Guys,<br><br>Have a look for my posting:<br><br>"How to keep Asterisk (1.2) out of the media path"<br><br>A
gentleman named Tony Jago provided some awesome info. I've
posted it below, but you might want to look at my posting for context:<br><br>1) Could someone confirm that I'll need to have canreinvite=yes in<br>sip.conf for both the Xlite and the Polycoms in order to bypass * from<br>
the media path?<br><br><span style="color: rgb(255, 0, 0);">This is correct.</span><br><br>2) Does the Polycom & XLite support reinvite?<br><br><span style="color: rgb(255, 0, 0);">I believe so.</span><br><br>3) Does reinvite work if you're behind a nat?
i.e. if I have nat=yes,<br>does this mean I _have_ to have canreinvite=no?<br><br><span style="color: rgb(255, 0, 0);">No. You need to have the nat set up correctly. This means you need to put</span><br style="color: rgb(255, 0, 0);">
<span style="color: rgb(255, 0, 0);">port forwarding rules in for each and every phone for it's sip and rtp</span><br style="color: rgb(255, 0, 0);"><span style="color: rgb(255, 0, 0);">ports. This means you will have to reconfigure each phone to use a different
</span><br style="color: rgb(255, 0, 0);"><span style="color: rgb(255, 0, 0);">port. eg.</span><br style="color: rgb(255, 0, 0);"><br style="color: rgb(255, 0, 0);"><span style="color: rgb(255, 0, 0);">phone <a href="http://10.0.0.1">
10.0.0.1</a>. SIP port 5060 and RTP ports 8001-8010.</span><br style="color: rgb(255, 0, 0);"><span style="color: rgb(255, 0, 0);">phone <a href="http://10.0.0.2">10.0.0.2</a> SIP port 5061 and RTP ports 8011-8020.</span>
<br style="color: rgb(255, 0, 0);"><span style="color: rgb(255, 0, 0);">phone <a href="http://10.0.0.3">10.0.0.3</a>. SIP port 5062 and RTP ports 8021-8030.</span><br style="color: rgb(255, 0, 0);"><span style="color: rgb(255, 0, 0);">
etc etc.</span><br style="color: rgb(255, 0, 0);"><br style="color: rgb(255, 0, 0);"><span style="color: rgb(255, 0, 0);">on your firewall, you need to map incoming ports 5060 -> <a href="http://10.0.0.1">10.0.0.1</a> and
</span><br style="color: rgb(255, 0, 0);"><span style="color: rgb(255, 0, 0);">8001-8010 -> <a href="http://10.0.0.1">10.0.0.1</a></span><br style="color: rgb(255, 0, 0);"><span style="color: rgb(255, 0, 0);">5061 ->
<a href="http://10.0.0.2">10.0.0.2</a> and 8011-8020 -> <a href="http://10.0.0.2">10.0.0.2</a> etc etc.</span><br style="color: rgb(255, 0, 0);"><br style="color: rgb(255, 0, 0);"><span style="color: rgb(255, 0, 0);">You need to turn on NAT support on each phone.
</span><br style="color: rgb(255, 0, 0);"><br style="color: rgb(255, 0, 0);"><span style="color: rgb(255, 0, 0);">What you are doing here is allowing each and every phone to work in its own</span><br style="color: rgb(255, 0, 0);">
<span style="color: rgb(255, 0, 0);">right across the NAT gateway. After you have finished. Each and every phone</span><br style="color: rgb(255, 0, 0);"><span style="color: rgb(255, 0, 0);">should be able to make and receive calls from anywhere on the internet
</span><br style="color: rgb(255, 0, 0);"><span style="color: rgb(255, 0, 0);">(without going through Asterisk).</span><br style="color: rgb(255, 0, 0);"><br style="color: rgb(255, 0, 0);"><span style="color: rgb(255, 0, 0);">
At this point, if you sacrifice a few chickens and a walrus you may get it</span><br style="color: rgb(255, 0, 0);"><span style="color: rgb(255, 0, 0);">all to work.</span><br><br>Finally:<br><br>I have a suspicion that using a NAT router will prevent me from
<br>eliminating Asterisk from the media path. I am currently running a<br>Linksys WRT54G with Talisman to get QOS. Any recommendations for an<br>alternate QOS router? Ideally it will also support multiple<br>sub-domains...
<br><br><span style="color: rgb(255, 0, 0);">You can do all sorts of stuff with your WRT54G. Running openser on your</span><br style="color: rgb(255, 0, 0);"><span style="color: rgb(255, 0, 0);">WRT54G could in theory do what your looking for. There are plenty of WRT54G
</span><br style="color: rgb(255, 0, 0);"><span style="color: rgb(255, 0, 0);">firmwares that let you do nifty VoIP things. You can even install asterisk</span><br style="color: rgb(255, 0, 0);"><span style="color: rgb(255, 0, 0);">
on your WRT54G. Check out <a href="http://www.openwrt.org">www.openwrt.org</a></span><br style="color: rgb(255, 0, 0);"><br style="color: rgb(255, 0, 0);"><span style="color: rgb(255, 0, 0);">Hope this is some help.</span>
<br style="color: rgb(255, 0, 0);"><br style="color: rgb(255, 0, 0);"><span style="color: rgb(255, 0, 0);">PS: I found a bug in asterisk's re-invite code that in some cases makes</span><br style="color: rgb(255, 0, 0);"><span style="color: rgb(255, 0, 0);">
asterisk push out an invalid SIP packet. If you see anything like this, let</span><br style="color: rgb(255, 0, 0);"><span style="color: rgb(255, 0, 0);">me know and I can send you the patch.</span><br style="color: rgb(255, 0, 0);">
<br>On 1/24/06, David Thomas <<a href="mailto:punknow@gmail.com">punknow@gmail.com</a>> wrote:<br>> That is the way way SER works. I too am very interested to know if<br>> this can be done with Asterisk.<br>>
<br>> David<br>> <br>> On 1/12/06, Pavel Jezek <<a href="mailto:pavel.jezek@i.cz">pavel.jezek@i.cz</a>> wrote:<br>> > Hi, I have asterisk on public IP and phones in two locations behind<br>> > firewall/nat,
<br>> > - when I have nat=yes and canreinvite=no, this is working fine, but rtp<br>> > stream must go _always_ through asterisk, even if phones talk inside<br>> > their locations<br>> > - when I have nat=yes and canreinvite=yes, phones can speak only inside
<br>> > their location and rtp stream is connected directly between phones (this<br>> > is, imho, correct and logical), but,<br>> > is possible to combine both, so do reinvite only "within" e.g
. one<br>> > context and disable reinvite when connecting phones between two context,<br>> > or any better option exist/planned how to solve?<br>> > thanks<br>> > PJ<br>> > _______________________________________________
<br>> > --Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com</a> --<br>> ><br>> > Asterisk-Users mailing list<br>> > To UNSUBSCRIBE or update options visit:<br>> >
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>> ><br>> _______________________________________________<br>> --Bandwidth and Colocation provided by
<a href="http://Easynews.com">Easynews.com</a> --<br>> <br>> Asterisk-Users mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users
</a><br>> <br>