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/[EMAIL PROTECTED]' (thanks to
SIP/to-bandwidth-bfb3)
Is there any configurations that can be set to prevent hairpin problem.
Can anyone help ?
Thanks
Ramki
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