[cisco-voip] Call Preserve SIP

[Bill Riley]

We did some SRST testing today and during the failover I had an active call
from my IP phone to a SIP provider on the local router. When I dropped the
connection to call manager and went to SRST mode my audio dropped but it
appeared that the call was still up on my phone. Is there a way to have the
call remain up during the transition to SRST?  I do have MTP required on the
SIP trunk and it is using the MTP at over the WAN which was down at the
time. Should I reconfigure it to use the MTP on the router? 

 

2921 CUBE with 151-2.T1

CUCM version 6.1

 

 

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