FAD Fedora Talk 2009 IRC meeting, 2009-10-06 UTC 1900

[Bruno Wolff III]

On Mon, Oct 12, 2009 at 10:56:10 -0700,
  John Poelstra <poelstra at redhat.com> wrote:
> Bruno Wolff III said the following on 10/11/2009 05:16 PM Pacific Time:
> >I was able to get the F13 version (asterisk-1.6.2.0-0.6.rc3.fc13.i686) of
> >asterisk working on F12. I was also able to get dahdi-linux working using
> >the trunk from Digium (some kernel build opts changed since 2.6.30 which
> >break 2.2.0.2) and a spec file from messinet.com with very minimal changes.
> >
> 
> Thanks for doing this!
> 
> Can you put the steps you took, etc. on a wiki page so the rest of
> us who haven't done this before don't have to start from scratch?

I think the process for that machine was specific to me. I had been running
asterisk to allow phones to be used within our house. We don't have a land
line any more and it's really convenient for my wife to pick up a phone
and have the one by my desk downstairs ring. I am also playing with sip
connections to fedora talk. I needed dahdi-linux to make the phones work.
Most of you (excepting Jeff) probably don't have the hardware and don't
want to play with dahdi-linux. The sip configuration is also specific to
me and needs some changes to fix some problems with the way it is working.
Hopefully I'll get that done and update my documentation on using asterisk
with Fedora Talk to include what I learn. (See
https://fedoraproject.org/wiki/User:Bruno/Using_Asterisk_with_Fedora_Talk
if you are curious.)

When I get the information about the configuration I'd like to replicate
that on another box I have available. I think it would be worthwhile to
document that process assuming I get the config information enough in
advance.

A brief summary of what is mostly needed is to install the asterisk 1.6.2
from F13 on an otherwise F12 system. You only need the plain asterisk rpm.
There are a bunch of extensions available as related packages, but none of
these is required for a basic asterisk setup. Asterisk is controlled by
initscripts so you use chkconfig to set which run levels you want it started
in. You can also use the service command to start/restart it for testing.
Warning if you run the asterisk binary as root some log files may get owned
as root which will cause you grief. Running it through the service command
does the right thing. You can run asterisk -r to connect to a running
asterisk instance and getting live status information. Inbound sip connections
are on port 5060. Some related data uses port 8000. I am not sure if there
is an iptables module that tracks that a data connection to port 8000 is
related to a connection on 5060. (5061 is used for secure SIP connections,
but I haven't been playing with that yet.) I just opened both ports for
my testing. The only config files that were necessary for me to get it
work were:
asterisk.conf
dahdi_channels.conf
indications.conf
modules.conf
sip.conf
chan_dahdi.conf
extensions.conf

And you shouldn't need either of dahdi_channels.conf or chan_dahdi.conf.