asterisk users - Sep 2005 - Hangup problem

i have a box running debian sarge with asterisk installed from distribution 
packages:

CLI> show version
Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k built by kk@nyx on a x86_64 running Linux

I have managed to configure a simple dialplan (the PBX task is quite simple as 
this is a small office with just a few phones)
I have one Zap (PSTN) line connected to it and one SIP to a local provider.

After some googling most things seem to work well, but i'm having problems 
with Hangup. This affects both the Zap interface and the SIP connection to 
the provider.

No matter who tries to hang up an established call, its not properly finished. 
In such case, with the other node just silence is heard (and not 
congestion/busy signal). This includes calls initiated both from outside and 
from inside with a voip phone connected to the asterisk. 

on asterisk console i'm getting these messages:

  == Spawn extension (incoming, s, 2) exited non-zero on
'SIP/1012082-8408'
  == Spawn extension (incoming, h, 1) exited non-zero on
'SIP/1012082-8408'

here is the relevant part of the dialplan:

[ Context 'incoming' created by 'pbx_config' ]
  'h' =>            1. Hangup()                                  
[pbx_config]
  's' =>            1. Ringing()                                 
[pbx_config]
                    2. Dial(SIP/11|5)                             [pbx_config]
                    3. Dial(SIP/11&SIP/21|20)     [pbx_config]
                    5. Hangup()                                   [pbx_config]
  't' =>            1. Hangup()                                  
[pbx_config]

please tell me what i'm doing wrong :)
thanks in advance,
Marek

On Thu, Sep 08, 2005 at 02:56:28PM +0200, Marek Zachara
wrote:
> i have a box running debian sarge with asterisk installed from distribution
> packages:
> 
> CLI> show version
> Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k built by kk@nyx on a x86_64 running
Linux
> 
> I have managed to configure a simple dialplan (the PBX task is quite simple
as
> this is a small office with just a few phones)
> I have one Zap (PSTN) line connected to it and one SIP to a local provider.
> 
> After some googling most things seem to work well, but i'm having
problems
> with Hangup. This affects both the Zap interface and the SIP connection to 
> the provider.
> 
> No matter who tries to hang up an established call, its not properly
finished.
> In such case, with the other node just silence is heard (and not 
> congestion/busy signal). This includes calls initiated both from outside
and
> from inside with a voip phone connected to the asterisk. 
> 
> on asterisk console i'm getting these messages:
> 
>   == Spawn extension (incoming, s, 2) exited non-zero on
'SIP/1012082-8408'
>   == Spawn extension (incoming, h, 1) exited non-zero on
'SIP/1012082-8408'
> 
> here is the relevant part of the dialplan:
> 
> [ Context 'incoming' created by 'pbx_config' ]
>   'h' =>            1. Hangup()                                 
[pbx_config]
>   's' =>            1. Ringing()                                
[pbx_config]
>                     2. Dial(SIP/11|5)                            
[pbx_config]
>                     3. Dial(SIP/11&SIP/21|20)     [pbx_config]
You dial. After a hangup you attempt to dial again. Right?

What do you try to do there? Ring the one that is free?
>                     5. Hangup()                                  
[pbx_config]
>   't' =>            1. Hangup()                                 
[pbx_config]
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