[Asterisk-Users] Tonight's CVS breaks Grandstream phone

[David Hindmarsh]

Hi Guys,

Tried the disallow=all and allow=all but still getting one way audio with
x-lite and messenger.

Any update on this problem.

Dave
----- Original Message -----
From: "John Todd" <jtodd at loligo.com>
To: <asterisk-users at lists.digium.com>
Sent: Friday, October 24, 2003 1:49 PM
Subject: Re: [Asterisk-Users] Tonight's CVS breaks Grandstream phone


> >FYI.  Haven't dug enough to be able to report any more, but
> >re-fetched CVS to verify that sometime in the last few days CVS
> >changes now break my GS phone.
> >
> >It appears to be at the RTP level.  It seems to set the call up just
> >fine, but no audio is passed back to the instrument.
> >
> >I reverted, and will try to play with this tomorrow unless someone
> >else tells us it's fixed.
> >
> >Thx.
> >
> >B.
>
> I am seeing the same error with CVS as of 02:00 today GMT.
> Grandstream phones will dial, the dialplan will seem to work, but
> after a few seconds the call fails.  Looking at the SIP debug, I see
> that
>
> There was a new feature added last night to allow for codec
> permission/denial on a per-peer basis in sip.conf.  This means that
> each SIP client can be forced to use particular codecs (at least,
> that is the intent.  more testing, anyone?)
>
> So, it seems that the Grandstreams do not elegantly handle some
> circumstances of codec presentation which were created by these new
> patches.  It is necessary for you to put the following lines in each
> Grandstream entry in your sip.conf, OR you can put the identical
> entries in [general] to have it work across all clients.  Note that
> both lines are required:
>
> disallow=all
> allow=all
>
>
> JT
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