VoIP Mailing List Archives

[lyle at lcrcomputer.net]

The two SIP devices can not share the same SIP registration to
accomplish what you want. You can dial both SIP devices from one dial
command.

For instance, You assign the user extension 120 and SIP device 120a and
120b and dial both devices when you call out to extension 1234.

Lyle
stephan schneider wrote:

[mwatson at becon.org]

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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[eric at fnords.org]

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:

[picstef at freenet.de]

Hey Matt, hey Lyle,

thanks for your suggestions... Thanks for you suggestions!

Unfortunately we're going to use elastix - and maybe changing
the extensions.conf isn't such a good idea...

What I've found out about the old system - where the multi-ring does
work - is that it is setup using SER...

So maybe SER is the solution... Has anyone experiences setting up
SER or OpenSER into an existing installation?
Thanks again,
Stefan


Matt Watson schrieb: