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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/3447/">https://reviewboard.asterisk.org/r/3447/</a>
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<p style="margin-top: 0;">On April 17th, 2014, 6:58 a.m. CDT, <b>Matt Jordan</b> wrote:</p>
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<pre style="white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">Quoting Pavel here, who sent this e-mail to the dev list. I'd hate for his comment to get missed here:
{quote}
Hi!
PLEASE, don't touch the screening indicator (screen=xxx). It has NOTHING to do with presentation of the number. It should be setup by the dialplan to reflect, whether the number is verified by the exchange and it's valid (i.e. belongs to a person , who sent it) or not. When the number comes from the trunk (again, it's on the dialplan logic to recognize, that the connection is categorized as a trunk), it should be kept intact, i.e. the value coming from the trunk must be preserved for the outgoing connection. Forcing the screening indicator direclty from the chan_sip without any preceding knowledge is really a BAD_THING.
With regards,
Pavel
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<pre style="white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">Well, I had no plans of making any changes to how party/privacy/screen are set, so he can rest easy on that one.</pre>
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<p>- Jonathan</p>
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<p>On April 16th, 2014, 4:23 p.m. CDT, Jonathan Rose wrote:</p>
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<div>Review request for Asterisk Developers, Joshua Colp, Matt Jordan, Mark Michelson, and wdoekes.</div>
<div>By Jonathan Rose.</div>
<p style="color: grey;"><i>Updated April 16, 2014, 4:23 p.m.</i></p>
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<b style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Bugs: </b>
<a href="https://issues.asterisk.org/jira/browse/AST-1301">AST-1301</a>,
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19465">ASTERISK-19465</a>
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<b style="color: #575012; font-size: 10pt;">Repository: </b>
Asterisk
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Description </h1>
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<pre style="margin: 0; padding: 0; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">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.</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Testing </h1>
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<pre style="margin: 0; padding: 0; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">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</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
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<li>/branches/1.8/configs/sip.conf.sample <span style="color: grey">(412438)</span></li>
<li>/branches/1.8/channels/sip/include/sip.h <span style="color: grey">(412438)</span></li>
<li>/branches/1.8/channels/chan_sip.c <span style="color: grey">(412438)</span></li>
<li>/branches/1.8/CHANGES <span style="color: grey">(412438)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/3447/diff/" style="margin-left: 3em;">View Diff</a></p>
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