Now that we've got base server stuff rewritten in ansible, remove the
old puppet versions.
Depends-On: https://review.openstack.org/588326
Change-Id: I5c82fe6fd25b9ddaa77747db377ffa7e8bf23c7b
This is no longer needed as pabelanger is now an infra-root.
Change-Id: I6939c2ee9e3662c08441dd0ddad3e84a47bd0192
Signed-off-by: Paul Belanger <pabelanger@redhat.com>
Centos6 is being deprecated so we need to move to something newer.
This will require pbx.o.o to be rebuilt.
Change-Id: Id3fc74bf58ba5febac79674e6fd23d6ade3e4bd1
Signed-off-by: Paul Belanger <pabelanger@redhat.com>
By parameterizing the selinux mode we can set the mode and pass the mode
to other classes so they can make decisions on whether or not to run
selinux commands as part of their configuration.
Depends-On: I21add092d9d09077f2b23760a384f5a5cb91d86a
Change-Id: I86a5bb006b6ab9d64f7fbd894c967428cfaed8f8
The gate-config-puppet-apply test will try to run everything on a
bare-precise node, so even though this doesn't occur in production it is
being tested.
Change-Id: I13b03a7476e30cbb26dc820bb4cf5b56c4018669
It may be useful to have a consumable log of every call the system has
processed. This enables a simple CSV based file for CDRs (Call Detail
Records).
Change-Id: I2086594a127e1377056fdb10af763ade3af4ad56
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
Also add myself to pbx.openstack.org to help provision the server.
Change-Id: Ib9fcc17b4aafe33ef8692baab248398338b2d499
Signed-off-by: Paul Belanger <paul.belanger@polybeacon.com>
Add the necessary bits to get Asterisk installed on the pbx node.
This is using Asterisk 11 from Digium's repo.
Change-Id: I200789c7ee7fc1fb2e0779a38db2ea35ead998ae
Ian Wienand