I an having a problem with the DTMF I am sending through my SIP trunk being detected by the devices on the other end of the call. When I do a capture of the call and look the DTMF events being sent by asterisk, it does not look correct when compared the the events received by asterisk from my Polycom phone. This is on a server running asterisk 1.2.13
The event packets sent by the Polycom phone have the duration value starting at 160 and incrementing another 160 for every event packet until the “end” packet for that digit is received. However the event packets for a digit originated by asterisk and sent to SIP trunk all have a duration of 0 until the the first end packet for that digit when the duration suddenly says 800.
Has anyone else experienced this? I could not find a setting in sip.conf that allows me to change the transmit duration of the DTMF digits. Is there a place to set the tone on and off durations of the DTMF digits sent via SIP?