[Serusers] Manipulate from_uri

Charles Wang lazy.charles at gmail.com
Wed Feb 23 21:01:54 CET 2005


Vitaly:

Your explain is very very clear for me to understand the RPID.

I am tring it on my server, but it seems a little bit problem at my
CISCO trunking.

Anyway, it is very useful for me to know what RPID is.

Thank you very much.

Best Regard
Charles


On Wed, 23 Feb 2005 12:56:21 -0500, Vitaly Nikolaev
<vitaly at voipsonic.com> wrote:
> Sure
> 
> RPID is Remote-Party-Id: field in SIP that tell to outgoing GW what value
> and setting put in CallerID when GW will terminate call to .. whatever
> 
> Read this:
> 
> Readme for module auth_db
> Readme for module Auth
> URL:
> http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122
> t/122t13/ftsipext.htm
> 
> May be it not best document, but it will tell you what is rpid and why you
> need it
> 
> In short:
> You put in ser:
> modparam("auth", "rpid_prefix", "Name <sip:")
> modparam("auth", "rpid_suffix",
> "@sip.frontlineglobal.net;user=phone>;party=calling;screen=yes;privacy=off")
> 
> Name can be any, screen=yes means CallerID is trusted, some carried sekip it
> if see screen=no, privacy=off means who CLI to remote party (on or full -
> caller id blocking)
> 
> This is for kind of static thing, and I did not find way (almost half year
> ago) to make it dynamic.., per customer, per call
> 
> Then u do:
> 
> append_rpid_hf(); on INVITE
> 
> and then SER will take value that It received from database during authorize
> and append prefix/suffix and send it in SIP header.
> 
> README is quite good, just read it and you will how exact idea how to use it
> 
> 
> -----Original Message-----
> From: serusers-bounces at iptel.org [mailto:serusers-bounces at lists.iptel.org] On
> Behalf Of Charles Wang
> Sent: Wednesday, February 23, 2005 12:41 PM
> To: serusers at lists.iptel.org
> Subject: Re: [Serusers] Manipulate from_uri
> 
> Dear Paul:
> 
> I am a new user of SER sip. And would you please tell me an explain
> about what the RPID is??
> 
> I want to control the call to PSTN with B2BUA of vovida or Asterisk.
> 
> Best Regard
> Charles
> 
> On Wed, 23 Feb 2005 10:33:26 -0500, Java Rockx <javarockx at gmail.com> wrote:
> > Vitaly,
> >
> > I believe you are correct that RPID is only helpful during SIP->PSTN
> calls.
> >
> > I'm not real familiar with B2BUA, but if I were to introduce one in to
> > my system, does that mean I need to have all the RTP traffic pass
> > through my network as well, or does the B2BUA only deal with SIP
> > messaging?
> >
> > Regards,
> > Paul
> >
> > On Wed, 23 Feb 2005 10:14:18 -0500, Vitaly Nikolaev
> > <vitaly at voipsonic.com> wrote:
> > >
> > > The answer is B2BUA
> > >
> > > There actually two ways,
> > >
> > > First is b2bua which is nice (for me ok.. I like it :)
> > >
> > > Second if you assign the CallerID number to device.. the real callerid,
> (for
> > > example in my network we use some fake numbers into devices and then
> change
> > > it do real callerid (did) in b2bua), then some devices, for example
> SIPURA,
> > > can send Anonymous as name in From and Contact and some GWs, will use it
> and
> > > hide callerid on outgoing call.
> > >
> > > And third:
> > >
> > > And there are also Remote-Party-Id: field in SIP that partially
> supported by
> > > SER and supported by most of GW and carriers, and that is probably RIGHT
> way
> > > to do that.
> > >
> > > When PSTN GW sends call to outside of ur network, it no just skip
> callerid,
> > > u can do  it but it is wrong, it make it private, so usual ppl will not
> see
> > > it but for example 911 and some toll free subscriber will see. That
> right
> > > way and that how everybody should do cli blocking.
> > >
> > > What is wrong in this method is that if you call goes from SIP to SIP
> and
> > > you use as me fake CLI on devices, customers will see fake numbers
> because
> > > as far as I know SIP devices does not support RPID (please correct me if
> I
> > > am wrong) so in my case I will have to use combination of N1 and N3..
> > >
> > >
> > > -----Original Message-----
> > > From: serusers-bounces at iptel.org [mailto:serusers-bounces at lists.iptel.org] On
> > > Behalf Of Java Rockx
> > > Sent: Wednesday, February 23, 2005 7:20 AM
> > > To: Klaus Darilion
> > > Cc: serusers at lists.iptel.org
> > > Subject: Re: [Serusers] Manipulate from_uri
> > >
> > > Klaus,
> > >
> > > I realize that RFC3261 says do not modify the From header, however, my
> > > question is this;
> > >
> > > When making SIP-to-SIP calls between two subscribers on the same SIP
> > > proxy, how can one implement Caller ID Block when RPID is not usually
> > > honored by SIP UAs - since this is really for PSTN gateways?
> > >
> > > It seems that altering the From header in this case is the only
> > > option. Have I missed something?
> > >
> > > Regards,
> > > Paul
> > >
> > > On Wed, 23 Feb 2005 12:00:14 +0100, Klaus Darilion
> > > <klaus.mailinglists at pernau.at> wrote:
> > > > Hi Mik!
> > > >
> > > > Do not modify the From: or To: URI - this is not allowed (RFC 3261)
> and
> > > > will cause problems.
> > > >
> > > > regards,
> > > > klaus
> > > >
> > > > Mik Cheez wrote:
> > > > > Since I'm able to modify the URI for relaying, I would assume there
> > > > > would be a way to modify the FROM_URI as well.  Has anyone managed
> to do
> > > > > this with SER acting as a proxy?  Any other possible avenues to
> modify
> > > > > the CALLID?
> > > > >
> > > > > Best regards
> > > > >
> > > > > _______________________________________________
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> > > > >
> > > > >
> > > >
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