Call transfer

call transfer from reception to other extensions.
Question: Details of Extensions

Reception - 2000
Sales - 2001
Accounts - 2002

any call comes it should be received by extenion 2000, n if person wants to talk to Sales, receptionist should put the caller on hold than connect to Sales i.e exten 2001, while on hold the caller should hear music on hold,now sale exten can take his call n talk to it.same with Accounts ext 2002.

####################################################################

vi /etc/asterisk/sip.conf

[general]
port = 5060
bindaddr = 0.0.0.0
context = others

[2000]
type=friend
context=reception
secret=1234
host=dynamic

[2001]
type=friend
context=sales
secret=1234
host=dynamic

[2001]
type=friend
context=accounts
secret=1234
host=dynamic
~
######################################################################

vi /etc/asterisk/extension.conf

[from-zaptel]
exten => s,1,wait(2)
exten => s,n,Dial(SIP/2000,20)

what to next…

Extensions 2001 and 2002 need to be defined and accessible from the reception context. If the SIP phones support attended transfers, you are there. Otherwise you need to add the t (or is it T) modifier on the dial command, make sure that you have configured features.conf correctly, and use the sequences defined in that file.

Note that phones often coalesce the hold and start of enquiry into a single user interface operation. (I seem to remember that the features mechanism always does that, but I’ve only used SIP transfers, in anger.)

Also note that you are over specifying, e.g. the separation of hold and enquiry is not actually needed, as all you really want is an attended transfer.

hi,
exten 2000 i.e reception can call exten 2001,2002 i.e sales n accounts.

exten 2001,2002 can call sales i.e 2000

extensions.conf

[from-zaptel]
exten => s,1,wait(2)
exten => s,n,Dial(SIP/2000,20)
exten => s,n,Hangup()

[reception]
exten => 2000,1,Dial(SIP/2000)
include = sales
include = accounts

[sales]
exten => 2001,1,Dial(SIP/2001)
include => reception

[accounts]
exten => 2002,1,Dial(SIP/2002)
include = reception

#################################################################################################

can u plz guide me more i.e in features.conf

thx

include => not include =

If you have sensible SIP phones, that should be enough.

However, should you need features.conf, it is documented in configs/features.conf.sample, in the source tree.

You don’t seem to have taken note of my mention of T/t for the Dial application. “core show application dial” will give you documentation for that. It should not be needed with the sensible SIP phone approach.

thx,
i am working on it.