* 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
After some more testing with the -infra team, it was found these changes
seem to provide ths best experience within a ConfBridge.
Change-Id: Ibfaa8ede94134e10a699f19913e682e9b042a5ce
Signed-off-by: Paul Belanger <paul.belanger@polybeacon.com>
*Grumble Grumble* Seems the Asterisk packaging from Digium is missing
asterisk functionality, as such we need to remove some modules from
loading otherwise we see the following warnings:
[2013-08-13 17:21:49.911] WARNING[20375] loader.c: Error loading module
'app_setcallerid.so': /usr/lib64/asterisk/modules/app_setcallerid.so:
cannot open shared object file: No such file or directory
[2013-08-13 17:21:49.914] WARNING[20375] loader.c: Error loading module
'codec_speex.so': /usr/lib64/asterisk/modules/codec_speex.so: cannot
open shared object file: No such file or directory
[2013-08-13 17:21:49.916] WARNING[20375] loader.c: Error loading module
'format_sln16.so': /usr/lib64/asterisk/modules/format_sln16.so: cannot
open shared object file: No such file or directory
[2013-08-13 17:21:49.917] WARNING[20375] loader.c: Error loading module
'func_curl.so': /usr/lib64/asterisk/modules/func_curl.so: cannot open
shared object file: No such file or directory
[2013-08-13 17:21:49.920] WARNING[20375] loader.c: Error loading module
'func_speex.so': /usr/lib64/asterisk/modules/func_speex.so: cannot open
shared object file: No such file or directory
[2013-08-13 17:21:49.922] WARNING[20375] loader.c: Error loading module
'res_curl.so': /usr/lib64/asterisk/modules/res_curl.so: cannot open
shared object file: No such file or directory
Change-Id: I0e148d05b1d73967b335912ffa208670003b44c7
Signed-off-by: Paul Belanger <paul.belanger@polybeacon.com>
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
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
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