[Asterisk-Users] Snom 200 Voice Call / Paging

[Christian Stredicke]

Hi Willy, 

I must admit I am clueless when it comes to configuration files for
Asterisk. I am talking from a SIP point of view. Maybe someone can translate
this into Asterisk configuration files....

Christian

> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com [mailto:asterisk-users-
> admin at lists.digium.com] On Behalf Of willy at yponeinc.com
> Sent: Tuesday, March 23, 2004 1:06 PM
> To: asterisk-users at lists.digium.com
> Subject: RE: [Asterisk-Users] Snom 200 Voice Call / Paging
> 
> Christian,
> I guess I am Confused about the 'header' stuff.
> 
> I am using the SIP strictly on a LAN as extensions to the
> [*].  Hence, I have a line in sip.conf like this:
> 
> [2200] ;snom 200
> callerid=Reception <2200>
> type = friend
> disallow=all
> allow=ulaw
> allow=alaw
> username = 2200
> secret = 2200
> host = dynamic
> dtmfmode = rfc2833
> context=intern
> mailbox = 2200
> 
> In extensions.conf I have
> exten => 2200,1,Dial(SIP/2200,20,tT)
> 
> Now, [*] is at 192.168.1.16. Where does the 'header' you
> refer to get sent?
> I tried adding intercom=true to the sip.conf but that is not
> it right?
> Lost ...
> Willy
> 
> ----- Original Message Follows -----
> > To use "Intercom" mode in the current releases of the snom
> > 200, you need to use an "intercom=true" flag in the
> > To-Header. Essentially that makes the phone to pick up the
> > call immediately.
> >
> > To: <sip:123 at bla.com;intercom=true>
> >
> > However, this mechanism is likely to change because of
> > security concerns and new interoperable methods.
> >
> > Christian
> >
> > > -----Original Message-----
> > > From: asterisk-users-admin at lists.digium.com
> > > [mailto:asterisk-users- admin at lists.digium.com] On
> > > Behalf Of willy at yponeinc.com Sent: Sunday, March 21,
> > > 2004 5:25 PM To: asterisk-users at lists.digium.com
> > > Subject: [Asterisk-Users] Snom 200 Voice Call / Paging
> > >
> > > To All,
> > > Several months (2003) ago there was a discussion
> > > regarding overhead paging & intercom functionality with
> > > SIP / Asterisk.  Jerry Gibson, John Todd and various
> > > others participated (from checking the archives).  One
> > > person even responded that they had the stuff working
> > > with the snom 200s.
> > > Voice Call (i.e. on-hook speaker/mic) is realy important
> > > in a lot of apps.  It would appear that the snom 200 and
> > > by extension the snom 105 support the functionality.
> > > I will be happy to make a wiki entry to explain & demo
> > > this functionality once I have it working properly.  I
> > > also understand that the (mis)use of conferencing is
> > > frowned upon as it wastes bandwidth and CPU.  However,
> > > until a better way comes around, that is not a problem
> > > as there are quite a few applications where (a) one
> > > needs Voice Call (which is 1 <-> 1) and / or an
> > > 'allPage' which can be limited to a subset of all
> > > phones.  Typically phones which are in designated or
> > > public areas, conference rooms, etc.  The BW/CPU issue
> > can be controlled. Better a limited solution than no
> > > solution at all ;)
> > > I am also allowing for the limitation that all
> > > participating phones are on the same LAN with the [*].
> > > Anyone who has this successfully working with snom,
> > > please respond ..  Using the [*] sound card for a
> > > separate PA system is NOT an option ;)
> > > As I said, I will be 'distilling' the info and turn it
> > > into a wiki entry.
> > > Cheers and TIA,
> > > Willy
> > >
> > > Willy Wouters
> > > ypOne Publishing
> > >
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> Willy Wouters
> ypOne Publishing
> 
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