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<DIV dir=ltr align=left><SPAN class=939081214-12092006><FONT face=Arial
color=#0000ff size=2>Thank you for your answer.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=939081214-12092006></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=939081214-12092006><FONT face=Arial
color=#0000ff size=2>You are probably right that I would get better answer
without the "off the wall" remark. Sorry if upset
anyone.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=939081214-12092006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=939081214-12092006></SPAN><SPAN
class=939081214-12092006><FONT face=Arial color=#0000ff size=2>Ok, now we are
getting some where. The 302s I get back from the UAs contains a diversion
header with the callee info and the contact header contains the destination for
the "forwarding". So is it possible to extract the info from these headers to
use when appending the branch?</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=939081214-12092006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=939081214-12092006><FONT face=Arial
color=#0000ff size=2>Kind regards</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=939081214-12092006><FONT face=Arial
color=#0000ff size=2>Roger</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=939081214-12092006> </SPAN></DIV>
<DIV dir=ltr align=left>
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</DIV>
<DIV dir=ltr align=left><FONT face=Tahoma size=2><B>Från:</B> Greger V. Teigre
[mailto:greger@teigre.com] <BR><B>Skickat:</B> den 12 september 2006
08:51<BR><B>Till:</B> Roger Lewau<BR><B>Kopia:</B> serusers@iptel.org;
jh@tutpro.com<BR><B>Ämne:</B> Re: SV: [Serusers] Handling 302
responses<BR></FONT><BR></DIV>
<DIV></DIV>Hi Roger,<BR>I think you are complicating the picture by trying to
join two quite different worlds: SIP world and the PSTN world. SIP was not
intended to be introduced into the PSTN world, it applied some basic
"Internet-type" principles to the problem of setting up media session over the
Internet. Even the new IMS specifications for 3G follow the intentions of SIP
(although there are some conflicts between "telco people" and "Internet
people". <BR><BR>By no means are you alone, many UAs have been implemented
with PSTN replacement services in mind. In PSTN, call forwarding is
associated with multiple call legs, each leg being charged to the A-number
(calling party). (Of course, tromboning can can be done to optimize the media
path, but signalling-wise the call legs are still present, and thus also in
billing). In SIP, a REFER is not the creation of a new call leg, it is
simple a message back to caller that callee can be reached elsewhere. Strictly
speaking, to simulate PSTN, but using SIP intentions, the SIP UA should
implement B2BUA and initiate a new INVITE to the forwarded address. The reason
is that SIP puts more responsibility on the endpoints than in PSTN (where the
endpoints are dumb). Of course, you don't want to do
this.<BR><BR> So, to your problem: You probably have incoming
PSTN calls to your UA, which you have decided should use 302 for "forwarding".
The 302 is a reply, not a failure, so you will receive it in your onreply
route. You can probably then test for 302, change the ruri and append a
branch. Depending on your billing handling, you should add a Diversion header,
CC-Diversion or store in acc module using avpairs (Jiri noted that this is
possible in a recent thread, I have never done it) the party who
redirected.<BR><BR>Maybe these call flows can help in understanding 302. It
doesn't cover 302 from clients though.<BR><A class=moz-txt-link-freetext
href="http://www.cisco.com/univercd/cc/td/doc/product/voice/sipsols/biggulp/bgsipcf.htm">http://www.cisco.com/univercd/cc/td/doc/product/voice/sipsols/biggulp/bgsipcf.htm</A><BR><BR>I
hope this makes sense to you.<BR><BR>BTW, I think you would received some
sensibles answers earlier if you hadn't used your "off the wall"
statement...<BR>g-)<BR>Roger Lewau wrote:
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<P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN lang=EN-GB
style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Hello
Greger<O:P></O:P></SPAN></FONT></P>
<P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN lang=EN-GB
style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><O:P></O:P></SPAN></FONT></P>
<P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN lang=EN-GB
style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">I am surprised that
you argue that 302s is not the standard call forward scenario when this is how
it is implemented in almost all sip equipment (haven’t seen any device who
does it differently). Customer will punch in cfwd (like *72<number>) in
their phone and expect the operator (me) to forward the call to the
destination at their expense. This is nothing new or strange. 302s signal that
an UA has temporarily been moved and can be reached somewhere else, how is
that not call forwarding? Also, I would never expect an UA to send back
an email address in an IP telephony system, even if it is theoretically
possible. I would expect it to send back a SIP address; everything else is not
reachable through the protocol of SIP and makes absolutely no
sense.<O:P></O:P></SPAN></FONT></P>
<P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN lang=EN-GB
style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><O:P></O:P></SPAN></FONT></P>
<P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN lang=EN-GB
style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Most Voip customers
are not techies and they are not rfc knowledgeable and could not care any less
about how the forwarding is made. They would not be upset that their 302
generate a cost, this is the expected behaviour for this
service.<O:P></O:P></SPAN></FONT></P>
<P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN lang=EN-GB
style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><O:P></O:P></SPAN></FONT></P>
<P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN lang=EN-GB
style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Kind
regards<O:P></O:P></SPAN></FONT></P>
<P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN lang=EN-GB
style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Roger<O:P></O:P></SPAN></FONT></P>
<P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN lang=EN-GB
style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><O:P></O:P></SPAN></FONT></P>
<DIV>
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face="Times New Roman" color=black size=3><SPAN
style="FONT-SIZE: 12pt; COLOR: windowtext">
<HR tabIndex=-1 align=center width="100%" SIZE=2>
</SPAN></FONT></DIV>
<P class=MsoNormal><B><FONT face=Tahoma color=black size=2><SPAN lang=EN-GB
style="FONT-WEIGHT: bold; FONT-SIZE: 10pt; COLOR: windowtext; FONT-FAMILY: Tahoma">Från:</SPAN></FONT></B><FONT
face=Tahoma color=black size=2><SPAN lang=EN-GB
style="FONT-SIZE: 10pt; COLOR: windowtext; FONT-FAMILY: Tahoma"> <A
class=moz-txt-link-abbreviated
href="mailto:serusers-bounces@lists.iptel.org">serusers-bounces@lists.iptel.org</A>
[<A class=moz-txt-link-freetext
href="mailto:serusers-bounces@lists.iptel.org">mailto:serusers-bounces@lists.iptel.org</A>]
<B><SPAN style="FONT-WEIGHT: bold">För </SPAN></B>Greger V. Teigre<BR><B><SPAN
style="FONT-WEIGHT: bold">Skickat:</SPAN></B> den 11
septembe</SPAN></FONT><FONT face=Tahoma color=black size=2><SPAN
style="FONT-SIZE: 10pt; COLOR: windowtext; FONT-FAMILY: Tahoma">r 2006
11:59<BR><B><SPAN style="FONT-WEIGHT: bold">Till:</SPAN></B> Roger
Lewau<BR><B><SPAN style="FONT-WEIGHT: bold">Kopia:</SPAN></B> <A
class=moz-txt-link-abbreviated
href="mailto:serusers@iptel.org">serusers@iptel.org</A>; <A
class=moz-txt-link-abbreviated
href="mailto:jh@tutpro.com">jh@tutpro.com</A><BR><B><SPAN
style="FONT-WEIGHT: bold">Ämne:</SPAN></B> Re: [Serusers] Handling 302
responses</SPAN></FONT><FONT color=black><SPAN
style="COLOR: windowtext"><O:P></O:P></SPAN></FONT></P></DIV>
<P class=MsoNormal><FONT face="Times New Roman" color=black size=3><SPAN
style="FONT-SIZE: 12pt"><O:P></O:P></SPAN></FONT></P>
<P class=MsoNormal><FONT face="Times New Roman" color=black size=3><SPAN
style="FONT-SIZE: 12pt">Hi Roger,<BR>I think that was Juha's point: we don't.
<BR>302 was created to enable a user agent to communicate back to the other
user agent that it can be reached somewhere else. Thus, your server should
relay the 302 and the receiving user agent should then decide what to do. Some
UAs immediate initiate a new call, while others (e.g.software agents) may pop
up a question to the user: "Callee is not available, but can be reached at
location" (which of course may well be an international PSTN call that can be
expensive).<BR>Some UAs also have options that can be set: How to handle
redirects<BR><BR>Server-centric forwarding can be better handled by user
preferences and loading av pairs.<BR><BR>That being said, I remember a thread
a while ago with a discussion on how to turn a 302 into a forwarding. I don't
remember the outcome, but it is probably possible, although not according to
the RFCs. You do have some problems though, e.g. if the UA sends back an
email uri etc.<BR><BR>And of course, as people tend to follow RFCs, you will
probably get one angry customer if he realizes that his 302 generates a cost.
If you have control over the UA and have decided to use 302 instead of the
more standardized call forward scenario, you really are making problems
for yourself.<BR>g-)<BR><BR>Roger Lewau wrote: <O:P></O:P></SPAN></FONT></P>
<DIV>
<P class=MsoNormal><FONT face=Tahoma color=black size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Tahoma">Hello Juha and
Andrey<O:P></O:P></SPAN></FONT></P></DIV>
<DIV>
<P class=MsoNormal><FONT face=Tahoma color=black size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Tahoma"><O:P></O:P></SPAN></FONT></P></DIV>
<DIV>
<P class=MsoNormal><FONT face=Tahoma color=black size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Tahoma">302 "Moved temporarily" is
definately about forwarding/redirecting calls. This is how the vast
majority of all IP phones and ATAs handle call forwarding. It might not be the
intended use of 302 according to RFCs, even if I see nothing that says
otherwise, but this is how it is used in end devices today. This brings us
back to my original question. How do you guys handle 302 redirection so that
costs are charged to the callee.<O:P></O:P></SPAN></FONT></P></DIV>
<DIV>
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style="FONT-SIZE: 10pt; FONT-FAMILY: Tahoma"><O:P></O:P></SPAN></FONT></P></DIV>
<DIV>
<P class=MsoNormal><FONT face=Tahoma color=black size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Tahoma">Kind
regards<O:P></O:P></SPAN></FONT></P></DIV>
<DIV>
<P class=MsoNormal><FONT face=Tahoma color=black size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Tahoma">Roger <O:P></O:P></SPAN></FONT></P></DIV>
<BLOCKQUOTE
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<P class=MsoNormal style="MARGIN-BOTTOM: 12pt"><FONT face="Times New Roman"
color=black size=3><SPAN style="FONT-SIZE: 12pt">-----Original
Message-----<BR>From: "Andrey Kouprianov" <A
href="mailto:andrey.kouprianov@gmail.com"><andrey.kouprianov@gmail.com></A><BR>To:
<A href="mailto:serusers@iptel.org">serusers@iptel.org</A><BR>Date: Mon, 11
Sep 2006 12:51:09 +0700<BR>Subject: Re: [Serusers] Handling 302
responses<O:P></O:P></SPAN></FONT></P>
<DIV>
<P class=MsoNormal><FONT face="Courier New" color=black size=3><SPAN
style="FONT-SIZE: 12pt; FONT-FAMILY: 'Courier New'">You can also use 302
responses to gather some information about the<BR>remote party. Contacts
returned in the response are not necessarily<BR>the SIP URI's. I've tried
using mail addresses, SIP tel: URI's and<BR>HTTP URLs too.<BR><BR>So, if the
remote party is Busy at the moment, but has other ways to<BR>let u contact
them, 302 is one of the answers to this.<BR><BR>On 9/11/06, Juha Heinanen <A
href="mailto:jh@tutpro.com"><jh@tutpro.com></A> wrote:<BR>> Roger
Lewau writes:<BR>><BR>> > In my mind that statement is
completely off the wall, it is not the<BR>> > requesting client
that should be responsible for establishing the forwarded<BR>> >
call, it never is in the rest of the telecom industry so why should it
be<BR>> > the case for SIP?<BR>><BR>> 302 is not about
"forwarded call". it just tells the caller that the<BR>> callee is
at some other uri, which the caller may or may not wish to<BR>> contact.
in many pstn networks, you can hear an announcement that the<BR>>
number you tried is not in use and you should try another number<BR>>
instead.<BR>><BR>> if callee wants to "forward" calls, he has other
means for that purpose,<BR>> for example, his phone can forward the
invite to some other uri or he<BR>> may configure his proxy to do
so.<BR>><BR>> -- juha<BR>>
_______________________________________________<BR>> Serusers mailing
list<BR>> <A
href="mailto:Serusers@lists.iptel.org">Serusers@lists.iptel.org</A><BR>>
<A href="http://lists.iptel.org/mailman/listinfo/serusers"
target=_blank>http://lists.iptel.org/mailman/listinfo/serusers</A><BR>><BR>_______________________________________________<BR>Serusers
mailing list<BR><A
href="mailto:Serusers@lists.iptel.org">Serusers@lists.iptel.org</A><BR><A
href="http://lists.iptel.org/mailman/listinfo/serusers"
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</SPAN></FONT></PRE><PRE><FONT face="Courier New" color=black size=2><SPAN style="FONT-SIZE: 10pt"><O:P> </O:P></SPAN></FONT></PRE><PRE><FONT face="Courier New" color=black size=2><SPAN style="FONT-SIZE: 10pt">_______________________________________________<O:P></O:P></SPAN></FONT></PRE><PRE><FONT face="Courier New" color=black size=2><SPAN style="FONT-SIZE: 10pt">Serusers mailing list<O:P></O:P></SPAN></FONT></PRE><PRE><FONT face="Courier New" color=black size=2><SPAN style="FONT-SIZE: 10pt"><A href="mailto:Serusers@lists.iptel.org">Serusers@lists.iptel.org</A><O:P></O:P></SPAN></FONT></PRE><PRE><FONT face="Courier New" color=black size=2><SPAN style="FONT-SIZE: 10pt"><A href="http://lists.iptel.org/mailman/listinfo/serusers">http://lists.iptel.org/mailman/listinfo/serusers</A><O:P></O:P></SPAN></FONT></PRE><PRE><FONT face="Courier New" color=black size=2><SPAN style="FONT-SIZE: 10pt"> <O:P></O:P></SPAN></FONT></PRE></DIV></BLOCKQUOTE></BODY></HTML>