what type of NAT do you have? sync? full cone? cone restricted, port restricted?<br>
any messages in asterisk verbose console?<br>
<br>
best regards<br><br><div><span class="gmail_quote">On 12/7/05, <b class="gmail_sendername">chawki hammoud</b> <<a href="mailto:cyhammoud@yahoo.com">cyhammoud@yahoo.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Hi list:<br>i have an asterisk box behind the NAT ,when i try to<br>send calls through Sip to the voip provider server the<br>call is answered but in a one way calling,I hear the<br>voice of the other side just for 4 seconds and then
<br>stop but the call do not hangup.<br><br>my sip.conf is:<br>[voip provider]<br>type=peer<br>host=<a href="http://213.112.50.8">213.112.50.8</a><br>username=XXXXXXX<br>secret=XXXXXX<br>fromuser=XXXXXXX<br>canreinvite=no
<br>nat=yes<br>insercure=invite<br>disallow=all<br>allow=gsm<br><br><br><br><br>__________________________________________<br>Yahoo! DSL – Something to write home about.<br>Just $16.99/mo. or less.<br><a href="http://dsl.yahoo.com">
dsl.yahoo.com</a><br><br>_______________________________________________<br>--Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com</a> --<br><br>Asterisk-Users mailing list<br>To UNSUBSCRIBE or update options visit:
<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></blockquote></div><br><br clear="all"><br>-- <br>"Su nombre es GNU/Linux, no solamente Linux, mas info en
<a href="http://www.gnu.org">http://www.gnu.org</a>"