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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><FONT SIZE="2">[root@asterisk asterisk]# cat sip.conf</FONT></I><BR>
<I><FONT SIZE="2">[general]</FONT></I><BR>
<I><FONT SIZE="2">bindport=5060</FONT></I><BR>
<I><FONT SIZE="2">bindaddr = 0.0.0.0</FONT></I><BR>
<BR>
<I><FONT SIZE="2">[BT201]</FONT></I><BR>
<I><FONT SIZE="2">type=friend</FONT></I><BR>
<I><FONT SIZE="2">context=intern</FONT></I><BR>
<I><FONT SIZE="2">host=192.168.4.210</FONT></I><BR>
<I><FONT SIZE="2">secret=testpaswoord</FONT></I><BR>
<BR>
<I><FONT SIZE="2">[GXP1200]</FONT></I><BR>
<I><FONT SIZE="2">type=friend</FONT></I><BR>
<I><FONT SIZE="2">context=intern</FONT></I><BR>
<I><FONT SIZE="2">host=192.168.4.211</FONT></I><BR>
<I><FONT SIZE="2">secret=testpaswoord</FONT></I>
<BR>
extensions.conf :
<BR>
<I><FONT SIZE="2">[root@asterisk asterisk]# cat extensions.conf</FONT></I><BR>
<I><FONT SIZE="2">[intern]</FONT></I><BR>
<I><FONT SIZE="2">exten => 210,1,Dial(SIP/BT201)</FONT></I><BR>
<I><FONT SIZE="2">exten => 211,1,Dial(SIP/GXP1200)</FONT></I>
<BR>
<FONT SIZE="2">Asterisk CLI shows me :</FONT>
<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"> == Parsing '/etc/asterisk/sip_notify.conf': Found</FONT></I><BR>
<I><FONT SIZE="2">asterisk*CLI> sip show peers</FONT></I><BR>
<I><FONT SIZE="2">Name/username Host Dyn Nat ACL Port Status </FONT></I><BR>
<I><FONT SIZE="2">GXP1200 192.168.4.211 5060 Unmonitored </FONT></I><BR>
<I><FONT SIZE="2">BT201 192.168.4.210 5060 Unmonitored </FONT></I><BR>
<I><FONT SIZE="2">2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]</FONT></I><BR>
<BR>
<I><FONT SIZE="2">asterisk*CLI> dialplan show intern</FONT></I><BR>
<I><FONT SIZE="2">[ Context 'intern' created by 'pbx_config' ]</FONT></I><BR>
<I><FONT SIZE="2"> '210' => 1. Dial(SIP/BT201) [pbx_config]</FONT></I><BR>
<I><FONT SIZE="2"> '211' => 1. Dial(SIP/GXP1200) [pbx_config]</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.
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