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

[Dan Austin]

> 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