> I ahve been using the RTP packetization patch for a while, and > its going great. I have a few questions: That is excellent.
> I always get this message: > 2006-08-31 22:11:22 WARNING[1278]: frame.c:1072 > ast_codec_pref_getsize: Framing not set for codec alaw, using > default 20 Not so excellent. > even though I set in sip.conf > [general] > context=default ; Default context for incoming calls > disallow=all ; First disallow all codecs > allow=ulaw:20 > allow=alaw:20 > allow=g729:80 > autoframing=yes > am I doing something wrong? That looks fine. Does it work with: allow:ulaw:20,alaw:20,g729:80 ? > Also, I am not sure if this is a bug. > If in sip.conf, if I set > [yusuf] > username=yusuf > secret=yusuf > type=friend > callerid=1002 > nat=yes > canreinvite=no > allow=all > host=dynamic > context=sip BUG! Which version of the patch and what SVN version? I suspect it has to do with one or more of the codecs that we could not find framing/packetization details about. Is alaw the codec used in the call that causes the crash? > then when asterisk calls, it says I have not set Framing (like above msg), > then asterisk just dies. > If I chane the line > allow=all to allow=alaw:20 > then its fine, and asterisk does not die. > Dont know if this is a bug, so I wont post debug/full messages now. Dan _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users