----------------------------------------------------------- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3447/ -----------------------------------------------------------
Review request for Asterisk Developers, Joshua Colp, Matt Jordan, Mark Michelson, and wdoekes. Bugs: AST-1301 https://issues.asterisk.org/jira/browse/AST-1301 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 412331 /branches/1.8/channels/chan_sip.c 412331 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