Asterisk as a SIP client. How to handle Re-INVITE?

Question

Asterisk

Hello!
I have a server with installed asterisk version 1.8.7.2 on FreeBDS 9.0.
I'm trying to connect Asterisk with someone else's (which is important because you can't reconfigure someone else's server) SIP server .
Registration does not cause problems, calls go, there is a voice.
There is only one problem that I can not overcome in any way - the SIP server is configured in such a way that it sends Re-INVITE.

This would not be a problem, but since the server is designed for automated testing, incoming calls are processed dynamically - an AGI script is executed. Actually, now the situation is such that upon the arrival of the second invite, a second channel is created and the script is launched a second time.

How to overcome the current situation and handle Re-INVITE in such a way that the second connection is not established?

Answer 1

Asterisk version 1.8.7 has a wonderful bug with the launch of AGI scripts, which was the cause of the problem.
Resolved by updating to 1.8.15.

Answer 2

insecure=port,invite
canreinvite=no
so Asterisk will not invait after the session expires.

Answer 3

also for complete beauty, isolation and anonymity, prohibit traffic from going directly between the end peers of RTP