Need to test DTMF sending through SIP trunks, I have an MX-One connceted to an SUT server, we verified DTMF are working from MX-One extension to SUT extension through establishing a confrence bridge on the SUT server and dialing 1 to join a confrence, any idea how I can test DTMF from the SUT extensions through the SIP trunks, what call services can be called through dialing DTMF digits while call is establsihed through a SIP trunk???? I don't have an VoiceMail on site.
TELEC3
TELEC3