Configuration of codecs

I want to configure a g729 codec for Asterisk, this codec use two 10 ms samples in each packet for default, in this way more than half of bandwith consumption is only header, in some way this can be change?
I try to see in codec.conf but any example I find it’s for speex codec, this is the only configurable codec?

In order to use the g729 codec with Asterisk, you must purchase licenses.

You must license each simultaneous connection to Asterisk that you wish to support.

So, if you have two phones, and each might connect to Asterisk at the same time (say, to check voicemail) then you must buy 2 licenses.

Check here for more details:

store.digium.com/productview.php … egory_id=5

Yes, I know the licenses things. But i need something about the configuration, I have the licenses already. I read a lot of the codec and in any place this codec is very configurable and it’s very inefficient in a WAN link use the default configuration.
I want to know how can I configure the g729 codec?

Yes, I know the licenses things. But i need something about the configuration, I have the licenses already. I read a lot of the codec and in any place this codec is very configurable and it’s very inefficient in a WAN link use the default configuration.
I want to know how can I configure the g729 codec?

Try this downloads.digium.com/pub/telepho … 729/README

Thak for everything but my question is the following:
I’d like to set up the codecs.conf file to g.729 codec. Where could I set up 3 or more frames per TX in Asterisk PBX?

Hi rogergscuall,
I hope you’ve already got an answer for your question … but if not … here it goes …

Talking about asterisk v.1.4 you can set up the codec’s packetization globally or by peer/friend/user in the allow statement, like this …

allow=g729:40 ; example for 40ms per packet …

you can set up the min of 10 until the maximum of 230ms (should consider your RTP max MTU). The default value, if not specified is 20ms as you said.

I hope it helped.

regards,

It works !!!
Thank you my friend, this is what I wanna know!!!

Is there a way of configuring allowed codecs when connecting to the Voicemail?

Let’s say I do want to allow clients to use g729 when they call each other, however I only want to accept g711 when they ring (or redirected) to the Voicemail on the Asterisk.

How can I do that? Any ideas would be appreciated!

Thanks!!!