<div>Hi,</div>
<div> </div>
<div>Thanks for that cool info. It will help me in the days to come</div>
<div> </div>
<div>Great going...</div>
<div> </div>
<div>Dan<br><br> </div>
<div><span class="gmail_quote">On 25/01/06, <b class="gmail_sendername">hugolivude</b> <<a href="mailto:hugolivude@gmail.com">hugolivude@gmail.com</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">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 onclick="return top.js.OpenExtLink(window,event,this)" href="http://10.0.0.1/" target="_blank">
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 onclick="return top.js.OpenExtLink(window,event,this)" href="http://10.0.0.2/" target="_blank">
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 onclick="return top.js.OpenExtLink(window,event,this)" href="http://10.0.0.3/" target="_blank">
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 onclick="return top.js.OpenExtLink(window,event,this)" href="http://10.0.0.1/" target="_blank">10.0.0.1</a> and </span><br style="COLOR: rgb(255,0,0)"><span style="COLOR: rgb(255,0,0)">
8001-8010 -> <a onclick="return top.js.OpenExtLink(window,event,this)" href="http://10.0.0.1/" target="_blank">10.0.0.1</a></span><br style="COLOR: rgb(255,0,0)"><span style="COLOR: rgb(255,0,0)">5061 -> <a onclick="return top.js.OpenExtLink(window,event,this)" href="http://10.0.0.2/" target="_blank">
10.0.0.2</a> and 8011-8020 -> <a onclick="return top.js.OpenExtLink(window,event,this)" href="http://10.0.0.2/" target="_blank">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 onclick="return top.js.OpenExtLink(window,event,this)" href="http://www.openwrt.org/" target="_blank">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>
<div><span class="e" id="q_108ff2443ae20dca_1"><br style="COLOR: rgb(255,0,0)"><br>On 1/24/06, David Thomas <<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:punknow@gmail.com" target="_blank">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 onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:pavel.jezek@i.cz" target="_blank">
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 onclick="return top.js.OpenExtLink(window,event,this)" href="http://easynews.com/" target="_blank">
Easynews.com</a> --<br>> ><br>> > Asterisk-Users mailing list<br>> > To UNSUBSCRIBE or update options visit:<br>> > <a onclick="return top.js.OpenExtLink(window,event,this)" href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">
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