This sets our testing to test pbx manifest on xenial and adds a hiera
group called pbx to the servers so we can properly do servernames with
digits and have common config in hiera.
Change-Id: I8c3096d18fe318c6ca206203de0ac984c8934566
Non instance variable representation is deprecated
so needs to be changed. This change changes varibles
to their instance variable representation.
See more details see:
http://docs.puppetlabs.com/guides/templating.html
Change-Id: Ib77827e01011ef6c0380c9ec7a9d147eafd8ce2f
Many SIP clients require an account. Add a simple account with a
username and password of "openstack" that people can plug in to their
client if it doesn't support making calls without it.
Change-Id: I08f73a6b1976de6c0bdd2f93cf7e36083fe8b12d
SIP is pretty much terrible at dealing with NAT. Asterisk has some
knobs that can be enabled to help deal with common issues. Turn them
on.
In passing, remove the videosupport=yes line. We don't have video
enabled for the conference application and we don't have any video
codecs allowed, so this option didn't do anything.
Change-Id: Ibc17ad3da9bbc110a8cb19daaea1655d0a208670