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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