Sip Calls Time out

[TelSpt]

We have deployed a Polycom Conf Bridge using Sip integration from our CS1K. We have multiple remote BCM sites that need to be able to access the Bridge. All of our remote BCMs are on 3.0 and use H323 trunking for 4-digit office to office dialing.Currently a remote user accesses the Bridge by dialing a PSTN number for the bridge as the H323 to SIP conversion does not address the RFC2833 DTMF issues and a user cannot input the conference ID etc.Testing of SIP Gateway trunks tandeming thru the CS1K from a BCM indicate that the RFC2833 DTMF issues are no longer a problem, but the caller gets disconnected from the Polycom after app 16 minutes exactly.SIP Traces show that the Polycom received the BYE from the CS1K, disconnecting the call.SIP calls originating directly from the CS1K have no issues

 

[absinthium]

So the Polycom is registered as an endpoint in NRS or the SS?
Dynamic or static registration?

Sounds like a SIP keep-alive interval issue.

If Dynamic, change to static.

Have a look at the keep-alive settings on the Polycom if possible.