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?



_______________________________________________
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to