[asterisk-users] 0005162: RTP Packetization : Few questions

[Yusuf]

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


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