I think for SIP the control channel can still go through the proxy while
the data is bridged natively allowing you to still account for the
call. I'm not sure of the details on how Asterisk does it.
MARK.
David Bandel wrote:
On 11/2/05, Mark Hulber <[EMAIL PROTECTED]> wrote:
I think this means that it attempted to create a native bridge, which is
that it was trying to have the call go directly between the two
endpoints instead of going through the asterisk server but that process
failed. So in that case, Asterisk continued to proxy the call data. If
that's the case, a better output might have been, "... was unsuccessful,
server will continue to bridge call," or something along those lines.
MARK.
Thanx, Mark. Makes sense since I deliberately put: canreinvite=no in
the configuration of both SIP phones. Tough to account for calls that
are not proxied.
Ciao,
David A. Bandel
--
Focus on the dream, not the competition.
- Nemesis Air Racing Team motto
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