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Tony Plack,<BR>
<BR>
this is the result form Asterisk CLI :<BR>
<BR>
<I><FONT SIZE="2">[root@asterisk asterisk]# /usr/sbin/asterisk -vvvvvr</FONT></I><BR>
<I><FONT SIZE="2">Asterisk 1.4.24, Copyright (C) 1999 - 2008 Digium, Inc. and others.</FONT></I><BR>
<I><FONT SIZE="2">Created by Mark Spencer <<A HREF="mailto:markster@digium.com">markster@digium.com</A>></FONT></I><BR>
<I><FONT SIZE="2">Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.</FONT></I><BR>
<I><FONT SIZE="2">This is free software, with components licensed under the GNU General Public</FONT></I><BR>
<I><FONT SIZE="2">License version 2 and other licenses; you are welcome to redistribute it under</FONT></I><BR>
<I><FONT SIZE="2">certain conditions. Type 'core show license' for details.</FONT></I><BR>
<I><FONT SIZE="2">=========================================================================</FONT></I><BR>
<I><FONT SIZE="2"> == Parsing '/etc/asterisk/asterisk.conf': Found</FONT></I><BR>
<I><FONT SIZE="2"> == Parsing '/etc/asterisk/extconfig.conf': Found</FONT></I><BR>
<I><FONT SIZE="2">Connected to Asterisk 1.4.24 currently running on asterisk (pid = 3895)</FONT></I><BR>
<I><FONT SIZE="2">Verbosity is at least 5</FONT></I><BR>
<I><FONT SIZE="2">asterisk*CLI> dialplan reload</FONT></I><BR>
<I><FONT SIZE="2">Dialplan reloaded.</FONT></I><BR>
<I><FONT SIZE="2"> == Parsing '/etc/asterisk/extensions.conf': Found</FONT></I><BR>
<I><FONT SIZE="2"> -- Registered extension context 'intern'</FONT></I><BR>
<I><FONT SIZE="2"> -- Added extension '210' priority 1 to intern</FONT></I><BR>
<I><FONT SIZE="2"> -- Added extension '211' priority 1 to intern</FONT></I><BR>
<I><FONT SIZE="2"> == Parsing '/etc/asterisk/users.conf': Found</FONT></I><BR>
<I><FONT SIZE="2">asterisk*CLI> sip reload</FONT></I><BR>
<I><FONT SIZE="2"> Reloading SIP</FONT></I><BR>
<I><FONT SIZE="2"> == Parsing '/etc/asterisk/sip.conf': Found</FONT></I><BR>
<I><FONT SIZE="2"> == Parsing '/etc/asterisk/users.conf': Found</FONT></I><BR>
<I><FONT SIZE="2"> == SIP Listening on 192.168.4.248:5060</FONT></I><BR>
<I><FONT SIZE="2"> == Using SIP TOS: none</FONT></I><BR>
<I><FONT SIZE="2"> == Parsing '/etc/asterisk/sip_notify.conf': Found</FONT></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:
<BLOCKQUOTE TYPE=CITE>
<PRE>
> bindaddr = 0.0.0.0
>
I would set this to the ethernet interface IP address, I believe this may be your issue.
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.
Tony Plack
</PRE>
</BLOCKQUOTE>
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