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@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@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@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.