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

[Gopalakrishnan N]

I think I got it, in other server C am using Sangoma Transcoding card, and
when I call from that server it uses the transcoding session and thats
where the voice files are not playing and DTMF also not recognized. But it
supposed to work.

Let me check .

Thanks.


On Tue, Mar 25, 2014 at 6:51 PM, Gopalakrishnan N <
gopalakrishnan.an at gmail.com> wrote:

> 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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