[asterisk-dev] SIP compliance question

[Michael Procter]

Klaus Darilion wrote:
> Roy Sigurd Karlsbakk wrote:
> > hi all
> >
> > trying to connect asterisk to a Swyx (dot com) server, I cannot send
the
> > call through to the swyx, always getting a 404. The swyx people tell
me:
> >
> >> These call is rejected on swyxware side, because the provider uses
a
> non
> >> rfc compatible kind of addressing in the sip invite packet:
> >> ~INVITE sip:+4721973540 at 80.239.107.95:
> >> using an ip-adresse instead of a realm is not allowed by RFC.
> >
> 
> This SIP URI is perfectly valid.

Actually it isn't.  The trailing colon (':') is wrong.  RFC3261 Sec 25
states (interesting bits quoted):

	SIP-URI	= "sip:" [ userinfo ] hostport
	hostport	= host [ ":" port ]
	port		= 1*DIGIT

So, either drop the colon, or add the port number.  But you can't have a
colon and no number.

Regards,

Michael Procter