[asterisk-users] transfer function

[Mojo with Horan & Company, LLC]

Possibly the called party is not sending their DTMF properly?  maybe 
experiment with inband/rfc2833/etc in the CALLED party's peer definition

Denis V. Gudtsov wrote:
> Hello!
> 
> I'm using asterisk 1.2.13, in extension.ael is set Dial(SIP/${EXTEN},12,tT)
> 
> but only calling party can do forward. How to configure '*' to take this
> possibility to called party?
> 
> ps
> both calling/called use sip
>