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

[Walter Klomp]

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.

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

Thanks
Walter