Asterisk,SIP Retransmission timeout

Question

I have created a sip trunk from One Asterisk(version 11.2.1) say \'A\' server to another Asterisk server(11.7.0) say \'B\', and I am getting sip response 200 ok.
But when I

Answer 1

By default Asterisk sends a RE-INVITE request after a call is established.

But most sip clients and sip servers in the market do not accept RE-INVITE requests. For this reason, when Asterisk sends a RE-INVITE after a call is established, the other side does not answer the request. So, after 32 seconds, Asterisk hangs up the call.

To solve the problem, you need to disable the RE-INVITE feature of Asterisk if your client software does not accept RE-INVITE requests. To do this, you need to edit the sip.conf file in Asterisk to include:

canreinvite = no