<div dir="ltr">See <a href="http://wiki.freeswitch.org/wiki/NDLB">http://wiki.freeswitch.org/wiki/NDLB</a><div><br></div><div>Try NDLB-connectile-dysfunction... it'll rewrite the contact header to where the REGISTER packet came from.</div>
</div><div class="gmail_extra"><br><br><div class="gmail_quote">On 7 January 2014 08:38, Mitja <span dir="ltr"><<a href="mailto:unau0@gmx.de" target="_blank">unau0@gmx.de</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hello there,<br>
<br>
Im trying to solve NAT related Problem which is caused by missbehaving<br>
SIP phone.<br>
Up Front: I searched the wiki and the web for all the NAT related<br>
sites/mailling list. I think its a error of the invoxia IPhone docking<br>
station (the missbehaving SIP phone), thus I did not open a jira for it.<br>
<br>
We have a freeswitch server with internet accessibility and a Phone<br>
behind a natting router (Fritz!Box).<br>
FreeSWITCH (1.2.3.4) <-> (10.11.12.13) NAT (192.168.0.1) <-><br>
(192.168.0.2) SIP Phone<br>
<br>
The Phone is able to register and initiate calls. But when someone else<br>
to call that endpoint, its unreachable.<br>
<br>
I did some research and the problem seems to be that the phone registers<br>
with a Contact containing the private IP Adress (192.168.0.2).<br>
<br>
freeswitch@1.2.3.4@internal> sofia status profile internal reg<br>
<br>
Registrations:<br>
=================================================================================================<br>
<br>
Call-ID: aeac7f28-a6a0-4ba1-9d7c-30934924726d<br>
User: <a href="mailto:270@1.2.3.4">270@1.2.3.4</a><br>
Contact: "user" <sip:270@192.168.0.2:52767;ob><br>
Agent: invoxia-lemonvoice-v6.17.4<br>
Status: Registered(UDP)(unknown) EXP(2014-01-06 13:06:35)<br>
EXPSECS(134)<br>
Host: ippbx-border<br>
IP: 10.11.12.13<br>
Port: 52767<br>
Auth-User: 270<br>
Auth-Realm: 1.2.3.4<br>
MWI-Account: <a href="mailto:270@1.2.3.4">270@1.2.3.4</a><br>
<br>
Other devices in the same Setup have their NAT IP [+Port]<br>
(<a href="http://10.11.12.13:52767" target="_blank">10.11.12.13:52767</a>) in their Contact Header which then works fine. In<br>
this setup however the FreeSWITCH tries to send the INVITE Message to<br>
192.168.0.2 and even when a NAT Table on the firewall manages to map it<br>
to <a href="http://10.11.12.13:52767" target="_blank">10.11.12.13:52767</a> the answer from 10.11.12.13 is discarded.<br>
<br>
Normally I use bridge with user/<a href="mailto:270@1.2.3.4">270@1.2.3.4</a> with the dialstring<br>
"{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(internal/${dialed_user}@${dialed_domain})}".<br>
I tried to force freeswitch to use the NAT IP address by calling bridge<br>
with "sofia/internal/sip:270@10.11.12.13:52767;ob" which worked fine.<br>
The IP address is of course not static but if I would be able to extract<br>
the IP information which is stored in sofia registry (see above) I could<br>
arrange my dialplan in a way that would fix that phones missbehaviour.<br>
<br>
So after all that describing text (sorry for that), heres my question:<br>
Is there a way (an API or Dialplan command) to get the IP Information<br>
which is stored the sofia registry.<br>
<br>
Thanks in advance<br>
<br>
Regards,<br>
Mitja<br>
<br>
<br>
<br>
<br>
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</blockquote></div><br></div>