[asterisk-dev] How to implement call transfer with Asterisk?

[Ding Peng]

Hi, everybody,

 

I want to implement the supplementary service, call transfer
unconditional/busy/NoAnswer through SIP in Asterisk. 

Does asterisk also already support it? What's the supported sip  message
flow?

How should I configure the sip.conf or extensions.conf?

I tried this way in extensions.conf, but there is no 181 message is sent to
calling party, which is expected as below picture.  

exten => 1010,1,Dial(SIP/1001); CFU 1010-->1001.



 

Thanks.

Ding Peng

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