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[asterisk-users] Custom Pickup and Transfer dial string


 
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mlist at itspecialist.it
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PostPosted: Tue Jan 22, 2008 6:11 am    Post subject: [asterisk-users] Custom Pickup and Transfer dial string Reply with quote

Hi to all,
i already searched the archive without finding a solution to my problem.
I have asterisk installation 1.2.18 to support multiple virtiual PBXs.
I use SIP peer in the format <ID>-<EXT> to let every virtual PBX to
share the same numbers of EXT.
Ex.

(PBX ID 10 Extensions)
10-101
10-102
10-103

(PBX ID 20 Extensions)
20-101
20-102
20-103

I use some rules in the dialplan to let users of this pbxs to dial
only EXT to call other extensions of their PBX.
So i add the pbx ID statically to the called number if it is an
extension.
Now i have to realize the pickup call and blind and attended call
transfer.
As i read in the documentation the standard codes *2 for transfer give
me the dial tone to dial the extension to
transfer at but i need to transfer the call not to the EXT but ID-EXT .
Have you any idea on how to realize this for Pickup and call transfer?
The end users don't have any idea of the ID part of their extension so
i have to let them dial something like *2101 and
transfer the call to 10-101 . For the pickup i need them to dial
something like *8101 and answer the call ringing for 10-101.
I already thought a solution for the Pickup to use the application
Pickup joined to a custom pickup code and a custom string in
the command it self, but i dont have a solution for the call transfer.
Thanks in advance for help.
Bye,
Marcello
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arielb27 at hotmail.com
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PostPosted: Tue Jan 22, 2008 7:03 am    Post subject: [asterisk-users] Custom Pickup and Transfer dial string Reply with quote

If your working with Virtual PBX then why not set your users with there own
rules and normal extension numbers in there own context. You can have many
context.

That way only extensions you allow to see the context there in will have
those options.

----- Original Message -----
From: "Marcello Lupo" <mlist at itspecialist.it>
To: <asterisk-users at lists.digium.com>
Sent: Tuesday, January 22, 2008 6:11 AM
Subject: [asterisk-users] Custom Pickup and Transfer dial string
Quote:
Hi to all,
i already searched the archive without finding a solution to my problem.
I have asterisk installation 1.2.18 to support multiple virtiual PBXs.
I use SIP peer in the format <ID>-<EXT> to let every virtual PBX to
share the same numbers of EXT.
Ex.

(PBX ID 10 Extensions)
10-101
10-102
10-103

(PBX ID 20 Extensions)
20-101
20-102
20-103

I use some rules in the dialplan to let users of this pbxs to dial
only EXT to call other extensions of their PBX.
So i add the pbx ID statically to the called number if it is an
extension.
Now i have to realize the pickup call and blind and attended call
transfer.
As i read in the documentation the standard codes *2 for transfer give
me the dial tone to dial the extension to
transfer at but i need to transfer the call not to the EXT but ID-EXT .
Have you any idea on how to realize this for Pickup and call transfer?
The end users don't have any idea of the ID part of their extension so
i have to let them dial something like *2101 and
transfer the call to 10-101 . For the pickup i need them to dial
something like *8101 and answer the call ringing for 10-101.
I already thought a solution for the Pickup to use the application
Pickup joined to a custom pickup code and a custom string in
the command it self, but i dont have a solution for the call transfer.
Thanks in advance for help.
Bye,
Marcello


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mlist at itspecialist.it
Guest





PostPosted: Tue Jan 22, 2008 7:23 am    Post subject: [asterisk-users] Custom Pickup and Transfer dial string Reply with quote

Hi,
it is already that way but my config is made this way because i need
unique SIP peers to dial at.
May be i'm missing something but as i know is not possible to restrict
SIP peers (in sip.conf) in different context.
It should me made under multidomain support (one domain for each
peer). This is the reason of my config.
Actually every Virtual PBX have his own contexts in dialplan and are
not visible between them, but the sip peers are
all in the same basket so i have to differentiate the sip peer to call
on PBX ID base.
Let me know if i'm missing something.
Thanks,
Bye,
Marcello

On Jan 22, 2008, at 1:03 PM, Ariel Batista Jr. wrote:

Quote:
If your working with Virtual PBX then why not set your users with
there own
rules and normal extension numbers in there own context. You can
have many
context.

That way only extensions you allow to see the context there in will
have
those options.

----- Original Message -----
From: "Marcello Lupo" <mlist at itspecialist.it>
To: <asterisk-users at lists.digium.com>
Sent: Tuesday, January 22, 2008 6:11 AM
Subject: [asterisk-users] Custom Pickup and Transfer dial string


Quote:
Hi to all,
i already searched the archive without finding a solution to my
problem.
I have asterisk installation 1.2.18 to support multiple virtiual
PBXs.
I use SIP peer in the format <ID>-<EXT> to let every virtual PBX to
share the same numbers of EXT.
Ex.

(PBX ID 10 Extensions)
10-101
10-102
10-103

(PBX ID 20 Extensions)
20-101
20-102
20-103

I use some rules in the dialplan to let users of this pbxs to dial
only EXT to call other extensions of their PBX.
So i add the pbx ID statically to the called number if it is an
extension.
Now i have to realize the pickup call and blind and attended call
transfer.
As i read in the documentation the standard codes *2 for transfer
give
me the dial tone to dial the extension to
transfer at but i need to transfer the call not to the EXT but ID-
EXT .
Have you any idea on how to realize this for Pickup and call
transfer?
The end users don't have any idea of the ID part of their extension
so
i have to let them dial something like *2101 and
transfer the call to 10-101 . For the pickup i need them to dial
something like *8101 and answer the call ringing for 10-101.
I already thought a solution for the Pickup to use the application
Pickup joined to a custom pickup code and a custom string in
the command it self, but i dont have a solution for the call
transfer.
Thanks in advance for help.
Bye,
Marcello


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To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users



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