[Asterisk-Users] canreinvite=yes

Tue Nov 15 07:19:15 MST 2005

Hi,

Just one question.  The documentation I have seen says that the RTP
audio stream is routed directly(if allowed ...), but never anything
about video streams? Is this just because documents are pre 1.2 or is it
true that audio can go directly, but video must pass through Asterisk?

Anyone?

Does anyone have experience with H263 on the 1.2.rc1 version? I think
there is a bug, and will trace and submit it to Bugzilla..??

Trond