[asterisk-users] Registration of multiple SIP-clients for the same extensions

[Eric Wieling]

We use the MAC of the device as it's SIP user ID with -a, -b, -c, etc 
appended to it to indicated the individual call appearances.

An extension is totally separate and different from a SIP peer.  An 
extension is a set of numbers you dial.  Those numbers, when received by 
the Asterisk tells Asterisk where in the dialplan to send the call.  The 
call can then be routed to an IVR, a SIP device, an IAX2 device, etc.

Matt Watson wrote:
> I think the way you are going to have to do this is by having 2 separate SIP peers for each user, 1 for the softphone, 1 for the hardphone.
> 
> Then your dialplan is going to be something like:
> 
> exten => 999,1,Dial(SIP/120&SIP/121)
> 
> where "999" is their extension number and "120" and "121" are the names of the SIP peers for the soft & hardphones.
> 
> --
> Matt
> http://www.mattgwatson.ca
> 
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of stephan schneider
> Sent: Monday, May 26, 2008 11:58 AM
> To: asterisk-users at lists.digium.com
> Subject: [asterisk-users] Registration of multiple SIP-clients for the same extensions
> 
> Hello,
> 
> we want to setup the following scenario:
> 
> - each user has a softphone AND a hardphone
> - the softphone is started with the operating system
> - the hardphone is connected all the time using SIP
> - only ONE extension for each user
> 
> Both phones should ring when the user is called.
> 
> We've setup an asterisk 1.4.18 and at the moment only
> the last registered client rings.
> 
> 
> In Asterisk 1.2 the setup worked, but it does not longer
> in 1.4...
> 
> # sip.conf
> 
>                                         [general]
> bindport = 5060           ; Port to bind to (SIP is 5060)
> 
> 
> bindaddr = 0.0.0.0    ; Address to bind to (all addresses on machine)
> 
> 
> disallow=all
> 
> 
> allow=ulaw
> 
> 
> allow=alaw
> 
> 
> tos=0x68
> 
> 
> notifyringing=yes
> 
> 
> notifyhold=yes
> 
> 
> limitonpeers=yes
> 
> [120]
> type=friend
> 
> 
> secret=secret
> 
> 
> record_out=Adhoc
> 
> 
> record_in=Adhoc
> 
> 
> qualify=yes
> 
> 
> port=5060
> 
> 
> pickupgroup=
> 
> 
> nat=yes
> 
> 
> mailbox=120 at default
> 
> 
> host=dynamic
> 
> 
> dtmfmode=inband
> 
> 
> disallow=
> 
> 
> dial=SIP/120
> 
> 
> context=from-internal
> 
> 
> canreinvite=no
> 
> 
> callgroup=
> 
> 
> callerid=device <120>
> 
> 
> allow=
> 
> 
> accountcode=
> 
> 
> call-limit=50
> 
> 
> Maybe someone has an idea how to setup the scenario without using
> ringgroups...
> 
> 
> Thanks a lot,
> Stefan
> 
> 
> 
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