[Freeswitch-users] Handling SIP 180 ringback to freetdm?

[Michael Jerris]

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?

On Mar 20, 2014, at 11:43 PM, Pete Ashdown <pashdown at xmission.com> wrote:

 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.

On 3/19/14, 2:10 PM, Michael Jerris wrote:

 https://wiki.freeswitch.org/wiki/Variable_ringback

 On Mar 19, 2014, at 4:01 PM, Pete Ashdown <pashdown at xmission.com> wrote:

I'm using Freeswitch as a PSTN gateway behind a Kamailio SIP proxy.  I
get no ringback audio if call a phone in this direction:

(PSTN freetdm) -> Freeswitch -> Kamailio -> Phone (registered to Kamailio)

Debugging it, I see that SIP 180 ringing is sent from the phone to
Kamailio, and then relayed properly to Freeswitch, which does recognize
the ring, but because there is no RTP audio associated with it, has
nothing to send into freetdm.

So my question is how do I turn SIP 180 into ringback audio generated on
the PSTN gateway itself?


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