-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3447/#review11659
-----------------------------------------------------------
So going with the legacy idea wdoekes posted, I've updated the expectations
charts:
sendrpid=pai
| pres=allowed |
pres=prohibited ┃
-------------------------+--------------------------------------+----------------------------------------------------┨
trust_id_outbound=legacy | PAI: "123" <sip:1...@xxx.xxx.xxx.xxx> | PAI:
"anonymous" <sip:anonymous@anonymous.invalid> ┃
-------------------------+--------------------------------------+----------------------------------------------------┨
trust_id_outbound=no | PAI: "123" <sip:1...@xxx.xxx.xxx.xxx> |
┃
-------------------------+--------------------------------------+----------------------------------------------------┨
trust_id_outbound=yes | PAI: "123" <sip:1...@xxx.xxx.xxx.xxx> | PAI: "123"
<sip:1...@xxx.xxx.xxx.xxx>, Privacy: id ┃
─────────────────────────┴──────────────────────────────────────┴────────────────────────────────────────────────────┚
sendrpid=rpid
| pres=allowed
| pres=prohibited
┃
-------------------------+--------------------------------------------------------------------------------------+------------------------------------------------------------------------------------------┨
trust_id_outbound=legacy | Remote-Party-ID: "123"
<sip:1...@xxx.xxx.xxx.xxx>;party=calling;privacy=off;screen=no |
Remote-Party-ID: "123"
<sip:123@anonymous.invalid>;party=calling;privacy=full;screen=yes ┃
-------------------------+--------------------------------------------------------------------------------------+------------------------------------------------------------------------------------------┨
trust_id_outbound=no | Remote-Party-ID: "123"
<sip:1...@xxx.xxx.xxx.xxx>;party=calling;privacy=off;screen=no |
┃
-------------------------+--------------------------------------------------------------------------------------+------------------------------------------------------------------------------------------┨
trust_id_outbound=yes | Remote-Party-ID: "123"
<sip:1...@xxx.xxx.xxx.xxx>;party=calling;privacy=off;screen=no |
Remote-Party-ID: "123"
<sip:1...@xxx.xxx.xxx.xxx>;party=calling;privacy=full;screen=yes ┃
─────────────────────────┴──────────────────────────────────────────────────────────────────────────────────────┴──────────────────────────────────────────────────────────────────────────────────────────┚
default setting of trust_id_outbound will be legacy for 1.8-12
no for trunk
I hope this is all correct...
- Jonathan Rose
On April 16, 2014, 4:23 p.m., Jonathan Rose wrote:
>
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/3447/
> -----------------------------------------------------------
>
> (Updated April 16, 2014, 4:23 p.m.)
>
>
> Review request for Asterisk Developers, Joshua Colp, Matt Jordan, Mark
> Michelson, and wdoekes.
>
>
> Bugs: AST-1301 and ASTERISK-19465
> https://issues.asterisk.org/jira/browse/AST-1301
> https://issues.asterisk.org/jira/browse/ASTERISK-19465
>
>
> Repository: Asterisk
>
>
> Description
> -------
>
> Walter Doekes pointed out that this might cause a less than ideal situation
> in which people who were expecting P-Asserted-Identity not to disclose party
> information will now be sending privacy information, so I pulled this patch
> from 1.8-trunk and we will now review it here.
>
> Without this patch, P-Asserted-Identity would always use anonymous for the
> caller ID information, and RFC-3325 seems to indicate that
> P-Asserted-Identity is something that should not be anonymized, but also only
> sent to trusted parties. The way this was presented to me, the intent here is
> that if you set callerpres to prohibited for a peer that receives
> P-Asserted-Identity, the P-Asserted-Identity shouldn't be anonymized, only
> the normal From/Contact headers would be anonymized. This apparently
>
> The obvious method for dealing with this mid-release change is to make the
> change into an option which defaults off in 1.8-12 while defaulting on in
> trunk. Also I'll need to add Upgrade notes for trunk since this might not
> always be a desired behavior as well as CHANGES notes throughout to indicate
> the new option if that's what we settle on.
>
>
> Diffs
> -----
>
> /branches/1.8/configs/sip.conf.sample 412438
> /branches/1.8/channels/sip/include/sip.h 412438
> /branches/1.8/channels/chan_sip.c 412438
> /branches/1.8/CHANGES 412438
>
> Diff: https://reviewboard.asterisk.org/r/3447/diff/
>
>
> Testing
> -------
>
> Call from SIP peer A to SIP peer B
> settings for both peers:
> sendrpid = pai
> callerpres = prohib
>
>
> Invite sent from Asterisk to the recipient of the call
> ------------------------------------------------------
> Prior to patch:
>
> Audio is at 19640
> Adding codec 0x4 (ulaw) to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
> Reliably Transmitting (NAT) to 10.24.18.240:5060:
> INVITE sip:123@10.24.18.240:5060 SIP/2.0
> Via: SIP/2.0/UDP 10.24.18.246:5060;branch=z9hG4bK2fb42910;rport
> Max-Forwards: 70
> From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as13075548
> To: <sip:123@10.24.18.240:5060>
> Contact: <sip:anonymous@10.24.18.246:5060>
> Call-ID: 762b8a5e5848d7997f38f71a770d4dd9@10.24.18.246:5060
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX SVN-branch-1.8-r410380
> Date: Tue, 11 Mar 2014 22:59:39 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH
> Supported: replaces, timer
> P-Asserted-Identity: "Anonymous" <sip:anonymous@anonymous.invalid>
> Content-Type: application/sdp
> Content-Length: 276
>
> v=0
> o=root 473543868 473543868 IN IP4 10.24.18.246
> s=Asterisk PBX SVN-branch-1.8-r410380
> c=IN IP4 10.24.18.246
> t=0 0
> m=audio 19640 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
>
> After patch:
>
> Audio is at 11822
> Adding codec 0x4 (ulaw) to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
> Reliably Transmitting (NAT) to 10.24.18.240:5060:
> INVITE sip:123@10.24.18.240:5060 SIP/2.0
> Via: SIP/2.0/UDP 10.24.18.246:5060;branch=z9hG4bK5d4a7db8;rport
> Max-Forwards: 70
> From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as181a14e3
> To: <sip:123@10.24.18.240:5060>
> Contact: <sip:anonymous@10.24.18.246:5060>
> Call-ID: 721bef28208f7633288e929c6e88824e@10.24.18.246:5060
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX SVN-branch-1.8-r410380M
> Date: Tue, 11 Mar 2014 22:57:39 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH
> Supported: replaces, timer
> P-Asserted-Identity: "Goldy Locks" <sip:6018@10.24.18.246>
> Privacy: id
> Content-Type: application/sdp
> Content-Length: 279
>
> v=0
> o=root 1606369071 1606369071 IN IP4 10.24.18.246
> s=Asterisk PBX SVN-branch-1.8-r410380M
> c=IN IP4 10.24.18.246
> t=0 0
> m=audio 11822 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
>
> Thanks,
>
> Jonathan Rose
>
>
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-dev
