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

[Gopalakrishnan N]

On top of this wanted to add one more point.

>From Server B (Asterisk) the number to reach the conference is 3054

and from Server C (Asterisk) the number to reach the conference is
5108249030

Will this make any difference?




On Tue, Mar 25, 2014 at 6:45 PM, Gopalakrishnan N <
gopalakrishnan.an at gmail.com> wrote:

> 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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