[asterisk-users] Asterisk-beginner : cannot make phonecallsusing Asterisk

[Danny Nicholas]

You are closing in.  what does users.conf look like?

 

  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of jonas kellens
Sent: Monday, April 13, 2009 3:40 PM
To: asterisk-users at lists.digium.com
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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