[Asterisk-Users] Incoming Calls

[C F]

---------- Forwarded message ----------
From: C F <shmaltz at gmail.com>
Date: Tue, 28 Dec 2004 13:21:45 -0500
Subject: Re: [Asterisk-Users] Incoming Calls
To: Rich Adamson <radamson at routers.com>


I didn't try Dial but I did try wait and it didn't help. I'll try dial
and see what happens. It might take a while until I do that, since I'm
waiting for a new TDM400 (the other one I installed by a client).


On Tue, 28 Dec 2004 10:32:23 -0600, Rich Adamson <radamson at routers.com> wrote:
> I'd have to guess that in your example, the exten=s entries are the
> root of the issue and is answering the call when you didn't expect it.
> Try something like this:
>  [default]
>  exten => s,1,Dial(SIP/3010)
>  exten => s,2,Hangup
> where 3010 is a valid extension. You should find that zap/4 is not
> answered until you pick up exten 3010.
>
> Also, based only on what you're showing below, it does not look like
> the contexts are working the way that you think they should. It
> appears the [default] either drops through and executes the statements
> in [incoming], or, there is something else going on in your specific
> case where the contexts aren't what you expect.
>
> ------------------------
> > I don't know what I did wrong but it didn't work. Here is how I
> > configured it (i have a TDM400, configured with 4 fxo, channel 4 was
> > the one I wanted to share):
> > zapata.conf
> > .......
> > context=incoming
> > channel=1-3
> > context=default
> > channel=4
> > ===========
> > extensions.conf
> > .......
> > [default]
> >
> > [incoming]
> > exten=s,1,do some code
> > ======================
> > I left the default context blank with no extensions b/c I didn't want
> > it to pick up. However * would pick up and in the console I get:
> > invalid extension s,1 in context default, then it would look for the t
> > extension. But it picked up after 2 rings. what am I doing wrong?
> >
> >
> >
> > On Tue, 28 Dec 2004 05:31:12 -0600, Rich Adamson <radamson at routers.com> wrote:
> > > Not sure why it didn't work for you unless we are talking about two
> > > different things. It does work for me and has been working just fine
> > > for over a year now.
> > >
> > > ------------------------
> > > > Just a note on this. I tried using an external device with the TDM400
> > > > configured as 4 FXO to ring even with asterisk. But no matter how I
> > > > configured it, asterisk always picked up. and the external device
> > > > didn't ring (just the first ring for CallerID to come in).
> > > >
> > > >
> > > > > > Here is where the problem is.
> > > > > >
> > > > > > When the call comes in, it will be ringing on 2 of the FXO ports,
> > > > > > and all the other phones in the office. I would like various / all
> > > > > > the IP phones to ring, however asterisk must not answer the call
> > > > > > while that is happening or else the normal extension would not
> > > > > > continue ringing. Obviously when an IP phone answers it will then
> > > > > > pick up the call and connect the 2. Is this possible, or is this
> > > > > > how it normally works by default?
> > > > >
> > > > > Maybe. Part of the answer is dependent upon exactly how your existing
> > > > > pbx handles the call.
> > > > >
> > > > > The approach I'd use for testing purposes is _not_ to ring both
> > > > > extensions to asterisk, but rather just one of them. When that
> > > > > extension rings, asterisk's fxo card will sense the ringing and
> > > > > the logic within your dialplan will have something like:
> > > > >  exten => s,1,Dial(${PHONE1}&${PHONE2})
> > > > > that will cause two sip phones to ring. You can add more sip phones
> > > > > to that statement if you'd like. If one of those sip phones answers
> > > > > the call, the fxo port will go off-hook (to your existing pbx),
> > > > > causing it to believe the call was answered; the existing pbx analog
> > > > > phones should then stop ringing.
> > > > >
> > > > > If an existing pbx analog extension answers the call, ringing to the
> > > > > asterisk fxo port will stop, and therefore ringing to the sip phones
> > > > > will stop a few seconds later.
> > > > >
> > > > > There will likely be a lag of time between ringing of analog phones
> > > > > and ringing of sip phones (by one or two rings), which might be
> > > > > somewhat disturbing to people that can hear both phones ringing.
> > > > > Should someone answer an analog extension first and someone answers
> > > > > a ringing sip phone seconds later, the sip phone user will hear
> > > > > nothing more then dialtone (depending upon how much lag actually
> > > > > exists).
> > > > >
> > > > > The above essentially says that one of the existing pbx to asterisk
> > > > > fxo interfaces must be dedicated to your special ringing arrangement.
> > > > >
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> > >
> > >
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