Faheem
2009-Aug-04 09:54 UTC
[asterisk-users] how to implement CLONED LINE Feature in asterisk?
how to implement CLONED LINE Feature in asterisk
Hey, I want to implement Clone Line feature in asterisk. I am using SPA-2100.
The feature should work in this way.
There are two ports in the SPA-2100 both are registered with asterisk with same
username/password, and have the same (phone number)
No
one on the phone
One
phone in use
Both
phones in use
Incoming
Calls
Both
phones ring
Phone
in use receives call waiting notification, unused phone rings
Both
phones receive call waiting notification
Outgoing
Calls
Both
phones can call out
The
unused phone can call out
Neither
phone can call out
?* Inbound:
????? - Both ports will ring.?Whichever port is picked up first, will field the
call.
????? - Any additional calls that come in would give call waiting notification
to the first line, and ring the second line.
????? - Once the second line is being utilized, all incoming calls will be
notifications in the form of call waiting beeps.
?* Outbound:
????? - You will have the ability to dial out from port one.
????? - You will be able to dial a different party on port two.
*** Note ***
???????? - If you have an active call on port one, and pick up port two, you
will NOT have the same call that is currently active on port one.?The Cloned
Line will share the same voice mail and will have the same telephone number as
the original telephone line.
? -? The Cloned Line is NOT a second telephone number.? The telephone number
that is assigned to the second phone port on the device is the same telephone
number as the number assigned to phone port one.
Thanks!
Faheem
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Stanisław Pitucha
2009-Aug-04 11:21 UTC
[asterisk-users] how to implement CLONED LINE Feature in asterisk?
Probably the easiest way: put an opensips box in front of asterisk. It can handle multiple registrations on the same username. If you have multiple registrations, it will do a parallel fork and work just like you wanted. You just have to make sure that phones register on opensips, not asterisk.
D Tucny
2009-Aug-05 06:06 UTC
[asterisk-users] how to implement CLONED LINE Feature in asterisk?
2009/8/4 Faheem <faheem_imt at yahoo.com>> how to implement CLONED LINE Feature in asterisk > > Hey, I want to implement Clone Line feature in asterisk. I am using > SPA-2100. > The feature should work in this way. > > There are two ports in the SPA-2100 both are registered with asterisk with > same username/password, and have the same (phone number) > > > > *No one on the phone * > > *One phone in use * > > *Both phones in use * > > *Incoming Calls * > > Both phones ring > > Phone in use receives call waiting notification, unused phone rings > > Both phones receive call waiting notification > > *Outgoing Calls * > > Both phones can call out > > The unused phone can call out > > Neither phone can call out > > > * Inbound: > - Both ports will ring. Whichever port is picked up first, will field > the call. > - Any additional calls that come in would give call waiting > notification to the first line, and ring the second line. > - Once the second line is being utilized, all incoming calls will be > notifications in the form of call waiting beeps. > > * Outbound: > - You will have the ability to dial out from port one. > - You will be able to dial a different party on port two. > > *** Note *** > - If you have an active call on port one, and pick up port two, > you will NOT have the same call that is currently active on port one. The > Cloned Line will share the same voice mail and will have the same telephone > number as the original telephone line. > > - The Cloned Line is NOT a second telephone number. The telephone > number that is assigned to the second phone port on the device is the same > telephone number as the number assigned to phone port one. >In sip.conf [line1] username=line1 secret=line1password type=friend host=dynamic context=outboundcalls mailbox=1234 at default [line2] username=line2 secret=line2password type=friend host=dynamic context=outboundcalls mailbox=1234 at default In extensions.conf [default] exten => 1234,1,NoOp(About to dial both phones) exten => 1234,n,Macro(stdexten,${EXTEN},SIP/line1&SIP/line2) exten => 1234,n,Hangup() or for trunk [default] exten => 1234,1,NoOp(About to dial both phones) exten => 1234,n,Gosub(stdexten(${EXTEN},SIP/line1&SIP/line2)) exten => 1234,n,Hangup() then stdexten would be default as comes in the sample configs... That should be everything you want if you configure the SPA-2100 to register line 1 with username line1 and line 2 with username line2... d -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090805/ee16bd63/attachment.htm
Faheem
2009-Aug-05 09:50 UTC
[asterisk-users] how to implement CLONED LINE Feature in asterisk?
By placing OPENSIP in front of Asterisk, we can register multiple
accounts, and we can successfully make call for Outgoing only. But in
case of incoming it fails.
If two users are registered with asterisk or OpenSIP then the user that
is registered latest is considered to be valid, and he is able to make
calls, other user with earlier registration can not make call.
My point here is in chain_sip.c what are variables or structure that
need to maintain so that we can consider all registered users as active
users.
Thanks!
Faheem
--- On Wed, 8/5/09, D Tucny <d at tucny.com> wrote:
From: D Tucny <d at tucny.com>
Subject: Re: [asterisk-users] how to implement CLONED LINE Feature in asterisk?
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Date: Wednesday, August 5, 2009, 11:06 AM
2009/8/4 Faheem <faheem_imt at yahoo.com>
how to implement CLONED LINE Feature in asterisk
Hey, I want to implement Clone Line feature in asterisk. I am using SPA-2100.
The feature should work in this way.
There are two ports in the SPA-2100 both are registered with asterisk with same
username/password, and have the same (phone number)
No
one on the phone
One
phone in use
Both
phones in use
Incoming
Calls
Both
phones ring
Phone
in use receives call waiting notification, unused phone rings
Both
phones receive call waiting notification
Outgoing
Calls
Both
phones can call out
The
unused phone can call out
Neither
phone can call out
?* Inbound:
????? - Both ports will ring.?Whichever port is picked up first, will field the
call.
????? - Any additional calls that come in would give call waiting notification
to the first line, and ring the second line.
????? - Once the second line is being utilized, all incoming calls will be
notifications in the form of call waiting beeps.
?* Outbound:
????? - You will have the ability to dial out from port one.
????? - You will be able to dial a different party on port two.
*** Note ***
???????? - If you have an active call on port one, and pick up port two, you
will NOT have the same call that is currently active on port one.?The Cloned
Line will share the same voice mail and will have the same telephone number as
the original
telephone line.
? -? The Cloned Line is NOT a second telephone number.? The telephone number
that is assigned to the second phone port on the device is the same telephone
number as the number assigned to phone port one.
In sip.conf
[line1]
username=line1
secret=line1password
type=friend
host=dynamic
context=outboundcalls
mailbox=1234 at default
[line2]
username=line2
secret=line2password
type=friend
host=dynamic
context=outboundcalls
mailbox=1234 at default
In extensions.conf
[default]
exten => 1234,1,NoOp(About to dial both phones)
exten => 1234,n,Macro(stdexten,${EXTEN},SIP/line1&SIP/line2)
exten => 1234,n,Hangup()
or for trunk
[default]
exten => 1234,1,NoOp(About to dial both phones)
exten => 1234,n,Gosub(stdexten(${EXTEN},SIP/line1&SIP/line2))
exten => 1234,n,Hangup()
then stdexten would be default as comes in the sample configs...
That should be everything you want if you configure the SPA-2100 to register
line 1 with username line1 and line 2 with username line2...
d
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