Hi FreeSWITCHers:

I configured a sip profile as bellow:

=================================================================================================
Name                    sipinterface_1
Domain Name             N/A
Auto-NAT                false
DBName                  sofia_reg_sipinterface_1
Pres Hosts
Dialplan                XML
Context                 context_1
Challenge Realm         auto_to
RTP-IP                  192.168.10.59
Ext-RTP-IP              X.Y.Z.W
SIP-IP                  192.168.10.59
Ext-SIP-IP               X.Y.Z.W
URL                     sip:mod_sofia@X.Y.Z.W:5060
BIND-URL                sip:mod_sofia@X.Y.Z.W:5060;maddr=192.168.10.59;transport=udp,tcp
HOLD-MUSIC              N/A
OUTBOUND-PROXY          N/A
CODECS IN               G722,PCMU,PCMA,GSM
CODECS OUT              G722,PCMU,PCMA,GSM
TEL-EVENT               101
DTMF-MODE               rfc2833
CNG                     13
SESSION-TO              0
MAX-DIALOG              0
NOMEDIA                 false
LATE-NEG                false
PROXY-MEDIA             false
ZRTP-PASSTHRU           false
AGGRESSIVENAT           true
CALLS-IN                0
FAILED-CALLS-IN         0
CALLS-OUT               0
FAILED-CALLS-OUT        0
REGISTRATIONS           1

for every INVITE that comes in via this profile, FS replys with this SDP:

   v=0
   o=FreeSWITCH 1407905843 1407905844 IN IP4 192.168.10.59
   s=FreeSWITCH
   c=IN IP4 192.168.10.59
   t=0 0
   m=audio 25142 RTP/AVP 0 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20

Although the Contact header of 200 OK is correct :
Contact: <sip:4000@X.Y.Z.W:5060;transport=udp>

Appreciate all comments,

-- Afshin