[asterisk-users] SIP Dial Command to a non-Asterisk url

[Alex Balashov]

Gavin,

   Does the Asterisk server's route to 192.168.45.18 traverse a firewall or
router that may be blocking non-SIP ports that are dynamically allocated?

   SDP -- part of the SIP INVITE transaction payload -- negotiates arbitrary 
ports between the two endpoints for actually passing media.  If these are
being dropped somewhere along the way, you'll have no audio in one or
more directions of the call path.

   Best thing to do is to is a packet capture on the Asterisk server and
filter on 192.168.45.183 to verify that you're seeing bidirectional media,
from and to that host.  Chances are something will be missing.

   Of course, it could be a non-IP problem of some sort as well, perhaps
even something fairly obvious.

-- Alex

--
Alex Balashov   <sasha at presidium.org>