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

[Rizwan Hisham]

How about using a SIP URI. I have not tested it but seems like it can work
in your scenarion. check these links:

http://www.voip-info.org/tiki-index.php?page=SIP%20URI
http://www.voip-info.org/wiki/view/Asterisk+tips+SIP+URI+Dial
http://www.ietf.org/rfc/rfc2396.txt


On Mon, May 26, 2008 at 8:57 PM, stephan schneider <picstef at freenet.de>
wrote:

> 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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-- 
Best Regards
Rizwan Hisham
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