I am very new to asterisk. I install asterisk and register tthree sip phone on it. I make a great effort to get the three sip phone work properly, they can call each other now. All of them are in the same LAN.
my question is that ,if I want to make call transfer(attended or not attended) and call forward(always,busy,no answer).what should I do.
I searched this subject on google .but it seems no result.
Can anyone tell me how to achieve this target?
Take a look at this link:
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
and make sure to take a look at the t option to transfer.
You can learn more about forwarding here:
http://www.voip-info.org/wiki-Asterisk+call+forwarding
Good Luck
Allan
[quote=“wilso027”]Take a look at this link:
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
and make sure to take a look at the t option to transfer.
You can learn more about forwarding here:
http://www.voip-info.org/wiki-Asterisk+call+forwarding
Good Luck
Allan[/quote]
Thanks very much
I add the t and T option in the exten , but when I press # and other sip phone nunber, there is no change at all. I install a standard asterisk on my redhat9 and three x-lite on other PCs, they all in the same LAN.
I add extentions bellow
exten => 1000,1,Dial(SIP/1000,10,rtT)
exten => 1001,1,Dial(SIP/1001,10,rtT)
exten => 1002,1,Dial(SIP/1002,10,rtT)
These three xlite can call each other ,but when I press #+number or number+# to do transfer(both attend and unattend), it does not work at all. Though there is a button on the xlite that can make attended transfer,
but I want to use the funtion asterisk supplied.
what’s the matter with my extention file .
BTW ,I changed parts of features.conf to following ,but it still not work when I press #1 or *2.
[featuremap]
blindxfer => #1 ; Blind transfer
disconnect => *0 ; Disconnect
;automon => *1 ; One Touch Record
atxfer => *2 ; Attended transfer
[quote=“wilso027”]Take a look at this link:
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
and make sure to take a look at the t option to transfer.
You can learn more about forwarding here:
http://www.voip-info.org/wiki-Asterisk+call+forwarding
Good Luck
Allan[/quote]
Thanks very much
I add the t and T option in the exten , but when I press # and other sip phone nunber, there is no change at all. I install a standard asterisk on my redhat9 and three x-lite on other PCs, they all in the same LAN.
I add extentions bellow
exten => 1000,1,Dial(SIP/1000,10,rtT)
exten => 1001,1,Dial(SIP/1001,10,rtT)
exten => 1002,1,Dial(SIP/1002,10,rtT)
These three xlite can call each other ,but when I press #+number or number+# to do transfer(both attend and unattend), it does not work at all. Though there is a button on the xlite that can make attended transfer,
but I want to use the funtion asterisk supplied.
what’s the matter with my extention file .
BTW ,I changed parts of features.conf to following ,but it still not work when I press #1 or *2.
[featuremap]
blindxfer => #1 ; Blind transfer
disconnect => *0 ; Disconnect
;automon => *1 ; One Touch Record
atxfer => *2 ; Attended transfer
can anyone help me why
[quote]exten => 1000,1,Dial(SIP/1000,10,rtT)
exten => 1001,1,Dial(SIP/1001,10,rtT)
exten => 1002,1,Dial(SIP/1002,10,rtT) [/quote]
You may add a “,” on your second extension 