[Asterisk-Users] Picking a channel (FXO port or SIP) for outbound calls

[Steven Critchfield]

On Mon, 2003-11-24 at 14:28, Chris Albertson wrote:
> Here is a little bit harder question:
> 
> I want my local outbound calls to use an FXO interface
> as described in this thread however...
> 
> If there is no available FXO interface then I'd like the
> called go through my SIP service provider who will gateway
> the call back to PSTN for me (for a small per minute fee and
> slight loss of quality)  I want to use SIP as a backup.
> 
> In some long distance cases SIP will be primary with the
> FXO as a backup (incase the Intrnet service is down or whatever)
> 
> So the generized question is how to do across channel type
> fail overs?

Well you just look at the examples for voicemail. If a line is busy you
jump to 100+n, where n is the current priority. So you could have;
exten => _NXXXXXX,1,Dial(Zap/g1/${EXTEN})
exten => _NXXXXXX,2,Hangup ;fall though prevention 
exten => _NXXXXXX,101,Dial(Sip/provider/${EXTEN})


Simple enough.

> --- Joe Kellman <joe at soulero.com> wrote:
> > first you would set up a group in zapata.conf
> >   [channels]
> >   signalling=fxs_ks
> >   group=1
> >   channel=1-2
> > 
> > 
> > then in your extensions.conf file replace your dial
> > line with this:
> >   exten => _9.,1,Dial(Zap/g1/${EXTEN:1},90,Tt)
> > 
> > ..Hope this helps...jak
> > 
> > --- Tony Kava <asterisk at pottcounty.com> wrote:
> > > Greetings:
> > > 
> > > I did some quick searching of my history of this
> > > list, and I tried a quick
> > > Google search as well, but perhaps someone on the
> > > list can quickly answer
> > > this question.  I have a very nicely working
> > > Asterisk system at home with
> > > two Digium X100P FXO cards.  When my SIP phones want
> > > to dial-out I have them
> > > setup to grab the first analog card (Zap/1) with the
> > > following
> > > extensions.conf segment:
> > > 
> > > ==== snippet ====
> > > 
> > > ; Outbound
> > > exten => _9.,1,Dial(Zap/1/${EXTEN:1},90,Tt)
> > > exten => _9.,2,Macro(fastbusy)
> > > exten => _9.,102,Macro(fastbusy)
> > >   
> > > ==== /snippet ====
> > > 
> > > Zap/1 and Zap/2 are analog phone lines.  What is the
> > > best method of picking
> > > an open line when someone tries to dial-out? i.e. if
> > > Zap/1 is in use how can
> > > I instruct Asterisk to use Zap/2 and vice versa? I
> > > know complex methods of
> > > making this happen, but I'm sure there is a very
> > > simple way to accomplish
> > > this task.
> > > 
> > > Thanks.
> > > 
> > > --
> > > Tony Kava
> > > Network Administrator
> > > Pottawattamie County, Iowa
> > > _______________________________________________
> > > Asterisk-Users mailing list
> > > Asterisk-Users at lists.digium.com
> > >
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
> =====
> Chris Albertson
>   Home:   310-376-1029  chrisalbertson90278 at yahoo.com
>   Cell:   310-990-7550
>   Office: 310-336-5189  Christopher.J.Albertson at aero.org
>   KG6OMK
> 
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-- 
Steven Critchfield  <critch at basesys.com>