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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:

  • 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


FreePBX Info

FreePBX users can use ## to do a blind call transfer. This is the preferred method of blind-transferring a call in FreePBX, because if a device-dependent implementation (e.g. flash-dial-hangup transfer with a Linksys/Sipura device) is used, the caller may not hear music-on-hold OR a ringing signal while waiting for the call to be picked up. Neither # nor *2 appear to work in FreePBX, at least not in all instances. Part of the reason *2 doesn't work is because the line that would normally enable it is commented out in /etc/asterisk/features.conf but if it is uncommented, it still does not appear to work. Uncommenting and changing it to #2 produces sporadic results, it works sometimes but not others. So, it may be best to use a device-dependent attended transfer if one is available (e.g. flash-dial-wait for answer-converse with transferee-hangup with a Linksys/Sipura device), but use ## when unattended (blind) transfer (or transfer to the parking lot) is desired.

FreePBX "General Options"

Make sure that you set "t" option for "Asterisk Dial Command Options" in FreePBX's "General Settings" Menu, and for "Outbound Dial Command option" always use "T". If you are using "trwkhTRWKH" for both the options (which i was using for sometime and then i accidently found out one day) then even the person dialling in your number, or the person you called can transfer your call to any extension in that context.This is also a feature, as you can have your Agent on mobile be able to transfer the call to someone in office, but still it is very dangerous to allow people to fork calls from outside.

See also



Comments

Comments Filter
222

333Making supervised transfer work

by jsollano, Thursday 07 of September, 2006 [00:04:59 UTC]
I have found that in order for supervised (or blind, or any other feature that can be configured in /etc/asterisk/features.conf) to work, one has to

include => featuremap

in your extensions.conf file
222

333

by aaaaal, Monday 21 of August, 2006 [10:21:13 UTC]

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?




222

333Re: Supervised call transfer

by CKitzmiller, Thursday 10 of August, 2006 [03:54:50 UTC]
I'm using BT-102 (fw 1.0.8.23) and whenever I do an attended transfer my phone hangs after I push the "transfer" button. My BT-200s do not have this problem. Running Asterisk 1.2-r34655 (SVN).
222

333supervised outside call transfer

by brown078, Wednesday 27 of July, 2005 [05:38:47 UTC]
Hello,

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
222

333Re: Supervised call transfer

by , Tuesday 11 of January, 2005 [14:14:48 UTC]
Hello.
"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
222

333 Supervised call transfer

by , Monday 29 of November, 2004 [12:56:31 UTC]
Hello.
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