I have run into a very strange issue with periodically losing Audio on Incomming calls I am receiving incomming calls to FreeSwitch via a DID from an ITSP. Once FreeSwitch receives the call, I am adding a header to pass the UUID (which I use later to inject some custom values into CDRs) and then bridge to an IVR application running on Microsoft Speech Server 2007 which is on a separate box, using the following dial plan <extension name="Inbound_FromSoTel"> <condition field="destination_number" expression="^1(\d+)$"> <action application="set" data="sip_h_X-FS_UUID=${uuid}" /> <action application="bridge" data="sofia/external/$1@173.14.17.213:5060;transport=tcp" /> </condition> </extension> Some calls are performing properly, but in some of the calls (about 30%) the IVR application doesn't seem to hear anything the caller is saying I have eliminated any Firewalls for testing, so I go straight from the ITSP to FreeSwitch and then to Speech Server. Thee is no NAT in play. There doesn't seem to be any rhyme or reason as to when a call is going to work, or when it isn't. I have also seen exactly the same behavior in a completely different environment with a different ITSP and a completely different IVR application. So I figure that the problem is somewhere between FreeSwitch and Speech Server, but I don't see anything wrong. Using WireShark I can see the SIP signalling creating/tearing down the call and I see RTP traffic flowing both ways between FreeSwitch and Speech Server The version of FreeSwitch that I am running is 'FreeSWITCH Version 1.0.head (git-38f06a3 2011-05-19 15-39-43 -0500)' I have pasted 2 FreeSwitch traces into pastebin. The first one is of a working call, the second one is of a call where the IVR application doesn't seem to hear anything the caller is saying Can anyone advise on what the issue might be. I am completely out of ideas. Thanks Brian Campbell