There is a workaround until “usereqphone” variable is available for From header.
You can use fromdomain variable on your sip*.conf.
For instance:
fromuser=yourSIPuser
fromdomain=yourdomain.com;user=phone
If you find something like this:
From: sip:yourSIPuser@yourdomain.com;user=phone:5160;tag=as695bc59b
You’ll have to change your Asterisk listening port for your trunk and for your extensions.
Now:
Trunk 5060
Extensions 5160
Trunk:
Edit your file for “bind” variable (in my case: pjsip.transports.conf)
Change bind=0.0.0.0:5060 to bind=0.0.0.0:5160
Extensions:
Edit your file for “bindport” variable (in my case: sip_general_additional.conf)
Change bindport=5160 to bindport=5060
Don’t forget to REGISTER your UA to new Extensions listening port (5060) and to reload these files
Now new From header will be as this (std.5060 port is avoided as is considered a well known port):
From: sip:yourSIPuser@yourdomain.com;user=phone;tag=as695bc59b
Hope to be useful