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 To :  yate@v...
 From :  Paul Chitescu <paulc@v...>
 Subject :  Re: [yate] sip <-> sip-t signaling proxy
 Date :  Tue, 8 Jun 2010 12:58:02 +0300
Hi!

Yate doesn't provide yet a specific decoder for fields of 
RedirectionInformation so it will use raw hex decoding or encoding.

You can set RedirectionInformation=03 31 (space separated octets in 
hexadecimal format, do not include parameter code or length)

Be very careful as Yate cannot check if the parameter is correctly formatted.

Paul

On Tuesday 08 June 2010 12:39:55 pm Alexandr Gevlichenko wrote:
> Hi!
> Thank you very much, I was able to use sip-t
> Last question:
> How can I define RedirectionInformation parameter?
> 
> I'm trying to do in regexproute.conf:
> RedirectionInformation = 0x0331
> 
> But in the log I see " Unwinding type storage for
> failed parameter RedirectionInformation"
> 
> I need to get this result (in Wireshark):
> ##################
> Redirection Information
> Optional Parameter: 19 (Redirection Information)
> Parameter length: 2
> .... .011 .... .... = Redirection indicator: call diverted (3)
> 0000 .... .... .... = Original redirection reason: unknown / not avaiable
> (0)
> .... .... .... .001 = Redirection counter: 1
> .... .... 0011 .... = Redirection reason: unconditional (national use) (3)
> ###################
> 
> 2010/6/2 Paul Chitescu 
> 
> > Hi!
> >
> > You can definitely add a RedirectingNumber ISUP parameter to the SIP
> > message.
> >
> > .*=;message-prefix=isup.;isup.message-type=IAM;isup.protocol-type=itu-t;\
> >
> >  
isup.CalledPartyNumber=${called};isup.CalledPartyNumber.nature=national;isup.CalledPartyNumber.plan=isdn;
\
> >  ...
> > ${sip_x-redirecting}.=;isup.RedirectingNumber=${sip_x-redirecting}
> >
> > Note that if you use sip-t then you should take care to create a complete
> > IAM populated with all and correct
> > parameters as required by the receiving end. Yate provides sensible
> > defaults (except for CalledPartyNumber
> > which defaults to empty) but those may not be adequate for the receiving
> > end.
> >
> > The mandatory IAM parameters are:
> > - NatureOfConnectionIndicators
> > - ForwardCallIndicators
> > - CallingPartyCategory
> > - TransmissionMediumRequirement (ITU only)
> > - UserServiceInformation (ANSI only)
> > - CalledPartyNumber
> >
> > You should take care that the SIP message headers + SDP body + ISUP body
> > fits in an UDP datagram.
> >
> > On the opposite direction you can check if an ISUP parameter is present 
and
> > add it as custom SIP headers:
> >
> > ${isup.RedirectingNumber}.=;osip_X-Redirecting=${isup.RedirectingNumber}
> >
> > ${isup.GenericNumber.qualifier}^redirecting$=;osip_X-
Redirecting2=${isup.GenericNumber}
> >
> > To enable sip-t the ysigchan.yate module needs to be loaded (happens
> > automatically in server mode).
> > In ysipchan.conf section [sip-t] set isup=on
> >
> > Paul
> >
> >
> > On Tuesday 01 June 2010 06:16:55 pm Alexandr Gevlichenko wrote:
> > > Hi all
> > > I need to configure sip <-> sip-t proxy with some unusual features:
> > > 1. Ability to implement in IAM (Attached to the INVITE) message
> > "redirecting
> > > number".
> > > For example, if the initial INVITE message contains an additional header
> > > "Redirecting number" must be added to the IAM message.
> > > 2. Normal work as a sip <-> sip-t signaling proxy. It is planned to be
> > used
> > > in the scheme ASTERISK - (sip) - YATE - (sip-t) - PSTN.
> > >
> > > YATE could be this proxy?
> > > Thanks.