stephan schneider
2008-May-26 15:57 UTC
[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
Lyle Giese
2008-May-26 18:36 UTC
[asterisk-users] Registration of multiple SIP-clients for the same extensions
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:> 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 >
Matt Watson
2008-May-26 18:51 UTC
[asterisk-users] Registration of multiple SIP-clients for the same extensions
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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Rizwan Hisham
2008-May-27 08:05 UTC
[asterisk-users] Registration of multiple SIP-clients for the same extensions
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 > > > > _______________________________________________ > -- 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 >-- Best Regards Rizwan Hisham -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080527/84f091de/attachment.htm