I am using FreePBX 184.108.40.206 with Asterisk 220.127.116.11. I am trying to get a conference number up and running. But when I have added a conference to FreePBX and I dial the conference number it hangs up immediatly.
I looked at the logfile and saw the following WARNING:
[2011-01-10 17:30:13] WARNING chan_sip.c: Retransmission timeout reached on transmission [email protected] for seqno 2 (Critical Response) – See doc/sip-retransmit.txt.
Packet timed out after 33600ms with no response
I did some looking on the internet and found that the module app_meetme.so is missing in the /usr/lib/asterisk/modules/.
Can somebody help to get this module installed? Don’t know how to start this.
Thanks to the post of tadpole in this thread: http://www.freepbx.org/support/documentation/faq/common-problems/invalid-conference. I came up with the following solution that makes my conference work:
As root on your Asterisk server:
tar zxvf dahdi-linux-complete-2.4.0+2.4.0.tar.gz
chkconfig dahdi on
wget wget http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.8-current.tar.gz
tar zxvf asterisk-1.8-current.tar.gz
cd asterisk-1.8.x.x (Replace X.X with the current version number)
contrib/scripts/get_mp3_source.sh (this may take a while)
Head into Channel Drivers and make sure chan_dahdi is selected
Head into Add-Ons and make sure app_mysql, app_saycountpl, cdr_mysql, format_mp3, res_config_mysql are selected
Head into Extra Sound Packages and make sure EXTRAS-SOUNDS-EN-GSM is selected