How Does Asterisk Choose SIPDOMAIN

Hi,
If asterisk have more than one interface How does asterisk choose SIPDOMAIN.

Exam:
eth1:1.1.1.1
eth2:2.2.2.2

[transport-udp-1]
type=transport
protocol=udp
bind=1.1.1.1:5060

[transport-udp-2]
type=transport
protocol=udp
bind=2.2.2.2:5060

What are you referring to when you say “SIPDOMAIN”?

I mean;
When I receive a call from transport-udp-1, I have below log

[Oct 14 13:35:13] VERBOSE[5871] pbx_variables.c: Setting global variable ‘SIPDOMAIN’ to ‘1.1.1.1’

When I receive a call from transport-udp-2, I have below log

[Oct 14 13:35:13] VERBOSE[5871] pbx_variables.c: Setting global variable ‘SIPDOMAIN’ to ‘2.2.2.2’

I want to set SIPDOMAIN 1.1.1.1 all call

That is set based on incoming traffic. It is set based on the host portion of the request URI.

is It mean: this variable is dynamic? Can I manipulate it?

You can overwrite it if you want, but it doesn’t change outgoing traffic.

Thanks for your reply. How can overwrite?

The same as any other dialplan variable. Set(SIPDOMAIN=blah)

Why do you still want to change it, given that you have been told it has no effect on outgoing traffic?

Actualy My Problem is; Asterisk Upgrade from 13.13 to 13.22 Issue

when i examined the log file, the only difference: if call coming from my sip provider, SIPDOMAIN was changed and then the phone ring but no voice.
That’s way I focus SIPDOMAIN.