Peter Dunkley writes:
> There is a much simpler WebSocket Kamailio configuration file in the
> examples directory in the source tree:
> http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob;f=examples/welcome.cfg;h=4bde0ae71be6f8da6a5bd3089d5af57569cc0178;hb=HEAD
the link to websocket
Sorry,
Wrong link. The correct one is:
http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob;f=examples/websocket.cfg;h=4176af0a86985dc88d768b31f4ebe4021abb093f;hb=HEAD
Peter
> Hi,
>
> There is a much simpler WebSocket Kamailio configuration file in the
> examples directory in the so
Hi,
There is a much simpler WebSocket Kamailio configuration file in the
examples directory in the source tree:
http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob;f=examples/welcome.cfg;h=4bde0ae71be6f8da6a5bd3089d5af57569cc0178;hb=HEAD
It doesn't have accounting or any of the other
Peter,
Thank you. By changing the "method_filtering" modparam to 0 (it was
actually 1), I am now able to make it past this, and the INVITE is
processed over WS transport. However, the audio call is still not
completing.
I am seeing a "180 Ringing" message for a while, followed by a "408 Request
Hello,
In SIP you can put an Allow: header in REGISTER requests to say which
methods the registering end-point is capable of receiving.
If you get a -2 returned from lookup() it means that the method for the
request (in this case INVITE) was not in the "Allow:" header in the
REGISTER.
You can ch
Hi,
New to Kamailio. I have my Kamailio 4.0 server with websocket support, and
the users can register using the JsSIP Tryit sample WebRTC application.
They can do 'chat' feature of JsSIP Tryit using kamailio 4.0 sip over
websockets module.
However, after registration, the users can't place an a
