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