed University for at least 2 years At least 6 months of inbound/outbound/customer service/back office work experience in a BPO setup Above-average communication skills
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 can shar...
we are looking for a very experienced Java senior developer, tech lead or architect. Second mandatory senior expertise and experience is about integrating SIP (VoIP) into a Java Applications. What will be required? We will define a call, we go hands on through best ways of integrating SIP / VoIP into a Java application, which may run on a backend but also on a frontend (desktop/laptop) You will discuss with our Java product owner about integration options of SIP into Java. You are a good fit, if you have: - used SIP already in multiple real life java projects - used SIP phone handling in real life java projects you are a bad fit, if you have never done tasks of integration SIP into Java A simple test task for seniors in SIP+Java: - listen to incoming calls (SIP provider can be provide...
Your task is to implement a prototype (in Java), which detects incoming SIP/VoIP calls and opens a browser with a URL. The URL contains some query parameters, like the callers number. As a UI-less (cmd line only) app on the desktop it will be so a simply app which does basically a call listening and triggering a browser based URL to show. The listener shall be configurable, means - what URL to call - which phone numbers to ignore - which phone numbers to track (regEx) Please recommend what to use SIP-RI/SIP-Jain to detected phone calls. You will get a SIP/VoIP account to be able to implement and test incoming calls (if you do not have your own) Your task is to implement Java classes to be able to access details of incoming calls as a listener. Implement JUnit test cases, which is s...
Witam, Interesuje nas pomoc przy ustawieniu centrali freepbx (z gui). Potrafimy zainstalować, wstępnie skonfigurować, uruchomić do działania. Nie potrafimy jej doustawić, tak aby połączenia wychodziły z odpowiednich zakupionych numerów, itp. Chcemy to zrobić z pomocą ekspertów własnymi rękoma (aby się nauczyć). Przydałoby się na to zapewne kilka godzin ze specjalistą, który nam wskaże co robimy źle.