Hi, I ahve been using the RTP packetization patch for a while, and its going great. I have a few questions:
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 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? 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 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. thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. _______________________________________________ --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