[asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk

[jonas kellens]

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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