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?)
Oh, and using ztmonitor, it shows the zap channel receiving all sorts of
sound, but no transmit.
Asterisk 0.7.1 (Debian/Unstable package)
zaptel 0.1.6
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