[asterisk-dev] SIP compliance question

[Klaus Darilion]

Michael Procter wrote:
> 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.


Yes, that's correct. I've missed this. If this isn't a copy/paste error 
then this is clearly a validation and a bug in Asterisk.

regards
klaus