[asterisk-users] g729 and polycoms problem

Alyed Tzompa alyed.tzompa at simitel.com
Thu Sep 21 14:12:50 MST 2006


		Sorry but I've ran out of ideas...

Anyone else out there with a successful Polycom g729 pass through-only experience?

Alyed 

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didn't work :(

Regards,
Santiago

On 9/20/06, Alyed Tzompa wrote:
> Not an expert at reading Polycom config files, but guess g729 and ulaw are
> both preference 1 isn't it?
>
> hey... you have in your sip.conf configuration "canreinvite=no"... think
> this may be a problem: since Asterisk will always stay in the path of the
> RTPs, I think it might need to have the proper transcoder, as it does not,
> then the error arises... at least that's what I think :)
>
> set "canreinvite=yes" (or just comment it since that's the default) on both
> parties and try again.
>
> Let me know if it works.
>
> Alyed
>
> ________________________________
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> Sep 20 12:38:41 2006
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>
> Still having the same problem. i modified the sip.cfg in order to make
> g729 the first choice:
>
>
>
> voice.codecPref.G711A="3" voice.codecPref.G729AB="1"
> voice.codecPref.IP_4000.G711Mu="1" voice.codecPref.IP_4000.G711A="2"
> voice.codecPref.IP_4000.G729AB=""/>
>
>
> Cheers,
> Santiago
>
> On 9/19/06, Alyed Tzompa wrote:
> > Make sure the codec used by the Polycom will be only g729 via the phone's
> > web interface, as far as I remember Polycom will try always to use ulaw or
> > alaw first unless it is configured to use only or as first choice the g729
> > codec.
> >
> > Alyed
> >
> > ________________________________
> > Return-Path: Tue
>
> > Sep 19 14:47:54 2006
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> > by lists.digium.com (Postfix) with ESMTP id AB0F03C1F4;
> >
> > Hi, I'm experiencing some problems with polycom phones, asterisk and g729
> > codec.
> >
> > As I understand, between polycom and polycom i can use g729 without
> > license at all as long as I'm using codec_g729.so module (i'm using
> > the Open Source Implementation (
> >
> http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
> > )
> > because it's pure pass-thru and there's no transcoding).
> >
> > My sip.conf has the following options:
> >
> > [general]
> > disallow=all
> > allow=g729
> > allow=ulaw
> >
> >
> > [voipuser]
> > type=friend
> > username=user
> > host=dynamic
> > callerid=user <202>
> > mailbox=202 at default
> > secret=gbvVf423
> > canreinvite=no
> > insecure=yes
> > disallow=all
> > allow=g729
> >
> >
> > so i force the voipuser to use g729 as main codec. The problem comes
> > when i try to connect to other polycom phone with the same config as
> > voipuser. The CLI shows the following:
> >
> > Sep 19 18:37:38 NOTICE[8226]: chan_sip.c:3691 process_sdp: No compatible
> > codecs!
> >
> > show modules doesnt show codec_g729.so but if i try to load it i get this:
> >
> > Unable to load module codec_g729.so
> > Sep 19 18:39:16 WARNING[10688]: loader.c:305 __load_resource: Module
> > 'codec_g729.so' already exists
> >
> >
> > Anyone had this issue?
> >
> > If you need more information, feel fre to ask for it :)
> >
> >
> > Thanks a lot!
> >
> > Santiago
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