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

[Michael Jerris]

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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