From Fedora Project Wiki


Name Remote? Fri Sat Sun Core Tasks
Alex Oviedo X X Travel Discussion leader
Bernardo Hermitaño X X Travel Discussion leader


Thursday, 2015-01-17
Time (EDT) Details
1500 Local team picks up Alex Oviedo from bus station
1515 Local team pick up Bernardoha from Juliaca airport
Saturday, 2009-10-24
Sunday, 2009-10-25

Use Cases

Explanations and proposed usage of different FedoraTalk functionality

Asterisk Packages & Installation Instructions

I think this is ready for review by the gurus as I have gone about as far as I can. I am able to complete twinkle to twinkle calls through the server.

  • Base system should be F12
  • wget for i686 systems.
  • yum install --nogpgcheck asterisk- 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:// 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 (
  • /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.
  • Note with this setup authentication (domain and realm in twinkle) uses but for the server setup and host being called you want to use the real domain name of the server.
  • If you edit system settings -> Network you can change the sip port and run an instance of twinkle on the same box as the server.

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):