I am using a JSSIP client and they claim to be implemented RFC-5626.
Following are the REGISTER and response messages. Is it possible to
confirm the JSSIP client has full implementation of SIP outbound?
If it supports fully then I can debug outbound and gruu functionality at
Kamailio(I have it
Hi,
Websocket module documentation has a code reference to use aliases for SIP
routing. However, aliases will not work in the following setup and
situation.
1. Kamailio is configured with active and standby node
2. Ping is implemented from webclient and kamailio responds with pong.
3. Two clients
> Do you see the same behaviour if you set the listen directive for SIP
> before the one for WS, in kamailio.cfg?
> i.e.
>
> listen=tcp:172.27.45.219:5060 advertise 172.27.45.198:5060
>
> before
>
> listen=tcp:172.27.45.219:MY_WS_PORT
>
> ?
>
> Thanks,
&g
Hi Daniel,
Sure, please check as per your availability.
I had tried the above option before adding ds_default socket in kamailio
.cfg but the result was same. Now also I have tried, but still kamailio is
declaring 8080 as received port .
Following is the dispatcher configuration I have tried now.
:
> Hello,
>
> the pcap shows only the OPTIONS requests, not the replies. Maybe you have
> to capture the traffic also on 8080 to catch it. It would important to see
> the reply and where it is sent back to kamailio.
>
> Cheers,
> Daniel
> On 24.06.21 09:24, Shahid Hussain wrote:
Hi,
I have configured the dispatcher default socket as below IP , the
expectation here is that kamailio should send all traffic from this IP
(port will be ephemeral) over TCP.
modparam("dispatcher", "ds_default_socket", "tcp:172.27.45.219:5060")
This system also listens on port 8080 which is
