<p>Yeah, now that is the the param that gave me headache as well. Contact mr off-list if u need offline message support. I carved a perl and lua config to queue messages if leg is not registered and deliver as soon as it reappears. Didnt find a stock way to do it.</p>

<div class="gmail_quote">On Mar 8, 2013 4:12 PM, &quot;mehroz&quot; &lt;<a href="mailto:mehroz.ashraf85@gmail.com">mehroz.ashraf85@gmail.com</a>&gt; wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Thanks Daniel.<br>
<br>
There is an update that i am having Secure signaling TLS. and in oder to<br>
deal with NAT issues , i had to put<br>
sip-force-contact=NDLB-tls-connectile-dysfunction to force rport and TLS<br>
issues. With this configuration chat does not work, BUT i just saw that* it<br>
works* when i remove the sip-force-contact parameter.<br>
:(<br>
<br>
Still for your observation, here is my profile<br>
<br>
Name                    198.84.61.52<br>
Domain Name             N/A<br>
Alias Of                internal<br>
Auto-NAT                false<br>
DBName                  sofia_reg_internal<br>
Pres Hosts              198.84.61.52,198.84.61.52<br>
Dialplan                XML<br>
Context                 rokacomm<br>
Challenge Realm         auto_from<br>
RTP-IP                  198.84.61.52<br>
SIP-IP                  198.84.61.52<br>
URL                     <a href="http://sip:mod_sofia@198.84.61.52:6000" target="_blank">sip:mod_sofia@198.84.61.52:6000</a><br>
BIND-URL                <a href="http://sip:mod_sofia@198.84.61.52:6000" target="_blank">sip:mod_sofia@198.84.61.52:6000</a><br>
TLS-URL                 <a href="http://sip:mod_sofia@198.84.61.52:5061" target="_blank">sip:mod_sofia@198.84.61.52:5061</a><br>
TLS-BIND-URL            sips:mod_sofia@198.84.61.52:5061;transport=tls<br>
HOLD-MUSIC              local_stream://moh<br>
OUTBOUND-PROXY          N/A<br>
CODECS IN               G722,PCMU,PCMA,GSM,AMR,H263,H264,H263-1998,H263-2000<br>
CODECS OUT              G722,PCMU,PCMA,GSM,AMR,H263,H264,H263-1998,H263-2000<br>
TEL-EVENT               101<br>
DTMF-MODE               rfc2833<br>
CNG                     13<br>
SESSION-TO              0<br>
MAX-DIALOG              0<br>
NOMEDIA                 false<br>
LATE-NEG                true<br>
PROXY-MEDIA             false<br>
ZRTP-PASSTHRU           true<br>
AGGRESSIVENAT           false<br>
STUN-ENABLED            true<br>
STUN-AUTO-DISABLE       false<br>
CALLS-IN                20<br>
FAILED-CALLS-IN         4<br>
CALLS-OUT               10<br>
FAILED-CALLS-OUT        4<br>
REGISTRATIONS           3<br>
<br>
and 9999 profile as:<br>
Call-ID:        c6dfab23429f90be4a57c6c39a3fd3ef@0:0:0:0:0:0:0:0<br>
User:           <a href="mailto:9999@198.84.61.52">9999@198.84.61.52</a><br>
Contact:        &quot;9999&quot;<br>
&lt;sip:9999@175.110.122.81:6000;transport=udp;registering_acc=198_84_61_52;fs_nat=yes&gt;<br>
Agent:          Jitsi1.0-build.3967Windows 7<br>
Status:         Registered(AUTO-NAT)(unknown) EXP(2013-03-08 15:07:12)<br>
EXPSECS(87)<br>
Host:           rokacomdev1<br>
IP:             175.110.122.81<br>
Port:           6000<br>
Auth-User:      9999<br>
Auth-Realm:     198.84.61.52<br>
MWI-Account:    <a href="mailto:9999@198.84.61.52">9999@198.84.61.52</a><br>
<br>
Anything else you need to see?<br>
<br>
But that is sure , the issue is due to sip_force_contact parameter in<br>
directory!<br>
<br>
<br>
<br>
--<br>
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