<div dir="ltr">Yes in both I have 20000 to 40000 start to end. </div><div class="gmail_extra"><br><br><div class="gmail_quote">On Mon, Mar 31, 2014 at 8:33 PM, Deon Vermeulen <span dir="ltr"><<a href="mailto:vermeulen.deon@gmail.com" target="_blank">vermeulen.deon@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div style="word-wrap:break-word">Have you checked that RTP min/max port range between FS and * is the same?<div><br></div>
<div>The defaults on both vary considerably and will need to be alligned.</div><div><br></div><div><br></div><div>Kind Regards</div><div><br></div><div><br><div><div><div class="h5"><div>On Mar 31, 2014, at 4:59 PM, Gopalakrishnan N <<a href="mailto:gopalakrishnan.an@gmail.com" target="_blank">gopalakrishnan.an@gmail.com</a>> wrote:</div>
<br></div></div><blockquote type="cite"><div><div class="h5"><div dir="ltr">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. <div>
<br></div><div>:(</div><div><br></div></div><div class="gmail_extra">
<br><br><div class="gmail_quote">On Wed, Mar 26, 2014 at 11:13 PM, Gopalakrishnan N <span dir="ltr"><<a href="mailto:gopalakrishnan.an@gmail.com" target="_blank">gopalakrishnan.an@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">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. <div>
<br></div><div>Let me check .</div><div><br></div><div>Thanks. </div></div><div><div><div class="gmail_extra"><br><br><div class="gmail_quote">On Tue, Mar 25, 2014 at 6:51 PM, Gopalakrishnan N <span dir="ltr"><<a href="mailto:gopalakrishnan.an@gmail.com" target="_blank">gopalakrishnan.an@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">On top of this wanted to add one more point. <div><br></div><div>From Server B (Asterisk) the number to reach the conference is 3054</div>
<div><br></div><div>and from Server C (Asterisk) the number to reach the conference is 5108249030</div>
<div><br></div><div>Will this make any difference?</div><div><br></div><div><br></div></div><div><div class="gmail_extra"><br><br><div class="gmail_quote">On Tue, Mar 25, 2014 at 6:45 PM, Gopalakrishnan N <span dir="ltr"><<a href="mailto:gopalakrishnan.an@gmail.com" target="_blank">gopalakrishnan.an@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Hi,<div><br></div><div>I have a setup as per the following,</div><div>Server A - FreeSWITCH (Location A)</div>
<div>Server B - Asterisk (Location A)</div><div>Server C - Asterisk (Location B)</div><div><br></div>
<div>Two Asterisk servers are trunked with FreeSWITCH.</div><div><br></div><div>In FreeSWITCH am establishing Conference via a Javascript. </div><div><br></div><div>From Server B (Asterisk) if I initiate the call, it works absolutely fine by entering into the conference room. </div>
<div><br></div><div>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. </div><div><br></div><div>
The network connection between Location A and Location B is MPLS. </div>
<div><br></div><div>My dialplan is pasted here <a href="http://pastebin.freeswitch.org/22228" target="_blank">http://pastebin.freeswitch.org/22228</a></div><div><br></div><div>Comments would be much appreciated. </div><div>
<br></div><div>
Thanks. </div><div><br></div></div>
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