You need to create an &quot;asterisk&quot; extension in your context.&nbsp; Assuming your callerid is set to your voicemail box, you could use something like:<br><br>exten =&gt; asterisk,1,VoiceMailMain(${CALLERIDNUM}@mb_tutorial)
<br><br>This is assuming you are trying to check your messages via a MWI message.<br><br>Anthony<br><br><br><div><span class="gmail_quote">On 9/14/06, <b class="gmail_sendername">Tanzeel serfaraz</b> &lt;<a href="mailto:tanzeelcs@yahoo.com">
tanzeelcs@yahoo.com</a>&gt; 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 users;<br><br>i am new to use asterisk so i am facing some problems.
<br>i have installed asterisk 1.2.4 and all the<br>requirements.<br>i have to implement the Method 3 of the link.<br><a href="http://www.voip-info.org/wiki/view/Asterisk+at+large">http://www.voip-info.org/wiki/view/Asterisk+at+large
</a>.<br><br>i am doing like that;<br><br>&lt;XLITE&gt;-----------&lt;OPENSER&gt;------------&lt;ASTERISK&gt;<br><a href="http://192.168.1.234">192.168.1.234</a>&nbsp;&nbsp;&nbsp;&nbsp;<a href="http://192.168.1.130">192.168.1.130</a>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <a href="http://192.168.1.131">
192.168.1.131</a><br><br>i have configured voice mail box on asterisk like:<br><br>1:voicemail.conf:<br>[mb_tutorial]<br>777=&gt;1212,ivan,ivan@localhost<br><br>2:sip.conf:<br><br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; [ivan]<br>type=friend
<br>username=ivan<br>secret=1212<br>insecure=yes<br>context=tutorial<br>mailbox=777@mb_tutorial<br>host=dynamic<br><br>3:extensions.conf:<br><br>[tutorial]<br>exten =&gt; 2222,1,Dial(SIP/ivan, 30)<br>exten =&gt; 2222,2,voiceMail(
777@mb_tutorial)<br>exten =&gt; 2222,3,PlayBack(vm-goodbye)<br>exten =&gt; 2222,4,wait(2)<br>exten =&gt; 2222,5,HangUp()<br><br><br>the user(ivan) is registered on my openser.when i m<br>dialling from my xlite ,i have 404 NOT FOUND error on
<br>my xlite.and my asterisk consule is showing like<br>screen like that:<br>NOTIFY <a href="mailto:sip:777@192.168.1.130">sip:777@192.168.1.130</a> SIP/2.0<br>v: SIP/2.0/UDP<br><a href="http://192.168.1.131:5060">192.168.1.131:5060
</a>;branch=z9hG4bK13dc61bb;rport<br>f: &quot;asterisk&quot; &lt;sip:asterisk@localhost&gt;;tag=as04f09bc4<br>t: &lt;<a href="mailto:sip:777@192.168.1.130">sip:777@192.168.1.130</a>&gt;<br>m: &lt;<a href="mailto:sip:asterisk@192.168.1.131">
sip:asterisk@192.168.1.131</a>&gt;<br>i: 4a9471541c2bc1331824afb7560fbe5a@localhost<br>CSeq: 102 NOTIFY<br>User-Agent: Asterisk PBX<br>Max-Forwards: 70<br>o: message-summary<br>c: application/simple-message-summary<br>l: 89
<br><br><br>Messages-Waiting: no<br>Message-Account: sip:asterisk@localhost<br>Voice-Message: 0/0 (0/0)<br><br><br>---<br>Scheduling destruction of call<br>'4a9471541c2bc1331824afb7560fbe5a@localhost' in 15000<br>ms<br><br>
<br>&lt;-- SIP read from <a href="http://192.168.1.130:5060">192.168.1.130:5060</a>:<br>SIP/2.0 404 Not Found<br>v: SIP/2.0/UDP<br><a href="http://192.168.1.131:5060">192.168.1.131:5060</a>;branch=z9hG4bK13dc61bb;rport=5060
<br>f: &quot;asterisk&quot; &lt;sip:asterisk@localhost&gt;;tag=as04f09bc4<br>t:<br>&lt;<a href="mailto:sip:777@192.168.1.130">sip:777@192.168.1.130</a>&gt;;tag=9ce625323ea050e3441580834adb7aaf.7343<br>i: 4a9471541c2bc1331824afb7560fbe5a@localhost
<br>CSeq: 102 NOTIFY<br>Server: OpenSer (1.0.0 (i386/linux))<br>Content-Length: 0<br>Warning: 392 <a href="http://192.168.1.130:5060">192.168.1.130:5060</a> &quot;Noisy feedback tells:<br> pid=4512<br>req_src_ip=<a href="http://192.168.1.131">
192.168.1.131</a> req_src_port=5060<br>in_uri=<a href="mailto:sip:777@192.168.1.130">sip:777@192.168.1.130</a><br>out_uri=<a href="mailto:sip:777@192.168.1.130">sip:777@192.168.1.130</a> via_cnt==1&quot;<br><br><br><br><br>
<br><br>Reliably Transmitting (no NAT) to <a href="http://192.168.1.234:5060">192.168.1.234:5060</a>:<br>NOTIFY sip:ivan@192.168.1.234:5060 SIP/2.0<br>v: SIP/2.0/UDP<br><a href="http://192.168.1.131:5060">192.168.1.131:5060
</a>;branch=z9hG4bK040fcabc;rport<br>f: &quot;asterisk&quot;<br>&lt;<a href="mailto:sip:asterisk@192.168.1.131">sip:asterisk@192.168.1.131</a>&gt;;tag=as0f3db050<br>t: &lt;sip:ivan@192.168.1.234:5060&gt;<br>m: &lt;<a href="mailto:sip:asterisk@192.168.1.131">
sip:asterisk@192.168.1.131</a>&gt;<br>i: <a href="mailto:62f1adf767b7cb3f4ef457c8422bf933@192.168.1.131">62f1adf767b7cb3f4ef457c8422bf933@192.168.1.131</a><br>CSeq: 102 NOTIFY<br>User-Agent: Asterisk PBX<br>Max-Forwards: 70
<br>o: message-summary<br>c: application/simple-message-summary<br>l: 93<br><br><br>Messages-Waiting: no<br>Message-Account: <a href="mailto:sip:asterisk@192.168.1.131">sip:asterisk@192.168.1.131</a><br>Voice-Message: 0/0 (0/0)
<br><br><br>---<br>Scheduling destruction of call<br>'<a href="mailto:62f1adf767b7cb3f4ef457c8422bf933@192.168.1.131">62f1adf767b7cb3f4ef457c8422bf933@192.168.1.131</a>'<br>in 15000 ms<br><br><br>&lt;-- SIP read from <a href="http://192.168.1.234:5060">
192.168.1.234:5060</a>:<br>SIP/2.0 200 Ok<br>Via: SIP/2.0/UDP<br><a href="http://192.168.1.131:5060">192.168.1.131:5060</a>;branch=z9hG4bK040fcabc;rport<br>From: &quot;asterisk&quot;<br>&lt;<a href="mailto:sip:asterisk@192.168.1.131">
sip:asterisk@192.168.1.131</a>&gt;;tag=as0f3db050<br>To: &lt;sip:ivan@192.168.1.234:5060&gt;;tag=2615819391<br>Contact: &lt;sip:ivan@192.168.1.234:5060&gt;<br>Call-ID:<br><a href="mailto:62f1adf767b7cb3f4ef457c8422bf933@192.168.1.131">
62f1adf767b7cb3f4ef457c8422bf933@192.168.1.131</a><br>CSeq: 102 NOTIFY<br>Server: X-Lite release 1103m<br>Content-Length: 0<br><br><br><br><br>--- (9 headers 0 lines)---<br>Destroying call<br>'<a href="mailto:62f1adf767b7cb3f4ef457c8422bf933@192.168.1.131">
62f1adf767b7cb3f4ef457c8422bf933@192.168.1.131</a>'<br>Destroying call<br>'4a9471541c2bc1331824afb7560fbe5a@localhost'<br><br>here is some output of command:<br><br>&gt;show voicemail users<br>Context&nbsp;&nbsp;&nbsp;&nbsp;Mbox&nbsp;&nbsp;User&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Zone
<br>NewMsg<br>other&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;1234&nbsp;&nbsp;Company2 User<br>&nbsp;&nbsp;&nbsp;&nbsp;0<br>mb_tutorial 777&nbsp;&nbsp; ivan<br>&nbsp;&nbsp;&nbsp;&nbsp; 0<br><br>&gt;sip show users<br>Username&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Secret<br>Accountcode&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Def.Context&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;ACL&nbsp;&nbsp;NAT<br>14082097788&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;1234
<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;testagi&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;No&nbsp;&nbsp; RFC3581<br>openser<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;tutorial,opense&nbsp;&nbsp;No&nbsp;&nbsp; RFC3581<br>ivan&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 1212<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;tutorial,mb_tut&nbsp;&nbsp;No&nbsp;&nbsp; RFC3581<br><br>PLZ HELP ME WHAT I AM DOING WRONG ,I SPEND A WEEK TO
<br>SOLVE ON MY WAY BUT UNABLE TO DO THAT:<br><br>HOPE SOMEONE WILL SOLVE MY PROBLEM:<br><br>THANKS AND REGARDS<br>TANZEEL<br><br><br><br><br><br><br><br><br><br><br><br><br><br><br><br>__________________________________________________
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</a><br></blockquote></div><br><br clear="all"><br>-- <br>Anthony D Cennami<br>