[asterisk-dev] Speex preprocessor and disconnection from Asterisk

Fadil Sutomo fsutomo at gmail.com
Thu Aug 16 11:19:18 CDT 2007


Now it works.

Thanks a lot, guys.
On 8/15/07, Donny Kavanagh <donnyk at gmail.com> 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 <fsutomo at gmail.com> 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 <donnyk at gmail.com> 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 <fsutomo at gmail.com > 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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-dev/attachments/20070816/cd1bb1c2/attachment-0001.htm 


More information about the asterisk-dev mailing list