[asterisk-users] snom led not working with asterisk 1.4.1

Andrew Latham lathama at lathama.com
Wed Mar 21 06:52:18 MST 2007


I think he meant DND, you can program the DND to send Asterisk a call
like *79 or something like that....



On 3/21/07, Steve Murphy <murf at digium.com> wrote:
> On Wed, 2007-03-21 at 13:55 +0100, Giorgio Incantalupo wrote:
> > Hi Steve,
> > thank you for your help, I set up the call-limit parameter  and SNOM
> > light are working good for ringing and busy status. I took a look at
> > sip.conf.sample but nothing about unavailable status. Should I set some
> > other parameter or there is some "trick"? Consider that my firmware
> > phone is updated to the last available version.
> >
> > TIA
> >
> > Giorgio Incantalupo
> >
> >
> Giorgio--
>
> no tricks, sorry!... I've got a snom360 here, and I've been slowly
> working my way thru the buttons myself. There's a config file option to
> make the "Retrieve" button work, you provide a name for an extension for
> it to use. You then provide
> that extension in the context for the phone, that does the
> VoiceMailMain() call.
>
> The "Record" button uses a SIP "INFO" message to asterisk, that isn't
> implemented, so that's not going to work at the moment.
>
> What does "unavailable" mean, and how do you get that way?
>
> murf
>
> >
> > Steve Murphy wrote:
> > > On Thu, 2007-03-15 at 15:33 +0100, Giorgio Incantalupo wrote:
> > >
> > >> Hi,
> > >> I'm testing Asterisk 1.4.1 with Snom phones but leds are not working to
> > >> show which devices are busy/not connected. The same phone worked with
> > >> Asterisk 1.2.9.1.
> > >> I would appreciate anyone who knows how to setup Asterisk 1.4.1 to
> > >> behave as 1.2.9.1.
> > >>
> > >
> > > Giorgio--
> > >
> > > That's a pretty generic question! But that aside, there's been a
> > > substantive change in the configs for SIP phones, that could easily
> > > affect your device state monitoring.
> > >
> > > So, suggestion: read the example sip config file in the src/configs dir,
> > > pay close attention to stuff like call-limit, the limitonpeers stuff,
> > > etc, and then make sure you update all your phone entries in sip.conf.
> > > Restart asterisk, or reload sip, and hopefully your lights will work.
> > >
> > > In general, EVERYONE, here's some advise: When you
> > > upgrade from version "1.x" to "1.(x+2)", always review ALL
> > > your config files against the new config file examples.
> > > Things change! Hopefully, for the better!
> > >
> > > murf
> > >
> > >
> >
> > _______________________________________________
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >    http://lists.digium.com/mailman/listinfo/asterisk-users
> --
> Steve Murphy
> Software Developer
> Digium
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>


-- 
---
Andrew Latham - AKA: LATHAMA (lay-th-ham-eh)
lathama at lathama.com - lathama at gmail.com
If any of the above are down we have bigger problems than my email!
Hind sight is most always 20/20 or better.
---


More information about the asterisk-users mailing list