FAD Fedora Talk 2009 game plan: Difference between revisions

Revision as of 20:45, 18 October 2009

Page description
This page is to lay out and plan the specifics the work we will complete at the FAD. The FAD Fedora Talk 2009 page explains the original goals and planning done for this event.

Participants

Travel days
Travel days are Thursday and Sunday. On-site participants are leaving early and no on-site hacking will take place on Sunday.

Name	Remote?	Fri	Sat	Sun	Core Tasks
John		X	X	Travel	Project Management + Documentation + Requirements
Paul		X	X		Hosting, chauffeuring, hacking when no one's looking
Ian		X	X	Travel
Jon		X	X	Travel	Testing, puppetizing, general hacking, learn Asterisk, moral support, etc :D
Jared		X	X		Be the Asterisk guru + Bring lots of VoIP equipment + Documentation
Jeff		X	X	Travel	Asterisk hacking
Mike	X				puppetizing and testing
Clint	X	X	X	X	puppetizing and testing (shadowing Mike)
Darren	X		X		testing
Bruno	X	X	X	AM	testing, possibly some asterisk config hacking

Asterisk Packages & Installation Instructions

I still need to test sip connections through the server, but I now have sip registration to the server working.

Base system should be F12
wget http://kojipkgs.fedoraproject.org/packages/asterisk/1.6.2.0/0.6.rc3.fc13/i686/asterisk-1.6.2.0-0.6.rc3.fc13.i686.rpm for i686 systems.
yum install --nogpgcheck asterisk-1.6.2.0-0.6.rc3.fc13.i686.rpm asterisk-sounds-core-en-wav
- This will pull in some other packages, including tex stuff (presumably for documentation).
- I think we can use the F12 asterisk-sounds-core until there is an F13 version available, but I haven't got that far in my testing.
- I am not sure which codecs we need and what issues installing multiple languages causes.
git clone git://fedorapeople.org/~jcollie/ftalk-asterisk-configs.git ftalk-asterisk-configs
cp -f ftalk-asterisk-configs/* /etc/asterisk
chown asterisk.asterisk /etc/asterisk/*
- There are a couple of config files in Jeff's set that are new and get owned by root instead of asterisk. Depending on your umask setting this might be a problem.
A line "type = peer" should get added into the [general] section of /etc/asterisk/users.conf . Without this registration as one of the users won't work.
If you want to test connecting with a sip client you can edit /etc/asterisk/users.conf to replace the modified md5secret with a correct md5secret or just secret.
/etc/asterisk/sip.conf has a fixed ip address in several commands. If you comment them out (with ;) asterisk will listen on all interfaces (0.0.0.0).
/etc/asterisk/asterisk.conf has a [directories] section that causes problems for sip. Probably something needed by sip is in a missing directory. Since we are keeping stuff in the standard places, this can just be commented out by adding (!) to the end of the line. So it should look like: [directories](!)
Open ports 5060/udp 5060/tcp for inbound SIP. Allowing related connections should take care of RTP.
- The way things are configured, only udp is being used for 5060. Unless that changes 5060/tcp doesn't need to be opened up.
- For example if you aren't using a firewall tool you can add the following to /etc/sysconfig/iptables and service iptables restart:
  - -A INPUT -m state --state NEW -m tcp -p tcp --dport 5060 -j ACCEPT
  - -A INPUT -m state --state NEW -m udp -p udp --dport 5060 -j ACCEPT
Open port 4569/udp for iax2, if we play with that.
chkconfig asterisk on
service asterisk restart
asterisk -r
- If you want to do some debugging you can run asterisk -r as root or asterisk.
- "core show help" will show you what commands are available.
- "sip set debug on" is one that might be particularly useful.

Targeted Trac Tickets

Primary targets:
- #309 -- Asterisk recording
- #395 -- Audio streaming of Fedora Board conf-calls
- #453 -- Asterisk/gstreamer solution for town hall meetings
- #1160 -- streaming and recording moderated conf-calls
Secondary targets (as time allows):
- #672 -- improve VoIP page on FAS
- #1723 -- improve talk.fp.o page content

Use Cases

Explanations and proposed usage of different FedoraTalk functionality

@@ Line 31: / Line 31: @@
 == Asterisk Packages & Installation Instructions ==
-I still need to do more testing, but the service is up and running. I am having trouble setting up registrations to it. I suspect this is because things are set up for a different domain than that that matches the IP address, but it might be something else.
+I still need to test sip connections through the server, but I now have sip registration to the server working.
 * Base system should be F12
 * wget http://kojipkgs.fedoraproject.org/packages/asterisk/1.6.2.0/0.6.rc3.fc13/i686/asterisk-1.6.2.0-0.6.rc3.fc13.i686.rpm for i686 systems.
@@ Line 42: / Line 42: @@
 * chown asterisk.asterisk /etc/asterisk/*
 ** There are a couple of config files in Jeff's set that are new and get owned by root instead of asterisk. Depending on your umask setting this might be a problem.
+* A line "type = peer" should get added into the [general] section of /etc/asterisk/users.conf . Without this registration as one of the users won't work.
+* If you want to test connecting with a sip client you can edit /etc/asterisk/users.conf to replace the modified md5secret with a correct md5secret or just secret.
+* /etc/asterisk/sip.conf has a fixed ip address in several commands. If you comment them out (with ;) asterisk will listen on all interfaces (0.0.0.0).
+* /etc/asterisk/asterisk.conf has a [directories] section that causes problems for sip. Probably something needed by sip is in a missing directory. Since we are keeping stuff in the standard places, this can just be commented out by adding (!) to the end of the line. So it should look like: [directories](!)
 * Open ports 5060/udp 5060/tcp for inbound SIP. Allowing related connections should take care of RTP.
 ** The way things are configured, only udp is being used for 5060. Unless that changes 5060/tcp doesn't need to be opened up.
@@ Line 48: / Line 52: @@
 *** -A INPUT -m state --state NEW -m udp -p udp --dport 5060 -j ACCEPT
 * Open port 4569/udp for iax2, if we play with that.
-* If you want to test connecting with a sip client you can edit /etc/asterisk/users.conf to replace the modified md5secret with a correct md5secret or just secret.
-* /etc/asterisk/sip.conf has a fixed ip address in several commands. If you comment them out (with ;) asterisk will listen on all interfaces (0.0.0.0).
-* /etc/asterisk/asterisk.conf has a [directories] section that causes problems for sip. Probably something needed by sip is in a missing directory. Since we are keeping stuff in the standard places, this can just be commented out by adding (!) to the end of the line. So it should look like: [directories](!)
 * chkconfig asterisk on
 * service asterisk restart

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