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

stephan schneider picstef at freenet.de
Tue May 27 06:46:48 CDT 2008


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:
> 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
> 
> 
> 
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users




More information about the asterisk-users mailing list