[Asterisk-Users] Incoming Calls

[C F]

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).


On Mon, 27 Dec 2004 07:04:00 -0600, Rich Adamson <radamson at routers.com> wrote:
> > 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.
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>