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