Now it works. Thanks a lot, guys. On 8/15/07, Donny Kavanagh <[EMAIL PROTECTED]> wrote: > > Fixes can no longer be applied to 1.2, try 1.4 and let us know what > happens. > > If it crashes with 1.4 then file a bug report. > > Donny > > On 8/15/07, Fadil Sutomo <[EMAIL PROTECTED]> wrote: > > I did that. > > Still disconnected from Asterisk. > > > > I also did make clean (just to be sure), but the result is still the > same. > > > > btw, I am using Asterisk-1.2.24. I don't know if this helps. > > > > Thank for your answer btw, > > Fadil > > > > > > On 8/15/07, Donny Kavanagh <[EMAIL PROTECTED]> wrote: > > > Try recompiling codec_speex.so. > > > > > > rm codecs/codec_speex.so > > > rm codecs/codec_speex.o > > > > > > from your source and then do a make install. > > > > > > On 8/15/07, Fadil Sutomo <[EMAIL PROTECTED] > wrote: > > > > Hi there, > > > > > > > > I am really interested in trying speex as my codec since I think it > > works > > > > best with my SIP clients (SIP Communicator). > > > > > > > > And I want to increase the voice quality of it by using > Preprocessor in > > > > Speex. That is, by setting 'preprocess' field to 'true' in > codecs.conf. > > > > > > > > Yes, I installed the latest version of Speex in my computer. That > is, > > when I > > > > type 'speexenc --version' in the terminal, it shows that I am using > > version > > > > 1.2-beta2. > > > > > > > > Here comes the problem. > > > > > > > > After I set 'preprocess' field to 'true', and reload codec_speex.so, > > > > everytime I place a call, then I lost connection with Asterisk > server as > > the > > > > first sound gets into the phone, whether it be my voice, or the > > background > > > > voice (maybe the 'meow' of your cat, back there). > > > > > > > > Any voice detected by the phone, Asterisk connection is gone! And > I'll > > see > > > > the disturbing message: > > > > "Disconnected from Asterisk server > > > > Executing last minute cleanups" > > > > > > > > So, anyone knows what's the problem and workaround for this ? > > > > > > > > Your help is appreciated. Thanks in advance. > > > > > > > > Fadil > > > > > > > > _______________________________________________ > > > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > > > > > > > asterisk-dev mailing list > > > > To UNSUBSCRIBE or update options visit: > > > > > > http://lists.digium.com/mailman/listinfo/asterisk-dev > > > > > > > > > > _______________________________________________ > > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > > > > > asterisk-dev mailing list > > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-dev > > > > > > > > > _______________________________________________ > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > > > asterisk-dev mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-dev > > > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev >
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