25 Aug
2016
25 Aug
'16
10:19 a.m.
Hi Matt,
Check out this post and config:
https://blog.voipxswitch.com/2016/08/11/kamailio-and-freeswitch-on-the-same…
It demonstrates how to handle NAT & forward a REGISTER to FreeSWITCH (on
localhost)
Thanks,
Emmanuel
On Thu, Aug 25, 2016 at 9:57 AM, Daniel Tryba <d.tryba(a)pocos.nl> wrote:
On Thu, Aug 25, 2016 at 09:38:12AM -0400, Mike
Patterson wrote:
I have configured Kamailio to pass a registration
request to another VoIP
provider. The reason I am doing this is to provide a sip port for users
where the ISP is blocking SIP. I am able to pass the registrations to
the
VoIP provider but I'd like to have inbound
calls working as well. I see
the
inbound invites coming into the Kamailio server
but it appears that the
server does not 'know' about the UA that is registered 'through' the
server
to the VoIP provider. Can anyone tell me how to
fix this problem?
Below is
my config.
You should look into the NAT handling tricks:
http://www.kamailio.org/docs/modules/stable/modules/
nathelper.html#nathelper.set_contact_alias
and
http://kamailio.org/docs/modules/stable/modules/
nathelper.html#nathelper.f.handle_ruri_alias
Take a look at the default config NATMANAGE routes and enable "natting"
for all sources except you upstream server.
Alternatively if the registrar of the voip provider supports Path,
enable that in kamailio use
http://kamailio.org/docs/modules/stable/modules/path.html#idp38122172
in the registers to add the header and set
http://kamailio.org/docs/modules/stable/modules/path.html#idp38033932
to handle the INVITEs
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users