[Asterisk-Users] How to disable sip Native bridge

[Aaron Daniel]

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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