[Asterisk-Users] SIP channel improvements

[Olle E. Johansson]

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.

This, in addition to the SIPDOMAIN variable, makes the SIP channel even more
useful.

Thank you, Mark, for your additions!

Now, ENUM/E.164 will propably work even better. I'll give it a try.

Now, to be the documentation-pain-in-the-*** I would like to get an explanation
of the autocreatepeer SIP.conf setting and functionality?

It's not in sip.conf.sample yet.

/Olle