9 Jun
2010
9 Jun
'10
12:20 a.m.
If that's really your complete Kamailio config, it's missing a number
of critical checks and handlers, such as loose_route(), which handles
sequential (in-dialog) requests differently from initial requests due
to the presence of a Route: header and a To header tag.
You can't just stick cr_route() in the main route { ... } block
without those canonical checks and expect everything to work as desired.
The reason that the proxy is not seeing the sequential requests
(reinvites, BYE, etc.) is because you don't have a Record-Route
header, which tells the endpoints to relay sequential requests through
the proxy on the network and transport level. Add this to your
initial request route prior to (cr_route())[1]:
if(!is_method("REGISTER|OPTIONS|MESSAGE"))
record_route();
That should take care of it.
-- Alex
[1] And ideally, add quite a few other things as well. See the
stock configuration file for more insight.
On 06/08/2010 05:13 PM, JR Richardson wrote:
Hi All,
I'm using kamailio with carrierroute to load balance calls to other servers.
I have 2 testing scenarios set up:
sipp><kamailio><asterisk server
In the above scenario all SIP transactions, dialogs go through
kamailio, INVITE, Trying, ACK, OK, BYE, OK
Here is another one:
phone><asterisk><kamailio><asterisk server
In this scenario only the initial INVITE, ACK and OK go through
kamailio, then the 2 asterisk servers finish the session directly to
each other with ACK, BYE and OK
In both asterisk servers, canreinvite=no is set in peers and general
section in sip.conf. The kamailio cfg is using t_relay. No errors
are coming up anywhere.
Here is a pastebin of the 2 asterisk traces and the kamailio config.
http://pastebin.com/Uk9qVhX2
Any ideas on why the call is not maintaining SIP session through the proxy?
Thanks.
JR
--
Alex Balashov - Principal
Evariste Systems LLC
1170 Peachtree Street
12th Floor, Suite 1200
Atlanta, GA 30309
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/