13 Feb
2008
13 Feb
'08
12:53 p.m.
Hi Dan/List,
I was reading the post below and trying to understand how your config
works. If
you are implementing this with something like a Cisco PSTN then you need all
of
these: PSTN, OpenSER, Mediaproxy and Yate involved in the SIP route? Does
the RTP
stream have to route via Yate and mediaproxy? :S
thanks for any help! cheers Andy.
Hey Marc,
I use Yate for doing that. It is simple and works out of the box (with
adding few
lines in configs of course).
I take Session timeout returned from connector and pass it to yate in a sip
header
Process that header in regex routing and define the
value as timeout for
session.
Yate knows by default that when a session has a
parameter "timeout"
returned
from routing to disconnect the call when timeout is
hit.
Let me know if you need further info, so I can send you some config files
if you
want to. You can contact me on IRC for live support
(DanB).
All the best,
DanB
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