Testing a SIP to SIP IP Trunking implementation with CUBE and came across an interesting problem. When a call is placed on hold and then resumed one side of the audio path breaks.
on an inbound call - no audio back to PSTN
on an outbound call - no audio back from PSTN
I was able to fix the inbound issue by enabling RFC 2543 Hold on the associated SIP Profile. Outbound calls still experience the same problem. I am running CUCM 6.1.2 and 12.4.20T on the CUBE router.