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Hi Tzafrir,<BR>
<BR>
yet with the first test, things get wrong :<BR>
<BR>
<BR>
<I><FONT SIZE="2">asterisk*CLI> logger show channels</FONT></I><BR>
<I><FONT SIZE="2">Channel Type Status Configuration</FONT></I><BR>
<I><FONT SIZE="2">------- ---- ------ -------------</FONT></I><BR>
<I><FONT SIZE="2">/var/log/asterisk/messages File Enabled - Warning Notice Error </FONT></I><BR>
<I><FONT SIZE="2"> Console Enabled - Warning Notice Error </FONT></I><BR>
<I><FONT SIZE="2">asterisk*CLI> </FONT></I><BR>
<I><FONT SIZE="2">asterisk*CLI> originate SIP/210 application playback demo-instruct</FONT></I><BR>
<I><FONT SIZE="2">[Apr 13 22:33:23] NOTICE[3481]: channel.c:3033 __ast_request_and_dial: Unable to request channel SIP/210</FONT></I><BR>
<I><FONT SIZE="2">asterisk*CLI> </FONT></I><BR>
<BR>
Instead of naming the phone BT201, I've named it after its internal telephone number. For clearity for myself <IMG SRC="cid:1239655102.14427.48.camel@localhost.localdomain" ALIGN="middle" ALT=":-)" 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:
<BLOCKQUOTE TYPE=CITE>
<PRE>
Hi
On Mon, Apr 13, 2009 at 06:18:58PM +0200, jonas kellens wrote:
> I pick up the phone of the BT201 and dial 211... nothing happens.
> I pick up the phone of the GXP1200 and dial 210... nothing happens.
>
> I would love to have your feedback on this. Where could this problem be
> situated ?
Your basic mistake at troubleshooting this is trying to test two things
at the same time. Let's test them separately.
1. A call from Asterisk to the phones:
In the Asterisk CLI:
originate SIP/BT201 application playback demo-instruct
And the other one:
originate SIP/GXP1200 application playback demo-instruct
Alternatively, use the echo-test aplication:
originate SIP/BT201 application echo
2. Next, test calling from the phones to Asterisk. Add those two extensions
to [intern]
exten => 250,1,Answer
exten => 250,n,Playback(demo-instruct)
exten => 250,n,Hangup
exten => 251,1,Answer
exten => 251,1,Echo
exten => 251,1,Hangup
Make sure you reload for that to take effect, and then try dialing 250
or 251.
Another useful tools: 'sip debug'. It tends to generate a very noisy
output that is normally not readable for mere mortals. However it does
indicate that "something is happening". If you call from a remote SIP
phone and there's nothing on the SIP debug, the problem is probably with
the settings of the phone, as it is not getting to you.
Last and not least: a sanity check as you "see nothing": what is the
output of: 'logger show channels' ?
</PRE>
</BLOCKQUOTE>
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