...able to pass voice calls incoming over SIP and forward them through Viber/whatsapp to complete the call to the called party number. The devolpment should run under GNU/linux (Asterisk,etc). 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
I will have an asterisk as a voicemail server. Another PBX will have all the extensions ans will forward to asterisk in case the user is unavailable (voicemail). Build a website to control an manage asterisk voicemails. The website must have one manager level to create backups, add, remove or edit mailboxes and another interface to users where they
I am looking for a Content writer (English) ,who has experience in writing Content for Telelcom / VoIP / International calling Service websites. Content writer should have a good understanding of SEO strategy too.
Asterisk is a multi-threaded telephony server. It already has channels for the JACK and ALSA sound systems. However, many Linux systems only come with Pulseaudio. Jack is difficult to install+configure, and ALSA frequently doesn't work correctly. Your job is to write a native Pulseaudio channel so that the Asterisk dialplan can call Dial("PULSE/Joe")
I need a sip-phone (IOS, Android, Windows) able to register into Asterisk and peer a SIM in a GSM gateway. Sip Client should be able to send/receive voice calls, SMSs and USSDs. Sip Client should be able to top-up the peered SIM.
I will give access to a linux box, you can install asterisk along with any other services, I want a web interface, where i will copy paste 200 phone numbers, also there should be option to upload a voice record (audio) , once i click play button. Each of the 200 numbers should get a call from their own numbers (spoof) simultaneously and all of them
VOIP E Commerce API to create SHOP folder in JASON for wordpress -- 2 AWAITING ACCEPTANCE
...Knowledge of WebRTC Gateway and not WebRTC web-client. Have developed SIP Servers applications (SBC or PBX) Good knowledge in WebRTC, Web Sockets, HTTP Knowledge of Asterisk server and Jitsi Media. Experience in product development and system engineering for WebRTC. Knowledge on JAVA will be good to have (Not mandatory). Min 1 year of experience
...with the httpd/virtual server configuration. Here's the install guides for both [url removed, login to view] https://wiki.freepbx.org/display/FOP/Installing+FreePBX+14+on+CentOS+7#InstallingFreePBX14onCentOS7-FirewalldBasicConfiguration I want Observium to continue operating at http://<hostname/ and asterisk admin panel to run at
Hello, All! We have ASTERISK with realtime. We need develop RestAPI service that will extract peer information from ASTERISK 1.6 and send it to client. This service must have limits for some parameters like queries per second, answer with care. so it should affect on system performance. Write your way to do this in your bid, please. Or use chat