You are closing in. what does users.conf look like?
_____ From: [email protected] [mailto:[email protected]] On Behalf Of jonas kellens Sent: Monday, April 13, 2009 3:40 PM To: [email protected] Subject: Re: [asterisk-users] Asterisk-beginner : cannot make phonecallsusing Asterisk Hi Tzafrir, yet with the first test, things get wrong : asterisk*CLI> logger show channels Channel Type Status Configuration ------- ---- ------ ------------- /var/log/asterisk/messages File Enabled - Warning Notice Error Console Enabled - Warning Notice Error asterisk*CLI> asterisk*CLI> originate SIP/210 application playback demo-instruct [Apr 13 22:33:23] NOTICE[3481]: channel.c:3033 __ast_request_and_dial: Unable to request channel SIP/210 asterisk*CLI> Instead of naming the phone BT201, I've named it after its internal telephone number. For clearity for myself :-). But when I dial the IP-phone from the CLI, I get the output of above... Thank for your reply ! Jonas. On Mon, 2009-04-13 at 23:11 +0300, Tzafrir Cohen wrote: 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' ?
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