[Asterisk-Users] Re: sip URL peering

[Wolfgang S. Rupprecht]

Chris Hills <chills at ne-worcs.ac.uk> writes:
> Perhaps you would be interested in TRIP (telephony routing over ip)?
> Each organisation can apply for an ITAD number, just like a
> domain. TRIP numbers take the form <extension>*<itad>, for example,
> 1234*222. As you can no doubt surmise, TRIP numbers can be dialled
> from a regular telephone handset. For more information, please see the
> following documents:-
>
> http://www.iana.org/assignments/trip-parameters
> http://www.ietf.org/rfc/rfc3219.txt

Thanks.  I'd entirely forgotten about that. 

Having the routing numbers coordinated by some benign central
authority like the IANA is certainly preferable to some enthusiast web
site which might or might not be around in a year.

Having just skimmed RFC 3219, it seems to add quite a bit of hair to
what is essentially just assigning an N-digit prefix (or suffix) to
every cooperating SIP server.  I'm sure that must have some advantages
for whatever situations they where concerned about, but I don't really
"get it".  SIP already does all the routing and redirect internally so
having the redirecting done at the top level seems redundant.

Ideally, for me at least, would be a simple ascii list in the style of
/etc/services or /etc/protocols that had an official mapping of
dialing-prefix and sip-server name (or domain name with _sip._udp SRV
entries).

If there is enough interest, maybe the greater asterisk community
could adopt some semi-official mapping tables.  I'd be willing to
periodically generate a flat mapping file and an extension.conf
dialplan snippet from sipbroker's list or whatever else is deemed more
neutral or useful if there was any interest in such.

-wolfgang
-- 
Wolfgang S. Rupprecht                http://www.wsrcc.com/wolfgang/