Here you can find a port for Asterisk trunk (rev. 377802) of this patch: http://lists.digium.com/pipermail/asterisk-dev/2011-September/051262.html.
This patch permits to set the codec of an outgoing multicast RTP stream via the variable MULTICAST_RTP_CODEC:

exten => 999,1,Answer()  
exten => 999,n,Set(MULTICAST_RTP_CODEC=alaw ...

Starting from Asterisk 1.8 you can send multicast rtp streams using the MulticastRTP channel driver. There is an open issue that breaks outgoing RTP if the source channel doesn't contains timing informations (Eg. playing an audio file with Dial(MulticastRTP/basic/239.255.255.245:5555,,A(my-announce ...

This little piece of extensions.conf implements a callback from a failed transfer (attendant and unattendant/bind)

exten => _1XX,1,Dial(SIP/${EXTEN},20,tT)
exten => _1XX,n,GotoIf($[ "a${BLINDTRANSFER}" = "a" ]?TransferFailed)
exten => _1XX,n,Set(CALLER=${CUT(BLINDTRANSFER,-,1)});
exten => _1XX,n,Goto(CallBack)
exten => _1XX,n(TransferFailed ...

Using directmedia=update or (canreinvite=update in old-sytle configuration) in chan_sip.conf Asterisk sends SIP UPDATE request to refresh SIP session.

If this UPDATE requests refers to an inactive call the phone reply with SIP/2.0 481 Call Leg/Transaction Does Not Exist.

In this situation Asterisk continue in ...

Asterisk use peername as username during SIP inbound digest authentication.
This patch add authuser parameter in SIP peer definition and use authuser in digest authentication:

Peer definition in /etc/asterisk/sip.conf

...
[pietro](sip-client-base)
authuser=MyUsername
secret=XXXXX
qualify=yes
nat=yes
...

Peer definition in my SIP client (Twinkle)

REGISTER ...