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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 16th, 2014, 6:02 p.m. CDT, <b>Joshua Colp</b> wrote:</p>
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<a href="https://reviewboard.asterisk.org/r/3447/diff/2/?file=57476#file57476line11420" style="color: black; font-weight: bold; text-decoration: underline;">/branches/1.8/channels/chan_sip.c</a>
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(Diff revision 2)
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<td colspan="4"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">static int add_rpid(struct sip_request *req, struct sip_pvt *p)</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span>if (ast_test_flag(&p->flags[1], SIP_PAGE2_TRUST_ID_OUTBOUND)) {</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="tb"> </span>/* If we trust, always use the true IP */</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="tb"> </span>fromdomain = ast_sockaddr_stringify_host_remote(&p->ourip);</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span>} else {</pre></td>
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<td bgcolor="#ffffff" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span>fromdomain = S_OR(p->fromdomain, ast_sockaddr_stringify_host_remote(&p->ourip));</pre></td>
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<td bgcolor="#ffffff" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="tb"> </span>fromdomain = S_OR(p->fromdomain, ast_sockaddr_stringify_host_remote(&p->ourip));</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span>}</pre></td>
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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;">If a legit fromdomain is specified you have to use it, or else you've just ignored the fromdomain configuration option.</pre>
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<pre style="margin-left: 1em; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">Hmmm. Since that's the case I need to figure out why I was seeing anonymous strings here with my test. This scares me a bit since I'm not sure how to deal with the anonymous.invalid address I was seeing in my baseline test for rpid (sendrpid=rpid, callingpres=prohib)
Remote-Party-ID: "123" <sip:123@anonymous.invalid>party=calling;privacy=full;screen=yes
If I need to send a non-anonymous address when using the trust_outbound_id setting, I need a way to get that, and right now it seems like one doesn't exist since the fromdomain will be invalid when callingpres is a prohibited value.</pre>
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<p style="margin-top: 0;">On April 16th, 2014, 6:02 p.m. CDT, <b>Joshua Colp</b> wrote:</p>
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<a href="https://reviewboard.asterisk.org/r/3447/diff/2/?file=57476#file57476line11430" style="color: black; font-weight: bold; text-decoration: underline;">/branches/1.8/channels/chan_sip.c</a>
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(Diff revision 2)
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<td colspan="4"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">static int add_rpid(struct sip_request *req, struct sip_pvt *p)</pre></td>
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<td bgcolor="#fdfebc" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="tb"> </span>if ((lid_pres & AST_PRES_RESTRICTION) <span class="hl">!</span>= AST_PRES_ALLOWED) {</pre></td>
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<td bgcolor="#fdfebc" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="tb"> </span>if ((lid_pres & AST_PRES_RESTRICTION) <span class="hl">=</span>= AST_PRES_ALLOWED) {</pre></td>
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<td bgcolor="#fdfebc" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="tb"> </span><span class="tb"> </span>ast_str_set(&tmp, -1, "<span class="hl">%s</span>", <span class="hl">anonymous_string</span>);</pre></td>
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<td bgcolor="#fdfebc" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="tb"> </span><span class="tb"> </span>ast_str_set(&tmp, -1, "<span class="hl">\"%s\" <sip:%s@%s></span>", <span class="hl">lid_name, lid_num, fromdomain</span>);</pre></td>
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<td bgcolor="#fdfebc" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="tb"> </span>} else {</pre></td>
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<td bgcolor="#fdfebc" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="tb"> </span><span class="tb"> </span>add_header(req, "P-Asserted-Identity", ast_str_buffer(tmp));</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="tb"> </span>} else if (ast_test_flag(&p->flags[1], SIP_PAGE2_TRUST_ID_OUTBOUND)) {</pre></td>
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<td bgcolor="#ffffff" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="tb"> </span><span class="tb"> </span>ast_str_set(&tmp, -1, "\"%s\" <sip:%s@%s>", lid_name, lid_num, fromdomain);</pre></td>
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<td bgcolor="#ffffff" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="tb"> </span><span class="tb"> </span>ast_str_set(&tmp, -1, "\"%s\" <sip:%s@%s>", lid_name, lid_num, fromdomain);</pre></td>
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<td bgcolor="#ffffff" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="tb"> </span>add_header(req, "P-Asserted-Identity", ast_str_buffer(tmp));</pre></td>
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<td bgcolor="#ffffff" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="tb"> </span><span class="tb"> </span>add_header(req, "P-Asserted-Identity", ast_str_buffer(tmp));</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="tb"> </span><span class="tb"> </span>add_header(req, "Privacy", "id");</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="tb"> </span>}</pre></td>
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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;">Previously if the presentation restriction was such that it wasn't allowed... then the anonymous header would get added. Reading this logic I don't see how the header would get added.</pre>
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<pre style="margin-left: 1em; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">The anonymous string would only get used under the following conditions:
1. sendrpid mode was pai
2. the lid_pres value was not an 'AST_PRES_ALLOWED' class of callingpres values.
Going by the chart for pai that wdoekes suggested:
| pres=allowed | pres=prohibited |
----------------------+-------------------------+-----------------------+
trust_id_outbound=no | PAI: 123, Privacy: none | | <-------------- It's this one
----------------------+-------------------------+-----------------------+
trust_id_outbound=yes | PAI: 123, Privacy: none | PAI: 123, Privacy: id |
----------------------+-------------------------+-----------------------+
Then we shouldn't be sending either header in this case, which is why the anonymous string was removed.</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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