<html>
<body>
<div style="font-family: Verdana, Arial, Helvetica, Sans-Serif;">
<table bgcolor="#f9f3c9" width="100%" cellpadding="8" style="border: 1px #c9c399 solid;">
<tr>
<td>
This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/4476/">https://reviewboard.asterisk.org/r/4476/</a>
</td>
</tr>
</table>
<br />
<pre style="white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">I'm not a big fan of this module for a couple of reasons:
1) The data in the request URI is intended to describe who the call is destined to be sent to, not who the call originated from. There are systems out there where the endpoint names in pjsip.conf match extensions in the dialplan, making it easy to route incoming calls to their destination. Therefore it may be possible for endpoint 100 to dial "200" to call endpoint 200, and have Asterisk think the incoming call is both from 200 and to 200. This can be mitigated through the use of endpoint identifier priorities, but this is not something the average user is expected to have to mess with in their setup.
2) Josh Colp actually already wrote an alternate means of dealing with this situation, and it's already in Asterisk trunk. See this review: https://reviewboard.asterisk.org/r/2964/ In trunk, you can specify "line = yes" in an outbound registration configuration. With this, each outbound registration from Asterisk carries a "line" parameter in the Contact header. This way, when the provider places a call into Asterisk, the To URI will have that line parameter, and we can use that as a way to match the correct endpoint.
So, my thought on this is that the problem you are trying to solve needs to be solved. Since we already have something that solves this issue, but only in trunk, the best way to handle this is to backport the "line = yes" feature from trunk into Asterisk 13.</pre>
<br />
<p>- Mark Michelson</p>
<br />
<p>On March 24th, 2015, 12:23 p.m. UTC, Dmitriy Serov wrote:</p>
<table bgcolor="#fefadf" width="100%" cellspacing="0" cellpadding="8" style="background-image: url('https://reviewboard.asterisk.org/static/rb/images/review_request_box_top_bg.ab6f3b1072c9.png'); background-position: left top; background-repeat: repeat-x; border: 1px black solid;">
<tr>
<td>
<div>Review request for Asterisk Developers, Matt Jordan and rnewton.</div>
<div>By Dmitriy Serov.</div>
<p style="color: grey;"><i>Updated March 24, 2015, 12:23 p.m.</i></p>
<div style="margin-top: 1.5em;">
<b style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Bugs: </b>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24852">ASTERISK-24852</a>
</div>
<div style="margin-top: 1.5em;">
<b style="color: #575012; font-size: 10pt;">Repository: </b>
Asterisk
</div>
<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Description </h1>
<table width="100%" bgcolor="#ffffff" cellspacing="0" cellpadding="10" style="border: 1px solid #b8b5a0">
<tr>
<td>
<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;">The main task: to find the right endpoint for an incoming call with a trunk that has outbound registration.
Simple example (a piece of pjsip.conf):
[trunk1]
type=registration
transport=udp-transport
outbound_auth=trunk1
server_uri=sip:sip.example.com
client_uri=sip:user1@sip.example.com
contact_user=trunk1-in
[trunk2]
type=registration
transport=udp-transport
outbound_auth=trunk2
server_uri=sip:sip.example.com
client_uri=sip:user2@sip.example.com
contact_user=trunk2-in
[trunk1-in]
type=endpoint
context=from-trunk
disallow=all
allow=ulaw
outbound_auth=trunk1
aors=trunk1
[trunk2-in]
type=endpoint
context=from-trunk
disallow=all
allow=ulaw
outbound_auth=trunk2
aors=trunk2
trunk1, trunk2 - outbound registrations to EXTERNAL sip server sip.example.com. One server and two registrations.
; "contact_user=" sets the SIP contact header's user portion of the SIP URI this will affect the extension reached in dialplan when the far end calls you at this registration.
In example option has values "trunk1-in" and "trink2-in".
Case: random user admin@sip.example.com calling to user2@sip.example.com.
External sip server redirect call to my server with packet:
INVITE sip:trunk2-in@8.8.8.8:5060 SIP/2.0
From: "PhonerLite" <sip:admin@sip.example.com>;tag=5F39A540-782390-DE41886A_kmbdctn-8A47
To: <sip:user2@sip.example.com>
This invite can be:
- anonymous. Very bad.
- identified by IP. Config has two endpoints with same IP.
- identified by username From. It cannot identify by "admin" (random)
res_pjsip_endpoint_identifier_request_user helps to identify this invite by uri username in request line (contact_user in registration).</pre>
</td>
</tr>
</table>
<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
<ul style="margin-left: 3em; padding-left: 0;">
<li>/trunk/res/res_pjsip_endpoint_identifier_request_user.c <span style="color: grey">(PRE-CREATION)</span></li>
</ul>
<p><a href="https://reviewboard.asterisk.org/r/4476/diff/" style="margin-left: 3em;">View Diff</a></p>
</td>
</tr>
</table>
</div>
</body>
</html>