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 To :  yate@v...
 From :  Paul Chitescu <paulc@v...>
 Subject :  Re: [yate] sip <-> sip-t signaling proxy
 Date :  Wed, 2 Jun 2010 12:43:38 +0300

You can definitely add a RedirectingNumber ISUP parameter to the SIP message.


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:


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


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.