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This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3447/
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(Updated April 21, 2014, 10:25 a.m.)
Status
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This change has been marked as submitted.
Review request for Asterisk Developers, Joshua Colp, Matt Jordan, Mark
Michelson, and wdoekes.
Changes
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Committed in revision 412744
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
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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
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/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
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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
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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:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.24.18.246:5060;branch=z9hG4bK2fb42910;rport
Max-Forwards: 70
From: "Anonymous" <sip:[email protected]>;tag=as13075548
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]: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:[email protected]>
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:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.24.18.246:5060;branch=z9hG4bK5d4a7db8;rport
Max-Forwards: 70
From: "Anonymous" <sip:[email protected]>;tag=as181a14e3
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]: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:[email protected]>
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
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