To fix some bugs on an existing Asterisk system that uses WebRTC for remote agents to take calls with their browser. This is not for beginners or opportunists ! You must know the subject well. Please see details in attachment.
...to perform some validation on these numbers to confirm if they are still valid or not. We use FreePBX/Asterisk, is there a way to automate something that from the PHP system, an API call is made to a number, the return code from the API/Asterisk is checked to confirm if valid or not, the record in MySQL then updated to reflect the numbers validity
Cloud Based VOIP Auto Dialer for Voice Broadcasting with 500 Concurrent channels no need of agents required, only ring and play message , should use VOIP SIP routes , multi campaign support [url removed, login to view] ID [url removed, login to view] upload [url removed, login to view] time setting [url removed, login to view] upload [url removed, login to view] campaign
I have a ByteSaver Installed Over Linux Server need to change The ByteSaver Sip Ip Inside To Another Sip IP, Need Someone Familiar With ByteSaver VOIP,
I need someone to get OpenBTS & Asterisk or Elastix set up and modified for my brand. Build Apps and other things too. I also need someone who can create NXT and Ethereum Block Chains. We can negotiate a price when we speak. I am not a programmer but I know how it all works, so I need someone who speaks good English and can explain how everything works
To fix and repair an existing system with Asterisk and WebRTC for agent registration and taking calls. System is made up of 3 servers for Apache Web Server, FreePBX/Asterisk and MySQL Data Base. Must be able to work on the servers through Teamviewer. This is not for beginners ! Please see requirements attached. Everything is well documented and must
I created a program in C# using Ozeki to create SIP server protocol. I didn't successed to create a remote database connection via MySQL. Looking for someone can fix my bugs and help me add more features.
...for iOS and Android. The call between SIP server and the webRTC client will be within VPN and the client has to then invoke a web based application using device internet passing on the caller number and some other parameters: The mobile application will facilitate both inbound and outbound calling on SF The inbound call lands on Sip Server Routed
SIP Program (Really simple) Create, edit, delete users & each user has one-more parameter named URL. when call inbound to user the program knows send GET request with callerid to specific assigned user's URL. GUI is not required, you can make it with NodeJS or C# or whatever. it should be SIP server for anything. (the program)
...need to develop a SIP to Viber/Whatsapp gateway. The gateway should be 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
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
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")