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

[Pete Ashdown]

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
> <mailto: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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