Most probably some NAT issue happens on the client side.<div><br><div>Router is not doing port translation as required.</div><div><br><div>FS replies to a port which is indicated in REGISTER request (correct), however client expects the reply on a different port.</div>
<div><br></div><div>Try to enable <span class="Apple-style-span" style="font-family: monospace; line-height: 11px; white-space: pre; background-color: rgb(255, 255, 255); "><param name="NDLB-force-rport" value="true"</span><span class="Apple-style-span" style="font-family: monospace; line-height: 11px; white-space: pre; background-color: rgb(255, 255, 255); ">/> in the profile.</span></div>
<div><span class="Apple-style-span" style="font-family: monospace; line-height: 11px; white-space: pre; background-color: rgb(255, 255, 255); "><br></span></div><div><a href="http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#NDLB_.28A.K.A._No_device_left_behind.29">http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#NDLB_.28A.K.A._No_device_left_behind.29</a></div>
<div><br></div><div><br></div><div>Please reply if this helped.</div><div><br></div><div>Vitalie<br><br><div class="gmail_quote">2011/11/21 Charlie Orford <span dir="ltr"><<a href="mailto:charlie.orford@attackplan.net">charlie.orford@attackplan.net</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div bgcolor="#FFFFFF" text="#000000">
<font face="Arial">Hello list<br>
<br>
I am an asterisk refugee and currently in the midst of moving our
voip platform across to freeswitch. The goal is to have FS in the
cloud (on a dedicated Linode virtual machine running Debian
Squeeze), with all office phones (Aastra 57i units) connecting via
the public internet.<br>
<br>
FS is compiled and running on the linode machine (using the latest
git build from a week ago). It is setup to listen on the public IP
only so there is no NAT happening at the server end. All relevant
firewall ports are open (tcp/udp 5060, tcp/udp 5080 and udp
16384:32768).<br>
<br>
Because our office net connection has a dynamic IP, we are using
(or trying to use) digest authentication rather than ACLs in order
to control user/extension access to the internal sip profile.<br>
<br>
The problem:<br>
<br>
For some reason, none of our phones are able to successfully
register with FS. Running fs_cli with logging at 7 and enabling </font>"sofia
global siptrace on" shows that the phones make contact and try to
REGISTER but when FS replies with a 401 Unauthorized and requests
the phone authenticate via digest, the phone seems to ignore this
and just repeatedly keeps sending the same original REGISTER request
with no accompanying Authorization header.<br>
<br>
My hunch is that the problem must lie with the phone or our router
rather than FS but I'm a little out of my depth with this problem
and so would appreciate any insight or advice.<br>
<br>
<br>
For a transcript of a failed registration between our FS server and
a phone at the office, please see: <a href="http://pastebin.com/1qRudrvE" target="_blank">http://pastebin.com/1qRudrvE</a>
(note: server and phone ip has been changed to protect the innocent).<br>
<br>
I also have a screen shot of the phone's SIP config here: <span style="font-size:10pt;font-family:"Arial","sans-serif""><a href="http://imgur.com/2lwiN" target="_blank">http://imgur.com/2lwiN</a>
(</span><span>we are running the latest publically available
Aastra firmware on the phones - v3.2.2.56)</span>.<br>
<span><br>
Finally, in case it is relevant, the router at the office is a
Draytek Vigor 2600 ADSL router (about 5 years old now but working
happily as far as we know).<br>
<br>
<br>
Thanks + Regards,<br><font color="#888888">
Charlie<br>
<br>
<br>
</font></span><br>
<br>
<font face="Arial"><br>
</font>
</div>
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