[Serusers] "3xx Redirection" - How Should This Work?

[Java Rockx]

Thanks Steve.

The only reason I was using 3xx responses was because I believed that
was the correct way to handle call forwarding.

The downside of just relaying the message after rewriting the
hostname/port is that validation must be done on the forwarded
destination.

For example, if a SIP subscriber __does not__ have the free-pstn ACL
and they forward to a PSTN number, they could circumvent the ACL and
make PSTN calls by dialing their own number.

sl_send_reply("302", "Redirect") causes a whole new INVITE to be
processed in the normal fashion.

I suppose if there is no way to failover to voicemail then I'll have
to validate forwarded numbers.

Another problem with sending out an INVITE is realized when dealing
with PSTN gateways. A nasty situation creeps up when a SIP phone is
forwared to a PSTN destination. Then when an incoming call from some
other PSTN phone to the SIP phone occurs, SER would try to send an
INVITE back out the PSTN to the forwarded destination and many PSTN
gateways will reply with "482 Loop Detected".

Regards,
Paul


On Mon, 28 Feb 2005 11:29:37 -0500, Steve Blair <blairs at isc.upenn.edu> wrote:
> 
> Why wouldn't you just rewrite the hostname and port to be that of the
> voicemail server in your "302 failure route handling logic" and relay the
> call?
> 
> Java Rockx wrote:
> 
> >Hi all.
> >
> >I'm trying to get my call working working and I have a question.
> >
> >Suppose a SIP subscriber has voicemail and call forwarding is enabled.
> >
> >When an incoming call is destined for this SIP user, SER will reply
> >with a 302 redirect message.
> >
> >Next the caller's phone should construct a brand new INVITE to the
> >redirected destination.
> >
> >My question is this:
> >
> >If the forwarded destination is busy, then the call should be sent to
> >the voicemail of the SIP subscriber that was originally called. This
> >is the way the RBOCs call forwarding works.
> >
> >If I'm using sl_send_reply("302", "Temporarily Moved") back to the
> >caller, how can I make the voicemail piece work?
> >
> >Regards,
> >Paul
> >
> >_______________________________________________
> >Serusers mailing list
> >serusers at lists.iptel.org
> >http://lists.iptel.org/mailman/listinfo/serusers
> >
> >
> 
> --
> 
> ISC Network Engineering
> The University of Pennsylvania
> 3401 Walnut Street, Suite 221A
> Philadelphia, PA 19104
> 
> voice: 215-573-8396
> 
>        215-746-8001
> 
> fax: 215-898-9348
> 
> sip:blairs at upenn.edu
> 
>