Ensure asterisknow-version is installed before asterisk

Change-Id: Ied523d40c9ec670f6ca1d0ca30adaa59b489e784
Enable inbound SIP calls

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
Enable basic conferencing support

Set up basic conferencing support.  Right now I have reserved 6000-6999
as conference rooms (not that we actually need that many, but whatever).

Change-Id: I9acddf4ffedc7f499740184778b8bd67e5b38a4f
Install additional sound prompts

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