I've imported a few puppet scripts from my asterisk package. This
will add reload support to asterisk until I get a chance to update
the module for CentOS.
Change-Id: I6d7f1d7b415de8fc9ccd55e887a6050f2e32f2a7
Signed-off-by: Paul Belanger <paul.belanger@polybeacon.com>
Telephone calls are generally 8 kHz. With VoIP, higher sample rates
are possible. G.722 is the most common, which is 16 kHz. Conferencing
supports these higher sample rates, as well. Make sure we have the
higher quality sound prompts installed as well in case we can make use
of them.
While we're at it, install the gsm prompts, as well. That is the low
bandwidth codec currently allowed.
Change-Id: I34921fa6a00720d05113a848bc9f1f94f2200c8b
Enable inbound SIP calls. There are a few steps to this.
1) iptables config. Open UDP and TCP port 5060 for SIP, as well as
UDP ports 10000-20000 for RTP.
2) Add a custom sip.conf which makes chan_sip listen on all address, including
IPv4 and IPv6. Also enable unauthenticated inbound calls and send them to the
'public' dialplan context.
3) Create the dialplan. Right now it just plays a sound prompt called 'spam'.
You'll have to call in to find out what it says. Note that this required
installing the extra sounds. There's a bunch of good stuff in there that
may be handy, other than just 'spam'.
Change-Id: I6b62511317603eedf9280b55a00ba5cee0611b62
This commit sets up the basic configuration for Asterisk. It will allow
Asterisk to run, but it won't do anything useful yet.
Change-Id: I7975082ff5351db4dc6e3c8cf9dd2d90675e3108
In addition to installing Asterisk itself, we want to install some sound
packages. This includes a large set of prompts, as well as some music on
hold files.
Change-Id: I197079cb2398f97ae82abf38a18d5cb8c377b5bc
Add the necessary bits to get Asterisk installed on the pbx node.
This is using Asterisk 11 from Digium's repo.
Change-Id: I200789c7ee7fc1fb2e0779a38db2ea35ead998ae
