[asterisk-users] Question on G.729 (was: H.264 *Not Patented*)

[Andy Davidson]

Hi,

I asked some questions here about G.729 earlier in the week, and it  
looks like it would fit the bill for compressing audio between my *  
server in colocation and sip phone at home.

This is what I want my setup to look like.
(Wont make sense unless you are using a fixed width font)


     [my phone]              [asterisk]       [third parties]
     Snom 360    <----------> v 1.4 <-------------> ???
                      SIP               IAX/SIP
                      G.729             Don't care (probably something
                                        other than G.729, my preferred
                                        supplier today likes ulaw and  
alaw)

My phone sees the * box over a relatively slow consumer connectivity  
link.  The * box is colocated and has excellent connectivity.   
Therefore the tighter compression between * and my phone is  
important, hence why I want to use g.729 here.

The config for my phone, and my preferred voice supplier looks like  
this :

[[[from sip.conf]]]
[andydesk]
type=friend
context=default
secret=xxx
host=dynamic
dtmfmode=rfc2833
username=andydesk
mailbox=1001
vmexten=500
disallow=all
allow=g729
allow=alaw
allow=ulaw
allow=gsm
regexten=1001
allowreinvite=no

[[[from iax.conf]]]
[thing]
type=friend
host=dynamic
username=thing
secret=xxx
trunking=off
bridging=on
context=thing
disallow=all
allow=ulaw
allow=alaw
allow=gsm



When I place a call, the other party's line rings as normal.  When  
the other party answers, I get a sip 'denied' packet, and the call is  
aborted.  Asterisk says :  No path to translate from SIP/mydeskphone  
to IAX/myprovider and   Had to drop call because I couldn't make SIP/ 
mydeskphone ompatible with IAX/myprovider.

This looks similar to this bug :
   http://bugs.digium.com/view.php?id=8781&nbn=4

What I would expect to happen, is that Asterisk would transcode  
between the ulaw/alaw party, and me, wanting to listen via g729.  Is  
this what *should* happen ?  Worth noting that my provider does not  
support G.729.  Is what is happening a bug ?  Any patches I can try  
to see if they work ?  Or is it my config which is broken ?

Inbound calls work ok, I guess this is because they are presented as  
alaw and asterisk is just passing them through (which of course isn't  
what i really want).

Thanks for any suggestions,
Andy