<html><head><meta http-equiv="Content-Type" content="text/html charset=us-ascii"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;"><a href="https://wiki.freeswitch.org/wiki/Variable_ringback">https://wiki.freeswitch.org/wiki/Variable_ringback</a><div><br><div style=""><div>On Mar 19, 2014, at 4:01 PM, Pete Ashdown <<a href="mailto:pashdown@xmission.com">pashdown@xmission.com</a>> wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite">I'm using Freeswitch as a PSTN gateway behind a Kamailio SIP proxy. I<br>get no ringback audio if call a phone in this direction:<br><br>(PSTN freetdm) -> Freeswitch -> Kamailio -> Phone (registered to Kamailio)<br><br>Debugging it, I see that SIP 180 ringing is sent from the phone to<br>Kamailio, and then relayed properly to Freeswitch, which does recognize<br>the ring, but because there is no RTP audio associated with it, has<br>nothing to send into freetdm.<br><br>So my question is how do I turn SIP 180 into ringback audio generated on<br>the PSTN gateway itself?<br><br><br>_________________________________________________________________________<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>