[Asterisk-Users] Hangup problem
Tzafrir Cohen
tzafrir at cohens.org.il
Thu Sep 8 06:07:59 MST 2005
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 at 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]
--
Tzafrir Cohen | tzafrir at jbr.cohens.org.il | VIM is
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