<html><head><meta http-equiv="content-type" content="text/html; charset=utf-8"></head><body dir="auto"><div>look at the trace live and see if it&#39;s a delay in us sending ring back indication or a delay in us receiving it?</div>
<div><br>On Mar 20, 2014, at 11:43 PM, Pete Ashdown &lt;<a href="mailto:pashdown@xmission.com">pashdown@xmission.com</a>&gt; wrote:<br><br></div><blockquote type="cite"><div>
  
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    <div class="moz-cite-prefix">Thank you Michael, this works. 
      However, the phone being called starts ringing a good 3 seconds
      before the ring audio is generated on the line.  Any idea how to
      start the ring audio at the same time, or at least minimize the
      delay?  I tried setting &quot;instant_ringback=true&quot; and it didn&#39;t make
      any difference.<br>
      <br>
      On 3/19/14, 2:10 PM, Michael Jerris wrote:<br>
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      <a href="https://wiki.freeswitch.org/wiki/Variable_ringback">https://wiki.freeswitch.org/wiki/Variable_ringback</a>
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          <div>On Mar 19, 2014, at 4:01 PM, Pete Ashdown &lt;<a href="mailto:pashdown@xmission.com">pashdown@xmission.com</a>&gt;
            wrote:</div>
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          <blockquote type="cite">I&#39;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) -&gt; Freeswitch -&gt; Kamailio -&gt; 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>
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    <br>
  

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