Hi,
I have configured Kamailio to register a SIP user over websocket. SIP users are getting registered over webphone like JSSIP fine but not able to establish the call scenario below.

From Twilio Trunk - INVITE - KAMAILIO - ASTERISK - play hello world - Dial SIP user 101@Kamailio_IP - Rings 101 and Answer - Hangup.

Here,
101 is registered with Kamailio IP, and I am sending a call from Twilio SIP trunk to Kamailio server IP.
Asterisk is on port 5080 and Kamailio is on port 5090 both are on the same host 45.118.163.244.

This above call scenario is NOT working with Webphones registered via JSSIP. I think the second call leg from Asterisk -> Kamailio is not getting set up properly and ACK is not getting confirmed here, call connects for some time and no voice transmits and then disconnects.

The same call scenario is working with users registered at Softphone like Zoiper perfectly fine. No idea why it is not working with a webphone.

I have attached kamailio.cfg, Kamailio log and Asterisk's pjsip logs here. So, please guide me on this as soon as possible.
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