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<TITLE>X-Lite, X100P, and Speex</TITLE>
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<P><FONT SIZE=2>I'm having a problem with using X-Lite to initiate a call via Asterisk out an X100P analog port, using the Speex codec. I've put in the registry fix for X-Lite and Speex so that works OK, and calling the echo test extension works. However, if I call out the analog port it appears that audio being initiated by X-Lite is being dropped, but audio being initiated from the analog line is being encoded and heard OK on X-Lite. /var/log/asterisk/messages keeps repeating "WARNING[XXXX]: Frame too large" and "WARNING[XXXX]: Out of buffer space" over and over again. Any ideas on what's wrong? (and if it's simply that one cannot use the speex codec with outbound calls, how would one configure asterisk to allow speex when it's a SIP to SIP call, but G.711 if it's a SIP to Analog call?)</FONT></P>
<P><FONT SIZE=2>Oh, and using ztmonitor, it shows the zap channel receiving all sorts of sound, but no transmit.</FONT>
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<P><FONT SIZE=2>Asterisk 0.7.1 (Debian/Unstable package)</FONT>
<BR><FONT SIZE=2>zaptel 0.1.6</FONT>
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