<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0//EN" "http://www.w3.org/TR/REC-html40/strict.dtd"><html><head><meta name="qrichtext" content="1" /><style type="text/css">p, li { white-space: pre-wrap; }</style></head><body style=" font-family:'Lucida Console'; font-size:9pt; font-weight:400; font-style:normal;">On Freitag, 8. Mai 2009, bhrugu mehta wrote:<br>
> I am new to openser.<br>
> I have register two sip user in openser (as register server) and call<br>
> handling in asterisk.<br>
> when 1001 user do a call to 1002 nothing happen.<br>
> call rejected.<br>
> If posible give a sip.conf and extension.conf snap of this scenario.<br>
><br>
> any suggestion?<br>
<p style="-qt-paragraph-type:empty; margin-top:0px; margin-bottom:0px; margin-left:0px; margin-right:0px; -qt-block-indent:0; text-indent:0px; -qt-user-state:0;"><br></p>Hi Bhrugu,<br>
<p style="-qt-paragraph-type:empty; margin-top:0px; margin-bottom:0px; margin-left:0px; margin-right:0px; -qt-block-indent:0; text-indent:0px; -qt-user-state:0;"><br></p>you could also add some "xlog" statements (take a look to the xlog module documentation how to use this function) to your configuration, and then take a look to the log file during call routing to get an idea how the message is routed and finally rejected.<br>
<p style="-qt-paragraph-type:empty; margin-top:0px; margin-bottom:0px; margin-left:0px; margin-right:0px; -qt-block-indent:0; text-indent:0px; -qt-user-state:0;"><br></p>Cheers,<br>
<p style="-qt-paragraph-type:empty; margin-top:0px; margin-bottom:0px; margin-left:0px; margin-right:0px; -qt-block-indent:0; text-indent:0px; -qt-user-state:0;"><br></p>Henning</p></body></html>