Sip Calls from BCM to Polycom 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

 

[oldestgeek]

That sounds like a tandem setting in the CS1k so you may want also post in that forum. May be a setting on the NRS.

 

[eng09]

is it exactly 16 minutes for all calls?

 

[TelSpt]

Yes, it is always the same, 16 Minutes.
I have also found that If I skip the Tandem through the CS1K, I dont have the timing issues.
Thought I had to Tandem, but apparently not.
Also thought that I had to have SIP Gateway Licensing on the BCMs, but I dont.

 

[oldestgeek]

With the setup you're using the BCM is speaking only H.323 so it doesn't need the SIP licenses. Look at your NRS logs to see if you see disconnect messages for the dropped calls. That may give you a reason for the disconnects.