Can anybody explain why open-source Asterisk is favourable in comparison with a traditional PBX system from a vendor such as NEC/Syntel/Matrix or Siemens.
Asterisk is an open source PBX and FreeSWITCH is an open source soft switch.
Source: http://www.freeswitch.org/node/117
Can you please explain the difference between PBX and soft switch in less technical terms
Examples illustrating the…
I start the red5,
and then i start red5phone
i try to register sip user , details i provide are
username = 999999
password = ****
ip = asteriskserverip
And I got -- Registering contact -- sip:999999@127.0.0.1:5072
The right contact…
I have been in IT for a long time now doing software development and some system/server administration, but all mostly software-related services. I would like to help set up a small business (~50 employees) with Asterisk, but I am not very familiar…
I'm using Asterisk 13.1.0 as packaged by Ubuntu Server 16.04 to run a pure-VoIP phone system. Asterisk has a module – phoneprov – that allows it to template out configuration files for specific lines and serve them from its builtin HTTP server. I'd…
We currently run FreePBX on a single node. I'd like to have a cluster of asterisk instances for sharing the load, but mainly for failover. I'm curious how other people have solved this problem. Ideally I'd like to use "free" (as in beer)…
Is it possible to configure Asterisk so that it sends RTP packets with audio from the receiver before the remote party actually picks up?
This seems to be required for a VoIP compliance test my setup needs to pass. They use a simulator to test…
So I read a lot of good things about Asterisk. I am not however looking to run a call center or small business setup. I am still interested what potential uses it has for me as a "power user" and what features I could harness for my communication…
Does anyone know if you can setup a Skype-to-SIP gateway on your network?
I tried Uplink, which works wonders, but only on Windows and with Skype on the same machine... It would be awesome if you could put that program on any machine and then use it…
After struggling with Asterisk for WebRTC for a few weeks now, I decided to put my problem on this forum. My Problem is as follows:
Im not getting audio from WebRTC to WebRTC clients. I work in a LAN environment.
Situation - Call from JSSIP to…
I am trying to set up color prompt in asterisk CLI. In the documentation I have found this:
%Cn[;n] Change terminal foreground (and optional background) color to
specified. A full list of colors may be found in include/asterisk/term.h*
But nowhere…
Does anybody have experience or running Asterisk on VMware(?) - specifically the latest ESX...?
I've got a proof of concept working but am reticent to roll it out without a bit more reassurance! I keep hearing about problems with timing...?
Anybody…
I'm having an intermittent issue where asterisk will play our greeting to the caller, and then drop the call instead of making our phones ring.
I'm unable to reproduce the problem with any phones I have here, and many callers get through just fine.…
I'm trying to set the caller id number for an outbound call.
My asterisk .call file looks like this:
Channel: SIP/flowroute/1234567890
Context: test
Extension: 1234567890
Priority: 1
Here's my extensions.conf:
[test]
exten =>…
I am pricing out hardware for 2 Asterisk (trixbox) systems, they are both smaller (one is a 4 line and the other is a 8 line, both analog TDM800p and TDM400p respectively) installs. And the one thing I have come across is the significant price…