[Freeswitch-users] Opensips integration questions

[Schneur Rosenberg]

Hi, I'm pretty new to Freeswitch, I built a complete solution using
OpenSIPS/Asterisk and I'm in the process of replacing Asterisk with
Freeswitch.

My current system is built that OpenSIPS act as a registrar and as a load
balancer, the rest is done on a cluster of Asterisk machines, which as I
said before is being rewritten to Freeswitch, in Freeswitch I'm using
xml_curl for my dialplan and call routing, I'm using Intralanman project to
handle the sip users.

With Asterisk I had OpensSIPS as a trunk and I know that everything coming
from that trunk, is either a DID or a authenticated outgoing call, Asterisk
and OpenSIPS shared the user table, Asterisk recognized the sip user name
from the FROM field and used all the variables etc from its own sip user
table as if the user has registered directly to itself, I had some issues
with blind transfers but I was able to do some manipulations and DB queries
to handle it fine.

I tried doing the same with Freeswitch, I let the ACL allow the ip from
Freeswitch, but Freeswitch would not use the variables from the user like
Asterisk did, for example accountcode, user_context etc, I was able to
partially fix that by having my web server do a db query and then using the
"set" application set all the variables, but I'm not sure if its the proper
solution, does anyone have a better solution? will setting my OpenSIPS as a
gateway instead of merely allowing it in with ACL work better? right now
I'm using the application "execute_extension" to send it to the proper
context that I've pulled from the DB, I dont like the hack and if there is
a better solution I would like to know.

One more major problem that I have is NAT related, when I place a outgoing
call for example from a phone which is behind a NAT (all of our customers
phones are behind NAT) the call connects and audio works 2 ways, but when
the called party hangs up, the carrier sends the BYE to Freeswitch,
freeswitch then sends it to my OpenSIPS which is supposed to send it to my
clients phone, but Freeswich writes the RURI with the natted address, the
RURI is BYE sip:101 at 192.168.0.11:5062 SIP/2.0. and my OpenSIPS does not
know what to do with it and ignores it, my Asterisk replaced the RURI with
the received address in the VIA, so same RURI would be BYE
sip:101 at 64.69.47.175:5062 SIP/2.0. which worked fine, same problem happens
with all subsequent requests  if user does not hang up, I need a solution
for this major issue.

thank you
S. Rosenberg
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