[SIPForum-discussion] Remote Party ID (RPID) missing for Restricted call (caller ID hide)

Pierre Pebay pierre.pebay at gmail.com
Wed Dec 17 10:58:14 UTC 2008


Hi Rashid

>From my experience, this will depend on peering convention between you (as
operator) and peer. Usually, when you have an operator national licence, you
are in charge of applying the Identity restriction at the latest step before
reaching the called party.

Pierre

On Tue, Dec 16, 2008 at 6:16 PM, Shakil, Rashid <rshakil at covad.com> wrote:

>  Hello all,
>
>
>
> Quick question please ...Can any SIP peer allowed to send an INVITE for
> restricted call without calling FROM information with no "Remote Party ID".
> If you look at the following information calling From number is restricted
> and "Remote Party ID (RPID)" is missing as well therefore no way to find out
> where the call is originating from. Is this acceptable if yes can you please
> reference a draft where I can read this in detail ?
>
>
>
>
> ============================================================================
>
> INVITE sip:2132560444 at 6.1.7.2:5060 SIP/2.0
>
> Via: SIP/2.0/UDP 6.6.19.72:5060
> ;branch=z9hG4bK61ed25f40d9a00534ea564d93dcad9e7-0
>
> *From: "Anonymous" <sip:Restricted at 6.1.7.2:5060
> >;tag=824f5ba36d11a58f9978aa130ae1eefd*
>
> To: <sip:2132560444 at 6.1.7.2:5060>
>
> Call-ID: 0a0ae76e8ada421e0669e2eb0cb79063-4947e03f at 6.1.7.2
>
> CSeq: 34992 INVITE
>
> Max-Forwards: 70
>
> Allow:
> INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,OPTIONS
>
> Accept: application/sdp, application/isup, application/dtmf,
> application/dtmf-relay,  multipart/mixed
>
> Contact: "Anonymous" <sip:Restricted at 6.6.19.72:5060;transport=udp>
>
> Anonymity: uri
>
> Supported: timer,100rel
>
> Session-Expires: 1800
>
> Min-SE: 1800
>
> Content-Length: 287
>
> Content-Disposition: session; handling=required
>
> Content-Type: application/sdp
>
>
>
> v=0
>
> o=Sonus_UAC 145400 14540000 IN IP4 62.62.96.28
>
> s=SIP Media Capabilities
>
> c=IN IP4 62.62.96.28
>
> t=0 0
>
> m=audio 43996 RTP/AVP 0 18 100
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:18 G729/8000
>
> a=fmtp:18 annexb=no
>
> a=rtpmap:100 telephone-event/8000
>
> a=fmtp:100 0-15
>
> a=sendrecv
>
> a=maxptime:20
>
> ==================================================================
>
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
> http://sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
>
>


-- 
Pierre Pebay

mailto:pierre.pebay at gmail.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://sipforum.org/pipermail/discussion/attachments/20081217/fdfd5568/attachment-0002.html>


More information about the discussion mailing list