[Asterisk-Dev] Re: multiple registrations of same credentials

[Jonathan k. Creasy]

I am reading this because I would like to have a shared line appearance,
or at least I think I would. The goal is to have several phones with a
single line and when it's busy on one, it's busy on all of them. It's
been requested by several users, most of them so that a secretary can
monitor the lines of 5 or 6 different people. 

Can someone respond to me off list and assist me in understanding how to
make use of this information? I'm a programmer but I haven't done custom
coding on Asterisk as of yet. 

I am also wondering if what I am trying to accomplish is already done
another way. I looked at the flash operator panel and will probably end
up just telling the customer that is their only way but it would make
them much happier to have a Snom 360 or Polycom IP601 with a
sidecar/module and a bunch of shared lines on it. 

Let me know if you can help me off list or we can take it to the users
list. 

-Jonathan

-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Mikael
Magnusson
Sent: Monday, October 24, 2005 9:12 AM
To: asterisk-dev at lists.digium.com
Subject: Re: [Asterisk-Dev] Re: multiple registrations of same
credentials

On Sun, Oct 23, 2005 at 10:51:26AM -0400, Jason Pyeron wrote:
> On Sat, 22 Oct 2005, Mikael Magnusson wrote:
> 
> >Olle E. Johansson wrote:
> >>
> >>Which means that the PBX has no idea that it is in fact calling
multiple
> >>devices. We have to implement *two* forking schemes, which makes
> >>everything much harder.
> >>
> >>Believe me, I've spent a lot of time thinking about how to make this
> >>work properly without breaking the Asterisk architecture.
> >>
> >
> >I think it can be solved by using both the peer (or AOR) and
registered 
> >contact address in the dial string. For example:
> >
> >Dial(SIP/400 at example.domain/sip:400 at a.b.c.d)
> >
> >I have implemented this scheme in a SIP channel based on eXosip and
the 
> >"forking" is done with a function (SIPPEER) which returns a dial
string 
> >that contains all current bindings of the peer.
> >
> >It can be used in the following way in the dial plan:
> >
> >exten 400,1,Dial(${SIPPEER(400 at example.domain})
> >
> >If both sip:400 at a.b.c.d and sip:400 at e.f.g.h have registered 
> >sip:400 at example.domain, then the above Dial statement will expand to
> >
> >exten 
>
>400,1,Dial(SIP/400 at example.domain/sip:400 at a.b.c.d&SIP/400 at example.domai
n/sip:400 at e.f.g.h)
> >
> >And it will use the following request URI:s and To headers in the 
> >generated INVITEs.
> >
> >
> >INVITE sip:400 at a.b.c.d SIP/2.0
> >..
> >To: sip:400 at example.domain
> >..
> >
> >
> >INVITE sip:400 at e.f.g.h SIP/2.0
> >..
> >To: sip:400 at example.domain
> >..
> 
> 
> so, have each phone have its own sub-domain and know all the phones
ahead 
> of time and the problem is solved?
> 
> Can I hear more about your implementation?
> 
> Thanks
> 

I used sip:400 at a.b.c.d and sip:400 at e.f.g.h in the above example to 
represent the contact uri:s for two different phones, a.b.c.d and 
e.f.g.h being two (different) IPv4 addresses.

The phone registers its contact uri in a REGISTER request as usual, 
and it doesn't have to be configured ahead of time. Actually the channel

doesn't support pre-configured bindings yet, but it's planned.

It's GPL licensed and an alpha release can be downloaded from:

http://www.hem.za.org/chan_exosip/

/Mikael

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