Please post a link with your network drawing. Your description is unclear.
Il 05/02/2014 20.19, Tony Ward ha scritto:
Hello,
I currently have a media server behind a nat firewall with calls
delivered via a PSTN Trunk. I want to add a 2nd media server and
route calls to either depending upon the dialed number. I've been
trying to do this using drouting in opensips 1.10.0, but cannot get a
configuration that works.
I started by generating the 'trunking script' using make menuconfig,
and populated mysql to accept my PSTN trunk and route to my media
server. When an incoming call arrives, it is directed to opensips,
and forwarded to media server with a record-route header containing my
private ip. This confuses my PSTN partner and we are unable to
establish the rtp stream.
After reviewing the mailing lists I tried setting alias and
advertised_address=my public ip. Now when an incoming call arrives it
is directed to opensips and forwarded to the media server with a
record-route header containing my public ip. Call setup completes
successfully. Call teardown initiated from PSTN trunk completes
successfully. Call teardown initiated from media server fails because
the media servers sends BYE to the public IP, and the NAT router does
not know what to do with it (destination unreachable).
It seems as though the invite to my media server needs to have a
record-route header with my private ip, while the ok response back to
my PSTN provider needs to have a record-route header with my public
ip. Is this the right approach? I've briefly toyed with rtpproxy and
also b2bua without much luck, and was hoping this simpler solution
could be made to work.
Thanks,
Tony
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