Hi guys, I have been running two freeswitch boxes (13754M)  that answer calls from a cisco 5300 (both on the same network) and records them to disk with a small lua application.  This has been working well for the past few months.  I decided to upgrade one of them to trunk (  git-3fbd9e2 2010-06-11 11-08-51 -0500 ) and have run into a problem.  Some calls will fail with a MEDIA_TIMEOUT  after a few minutes, the time it takes to fail ranges from 4 minutes to 10 minutes,  I don't have a full sip trace or pcap dump yet, I reverted back to the old freeswitch version (on the same hardware) and have not been able to reproduce it in a test environment yet ( I continue to try).   Below are the relevant lines from the log files for one of the calls:

2010-06-23 07:42:19.033466 [DEBUG] switch_channel.c:2257 (sofia/external/nobody@192.168.21.4) Callstate Change ACTIVE -> HANGUP
2010-06-23 07:42:19.033466 [NOTICE] mod_sofia.c:884 Hangup sofia/external/nobody@192.168.21.4 [CS_EXECUTE] [MEDIA_TIMEOUT]
2010-06-23 07:42:19.033466 [DEBUG] switch_channel.c:2273 Send signal sofia/external/nobody@192.168.21.4 [KILL]
2010-06-23 07:42:19.033466 [DEBUG] switch_core_session.c:1023 Send signal sofia/external/nobody@192.168.21.4 [BREAK]
2010-06-23 07:42:19.033466 [DEBUG] switch_core_codec.c:146 sofia/external/nobody@192.168.21.4 Restore previous codec PCMU:0.

My configuration is bone stock, so the rtp timeout value is at 300,  but I have some calls that have lasted only 4 minutes.  One other piece of information is that on one of the recordings that was hung up after 4 minutes and 17 seconds the recorded file was only 24 seconds long (it stopped recording after the first 24 seconds) , so I'm assuming freeswitch did not think there were any rtp packets to record.

Any ideas on where to start debugging this?  I have setup a new freeswitch box connected to the same 5300 to reproduce, but have not been able to generate the call volume ( there where around 30 calls being recorded) yet, but I'm working on it.

Thanks!