Hi, When we make a call using asterisk to call one of the numbers which is again configured to be terminated in our server, the call is not getting bridged but instead its getting joined. As a result some of our configurations does not work. Our application logs show the following : Jul 28 15:01:58 VERBOSE[12767] logger.c: -- Got SIP response 482 "Loop Detected" back from 4.79.212.236 Jul 28 15:01:58 DEBUG[12767] chan_sip.c: Hairpin detected, setting up call forward for what it's worth Jul 28 15:01:58 VERBOSE[20128] logger.c: -- Now forwarding SIP/to- bandwidth-00c5 to 'Local/+18666301940@from-sip-bandwidth' (thanks to SIP/to-bandwidth-bfb3) Is there any configurations that can be set to prevent hairpin problem. Can anyone help ? Thanks Ramki