[Asterisk-Users] sharing a line w/multiple extensions

[Rich Adamson]

> >> I'd like to configure Asterisk so an incoming call from one POTS line 
> >> is shared amongst multiple extensions - both SIP and analog.  i.e.  
> >> If one SIP phone answers the call, another SIP or analog extension 
> >> phone can pick up and join the conversation.  How do I configure 
> >> this?  Is it all in extensions.conf?
> >
> > Asterisk is not a key system. It does not behave this way.
> >
> > What do you mean by 'another SIP phone can pick up (...) the 
> > conversation'? Exactly what would the SIP phone user do to accomplish 
> > that?
> >
> Think residential installation where someone picks up the phone in one 
> room but someone in another room wants to join the conversation.  
> Ideally, I'd like to have "Line 1" on every phone (SIP or analog) behave 
> this way.  Another poster pointed out a good potential approach using 
> meetme.  When an incoming call comes in, it dials all SIP + analog 
> phones.  When someone picks up (don't know how I can detect this), it 
> could transfer both parties to a meetme room.  When additional 
> extensions pickup, they go to the meetme room.  When everyone hangs up, 
> the call ends.  Can this be done?

There might be a way for you to address your objective depending upon
exactly what you're trying to do.

The previous responses to your question _assume_ that each room in
your case has a pbx extension (regardless of whether its a sip or analog
phone). If their assumption is correct, then the responses are correct.

However, if you want to use your existing analog phones and you group
them together, several analog phones "can" share a single extension
and those phones in the group can pick up and join the conversation
whenever they want. Think in terms of using something like a Sipura
sip adapter (or the equivalent from other vendors), and connecting all
analog phones within your defined group to the rj11 analog jack of
the adapter.

For example, I have four analog pstn lines and multiple iax connections to
various itsp's and clients. One of the analog pstn lines is a house line
and connects directly to * via a TDM04b card. When an incoming call occurs
on that line, it rings multiple sip phone/adapters. One of those happens
to be a Sipura spa3000 that has most of the analog house phones attached.
Anyone one of those phones can answer the call, others can join in, etc.

The approach can work "if" you can define specific groups of interest
such as kids vs adults, sales vs support, home vs business, etc.
Combine that approach with carefull selection of analog phones (those
with some form of "line in use" LED), and you end up with an approach
that sort of looks like a poor-man's key system behind a pbx. Pay attention
to the features within the sip adapter (eg, Sipura) and you're likely to
find additional options that might address your needs.

All depends upon exactly what it is that you're trying to engineer.