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