[asterisk-users] How to enter bridge_native_loop???

[Santosh Raghuram]

Hi,

With canreinvite=yes, all the media/rtp traffic for the call typically flows
directly between the two peers. So how is the code in bridge_native_loop
called and when? Is it called and used for any further sip signalling and
not rtp?


Thanks for your prompt reply.

Regards,
Santosh.

> Hi,
>
> I am using asterisk-1.4.0.
>
> I am inquisitive of what Packet2Packet bridge (bridge_p2p_loop)
> does and what Native bridge (bridge_native_loop) does.
>
> I have configured my dial plans and options such that I can enter
> bridge_p2p_loop. However, I am unable to enter bridge_native_loop
> for some reason.
>
> I have the following extensions:
>
> exten => 7126,1,Dial(SIP/lin_santosh)
> exten => 7126,s+1,Hangup
>
> exten => 7140,1,Dial(SIP/win_test)
> exten => 7140,s+1,Hangup
>
> My sip.conf is as:
>
> [lin_santosh]
> type=friend
> regexten=7126
> callerid="LIN Santosh" <7126>
> host=dynamic
> nat=yes
> canreinvite=no
> allow=all
>
You have set canreinvite to no, thus disabling native briding.

/O
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