[Asterisk-Users] How to disable sip Native bridge

[Kevin P. Fleming]

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.