...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
Hello guys. A friend have an e-shop to a VPS... I don't know what he did but his website it's offline. I saw from plesk panel that apache was off and i couldn't start it... I wasn't able to click "start" apache. I tried also from ssh and i saw an error: unit not found. I'm not ssh expert so i can't do something. I know that it's 5 minutes job for s...
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
Site has become too slow since couple of hours
Hi, I have a Magento Store (1.9.3) which is very slow and not optimised for speed. The site is with over 200K products and 1500 categories and many installed extensions. SERVER SIDE: Intel® Xeon® E3-1275 v5 Quad-Core incl. Hyper-Threading Technology RAM: 64 GB DDR4 ECC Hard Drive: 2 x 512 GB NVMe Gen3 x4 SSD (Software-RAID 1) ================ PHP7, HHVM, redis session and backe...
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
Hello i have Centos VPN Servers and i have Cisco VPN Routers i need to link this routers to the vps as vpn clients
...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
I need a system admin to upgrade the HAPROXY on my centos 7, 64 bits to the latest version. Current version running and in production on the server is : 1.5.18 I need this done urgently .... PM me if only you have installed HAPROXY before and have experience running it .
Dear Nerds, I have a CentOS 7 system that has been running Observium (SNMP service) without issue. I wanted to add FreePBX on to the same host. Both services are now installed and are running, but I need a helping hand with the httpd/virtual server configuration. Here's the install guides for both [url removed, login to view] https://wiki