We have a 3cx V20 update 9 with a generic sip trunk to orange. Sometimes calls are disconnected during a consult transfer. The user puts the customer on hold to talk with his colleague and before he can transfer the call, the customer is gone. We receive a bye from orange with SIP;cause=200;text="RTCP Timeout". The customer still hears music on hold until call is disconnected, so there is RTP.
How can we check if rtcp packets are send? How can we configure/change our sip trunk to solve this problem. We can't start a sip trace, because it doesn't happen during the test.
settings sip trunk:
How can we check if rtcp packets are send? How can we configure/change our sip trunk to solve this problem. We can't start a sip trace, because it doesn't happen during the test.
settings sip trunk:
