[Asterisk-Users] Incoming Calls

[Rich Adamson]

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. 

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> 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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