Re: [asterisk-users] how to implement CLONED LINE Feature in asterisk?
2009/8/4 Faheem faheem_...@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=1...@default
[line2]
username=line2
secret=line2password
type=friend
host=dynamic
context=outboundcalls
mailbox=1...@default
In extensions.conf
[default]
exten = 1234,1,NoOp(About to dial both phones)
exten = 1234,n,Macro(stdexten,${EXTEN},SIP/line1SIP/line2)
exten = 1234,n,Hangup()
or for trunk
[default]
exten = 1234,1,NoOp(About to dial both phones)
exten = 1234,n,Gosub(stdexten(${EXTEN},SIP/line1SIP/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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Re: [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...@tucny.com wrote:
From: D Tucny d...@tucny.com
Subject: Re: [asterisk-users] how to implement CLONED LINE Feature in asterisk?
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Wednesday, August 5, 2009, 11:06 AM
2009/8/4 Faheem faheem_...@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=1...@default
[line2]
username=line2
secret=line2password
type=friend
host=dynamic
context=outboundcalls
mailbox=1...@default
In extensions.conf
[default]
exten = 1234,1,NoOp(About to dial both phones)
exten = 1234,n,Macro(stdexten,${EXTEN},SIP/line1SIP/line2)
exten = 1234,n,Hangup()
or for trunk
[default]
exten = 1234,1,NoOp(About to dial both phones)
exten = 1234,n,Gosub(stdexten(${EXTEN},SIP/line1SIP/line2))
exten = 1234,n,Hangup
Re: [asterisk-users] how to implement CLONED LINE Feature in asterisk?
I'd suggest using different user names and getting asterisk to handle the
cleverness... And, well, doing it this way is pretty simple, straight
forward, basic asterisk functionality...
Trying to get two different instances registered as the same user, is, as
you've found out, not going to be trivial to implement... It's just not how
it works...
If you implemented what I suggested below, using two different usernames for
the two ports on the SPA, it would just work...
d
2009/8/5 Faheem faheem_...@yahoo.com
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...@tucny.com* wrote:
From: D Tucny d...@tucny.com
Subject: Re: [asterisk-users] how to implement CLONED LINE Feature in
asterisk?
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Wednesday, August 5, 2009, 11:06 AM
2009/8/4 Faheem
faheem_...@yahoo.comhttp://mc/compose?to=faheem_...@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=1...@default
[line2]
username=line2
secret=line2password
type=friend
host=dynamic
context=outboundcalls
mailbox=1...@default
In extensions.conf
[default]
exten = 1234,1,NoOp(About to dial both phones)
exten = 1234,n,Macro(stdexten,${EXTEN},SIP/line1SIP/line2)
exten = 1234,n,Hangup()
or for trunk
[default]
exten = 1234,1,NoOp(About to dial both phones)
exten = 1234,n,Gosub(stdexten(${EXTEN},SIP/line1SIP/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
-Inline Attachment Follows-
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[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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [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. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
