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

[Gopalakrishnan N]

Yes in both I have 20000 to 40000 start to end.


On Mon, Mar 31, 2014 at 8:33 PM, Deon Vermeulen <vermeulen.deon at gmail.com>wrote:

> Have you checked that RTP min/max port range between FS and * is the same?
>
> The defaults on both vary considerably and will need to be alligned.
>
>
> Kind Regards
>
>
> On Mar 31, 2014, at 4:59 PM, Gopalakrishnan N <gopalakrishnan.an at gmail.com>
> wrote:
>
> 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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