<html><head><meta http-equiv="Content-Type" content="text/html charset=iso-8859-1"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;">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>On Mar 31, 2014, at 4:59 PM, Gopalakrishnan N <<a href="mailto:gopalakrishnan.an@gmail.com">gopalakrishnan.an@gmail.com</a>> wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><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 class="HOEnZb"><div class="h5"><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>
</blockquote></div><br></div>
</div></blockquote></div><br></div>
</div></div></blockquote></div><br></div>
_________________________________________________________________________<br>Professional FreeSWITCH Consulting Services:<br><a href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a><br>http://www.freeswitchsolutions.com<br><br>FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>http://www.cudatel.com<br><br>Official FreeSWITCH Sites<br>http://www.freeswitch.org<br>http://wiki.freeswitch.org<br>http://www.cluecon.com<br><br>FreeSWITCH-users mailing list<br>FreeSWITCH-users@lists.freeswitch.org<br>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>http://www.freeswitch.org<br></blockquote></div><br></div></body></html>