Thank you.<br><br clear="all"> <b style="color:rgb(153,153,153)"><span style="color:rgb(153,153,153)"><font><font><span style="font-family:verdana,sans-serif"><span style="font-family:trebuchet ms,sans-serif"><span style="font-family:verdana,sans-serif">José Pablo Méndez</span></span></span></font><br>
</font></span></b><b style="color:rgb(153,153,153)"><span style="color:rgb(153,153,153)"></span></b><b style="color:rgb(153,153,153)"><span style="color:rgb(153,153,153)"></span></b><b style="color:rgb(153,153,153)"><span style="color:rgb(153,153,153)"></span></b><b style="color:rgb(153,153,153)"><span style="color:rgb(153,153,153)"></span></b><br>
<br><br><div class="gmail_quote">On Sun, Dec 18, 2011 at 8:23 PM, Kevin P. Fleming <span dir="ltr"><<a href="mailto:kpfleming@digium.com">kpfleming@digium.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div class="im">On 12/18/2011 01:22 PM, José Pablo Méndez Soto wrote:<br>
<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Embarrassingly enough, I just tried the nat=no again both in the<br>
general and peer sections and the blessed phone registered.... My<br>
apologies, again, I wrote the thread late at night probably this blinded me.<br>
</blockquote>
<br></div>
No problem, we've all done that :-)<br>
<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div class="im">
Now, one question about a previous answer from you ("It is exactly that;<br>
'force_rport' is now the default....."):<br>
<br>
is the trigger for using the source UDP port from the REGISTER, inside<br>
the rport field and inside the destination UDP port of the 200 OK:<br>
<br></div>
1. The mismatch between the UDP source port of the REGISTER and the VIA<br>
port? Or<br>
2. The fact that the other entity sends an empty rport?<br>
3. Or any of the above?<div class="im"><br>
<br>
Its a difficult question to ask/describe, so if I am not asking<br>
correctly please let me know. Thanks a lot, really.<br>
</div></blockquote>
<br>
Not at all. The trigger for Asterisk to respond to the port that the request was sent from, instead of the port listed in the top-most Via header, is *exactly* 'force_rport'. This causes Asterisk to behave as if the 'rport' parameter was included in the top-most Via header, which would be an explicit request from the sending UA for Asterisk to respond to the sending port (and also report back what the sending port was, but that's not part of the problem here).<br>
<br>
So, if the sending UA include an empty 'rport' parameter in its top-most Via header, *or* if the Asterisk has been told to act as if one had been included even if it wasn't, then Asterisk will respond to the perceived sending port; otherwise, it will respond to the port listed in the top-most Via header.<br>
<br>
As far as we know from our research before making this change, the Cisco phones in question are the only ones that send their requests from one port and require the responses to go back to a different port. All other phones that we (and others) use with Asterisk use the same port for both, which makes them quite easy to use behind NAT devices. The Cisco phone models you are dealing with won't work behind a NAT device unless that NAT device has a 'helper' that understands SIP and can fix up this situation (and of course many Cisco phone users have Cisco routers that do exactly this).<div class="HOEnZb">
<div class="h5"><br>
<br>
-- <br>
Kevin P. Fleming<br>
Digium, Inc. | Director of Software Technologies<br>
Jabber: <a href="mailto:kfleming@digium.com" target="_blank">kfleming@digium.com</a> | SIP: <a href="mailto:kpfleming@digium.com" target="_blank">kpfleming@digium.com</a> | Skype: kpfleming<br>
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br>
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