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
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?