[asterisk-users] propagate sip reinvites with directrtpsetup=yes

[Eugene Oden]

is there a trick to get asterisk (1.6.2.13) to propagate
codec-changing sip reinvites when directrtpsetup=yes?

i'm trying to route calls to a gateway without keeping asterisk in the
rtp stream.

the gateway is first routing the call to a media server.  when
connecting the call to the downstream carrier a different codec is
selected.

the reinvite makes it to asterisk but asterisk isn't sending it along
to the originator so the transmit/receive codecs are mismatched
causing one-way audio.

thanks,

gene