The opendev project has been moving away from puppet and this is one of
the puppet modules that is no longer used. To simplify things for us we
are taking the extra step of retiring this repo.
Change-Id: I46c786779b8f8d5b7dd836250de10f259744b5b5
The new resource ordering algorithm in puppet 4 may cause the asterisk
service to be refreshed before the module directories are in place,
which then causes the module reload exec to fail. This patch adds an
explicit ordering to the customdir defined type to ensure that the
module directory exists before the asterisk service is refreshed.
Change-Id: Ic892983476c7da60e4801779dfecc50587a2404d
This is needed to migrate pbx.openstack.org to Ubuntu Trusty. The
changes will not affect centos6.
Change-Id: I74094e6777ec768bb1ab3f2d480e3eecdb15c363
Depends-On: Id3fc74bf58ba5febac79674e6fd23d6ade3e4bd1
Signed-off-by: Paul Belanger <pabelanger@redhat.com>
Discussion following https://review.openstack.org/#/c/200854/ confirmed
that we do want to enforce this check. This patch re-enables the check
and fixes lint errors that fail the check.
Change-Id: I39cdaa5c26668f7815460c489fcde543a57e9a01
In anticipation of puppet 4, start trying to deal with puppet 4 things
that can be helpfully predicted by puppet lint plugins. This patch also
corrects lint errors caught by the puppet-lint-absolute_classname-check
and puppet-lint-empty_string-check gems.
We disable the disable_arrow_alignment check in the Rakefile.
Previously, the system version of puppet-lint that was being run by
'rake lint' was too old to catch this. When using 'bundle exec rake
lint' the gem is new enough to catch this. Whether we want to actually
fix these lint errors is a different discussion, so this patch disables
the check for now.
Change-Id: Ib44f2d6ac941e6fed118a5ff78bddd3d0f0a40b5
* modules/asterisk/manifests/init.pp
* modules/jenkins/manifests/master.pp
* modules/jenkins/manifests/slave.pp
* modules/openstack_project/manifests/gerrit.pp
* modules/openstack_project/manifests/jenkins.pp
* modules/openstack_project/manifests/nodepool.pp
* modules/openstack_project/manifests/static.pp: When a directory is
puppet-managed for content using recurse, replace and purge you also
need force or empty subdirectories will fail to be removed. What's
worse, subscribing to that directory will cause a refresh to be
triggered for it on every agent run.
Change-Id: I232d6ba98475522f391f469c194a4450c7a0b2e1
Rather then autoloading everything, we explicitly load what we need. I
find this give the user better control of what is installed by default.
Additionally, upstream (my) puppet modules will likely expect this.
Change-Id: Ib572c54053bd5b5f9a3a513f6f8696db87ea0864
Signed-off-by: Paul Belanger <paul.belanger@polybeacon.com>
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
Clark Boylan