[Asterisk-Users] speex codec problem

[John Chester]

I have Asterisk 0.7.2 running on a RedHat 8.0 box.  Before installing 
Asterisk, I installed libogg-1.1 and speex-1.0.3.  speexenc and speexdec 
work OK from the command line.  I see in Asterisk's startup messages that 
it's registered the translators lintospeex and speextolin.

I'm using a mixture of hardware phones and Xten softphones.  A call from 
one Xten phone to another using speex works OK, but in that case Asterisk 
is not handling the RTP data.  (I did have to change the speex magic number 
to 110 in the Xten phone.)

A call from a hardware phone using ulaw to an Xten phone using gsm works 
OK; in that case, Asterisk is doing the transcoding.

A call from a hardware phone using ulaw to an Xten phone using speex 
fails.  When the Xten phone answers the call, Asterisk produces an endless 
stream of error messages:
WARNING[311313]: codec_speex.c:167 speextolin_framein: Out of buffer space
This continues until I shut Asterisk down.

Has anyone else out there run into this problem?   Is anyone successfully 
using speex transcoding?  If so, what versions of Asterisk and speex are 
you using?