[Asterisk-Users] Re: Asterisk Conferencing using g729

[box100]

Thanks, Tony, you answered a question about g729 licencing and * conferences that I wanted ask. Very enlightening. I was wondering about that because it seemed to be using a license for each connections, despite the Sipura natively supporting g729, but I wasn't sure that that is the way it has to be.
 
Another related question: Is there a way to just use g729 for the conference and for nothing else. The problem I have is that I have Broadvoice ( BV rocks, by the way) which requires ULAW and sends DTMF inband. If I allow g729 in the sip.conf, Asterisk complains that inband dtmf is only supported under ULAW and incoming dtmf does not work through Asterisk, something I must have.
 
Well I have partially solved the problem in the paragraph above. It appears that if I leave g729 out of the general section of sip.conf, but add allow=g729 to my SPA-2000 device section in the sip.conf file I still get BV incoming dtmf to work and I get the SPA-2000 to use g729. If, however. I add allow=ulaw to the SPA-2000 section, it uses ulaw even though I am using setvar=g729 right before the redirect to the conference room as below:
 
exten => 3001,1,setvar(SIP_CODEC=g729)
exten => 3001,2,Meetme,1000,Maps

Asterisk does say that the codec is being changed to g729 but the SIP  SHOW CHANNELS command tells me the channel is using only ULAW. The real interest in g729 is saving OUTBOUND bandwidth and thus forcing g729 to be used with any device outside the firewall/router. At the same time I need ulaw. I would think that is what the setvar command as used above is for but it doesn't seem to have any effect. How do I force anyone from the outside to use only g729 to connect to my conferences but allow them to access the internal extensions using ulaw if they have it available?
 
Here and excerpt from my log:
Sep  7 00:03:13 NOTICE[-174232656]: chan_sip.c:1834 sip_answer: Changing codec to 'g729' for this call because of ${SIP_CODEC) variable
sip show channels
Peer             User/ANR    Call ID      Seq (Tx/Rx)   Format
192.168.1.5      2201        91e0706e-51  00101/00102   ULAW

Thanks,
Roger

________________________________

From: asterisk-users-bounces at lists.digium.com on behalf of Tony Mountifield
Sent: Mon 9/6/2004 16:15
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Re: Asterisk Conferencing using g729



In article <6b65470d040906125254620f2e at mail.gmail.com>,
William Suffill <william.suffill at gmail.com> wrote:
> Good call Daniel I didn't even notice that.
>
> As far as number of license it really depends on how many concurrent
> calls you will be doing and if asterisk needs to transcode at all. If
> you call from g729 device to g729 you are fine but g729 to vm would be
> 1 license etc.

And if you are conferencing, you need one G.729 licence for each
conference participant, because Asterisk can't mix G.729 natively,
so it transcodes each channel from G.729 to Signed Linear.

Cheers
Tony
--
Tony Mountifield
Work: tony at softins.co.uk - http://www.softins.co.uk
Play: tony at mountifield.org - http://tony.mountifield.org
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