[asterisk-users] two SIP phones as one line

[Noah Miller]

Hi Mark -

> > > PhoneA is a SIP hard phone, phoneB is an analogue phone connected to a
> > > SIP ATA. When an incoming call comes in, I would like to ring both
> > > phones, but if phoneA is answered first, I would like phoneB to be
> > > answered as well and left in a "off hook" state so that when someone
> > > picks up the receiver of phoneB, they can hear and participate in the
> > > conversation between the calling party and phoneA.
> > >
> > > I believe I would have to put both phones in a MeetMe conference, but
> > > how to I "auto-answer" phoneB when phoneA has answered the call?

Two Questions:

1. On the SIP phone, will this special conference function be needed
on both incoming and outgoing calls, or just one of those?

2. Does the analog phone have to do anything else?  Should it work
like a normal phone when it's not doing this special conference
function?

You should be able to do this, like you said, by dumping both phones
into a meetme conference.  There would be two tricky things here A)
getting the analog phone to automatically go to a meetme conference
whenever it is off-hook, B) getting outgoing calls from the sip phone
into a meetme conference (incoming calls would be easy).

I think A) is probably not possible, given that you are using an
external ATA device.  That device would somehow have to send the
off-hook status back to asterisk via sip messages (I think there's
actually a bounty for this).  This should be possible if you were
using an internal zaptel card rather than an external gateway.  If the
answer to question 2) above is yes, you would have other problems,
too.

A good compromise to the problems of both 2) and A) would be to put
the analog phone into a special context where you'd have a one digit
press for each function (e.g. press 1 for normal phone, press 2 for
conference).

Still, unless these users are really ornery, I'd probably just make
them learn to transfer and dial into a conference.


- Noah


On 10/15/06, Marc Heckmann <mh at nadir.org> wrote:
> On Sun, 2006-15-10 at 05:09 -0400, Henry.L.Coleman wrote:
> > The quirk of your old PBX is in fact exactly what happens when you put any
> > two analog phones on the same line. The easiest way to duplicate this is
> > to connect another analog phone to your ATA. Some analog phones can
> > indicate when the other is on the line and can put a call on hold locally.
>
> In fact no, I should have explained better, but in the old system one
> phone was analogue and the other was a multi-line digital Nortel
> Meridian phone. The one phone has to be analogue because it interfaces
> with a radio broadcast phone patch.
>
> -m
>
> >
> > > Hi,
> > >
> > > I am looking to replace a quirk of our old PBX system functionality with
> > > asterisk but after searching, archives, wiki, etc.. I cannot figure out
> > > how.
> > >
> > > Here is what I would like to do:
> > >
> > > PhoneA is a SIP hard phone, phoneB is an analogue phone connected to a
> > > SIP ATA. When an incoming call comes in, I would like to ring both
> > > phones, but if phoneA is answered first, I would like phoneB to be
> > > answered as well and left in a "off hook" state so that when someone
> > > picks up the receiver of phoneB, they can hear and participate in the
> > > conversation between the calling party and phoneA.
> > >
> > > I believe I would have to put both phones in a MeetMe conference, but
> > > how to I "auto-answer" phoneB when phoneA has answered the call?
> > >
> > > I suspect that this may not be possible with asterisk, but would like
> > > confirmation of that.
> > >
> > > Thanks in advance.
> > >
> > > -m
> > >
>
>
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