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Danny Nicholas wrote:
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<p class="MsoNormal"><font color="navy" face="Arial" size="2"><span
style="font-size: 10pt; font-family: Arial; color: navy;">Do you have
include=intern in the default
context? If no, * will come back with can’t find peer 210 (or 211).<o:p></o:p></span></font></p>
<p class="MsoNormal"><font color="navy" face="Arial" size="2"><span
style="font-size: 10pt; font-family: Arial; color: navy;"><o:p> </o:p></span></font></p>
<p class="MsoNormal"><font color="#1f497d" face="Calibri" size="2"><span
style="font-size: 11pt; font-family: Calibri; color: rgb(31, 73, 125);"><o:p> </o:p></span></font></p>
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<p class="MsoNormal"><b><font face="Tahoma" size="2"><span
style="font-size: 10pt; font-family: Tahoma; font-weight: bold;">From:</span></font></b><font
face="Tahoma" size="2"><span
style="font-size: 10pt; font-family: Tahoma;">
<a class="moz-txt-link-abbreviated" href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>
[<a class="moz-txt-link-freetext" href="mailto:asterisk-users-bounces@lists.digium.com">mailto:asterisk-users-bounces@lists.digium.com</a>] <b><span
style="font-weight: bold;">On Behalf Of </span></b>jonas kellens<br>
<b><span style="font-weight: bold;">Sent:</span></b> Monday, April
13, 2009 11:19
AM<br>
<b><span style="font-weight: bold;">To:</span></b>
<a class="moz-txt-link-abbreviated" href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><br>
<b><span style="font-weight: bold;">Subject:</span></b>
[asterisk-users]
Asterisk-beginner : cannot make phonecalls usingAsterisk<o:p></o:p></span></font></p>
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<p class="MsoNormal"><font face="Times New Roman" size="3"><span
style="font-size: 12pt;"><o:p> </o:p></span></font></p>
<p class="MsoNormal"><font face="Times New Roman" size="3"><span
style="font-size: 12pt;">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>
</span></font><i><font size="2"><span
style="font-size: 10pt; font-style: italic;">[root@asterisk
asterisk]# cat sip.conf</span></font></i><br>
<i><font size="2"><span style="font-size: 10pt; font-style: italic;">[general]</span></font></i><br>
<i><font size="2"><span style="font-size: 10pt; font-style: italic;">bindport=5060</span></font></i><br>
<i><font size="2"><span style="font-size: 10pt; font-style: italic;">bindaddr
=
0.0.0.0</span></font></i><br>
<br>
<i><font size="2"><span style="font-size: 10pt; font-style: italic;">[BT201]</span></font></i><br>
<i><font size="2"><span style="font-size: 10pt; font-style: italic;">type=friend</span></font></i><br>
<i><font size="2"><span style="font-size: 10pt; font-style: italic;">context=intern</span></font></i><br>
<i><font size="2"><span style="font-size: 10pt; font-style: italic;">host=192.168.4.210</span></font></i><br>
<i><font size="2"><span style="font-size: 10pt; font-style: italic;">secret=testpaswoord</span></font></i><br>
<br>
<i><font size="2"><span style="font-size: 10pt; font-style: italic;">[GXP1200]</span></font></i><br>
<i><font size="2"><span style="font-size: 10pt; font-style: italic;">type=friend</span></font></i><br>
<i><font size="2"><span style="font-size: 10pt; font-style: italic;">context=intern</span></font></i><br>
<i><font size="2"><span style="font-size: 10pt; font-style: italic;">host=192.168.4.211</span></font></i><br>
<i><font size="2"><span style="font-size: 10pt; font-style: italic;">secret=testpaswoord</span></font></i>
<br>
extensions.conf : <br>
<i><font size="2"><span style="font-size: 10pt; font-style: italic;">[root@asterisk
asterisk]# cat extensions.conf</span></font></i><br>
<i><font size="2"><span style="font-size: 10pt; font-style: italic;">[intern]</span></font></i><br>
<i><font size="2"><span style="font-size: 10pt; font-style: italic;">exten
=>
210,1,Dial(SIP/BT201)</span></font></i><br>
<i><font size="2"><span style="font-size: 10pt; font-style: italic;">exten
=>
211,1,Dial(SIP/GXP1200)</span></font></i> <br>
<font size="2"><span style="font-size: 10pt;">Asterisk CLI shows me :</span></font>
<br>
<i><font size="2"><span style="font-size: 10pt; font-style: italic;">asterisk*CLI>
sip reload</span></font></i><br>
<i><font size="2"><span style="font-size: 10pt; font-style: italic;">Reloading
SIP</span></font></i><br>
<i><font size="2"><span style="font-size: 10pt; font-style: italic;">
==
Parsing '/etc/asterisk/sip.conf': Found</span></font></i><br>
<i><font size="2"><span style="font-size: 10pt; font-style: italic;">
==
Parsing '/etc/asterisk/users.conf': Found</span></font></i><br>
<i><font size="2"><span style="font-size: 10pt; font-style: italic;">
==
Parsing '/etc/asterisk/sip_notify.conf': Found</span></font></i><br>
<i><font size="2"><span style="font-size: 10pt; font-style: italic;">asterisk*CLI>
sip show peers</span></font></i><br>
<i><font size="2"><span style="font-size: 10pt; font-style: italic;">Name/username
Host
<st1:place w:st="on"><st1:PlaceName w:st="on">Dyn</st1:PlaceName> <st1:PlaceName
w:st="on">Nat</st1:PlaceName> <st1:PlaceName w:st="on">ACL</st1:PlaceName>
<st1:PlaceType w:st="on">Port</st1:PlaceType></st1:place>
Status </span></font></i><br>
<i><font size="2"><span style="font-size: 10pt; font-style: italic;">GXP1200
192.168.4.211 5060
Unmonitored </span></font></i><br>
<i><font size="2"><span style="font-size: 10pt; font-style: italic;">BT201
192.168.4.210
5060
Unmonitored </span></font></i><br>
<i><font size="2"><span style="font-size: 10pt; font-style: italic;">2
sip peers
[Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]</span></font></i><br>
<br>
<i><font size="2"><span style="font-size: 10pt; font-style: italic;">asterisk*CLI>
dialplan show intern</span></font></i><br>
<i><font size="2"><span style="font-size: 10pt; font-style: italic;">[
Context
'intern' created by 'pbx_config' ]</span></font></i><br>
<i><font size="2"><span style="font-size: 10pt; font-style: italic;">
'210'
=> 1. Dial(SIP/BT201)
[pbx_config]</span></font></i><br>
<i><font size="2"><span style="font-size: 10pt; font-style: italic;">
'211'
=> 1.
Dial(SIP/GXP1200)
[pbx_config]</span></font></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>
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</blockquote>
This is not the case since both of his phones are configured to come in
to the "intern" context by default. In the real world, if you intern
context had access to outside calls and you included it in the
"default" context and happened to allow guest access, then anybody
coming in to your box could make outbound calls.<br>
<br>
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