[Date Prev][Date Next][Thread Prev][Thread Next][Date Index][Thread Index]
RE: AW: AW: [Sip] SIPIT Interop problem with ;user=phone
Many phones do this.
Some phones have the local dial plan interpreted in the phone,
and only emit "complete" (within the domain) phone numbers.
Some will send a tel uri, some will send the number in a sip
uri in the local (configured) domain.
I think that for most dial plans you can safely translate such
a number again by the proxy, but I worry that that is not ALWAYS
true. So, I think it might be useful to mark a sip uri with
an uninterpreted dial string differently from a sip uri with
an interpreted phone number. I also wonder if it might be
helpful in other circumstances; for example, where there are
several proxies in the path which have the capability to
translate dial strings to phone numbers. Generally, I think
it's always the case that the first proxy encountered that
has the capability should do the translation. I think it might
be useful for a succeeding proxy to know that it should NOT
translate (because it has already been translated).
Brian
> -----Original Message-----
> From: Juha Heinanen [mailto:jh@tutpro.com]
> Sent: Tuesday, September 02, 2003 12:55 PM
> To: Rosen, Brian
> Cc: 'hisham.khartabil@nokia.com'; Richard.Stastny@oefeg.at;
> fluffy@cisco.com; dean.willis@softarmor.com; sip@ietf.org
> Subject: RE: AW: AW: [Sip] SIPIT Interop problem with ;user=phone
>
>
> Rosen, Brian writes:
>
> > I think your example not ideal only because a UA could
> > have a trivial rule that a number starting with + is known to
> > be a globally dialable number, and thus COULD be put in a tel
> > uri.
>
>
> no matter what number the user dials (with or without +), my sip phone
> will do the same thing: append its own domain to the number
> and forward
> the call to the proxy of the domain. the proxy of the phone's domain
> will then interpret the number according to the conventions that exist
> within the domain.
>
> -- juha
>
_______________________________________________
Sip mailing list https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip