How do I get setup for g729 encoding with the FreePBX iso?
So far I’ve gathered I buy the licenses from digium, and then follow the install intructions, and then change ulaw to g729&ulaw inside freepbx.
Does that sound correct? Anything I’m missing?
Any tricks I should know when doing this?
Its a little more involved, but you have the basic gist of it.
Once you buy the licenses, you download an activation utility that marries your server with your license and authorizes the asterisk module to run. You are strongly encouraged to follow the instructions after installing the license to backup your new license for disaster purposes.
Then you download a testing utility from Digium that tries several builds of the module against your server’s architecture to find the optimal build that compresses/decompresses the fastest against your processor/kernel/architecture combination.
Then you download and copy the modules to the asterisk module folder, then restart asterisk, and then, you can rearrange your codec priorities in the “allow” section for trunks and extensions to indicate G729 as the first priority if you wish.
Once a SIP call is established, you can issue a “sip show channels” at the Asterisk CLI and see the codecs between SIP devices. You will even see if Asterisk is transcoding between g729 and others like ulaw, should a friend/peer device not support g729. Kinda cool.
Finally from the Asterisk command line, “core show translation recalc 10” will show you the amount of time in microseconds a piece of voip would take to transcode between codecs. (in a cool looking chart)
Awesome! Thanks for the tips.