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
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