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

[Paul Belanger]

On 13-02-15 10:21 AM, Ding Peng wrote:
> 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.
>
Please do not cross post the mailing lists. This post only needs to 
exist on asterisk-users.

-- 
Paul Belanger | PolyBeacon, Inc.
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