I know it's not a NAT environment, but the way we got around that was
by setting nat=yes in the sip.conf. nat=yes basically just tells the
server to stick around during the conversation so you don't lose the
rtp stream.
Aaron
On Dec 14, 2005, at 9:12 PM, Kevin P. Fleming wrote:
Jean-François Rousseau wrote:
I've tried putting canreinvite=no everywhere in my config, but
asterisk is
still trying a native bridge on the call. The problem is that when
this
happen, the native bridge fail but one phone (Sipura 2000) think
that the
bridging was done and the BYE is not received by asterisk when the
call end.
native bridge != reinvite
Native bridge means that the RTP packets never leave the RTP core
in Asterisk, they are forwarded directly back to the endpoints.
Reinvite is something entirely different; if you use 'sip debug'
and you see Asterisk sending re-INVITE requests to the phones with
'canreinvite=no' in place, then that is a bug.
But I will repeat (since this comes up all the time) native bridge !
= reinvite.
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