PBX CallTransfer
Created by: steveh,Last modification on Fri 29 of Feb, 2008 [13:09 UTC] by bharat_samaria
Call Transfer is used to transfer a call in progress to some other destination.
There are two types of call transfer:
exten => 3001,1,SetVar(GOTO_ON_TRANSFER=woohoo^s^1)
exten => 3001,2,Dial(SIP/10.3.3.6|60|Tt)
If you now perform a # transfer you will end up at woohoo,s,1
There are two types of call transfer:
- Supervised call transfer - Where the call is placed on hold, a call is placed to another party, a conversation can take place privately before the caller on hold is connected to the new destination. This is also called "Attended Call Transfer" elsewhere in this website.
- Blind call transfer - Where the call is transferred to the other destination with no intervention (the other destination could ring out and not be answered for instance).
Asterisk Info
- # - Blind call transfer. Note that the user might end up being forwarded to voicemail or elsewhere.
- *2 - Attended call transfer.
- This patch looks for the variable GOTO_ON_TRANSFER in a # transferring channel and sends the transferrer to that context|exten|pri (you can use ^ to represent | to avoid escapes)
exten => 3001,1,SetVar(GOTO_ON_TRANSFER=woohoo^s^1)
exten => 3001,2,Dial(SIP/10.3.3.6|60|Tt)
If you now perform a # transfer you will end up at woohoo,s,1
- Asterisk cmd Transfer
- bug 8413 - Supervised transfer extension
- Asterisk config features.conf.

Comments
333Making supervised transfer work
include => featuremap
in your extensions.conf file
333
in my features.conf
featuremap
blindxfer => #1 ; Blind transfer
disconnect => *0 ; Disconnect
;automon => *1 ; One Touch Record
atxfer => *2 ; Attended transfer
and I set canreinvite=no
but when I press #1 or *2 to transfer call there is no response at all
though there is a xfer key on xlite and it can work, but I wonder why there is no response when I use the key #1 and *2
my email is liu.renfeng@gmail.com
can anyone tell me why?
333Re: Supervised call transfer
333supervised outside call transfer
I have a system with 10 POTS lines connected to a rhino box. I have 12 SIP clienst calling out through them. I am trying to set up call transfer on the asterisk side because the SIP clients dont support call tranfer. What steps do i take to configure this to outside lines?
Thanks
D
333Re: Supervised call transfer
"supervised call transfer" - BT100(fw 1.0.5.18) work with asterisk v1.0.1 only. With current CVS don't. I don't know why.(:confused:)
JJ
333 Supervised call transfer
I'm using Asterisk with the BudgeTone 100 SIP phone and I'm not able to implement the "supervised call transfer" feature. Could you help me on this issue?(:frown:)
Regards,
Plishu