It's acting like the remote end is getting multiple Keypresses for each digit. I used the command 'debug voice dsp voip 0/1 channel verbose rfc2833'īelow is the example when Entering my zip code into an IVR out on the PSTN (60505) It looks like the Adtran is seeing the DTMF Event over the PRI correctly, but I'm not sure exactly what it's telling me. I'm running the DTMF Debug options on both the Asterisk box and the Adtran, but I'm having some issues deciphering what the Adtran Output is telling me. I'm testing this to be able to provide Voip Termination via PRI to legacy PBX Products. This is a test setup, that is why the Sip Carrier does not connect directly to the Asterisk box. Calls Inbound, from the carrier to the Asterisk box can navigate my IVR Properly. I'm having DTMF Issues when calling from the asterisk box out to the carrier. My Setup is Handset ->SIP -> Asterisk -> PRI -> Adtran - > SIP -> Carrier. I'm using a TA904 as a SIP to PRI Gateway. Here is a sample of our gather code: Īny advice or direction to point us in is helpful.Hello, I'm relatively new to the Adtran products. In our TwiML voice scripting we are using GATHER. Because we can dial other IVR systems and enter codes fine with Skype, it doesn't appear to be a problem with Skype. We believe this to be a problem with either Twilio accepting DTMF tones from voip phones OR something that is running inconsistently in our Twilio scripting. The other 80% of the time the calls are not recognizing the tones. To make matters more confusing for us, about 20% of the calls that we make via Skype it DOES accept the tones. Skype version 8.11.0.4 on MAC OSX Sierra.Versions of Skype tested (at various different physical locations):
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