I just discovered that the SIP channel has undergone some major improvements.
I'm now able to dial any SIP URL with dial, couldn't get it to work earlier,
all domains had to be defined in SIP.conf.
...and I'm able to call any SIP URL with Xlite, with Asterisk resolving the domain part according to DNS SRV records, contacting the right SIP proxy for the DOMAIN, setting up the call.
This is brilliant, a major step forward for the SIP support in Asterisk!
/O
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