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

[Gopalakrishnan N]

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.
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