> Hmm... but what if the DTMF is in band. Is supporting out of > band DTMF a prerequisite for SIP re-invite?
On a SIP proxy, this would not be the case because the SIP proxy would never need to know what digits were dialed. However, if the DTMF digits were important to the SIP proxy, the call could be set up to send them out-of-band as a SIP INFO message. Now, for certain Asterisk features, the DTMF digits need to be delivered to Asterisk and not the remote endpoint. Ideally, SIP INFO headers should have allowed the audio path to be directly end-to-end without Asterisk in the middle and still allow for the features, but Asterisk does not support that. Therefore, if you have a t or T in your Dial() command, Asterisk will continue to stay in the audio path. Nabeel
