[Asterisk-Users] Speex codec problem (unresolved ?)

[David Uzzell]

Walter Klomp wrote:
> Hi,
> 
> I'm sorry to bring this up again, but I have been googling forever and
> whatever solutions are offered don't work for me.
> 
> I am using x-lite (the latest build) and trying to use Speex.
> 
> When I do call from the x-lite to another SIP phone or PSTN (through Cisco
> gateway) My asterisk fills up with this message:
> WARNING[1007]: codec_speex.c:196 speextolin_framein: Out of buffer space
> 
> The x-lite client can hear the remote end (SIP or PSTN call) quite clearly,
> but what comes from the X-Lite is completely garbled and mixed with DTMF
> tones.
> 
> I had tried the registry fix (which only changes the magic number from 97 to
> 110 and apparently didn't do anything else), didn't work.
> 
> After looking at the source I had also tried to increase the buffer size
> from 8000 to 16000, but that made other codecs (like lin_to_g729) choke, and
> I still had the problem...
> 
> I like speex and would like to use it (as I find ilbc a bit too scratchy)
> 
> I am running Asterisk CVS-HEAD-11/16/04-17:19:53 and speex-1.0.4 libraries
> on Gentoo Linux.

The best sugestion that I can offer is that I saw the same problem and 
could not resolve it but after upgrading * to CVS after the 12/10 it 
went away. Never did find a solution and gave up looking as it solved it.

It also fixed some SIP issues I had and they went away aswell.

Sorry that might not be the answer you are looking for but thats what 
worked for me.

David

> 
> Can anybody help me further on how to resolve this problem ?
> 
> Thanks
> Walter
> 
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