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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'>You are closing in… what does users.conf
look like?<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 3:40
PM<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> Re: [asterisk-users]
Asterisk-beginner : cannot make phonecallsusing Asterisk</span></font><o:p></o:p></p>
</div>
<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 Tzafrir,<br>
<br>
yet with the first test, things get wrong :<br>
<br>
<br>
</span></font><i><font size=2><span style='font-size:10.0pt;font-style:italic'>asterisk*CLI>
logger show channels</span></font></i><br>
<i><font size=2><span style='font-size:10.0pt;font-style:italic'>Channel
Type Status Configuration</span></font></i><br>
<i><font size=2><span style='font-size:10.0pt;font-style:italic'>-------
---- ------ -------------</span></font></i><br>
<i><font size=2><span style='font-size:10.0pt;font-style:italic'>/var/log/asterisk/messages
File Enabled - Warning Notice Error </span></font></i><br>
<i><font size=2><span style='font-size:10.0pt;font-style:italic'>
Console Enabled - Warning Notice Error </span></font></i><br>
<i><font size=2><span style='font-size:10.0pt;font-style:italic'>asterisk*CLI>
</span></font></i><br>
<i><font size=2><span style='font-size:10.0pt;font-style:italic'>asterisk*CLI>
originate SIP/210 application playback demo-instruct</span></font></i><br>
<i><font size=2><span style='font-size:10.0pt;font-style:italic'>[Apr 13
22:33:23] NOTICE[3481]: channel.c:3033 __ast_request_and_dial: Unable to
request channel SIP/210</span></font></i><br>
<i><font size=2><span style='font-size:10.0pt;font-style:italic'>asterisk*CLI>
</span></font></i><br>
<br>
Instead of naming the phone BT201, I've named it after its internal telephone
number. For clearity for myself <img width=16 height=16 id="_x0000_i1025"
src="cid:image001.gif@01C9BC4F.3EC49A40" alt=":-)" align=middle border=0>.<br>
<br>
But when I dial the IP-phone from the CLI, I get the output of above...<br>
<br>
Thank for your reply !<br>
<br>
Jonas.<br>
<br>
<br>
On Mon, 2009-04-13 at 23:11 +0300, Tzafrir Cohen wrote: <o:p></o:p></p>
<pre><font size=2 face="Courier New"><span style='font-size:10.0pt'><o:p> </o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'>Hi<o:p></o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'><o:p> </o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'>On Mon, Apr 13, 2009 at 06:18:58PM +0200, jonas kellens wrote:<o:p></o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'><o:p> </o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'>> I pick up the phone of the BT201 and dial 211... nothing happens.<o:p></o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'>> I pick up the phone of the GXP1200 and dial 210... nothing happens.<o:p></o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'>> <o:p></o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'>> I would love to have your feedback on this. Where could this problem be<o:p></o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'>> situated ?<o:p></o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'><o:p> </o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'>Your basic mistake at troubleshooting this is trying to test two things<o:p></o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'>at the same time. Let's test them separately.<o:p></o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'><o:p> </o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'>1. A call from Asterisk to the phones:<o:p></o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'><o:p> </o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'><o:p> </o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'>In the Asterisk CLI:<o:p></o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'><o:p> </o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'> originate SIP/BT201 application playback demo-instruct<o:p></o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'><o:p> </o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'>And the other one:<o:p></o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'><o:p> </o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'> originate SIP/GXP1200 application playback demo-instruct<o:p></o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'><o:p> </o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'>Alternatively, use the echo-test aplication:<o:p></o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'><o:p> </o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'> originate SIP/BT201 application echo<o:p></o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'><o:p> </o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'><o:p> </o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'>2. Next, test calling from the phones to Asterisk. Add those two extensions<o:p></o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'>to [intern]<o:p></o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'><o:p> </o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'>exten => 250,1,Answer<o:p></o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'>exten => 250,n,Playback(demo-instruct)<o:p></o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'>exten => 250,n,Hangup<o:p></o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'><o:p> </o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'>exten => 251,1,Answer<o:p></o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'>exten => 251,1,Echo<o:p></o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'>exten => 251,1,Hangup<o:p></o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'><o:p> </o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'>Make sure you reload for that to take effect, and then try dialing 250<o:p></o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'>or 251.<o:p></o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'><o:p> </o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'>Another useful tools: 'sip debug'. It tends to generate a very noisy <o:p></o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'>output that is normally not readable for mere mortals. However it does <o:p></o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'>indicate that "something is happening". If you call from a remote SIP <o:p></o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'>phone and there's nothing on the SIP debug, the problem is probably with <o:p></o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'>the settings of the phone, as it is not getting to you.<o:p></o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'><o:p> </o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'>Last and not least: a sanity check as you "see nothing": what is the<o:p></o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'>output of: 'logger show channels' ?<o:p></o:p></span></font></pre><pre><font
size=2 face="Courier New"><span style='font-size:10.0pt'><o:p> </o:p></span></font></pre></div>
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