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

[Gopalakrishnan N]

I forced Sangoma configuration in Asterisk not to use ulaw from Sangoma,
but still it uses to establish the audio and there is no audio @FreeSWITCH
end.

:(



On Wed, Mar 26, 2014 at 11:13 PM, Gopalakrishnan N <
gopalakrishnan.an at gmail.com> wrote:

> 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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