<font size=2 face="sans-serif">David,</font>
<br>
<br><font size=2 face="sans-serif">you obviously have to test for your
situation, but the short answer is that it should. The connection will
start with running through Asterisk, but very quickly the phones will see
that they can talk directly and take the Asterisk server out of the media
path. There are a couple of gotchas that can happen based on your dial
options, so check out this page:</font>
<br><a href="http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite"><font size=3 color=blue><u>http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite</u></font></a><font size=3>
</font>
<br>
<br><font size=2 face="sans-serif">canreinvite was renamed to directmedia
in Asterisk 1.6.2, but the page is still pretty good with regards to the
options that are available.</font>
<br><font size=2 face="sans-serif"><br>
Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208</font>
<br>
<br>
<br>
<br><font size=1 color=#5f5f5f face="sans-serif">From:
</font><font size=1 face="sans-serif">David Wessell <david@ringfree.biz></font>
<br><font size=1 color=#5f5f5f face="sans-serif">To:
</font><font size=1 face="sans-serif">Asterisk Users Mailing
List - Non-Commercial Discussion <asterisk-users@lists.digium.com>,
</font>
<br><font size=1 color=#5f5f5f face="sans-serif">Date:
</font><font size=1 face="sans-serif">04/25/2013 10:49 AM</font>
<br><font size=1 color=#5f5f5f face="sans-serif">Subject:
</font><font size=1 face="sans-serif">Re: [asterisk-users]
Sip and the media path</font>
<br><font size=1 color=#5f5f5f face="sans-serif">Sent by:
</font><font size=1 face="sans-serif">asterisk-users-bounces@lists.digium.com</font>
<br>
<hr noshade>
<br>
<br>
<br><font size=2 face="Calibri">Kevin,</font>
<br>
<br><font size=2 face="Calibri">Thanks for the info. Clarification. The
asterisk server is NOT on the same LAN as the phones. The asterisk server
is in a datacenter only accessible via WAN.</font>
<br>
<br><font size=2 face="Calibri">However, all of the phones are in side
of the same LAN. Will directmedia still function that way?</font>
<br>
<br><font size=2 face="Calibri">Thanks</font>
<br><font size=2 face="Calibri">David</font>
<br>
<br><font size=2 face="Calibri"><b>From: </b>Kevin Larsen <</font><a href=mailto:kevin.larsen@pioneerballoon.com><font size=2 color=blue face="Calibri"><u>kevin.larsen@pioneerballoon.com</u></font></a><font size=2 face="Calibri">><b><br>
Reply-To: </b>Asterisk Users Mailing List - Non-Commercial Discussion <</font><a href="mailto:asterisk-users@lists.digium.com"><font size=2 color=blue face="Calibri"><u>asterisk-users@lists.digium.com</u></font></a><font size=2 face="Calibri">><b><br>
Date: </b>Thursday, April 25, 2013 9:16 AM<b><br>
To: </b>Asterisk Users Mailing List - Non-Commercial Discussion <</font><a href="mailto:asterisk-users@lists.digium.com"><font size=2 color=blue face="Calibri"><u>asterisk-users@lists.digium.com</u></font></a><font size=2 face="Calibri">><b><br>
Subject: </b>Re: [asterisk-users] Sip and the media path</font>
<br>
<br><font size=2 face="sans-serif">You will want to look at the directmedia
option. You will want all the phones on the same lan as the Asterisk server
to be directmedia=yes and the ones on the wan to be directmedia=no. Then,
internal calls will send the media between themselves without involving
Asterisk, but ones outside on the wan will be forced to talk directly to
the Asterisk server for everything. You might also want to look at the
nonat option of directmedia.<br>
<br>
Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208</font><font size=2 face="Calibri">
<br>
<br>
<br>
</font><font size=1 color=#5f5f5f face="sans-serif"><br>
From: </font><font size=1 face="sans-serif">David
Wessell <</font><a href=mailto:david@ringfree.biz><font size=1 color=blue face="sans-serif"><u>david@ringfree.biz</u></font></a><font size=1 face="sans-serif">></font><font size=1 color=#5f5f5f face="sans-serif"><br>
To: </font><font size=1 face="sans-serif">Asterisk
Users Mailing List - Non-Commercial Discussion <</font><a href="mailto:asterisk-users@lists.digium.com"><font size=1 color=blue face="sans-serif"><u>asterisk-users@lists.digium.com</u></font></a><font size=1 face="sans-serif">>,
</font><font size=1 color=#5f5f5f face="sans-serif"><br>
Date: </font><font size=1 face="sans-serif">04/25/2013
07:33 AM</font><font size=1 color=#5f5f5f face="sans-serif"><br>
Subject: </font><font size=1 face="sans-serif">[asterisk-users]
Sip and the media path</font><font size=1 color=#5f5f5f face="sans-serif"><br>
Sent by: </font><a href="mailto:asterisk-users-bounces@lists.digium.com"><font size=1 color=blue face="sans-serif"><u>asterisk-users-bounces@lists.digium.com</u></font></a><font size=2 face="Calibri"><br>
</font>
<hr noshade><font size=2 face="Calibri"><br>
<br>
</font><font size=2 face="Calibri"><br>
We're running asterisk 1.8 in the DC on a public IP address.</font><font size=2 face="Calibri"><br>
</font><font size=2 face="Calibri"><br>
Connecting to it are about 200 phones behind a LAN in a remote location.</font><font size=2 face="Calibri"><br>
</font><font size=2 face="Calibri"><br>
Is there a way to reliably keep asterisk out of the media stream on internal
calls inside that LAN? All phones are Polycom Soundpoint phones.</font><font size=2 face="Calibri"><br>
</font><font size=2 face="Calibri"><br>
Asterisk would say in the media stream for any calls that traverse from
LAN to WAN. However it would step out for LAN to LAN calls.</font><font size=2 face="Calibri"><br>
</font><font size=2 face="Calibri"><br>
Thanks</font><font size=2 face="Calibri"> </font><font size=2 face="Calibri"><br>
David</font><font size=2 face="Calibri"> </font><font size=2 face="Calibri"><br>
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