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Hi<br>
<br>
I've seen exactly the same behaviour and even used gdb breakpoints
to understand why is this happening (the only mention-worthy
difference in SIP/SDP between INVITE and re-INVITE was the ;tag
added to To: header)<br>
<br>
Unfortunately, I did not save the results, but if I remember
correctly, that happened simply because a channel was added to a
bridge, and bridge was calling "update_connectedline" function on
every of the channels involved (including the newly added channel
itself) <br>
<br>
That was the most basic case we did with ARI, so we were a little
surprised of course, but somehow we've decided that this is "how ARI
works" so we stopped further research on this. <br>
<br>
Kirill<br>
<br>
<div class="moz-cite-prefix">26.04.2016 21:57, Nitesh Bansal пишет:<br>
</div>
<blockquote
cite="mid:CAOLsin7PQn9hTUHEjQoA4i54OA-ma9K4qJ_yh63VpFSQhc+M9A@mail.gmail.com"
type="cite">
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<div>
<div>Hi,<br>
<br>
</div>
c-line in SDP remains the same, only SDP version in the
o-line changes.<br>
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<br>
</div>
<div>Thanks,<br>
</div>
Nitesh<br>
</div>
<div class="gmail_extra"><br>
<div class="gmail_quote">On Tue, Apr 26, 2016 at 4:45 PM, Joshua
Colp <span dir="ltr"><<a moz-do-not-send="true"
href="mailto:jcolp@digium.com" target="_blank">jcolp@digium.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex"><span
class="">Nitesh Bansal wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">
Hello,<br>
<br>
I'm building an ARI based conference with Asterisk 13.<br>
<br>
Scenario:<br>
Peer A dials into Asterisk, mixing bridge is created and
channel 1 put<br>
into the bridge.<br>
Asterisk is also told to initiate call to a recording
server, so<br>
recording server is<br>
also added into the bridge.<br>
I have noticed that after the initial INVITE completes
with the<br>
Recording Server,<br>
Asterisk is doing a REINVITE towards Recording server,
this REINVITE has the<br>
same media IP, media port though SDP version number
increases.<br>
<br>
I'm really curious why is Asterisk sending this REINVITE
on the outbound<br>
leg to<br>
the Recording server.<br>
Any logical rational for doing that?<br>
</blockquote>
<br>
</span>
Is it updating connected line information?<span
class="HOEnZb"><font color="#888888"><br>
<br>
-- <br>
Joshua Colp<br>
Digium, Inc. | Senior Software Developer<br>
445 Jan Davis Drive NW - Huntsville, AL 35806 - US<br>
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rel="noreferrer" target="_blank"><a class="moz-txt-link-abbreviated" href="http://www.asterisk.org">www.asterisk.org</a></a><br>
<br>
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