<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'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 <<a href="mailto:pashdown@xmission.com">pashdown@xmission.com</a>> 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 "instant_ringback=true" and it didn'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 <<a href="mailto:pashdown@xmission.com">pashdown@xmission.com</a>>
wrote:</div>
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<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>
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