<html xmlns:v="urn:schemas-microsoft-com:vml" xmlns:o="urn:schemas-microsoft-com:office:office" xmlns:w="urn:schemas-microsoft-com:office:word" xmlns:m="http://schemas.microsoft.com/office/2004/12/omml" xmlns="http://www.w3.org/TR/REC-html40">
<head>
<meta http-equiv=Content-Type content="text/html; charset=utf-8">
<meta name=Generator content="Microsoft Word 12 (filtered medium)">
<style>
<!--
/* Font Definitions */
@font-face
{font-family:Calibri;
panose-1:2 15 5 2 2 2 4 3 2 4;}
@font-face
{font-family:Tahoma;
panose-1:2 11 6 4 3 5 4 4 2 4;}
@font-face
{font-family:Consolas;
panose-1:2 11 6 9 2 2 4 3 2 4;}
@font-face
{font-family:"Brush Script MT";
panose-1:3 6 8 2 4 4 6 7 3 4;}
@font-face
{font-family:"Bradley Hand ITC";
panose-1:3 7 4 2 5 3 2 3 2 3;}
/* Style Definitions */
p.MsoNormal, li.MsoNormal, div.MsoNormal
{margin:0in;
margin-bottom:.0001pt;
font-size:12.0pt;
font-family:"Times New Roman","serif";}
a:link, span.MsoHyperlink
{mso-style-priority:99;
color:blue;
text-decoration:underline;}
a:visited, span.MsoHyperlinkFollowed
{mso-style-priority:99;
color:purple;
text-decoration:underline;}
span.EmailStyle17
{mso-style-type:personal-reply;
font-family:"Calibri","sans-serif";
color:#1F497D;}
.MsoChpDefault
{mso-style-type:export-only;
font-size:10.0pt;}
@page Section1
{size:8.5in 11.0in;
margin:1.0in 1.0in 1.0in 1.0in;}
div.Section1
{page:Section1;}
-->
</style>
<!--[if gte mso 9]><xml>
<o:shapedefaults v:ext="edit" spidmax="1026" />
</xml><![endif]--><!--[if gte mso 9]><xml>
<o:shapelayout v:ext="edit">
<o:idmap v:ext="edit" data="1" />
</o:shapelayout></xml><![endif]-->
</head>
<body lang=EN-US link=blue vlink=purple>
<div class=Section1>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>What do you see when you run asterisk –r and dial 210 or 211
from one of the phones<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>
<div>
<p class=MsoNormal><span style='font-size:18.0pt;font-family:"Brush Script MT";
color:#1F497D'>James Shigley<o:p></o:p></span></p>
<p class=MsoNormal><b><span style='font-size:18.0pt;font-family:"Bradley Hand ITC";
color:#1F497D'>Monroe Telephone Answering Service<o:p></o:p></span></b></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>409-981-9213</span><b><span style='font-size:18.0pt;font-family:
"Bradley Hand ITC";color:#1F497D'><o:p></o:p></span></b></p>
<p class=MsoNormal><span style='font-size:8.0pt;font-family:"Arial","sans-serif";
color:#1F497D'>Infinity 5.5,UC 4.02.3803, Blink 3.0.104<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:8.0pt;font-family:"Arial","sans-serif";
color:#1F497D'>Ecreator:2.21, eResponse 1.1.7<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:8.0pt;font-family:"Arial","sans-serif";
color:#1F497D'>Webportal,WebApps,<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'> </span><span style='color:#1F497D'><o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>CONFIDENTIALITY NOTICE: </span><span style='font-size:7.5pt;
font-family:"Calibri","sans-serif";color:#1F497D'>This email, including any
attachments, contains information which may be confidential or privileged. The
information is intended to be for the use of the individual or entity named
above. If you are not the intended recipient, be aware that any disclosure,
copying, distribution or use of the contents of this information is prohibited.
If you have received this email in error, please notify the sender immediately
by "reply to sender only" message and destroy all electronic and hard
copies of the communication, including attachments.</span><span
style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> <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>
<p class=MsoNormal><span style='font-size:10.5pt;font-family:Consolas;
color:#1F497D'>"Common sense is the collection of prejudices acquired by
age eighteen." -- Albert Einstein <o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.5pt;font-family:Consolas;
color:#1F497D'>"Once you can accept the universe as matter expanding into
nothing that is something,wearing stripes with plaid comes easy." --
Albert Einstein<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.5pt;font-family:Consolas;
color:#1F497D'>"I know a little of everything, but a lot of nothing"<o:p></o:p></span></p>
</div>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<div>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>
<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 11:19 AM<br>
<b>To:</b> asterisk-users@lists.digium.com<br>
<b>Subject:</b> [asterisk-users] Asterisk-beginner : cannot make phonecalls
usingAsterisk<o:p></o:p></span></p>
</div>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>Hi there,<br>
<br>
this is the first time that I'm building an Asterisk-server.<br>
<br>
I have compiled Asterisk together with Zaptel on an CentOS 5.3-system.<br>
Zaptel is for later, when configuring the POTS-line. Now first internal
communication with SIP.<br>
<br>
Thought it would go easier...<br>
<br>
I have 2 Grandstream IP-phones : BT-201 and GXP-1200.<br>
<br>
These are my settings :<br>
<br>
sip.conf : <br>
<i><span style='font-size:10.0pt'>[root@asterisk asterisk]# cat sip.conf</span></i><br>
<i><span style='font-size:10.0pt'>[general]</span></i><br>
<i><span style='font-size:10.0pt'>bindport=5060</span></i><br>
<i><span style='font-size:10.0pt'>bindaddr = 0.0.0.0</span></i><br>
<br>
<i><span style='font-size:10.0pt'>[BT201]</span></i><br>
<i><span style='font-size:10.0pt'>type=friend</span></i><br>
<i><span style='font-size:10.0pt'>context=intern</span></i><br>
<i><span style='font-size:10.0pt'>host=192.168.4.210</span></i><br>
<i><span style='font-size:10.0pt'>secret=testpaswoord</span></i><br>
<br>
<i><span style='font-size:10.0pt'>[GXP1200]</span></i><br>
<i><span style='font-size:10.0pt'>type=friend</span></i><br>
<i><span style='font-size:10.0pt'>context=intern</span></i><br>
<i><span style='font-size:10.0pt'>host=192.168.4.211</span></i><br>
<i><span style='font-size:10.0pt'>secret=testpaswoord</span></i> <br>
extensions.conf : <br>
<i><span style='font-size:10.0pt'>[root@asterisk asterisk]# cat extensions.conf</span></i><br>
<i><span style='font-size:10.0pt'>[intern]</span></i><br>
<i><span style='font-size:10.0pt'>exten => 210,1,Dial(SIP/BT201)</span></i><br>
<i><span style='font-size:10.0pt'>exten => 211,1,Dial(SIP/GXP1200)</span></i>
<br>
<span style='font-size:10.0pt'>Asterisk CLI shows me :</span> <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'> == Parsing
'/etc/asterisk/sip_notify.conf': Found</span></i><br>
<i><span style='font-size:10.0pt'>asterisk*CLI> sip show peers</span></i><br>
<i><span style='font-size:10.0pt'>Name/username
Host Dyn Nat ACL
Port Status
</span></i><br>
<i><span style='font-size:10.0pt'>GXP1200
192.168.4.211
5060 Unmonitored
</span></i><br>
<i><span style='font-size:10.0pt'>BT201
192.168.4.210
5060 Unmonitored
</span></i><br>
<i><span style='font-size:10.0pt'>2 sip peers [Monitored: 0 online, 0 offline
Unmonitored: 2 online, 0 offline]</span></i><br>
<br>
<i><span style='font-size:10.0pt'>asterisk*CLI> dialplan show intern</span></i><br>
<i><span style='font-size:10.0pt'>[ Context 'intern' created by 'pbx_config' ]</span></i><br>
<i><span style='font-size:10.0pt'> '210' =>
1. Dial(SIP/BT201)
[pbx_config]</span></i><br>
<i><span style='font-size:10.0pt'> '211'
=> 1.
Dial(SIP/GXP1200)
[pbx_config]</span></i> <br>
<br>
I pick up the phone of the BT201 and dial 211... nothing happens.<br>
I pick up the phone of the GXP1200 and dial 210... nothing happens.<br>
<br>
I would love to have your feedback on this. Where could this problem be
situated ?<br>
<br>
I notice (on the Asterisk CLI) that my SIP-phones do not register. They have a
fixed IP and there account information is set via the web interface.<br>
<br>
Greetingz,<br>
Jonas. <o:p></o:p></p>
</div>
</body>
</html>