asterisk users - Sep 2003 - Possible FAQ: IAX2 -> SIP with G729 and no licence

Assuming I've got a setup where calls entering Asterisk on SIP leave on IAX2
( and the reverse), i.e. a SIP user might dial '1234' where we then have

extern => 1234,1,Dial(IAX2/somewhereelse)

Now, we don't have any G.729 functionality on this server, so what happens
if the SIP user calls with G.729 only available?

Assuming the remote IAX2 server does have G.729 can it be passed through to
it?

Linus

> Assuming I've got a setup where calls entering Asterisk on
> SIP leave on IAX2
> ( and the reverse), i.e. a SIP user might dial '1234' where
> we then have
>
> extern => 1234,1,Dial(IAX2/somewhereelse)
>
> Now, we don't have any G.729 functionality on this server, so
> what happens
> if the SIP user calls with G.729 only available?
>
> Assuming the remote IAX2 server does have G.729 can it be
> passed through to
> it?
>
Linus,

Theoretically (in network terms), there shouldn't be an issue as G.729 is a
codec, whereas the process you are referring to describes
"transporting" the
codecs from A to B. The transporting is handled by the transport protocols
(SIP,IAX2,etc).

Whether this theory applies to Asterisk or not - I don't know. My current
understanding is that Asterisk acts like a router in a sense, transmitting
packets along channels to the client which in turn reads the audio stream
using the codec selected. So unless Asterisk performs some other tasks with
the codecs your suggestion should work fine.

Dave