...any number in any list: show a different calling number on customer phone for each attempt. Will give you 10 calling numbers that we have 3) Set up load balancing on 3 trunks (sip provider). When dialing on lists, send one call to one trunk, next to the second trunk and next to third trunk and so on... 4) Failover : after dialing a number on a trunk, if
...to launch our new product to the world Naturally flavoured Coffee beans. Imagine its early morning your somewhere between being asleep and being awake and you have that first sip of the freshest, best quality, real Arabica coffee with a hint of rich Caribbean vanilla rum and tropical spices or smooth almond marzipan. Ahhh absolute haven, aromatherapy
Hi All I have a SIP client on my network, its a Intercom. It registers to my FreePBX/Asterisk solution and makes calls to my mobile phone fine. I want to intergrate Skype because Amazon Echo's can answer/make calls using Skype. The intention is to receive the Intercom call on my home network to an Echo Dot, which I believe will work when the FreePBX
...allow only Checked In guests to sign in using surname and Room number. once signed in also allow guest to connect to the Hotel WIFI also within the APP we will need a build SIP/VOIP on the App which will interface with hotel PABX and the app will become the guest extension as long as his signed in and for the duration of his stay. This app will need
The client is using astrix server. They will give us a vpn connection to their network And we need to setup WebRTC client to talk to astrix server. We need to use open source functionality working which it can't currently get a function to work within a type. [fazer login para ver a URL] So that they can make a 1 way audio call which is having issues with local and remote media streams to wor...
Looking for freelancer to configure FusionPBX with PowerPBX hosting. [fazer login para ver a URL], [fazer login para ver a URL], Debian ...Plans, PBX Extensions, I/O Trunks, Voicemail, DISA, IVR & local number display on the go, Call Forwarding, Call waiting, Linphone setup and instructions, call recording, setup sip trunks. for 4 - 6 lines. PHP or LUA scripting, VOIP, interconnect
... Predictive dialing, IVR, Incoming calls, and potentially GSM hardware upgrade (need to evaluate solution) with Asterisk expert. - Server health - IP phone connection - SIP line - Bandwidth and latency of system to network - Connectivity Other task related to asterisk server that will be needed in future: We need to have a setup, where we
I need help you configure my ejabberd server with sylkserver for translate jingle/sip between xmpp client and SIP proxy and/or PBX. Goal is that XMPP client can make a call through the PBX to the outside work (as a regular UA) and that the XMPP client can be called via the PBX.