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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Alright again, what do you see on the CLI when you make a call
to 210/211?<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
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<p class=MsoNormal><span style='font-size:18.0pt;font-family:"Brush Script MT";
color:#1F497D'>James Shigley<o:p></o:p></span></p>
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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
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<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>jonas
kellens<br>
<b>Sent:</b> Monday, April 13, 2009 12:07 PM<br>
<b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br>
<b>Subject:</b> Re: [asterisk-users] Asterisk-beginner : cannot make phonecalls
using Asterisk<o:p></o:p></span></p>
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<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>Tony Plack,<br>
<br>
this is the result form Asterisk CLI :<br>
<br>
<i><span style='font-size:10.0pt'>[root@asterisk asterisk]# /usr/sbin/asterisk
-vvvvvr</span></i><br>
<i><span style='font-size:10.0pt'>Asterisk 1.4.24, Copyright (C) 1999 - 2008
Digium, Inc. and others.</span></i><br>
<i><span style='font-size:10.0pt'>Created by Mark Spencer <<a
href="mailto:markster@digium.com">markster@digium.com</a>></span></i><br>
<i><span style='font-size:10.0pt'>Asterisk comes with ABSOLUTELY NO WARRANTY;
type 'core show warranty' for details.</span></i><br>
<i><span style='font-size:10.0pt'>This is free software, with components
licensed under the GNU General Public</span></i><br>
<i><span style='font-size:10.0pt'>License version 2 and other licenses; you are
welcome to redistribute it under</span></i><br>
<i><span style='font-size:10.0pt'>certain conditions. Type 'core show license'
for details.</span></i><br>
<i><span style='font-size:10.0pt'>=========================================================================</span></i><br>
<i><span style='font-size:10.0pt'> == Parsing
'/etc/asterisk/asterisk.conf': Found</span></i><br>
<i><span style='font-size:10.0pt'> == Parsing
'/etc/asterisk/extconfig.conf': Found</span></i><br>
<i><span style='font-size:10.0pt'>Connected to Asterisk 1.4.24 currently
running on asterisk (pid = 3895)</span></i><br>
<i><span style='font-size:10.0pt'>Verbosity is at least 5</span></i><br>
<i><span style='font-size:10.0pt'>asterisk*CLI> dialplan reload</span></i><br>
<i><span style='font-size:10.0pt'>Dialplan reloaded.</span></i><br>
<i><span style='font-size:10.0pt'> == Parsing
'/etc/asterisk/extensions.conf': Found</span></i><br>
<i><span style='font-size:10.0pt'> -- Registered extension
context 'intern'</span></i><br>
<i><span style='font-size:10.0pt'> -- Added extension '210'
priority 1 to intern</span></i><br>
<i><span style='font-size:10.0pt'> -- Added extension '211'
priority 1 to intern</span></i><br>
<i><span style='font-size:10.0pt'> == Parsing '/etc/asterisk/users.conf':
Found</span></i><br>
<i><span style='font-size:10.0pt'>asterisk*CLI> sip reload</span></i><br>
<i><span style='font-size:10.0pt'>Reloading SIP</span></i><br>
<i><span style='font-size:10.0pt'> == Parsing '/etc/asterisk/sip.conf':
Found</span></i><br>
<i><span style='font-size:10.0pt'> == Parsing '/etc/asterisk/users.conf':
Found</span></i><br>
<i><span style='font-size:10.0pt'> == SIP Listening on 192.168.4.248:5060</span></i><br>
<i><span style='font-size:10.0pt'> == Using SIP TOS: none</span></i><br>
<i><span style='font-size:10.0pt'> == Parsing
'/etc/asterisk/sip_notify.conf': Found</span></i><br>
<br>
So I've changed the bindaddr... Still no change I'm afraid...<br>
<br>
Thanks for your reply !<br>
<br>
Please help me a bit further cause this a work I'm doing as thesis.<br>
<br>
Jonas.<br>
<br>
<br>
On Mon, 2009-04-13 at 11:34 -0500, Anthony Plack wrote: <o:p></o:p></p>
<pre><o:p> </o:p></pre><pre>> bindaddr = 0.0.0.0<o:p></o:p></pre><pre>><o:p> </o:p></pre><pre><o:p> </o:p></pre><pre>I would set this to the ethernet interface IP address, I believe this may be your issue.<o:p></o:p></pre><pre><o:p> </o:p></pre><pre>Registration is only for receiving calls, if you are not seeing information on the dial, then the phone is not talking to the server. I would make sure of the settings in the web-interface as well.<o:p></o:p></pre><pre><o:p> </o:p></pre><pre>Tony Plack<o:p></o:p></pre></div>
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