All my asterisk systems use only IPv4 currently. I have one phone which is on T-Mobile network, and this network is only IPv6 now.
The phone can register fine, because T-Mobile does NAT64 and it connects fine to my IPv4 asterisk server.
But in the SDP for a call setup, this phone sends only an IPv6 address as a contact, so RTP fails.
I have nat=yes already set on this chan_sip extension, I thought this would ignore the IPv6 in the SDP and use the *apparent* IPv4 instead, but apparently not?
Any help appreciated, thanks all.