[Freeswitch-users] Audio(RTP) Stops after first message played

Hi, I have a setup as per the following, Server A - FreeSWITCH (Location A) Server B - Asterisk (Location A) Server C - Asterisk (Location B) Two Asterisk servers are trunked with FreeSWITCH. In FreeSWITCH am establishing Conference via a Javascript.

From Server B (Asterisk) if I initiate the call, it works absolutely fine by entering into the conference room.

From Server C (Asterisk) if I initiate the call, am able to hear the first word (Please) from the message "Please enter your conference number" and then its blank.

The network connection between Location A and Location B is MPLS. My dialplan is pasted here http://pastebin.freeswitch.org/22228 Comments would be much appreciated. Thanks.