25 Nov
2003
25 Nov
'03
12:31 p.m.
SRATCH THAT!!! it only bloomin' works!!! I upgraded asterisk on the target box
and the grandstream now works, kphone is still not playing ball, I'm not
entirely sure why there, but getting the grandstream going is a big step
forward.
On Tue, Nov 25, 2003 at 06:23:35AM -0800, Gregory Sandul wrote:
Tristian,
When INVIVE arrives to you try to modify SIP body -
SDP "c=" with means of "replace" function of
textopt.so module. The decision to modify body may be
taken based of source IP and uri.
On reply to route you will need to modify "c=" again
and put another NIC's IP.
I've tryed to do it today but with some errors. I'll
try tomorow again.
It will be the best if force_rtp_proxy() funtion have
optional parameter - IP address which it will put in
SDP body when function is called.
Regards,
Greg.
--- Tristan Colgate <tristan(a)inuxtech.co.uk> wrote:
--
Tristan Colgate
Inux Technologies
E-Mail: tristan(a)inuxtech.co.uk
Mobile: 07900 690 912
Sorry to reply to myself,
OK, I've started looking at this but there seems to
be a basic, pretty much
unsovelable problem. In the case of an invite we
don't actually know for sure
where the other end of the conversation is going to
be, since we havent seen
the sdp in the OK by that stage.
The only thing I can think of, is to listen on the
interface that the INVITE
request itself will go out on, this isn't the same
thing obviously. It will
work for the setup I have, infact I already make
similar assumptions when
deciding whether or not to rtp proxy in the first
place.
Any thaughts?
--
Tristan Colgate
Inux Technologies
E-Mail: tristan(a)inuxtech.co.uk
Mobile: 07900 690 912
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