SIP Phone Registered with CUCM Not Working with AA

[b0otable7]

I have some Polycom 335 sip phones that do not seem to pass their dtmf tones correctly when the digits are press for an AA. All of my other devices including a sip ata that is trunked to CM are able to talk with the AA just fine. I can't seem to locate any section in CM where I can adjust the DTMF settings (which is what I think is the problem, but I'm open to suggestions if there are other ideas)

TL;DR: Polycom 335 is able to call AA, but can't do anything afterwards

[jtesar]

Is this AA on Unity?  If so, is it integrated via SCCP?

SCCP, H323, and MGCP send DTMF out of band, while SIP devices typically use RFC2833 to send DTMF in the RTP stream.  Connecting between SIP devices and non-sip result in a DTMF mismatch.  CM will attempt to dynamically allocate a MTP to perform DTMF interworking.  However, if for some reason the MTP allocation fails, the default behavior of CM is to connect the call w/o the MTP.  The result is no DTMF.

I would look at your MRGL for the SIP device and make sure it contains a MTP with appropriate codecs to connect to both the SIP phone and the AA port.

Hope this helps!

-jt