[asterisk-users] Goto in Realtime extensions

Atis Lezdins atis at iq-labs.net
Fri Feb 8 09:40:02 CST 2008


On 2/8/08, Yves Räber <yraber at mailup.net> wrote:
> That's very unfortunate.
>
> I use now a workaround : I'm just switching (with gotos) between
> extensions and use some macros but always within the same context.

Well, you should create contexts for your main features, and you can
write few of them in extensions.conf - it's a small trouble when
compared to gain from separation of different functionality.

Regards,
Atis

>
> I'll try to remeber it for next time :)
>
> Cheers,
>
> Yves.
>
> On Fri, 2008-02-08 at 14:36 +0200, Atis Lezdins wrote:
> > On 2/8/08, Yves Räber <yraber at mailup.net> wrote:
> > > * Version: Asterisk 1.4.14
> > >
> > > * Commas instead of pipes => already tried, this is not working at all
> > >
> > > * Realtime switch for script_13_0 => No, should I ? This would be really
> > > a shame, I want to use realtime BECAUSE I don't want to play with my
> > > extensions.conf file. (I'm building a web interface that has to generate
> > > the contexts).
> >
> > Yes, unfortuneately that's the thing you have to do. You have to add
> > each context you want - in static conf file like this:
> >
> > [db_na]
> > switch => Realtime/db_na
> >
> > [db_busy]
> > switch => Realtime/db_busy
> >
> > You can have as many extensions you like with whatever commands, but
> > contexts still should be registered. Generally editing and debugging
> > of complete dialplan in DB is not so easy - so you should keep your
> > main logic in static, but use realtime for data that actually changes.
> >
> > Regards,
> > Atis
> >
> > >
> > > * Using numbers instead of 's' => already tried, no changes
> > >
> > > * Renaming contexts without underscores => tried it right now, no
> > > changes
> > >
> > > Thanks for all those ideas.
> > >
> > > Yves.
> > >
> > > On Thu, 2008-02-07 at 17:39 +0200, Atis Lezdins wrote:
> > > > On 2/7/08, Yves Räber <yraber at mailup.net> wrote:
> > > > > I would have been happy ... but it's not that. This query gives me the
> > > > > right row (I double checked).
> > > > >
> > > > > On Thu, 2008-02-07 at 08:36 -0600, Tilghman Lesher wrote:
> > > > > > On Thursday 07 February 2008 08:05:40 Yves Räber wrote:
> > > > > > > Hello,
> > > > > > >
> > > > > > > I'm having troubles while using the "Goto" function in a realtime
> > > > > > > extension. Here is the error message :
> > > > > > >
> > > > > > > -- Executing Goto("SIP/siemens1-081f56b0", "script_13_0|s|1")
> > > > > > > -- Goto (script_13_0,s,1)
> > > > > > > [Feb  7 13:24:21] WARNING[28666]: pbx.c:2455 __ast_pbx_run: Channel
> > > > > > > 'SIP/siemens1-081f56b0' sent into invalid extension 's' in context
> > > > > > > 'script_13_0', but no invalid handler
> > > > > > >
> > > > > > > And I definitively have a row in my extensions table with context
> > > > > > > script_13_0, exten s and priority 1 !
> > > > > > >
> > > > > > > I also tried to goto in another context that is in my extensions.conf
> > > > > > > file, and it works.
> > > > > > >
> > > > > > > Is this a restriction or a bug ? It seems that it's not possible to
> > > > > > > "Goto" to another context within the realtime extensions.
> > > > > >
> > > > > > It's impossible to guess what might be wrong, because you haven't included
> > > > > > a dump from your table.  Try a:
> > > > > >
> > > > > > SELECT * FROM extensions_table WHERE exten='s' AND context='script_13_0'
> > > > > > AND priority='1'
> > > > > >
> > > > > > If that fails, you have your answer.
> > > > > >
> > > >
> > > > What version? You could try replacing pipes with commas. Do you have
> > > > realtime switch statement for script_13_0? Can you try renaming
> > > > context to not use underscores? Try using not "s" but any number (and
> > > > create extension _X.)
> > > >
> > > > Regards,
> > > > Atis
> > > >
> > >
> > >
> > > _______________________________________________
> > > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> > >
> > > asterisk-users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > >    http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
>
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users


-- 
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
atis at iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835



More information about the asterisk-users mailing list