Thank you..<br><br>Do you know why? and solution?<br><br>Thank you<br>Fadil<br><br><div><span class="gmail_quote">On 8/15/07, <b class="gmail_sendername">Sergey Okhapkin</b> <<a href="mailto:sos@sokhapkin.dyndns.org">sos@sokhapkin.dyndns.org
</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Seems like your asterisk crashes.<br><br>On Wednesday 15 August 2007 14:34, Fadil Sutomo wrote:
<br>> Hi there,<br>><br>> I am really interested in trying speex as my codec since I think it works<br>> best with my SIP clients (SIP Communicator).<br>><br>> And I want to increase the voice quality of it by using Preprocessor in
<br>> Speex. That is, by setting 'preprocess' field to 'true' in codecs.conf.<br>><br>> Yes, I installed the latest version of Speex in my computer. That is, when<br>> I type 'speexenc --version' in the terminal, it shows that I am using
<br>> version 1.2-beta2.<br>><br>> Here comes the problem.<br>><br>> After I set 'preprocess' field to 'true', and reload codec_speex.so,<br>> everytime I place a call, then I lost connection with Asterisk server as
<br>> the first sound gets into the phone, whether it be my voice, or the<br>> background voice (maybe the 'meow' of your cat, back there).<br>><br>> Any voice detected by the phone, Asterisk connection is gone! And I'll see
<br>> the disturbing message:<br>> "Disconnected from Asterisk server<br>> Executing last minute cleanups"<br>><br>> So, anyone knows what's the problem and workaround for this ?<br>><br>> Your help is appreciated. Thanks in advance.
<br>><br>> Fadil<br><br>_______________________________________________<br>--Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--">http://www.api-digital.com--</a><br><br>asterisk-dev mailing list
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