*67/68 should be for all following calls. *81/82 are per call. After *67,
all calls should have anonymous user name if it is supported.
Richard
-----Original Message-----
From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On
Behalf Of Java Rockx
Sent: Monday, January 24, 2005 3:03 PM
To: ser users
Subject: Re: [Serusers] Grandstream BT100 Shows Caller ID Even With
RPIDprivacy=full
Jiri,
I started down that road of thinking until I realized that having the SIP
proxy alter the FROM
header was a bad thing.
I wish there were some way, from the ser proxy, to guarantee blocking
caller ID for both SIP->PSTN
and SIP->SIP calls.
Our situation is that we need to ship IADs/ATAs to customers preconfigured
and locked down to keep
customers from messing up their settings and to keep them from knowing
their SER
username/password. So we can't allow users to permanently enable caller
id blocking on their ATA.
I guess *67 is my only choice for now, but this has to be dialed every
time a subscriber calls
someone :-(
Regards,
Paul
--- Jiri Kuthan <jiri(a)iptel.org> wrote:
Well, it seems to assume model in which content
of caller id is
intended for use by PSTN gateways that see just the caller id
and don't care about content of From then. So all it does is
it enables privacy in the caller id. I think it would make sense
to use "anonymous" From in that case too.
-jiri
At 10:49 PM 1/24/2005, Java Rockx wrote:
>Hi All.
>
>Does anyone know why my grandstream BT100 firmware 1.0.5.16 shows the
caller
ID rather than
> >"anonymous" even though I have Remote-Part-ID enabled with
>
>Below is the INVITE message sent to the BT100 from a UTstarcom ATA.
>
>Regards,
>Paul
>
>U 2005/01/24 16:43:55.073424 68.84.xxx.xxx:5060 -> 68.84.xxx.xxx:13721
>INVITE sip:3215551212@68.84.xxx.xxx:13721;user=phone SIP/2.0.
>Record-Route: <sip:68.84.xxx.xxx;ftag=2590080990;lr=on>.
>Via: SIP/2.0/UDP 68.84.xxx.xxx;branch=z9hG4bK5082.72060543.0.
>Via: SIP/2.0/UDP
192.168.0.86:5060;rport=1041;received=68.84.xxx.xxx;branch=z9hG4bK14034648
25.
From:
UTStarcom <sip:4075551212@mycompany.com>;tag=2590080990.
To: <sip:3215551212@mycompany.com>.
Call-ID: 1980470039(a)192.168.0.86.
CSeq: 1301 INVITE.
Contact: <sip:4075551212@68.84.xxx.xxx:1041>.
max-forwards: 16.
subject: sipcall.
Allow: INVITE, ACK, CANCEL, BYE, REFER, NOTIFY.
Content-Type: application/sdp.
Content-Length: 178.
Remote-Party-ID: "Anonymous"
<sip:unknown@68.84.xxx.xxx;user=phone>;party=calling;screen=yes;id-
type=subscriber;privacy=full.
.
v=0.
o=- 1363 3077 IN IP4 192.168.0.86.
s=-.
c=IN IP4 68.84.xxx.xxx.
t=0 0.
m=audio 36526 RTP/AVP 18 0 8 2 4 101.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:20.
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http://iptel.org/~jiri/
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