Could anyone who has successfully configured Asterisk to use g729 to conference 10-20 people please post their configs. I purchased and successfully installed 2 g729 licenses and but when I dial into my conference number on the Asterisk box from a SPA-2000 set to allow all codecs, it always appears to connect using ULAW. My iax.conf file includes the following under the general section disallow=all bandwidth=low allow=g729 allow=ulaw Thanks, Roger Easlick -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040905/f757cb65/attachment.htm
Roger, I haven't had any problems doing confs w/ g729. My guess is the Sipura is asking for ulaw first. Try adjusting the codec priority on the sipura side. IF you still have problems I an get my spa-3000 out and trying and solve it for you. -- William ----- Original Message ----- From: box100 <box100@easlick.com> Date: Mon, 6 Sep 2004 00:28:50 -0400 Subject: [Asterisk-Users] Asterisk Conferencing using g729 To: asterisk-users@lists.digium.com Could anyone who has successfully configured Asterisk to use g729 to conference 10-20 people please post their configs. I purchased and successfully installed 2 g729 licenses and but when I dial into my conference number on the Asterisk box from a SPA-2000 set to allow all codecs, it always appears to connect using ULAW. My iax.conf file includes the following under the general section disallow=all bandwidth=low allow=g729 allow=ulaw Thanks, Roger Easlick _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Thanks for the quick response, William. Yes, I was able to force it by removing ULAW from the sip.conf but I needed that in there. If it is there, then the Sipura as well as X-PRO use ulaw. I didn't really want to adjust the clients because I am planning on having random clients accessing the server and don't want to instruct each one to change whatever hard or softphone they may have over to g729. I wanted that process transparent to the user. I also seem to need more than 2 licenses although the Sipura support g729, so I wouldn't think I would also need a license on the Asterisk box for each one. It looks like the choices are either use a dedicated box only for conferencing and eliminate the possiblity of anyone using ulaw, or just getting more bandwidth so that there is enough for as the 20 clients can connect. Just seems like a lot of bandwidth to have, that's all. The ideal situation would be the ability to only use g729 for the conference rooms and ulaw for everything else, but the bandwidth=low setting doesn't seem to work if ulaw is allowed. Thanks again. ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of William Suffill Sent: Mon 9/6/2004 00:35 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Conferencing using g729 Roger, I haven't had any problems doing confs w/ g729. My guess is the Sipura is asking for ulaw first. Try adjusting the codec priority on the sipura side. IF you still have problems I an get my spa-3000 out and trying and solve it for you. -- William ----- Original Message ----- From: box100 <box100@easlick.com> Date: Mon, 6 Sep 2004 00:28:50 -0400 Subject: [Asterisk-Users] Asterisk Conferencing using g729 To: asterisk-users@lists.digium.com Could anyone who has successfully configured Asterisk to use g729 to conference 10-20 people please post their configs. I purchased and successfully installed 2 g729 licenses and but when I dial into my conference number on the Asterisk box from a SPA-2000 set to allow all codecs, it always appears to connect using ULAW. My iax.conf file includes the following under the general section disallow=all bandwidth=low allow=g729 allow=ulaw Thanks, Roger Easlick _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 5846 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040906/aff7487c/attachment.bin
box100 wrote:> My iax.conf file includes the following under the general sectionA SIPURA is a SIP device, configure the codecs under sip.conf not IAX.conf.> disallow=all > bandwidth=low > allow=g729 > allow=ulaw > > Thanks, > Roger Easlick > > > ------------------------------------------------------------------------ > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-- Daniel Jimenez <djimenez[at]pobox[dot]com>
Yes, thanks for the comment, and I did configure the sip.conf for the Sipura -- sorry for the confusion. The reference to iax.conf is because I am running FWD through IAX so I would need to configure iax for those connecting through FWD, wouldn't I? Roger ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of William Suffill Sent: Mon 9/6/2004 15:52 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Conferencing using g729 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. On Mon, 06 Sep 2004 04:51:26 -0500, Daniel Jimenez <djimenez@pobox.com> wrote:> > > box100 wrote: > > > My iax.conf file includes the following under the general section > > A SIPURA is a SIP device, configure the codecs under sip.conf not IAX.conf. > > > disallow=all > > bandwidth=low > > allow=g729 > > allow=ulaw > > > > Thanks, > > Roger Easlick > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > Daniel Jimenez <djimenez[at]pobox[dot]com> > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >_______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 5350 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040906/45e8d35f/attachment.bin
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@lists.digium.com on behalf of Tony Mountifield
Sent: Mon 9/6/2004 16:15
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: Asterisk Conferencing using g729
In article <6b65470d040906125254620f2e@mail.gmail.com>,
William Suffill <william.suffill@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@softins.co.uk - http://www.softins.co.uk
Play: tony@mountifield.org - http://tony.mountifield.org
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