[Asterisk-Users] Passthrough and reInvite

[Tom Samplonius]

  It is not clear how exactly g729 pass-through can be enabled.  I
have a SIP call off a gateway come into an Asterisk menu, and then I
send the SIP call to another SIP gateway using Dial().   Even though
codec preferences have g729 listed first, it never gets used.

  Both gateways have separate peer entries in sip.conf, and both have
canreinvite=yes set.  Can Asterisk change the media type during a
re-Invite?  The call is answered as g711u initially, and then Asterisk
plays a menu, and then does a Dial().  I can see Asterisk doing the
reInvite, but the protocol stays at g711u.

Tom