Hi Jayesh,
Could you share a pcap with the call to check ? A wild guess is that
OverSIP is (by mistake) removing the Route header of OpenSIPS too
(instead of removing his own Route only).
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 16.05.2014 18:48, Jayesh Nambiar wrote:
Hi All,
I have a Freeswitch, OpenSIPS and Oversip all on the same server and
different ports. Freeswitch originates a call, goes to OpenSIPS,
OpenSIPS looks up into location and figures that it supposed to be
routed to Oversip and routes it to Oversip and the call is connected
with a WebRTC JsSip client.
But when the call is disconnected from the WebRTC client, the Oversip
directly routes the BYE to Freeswitch instead of going Via OpenSIPS.
Basically the BYE originated from JS-SIP itself has Freeswitch IP and
Port in the URI :(
But if I run Oversip on a different server, the routing of BYE from
WebRTC client is as expected which is from Oversip to OpenSIPS to
Freeswitch.
Can someone point me to why this happens when the application runs on
the same host !! Should I add some additional functions when running
applications on same host.
My default route contains this for all calls:
# record routing
if (!is_method("REGISTER|MESSAGE")) {
record_route();
}
Thanks in advance;
--- Jayesh
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