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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Put register=yes in the BT201 and GXP1200
contexts of sip.conf<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
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<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <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>
asterisk-users@lists.digium.com<br>
<b><span style='font-weight:bold'>Subject:</span></b> [asterisk-users]
Asterisk-beginner : cannot make phonecalls usingAsterisk</span></font><o:p></o:p></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>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:10.0pt;font-style:italic'>[root@asterisk
asterisk]# cat sip.conf</span></font></i><br>
<i><font size=2><span style='font-size:10.0pt;font-style:italic'>[general]</span></font></i><br>
<i><font size=2><span style='font-size:10.0pt;font-style:italic'>bindport=5060</span></font></i><br>
<i><font size=2><span style='font-size:10.0pt;font-style:italic'>bindaddr =
0.0.0.0</span></font></i><br>
<br>
<i><font size=2><span style='font-size:10.0pt;font-style:italic'>[BT201]</span></font></i><br>
<i><font size=2><span style='font-size:10.0pt;font-style:italic'>type=friend</span></font></i><br>
<i><font size=2><span style='font-size:10.0pt;font-style:italic'>context=intern</span></font></i><br>
<i><font size=2><span style='font-size:10.0pt;font-style:italic'>host=192.168.4.210</span></font></i><br>
<i><font size=2><span style='font-size:10.0pt;font-style:italic'>secret=testpaswoord</span></font></i><br>
<br>
<i><font size=2><span style='font-size:10.0pt;font-style:italic'>[GXP1200]</span></font></i><br>
<i><font size=2><span style='font-size:10.0pt;font-style:italic'>type=friend</span></font></i><br>
<i><font size=2><span style='font-size:10.0pt;font-style:italic'>context=intern</span></font></i><br>
<i><font size=2><span style='font-size:10.0pt;font-style:italic'>host=192.168.4.211</span></font></i><br>
<i><font size=2><span style='font-size:10.0pt;font-style:italic'>secret=testpaswoord</span></font></i>
<br>
extensions.conf : <br>
<i><font size=2><span style='font-size:10.0pt;font-style:italic'>[root@asterisk
asterisk]# cat extensions.conf</span></font></i><br>
<i><font size=2><span style='font-size:10.0pt;font-style:italic'>[intern]</span></font></i><br>
<i><font size=2><span style='font-size:10.0pt;font-style:italic'>exten =>
210,1,Dial(SIP/BT201)</span></font></i><br>
<i><font size=2><span style='font-size:10.0pt;font-style:italic'>exten => 211,1,Dial(SIP/GXP1200)</span></font></i>
<br>
<font size=2><span style='font-size:10.0pt'>Asterisk CLI shows me :</span></font>
<br>
<i><font size=2><span style='font-size:10.0pt;font-style:italic'>asterisk*CLI>
sip reload</span></font></i><br>
<i><font size=2><span style='font-size:10.0pt;font-style:italic'>Reloading SIP</span></font></i><br>
<i><font size=2><span style='font-size:10.0pt;font-style:italic'> ==
Parsing '/etc/asterisk/sip.conf': Found</span></font></i><br>
<i><font size=2><span style='font-size:10.0pt;font-style:italic'> ==
Parsing '/etc/asterisk/users.conf': Found</span></font></i><br>
<i><font size=2><span style='font-size:10.0pt;font-style:italic'> ==
Parsing '/etc/asterisk/sip_notify.conf': Found</span></font></i><br>
<i><font size=2><span style='font-size:10.0pt;font-style:italic'>asterisk*CLI>
sip show peers</span></font></i><br>
<i><font size=2><span style='font-size:10.0pt;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:10.0pt;font-style:italic'>GXP1200
192.168.4.211 5060
Unmonitored </span></font></i><br>
<i><font size=2><span style='font-size:10.0pt;font-style:italic'>BT201
192.168.4.210 5060
Unmonitored </span></font></i><br>
<i><font size=2><span style='font-size:10.0pt;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:10.0pt;font-style:italic'>asterisk*CLI>
dialplan show intern</span></font></i><br>
<i><font size=2><span style='font-size:10.0pt;font-style:italic'>[ Context
'intern' created by 'pbx_config' ]</span></font></i><br>
<i><font size=2><span style='font-size:10.0pt;font-style:italic'> '210'
=> 1. Dial(SIP/BT201)
[pbx_config]</span></font></i><br>
<i><font size=2><span style='font-size:10.0pt;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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