This is needed because the centos packages install the sound files in a
different directory then the asterisk defaults too.
Change-Id: I97bd50439fad97906b3ee77a21672cc1114088c2
Signed-off-by: Paul Belanger <paul.belanger@polybeacon.com>
conf_bridge is needed for app_confbridge to be able to play sounds into
an active conference bridge.
Change-Id: I82aa5962c6e5f21909d7f30334d01a2ee711d02f
This was a missing (+) from the context header, this failed to properly
load the modules listed in modules.conf.
Change-Id: Ic49216f9e059267ce5a13a8304bc703720e128b8
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>
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
Add a wait of 1 second after answering and before playing a prompt.
Sometimes it takes a little bit of time to set up the audio path.
If you don't wait a bit, the caller will miss the beginning of the
prompt.
Change-Id: Ied25fd6d90638d938b0cd04562ad15de1e7d0426
This patch updates the default extension. This is what a caller would
hit if they called sip:openstack.org (as opposed to
sip:6000@openstack.org), or when they are calling in via the PSTN. It
asks them to enter a conference number and then hit #. It will give up
after 3 tries and just hang up.
I moved the 'spam' playback that I originally had to only play if you
enter the digits for spam (7726).
Change-Id: Ia82b9b52766ee191e3fd6f20d4e3b7fbb56f1f5b
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
