Hi Miguel,

I do not understand how the ACL (if you mean apply-nat-acl) can effects on using public IP in SDP when I specified both ext-sip-ip and ext-rtp-ip in sip profile!

I changed the apply-nat-acl to a "always allow" and "rfc1918" but does not make difference.

Afshin

On Thu, Aug 14, 2014 at 3:18 PM, Miguel Oyarzo <miguelaustro@gmail.com> wrote:

Wrong ACL?

On Wed, Aug 13, 2014 at 10:06 PM, afshin afzali <a.afzali2003@gmail.com> wrote:

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

_________________________________________________________________________
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--

"Efficiency is doing things right; effectiveness is doing the right things (Peter Drucker)"

Miguel Oyarzo
DevOps & VoIP Engineer
Linux User: # 483188 - counter.li.org
http://au.linkedin.com/in/mikeaustralia 
Melbourne, Australia

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org