Re: VoiceMail() stops dialplan processing

Antony Stone <[email protected]>
Newsgroups gmane.comp.telephony.pbx.asterisk.user
Message-ID <[email protected]>
On Monday 03 October 2022 at 14:14:54, Joshua C. Colp wrote:

> On Mon, Oct 3, 2022 at 9:11 AM Antony Stone <
> 
> [email protected]> wrote:
> > Hi.
> > 
> > I have a dialplan which calls the VoiceMail() application, and I'm
> > getting the following behaviour:
> >  - if the inbound caller leaves a message, then presses #, and then
> > presses 1 to accept the recording, everything works as expected and the
> > dialplan continues processing after the line containing VoiceMail()
> > 
> >  - if the inbound caller leaves a message and then hangs up, the diaplan
> > simply stops executing with a message such as:
> > 
> > [2022-10-03 13:02:23.355976] pbx VERBOSE[19022][C-00000556]: pbx.c:4413
> > in __ast_pbx_run: Spawn extension (RecordVM, 00xx74xx88xx90, 2) exited
> > non-zero on 'SIP/TrunkOne-00000c12'
> > 
> > The subsequent commands in the dialplan do not get processed.
> 
> This is fundamentally how dialplan works. If a channel hangs up, then
> normal dialplan execution stops.

I suppose that fits other situtations, yes.

> > Can anyone suggest either why this would happen and how to get the
> > dialplan to continue processing under all circumstances, or at least how
> > to investigate futher what is causing this to happen?
> > 
> > I'm sure that leaving a message and hanging up the call should be valid
> > because that's what the default greeting message tells people they can
> > do.
> 
> It is. If you're needing to do something afterwards, then the 'h' extension
> or hangup handlers are used to execute logic when the channel is hung up.

Okay, sounds simple enough - thanks,


Antony.

-- 
RTFM may be the appropriate reply, but please specify exactly which FM to R.

                                                   Please reply to the list;
                                                         please *don't* CC me.

-- 
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
      https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
lmpx.com only provides a reader for public news (NNTP) servers. It is not affiliated with the servers or forums shown here and is not responsible for the content of articles, which is written by their respective authors.