[Freeswitch-users] No Ringing tone when call is forwarded to PSTN

Peter P GMX Prometheus001 at gmx.net
Mon Dec 21 04:02:51 PST 2009

I just crosschecked the dialplan which is used. We do not anwer the
call, we bridge it directly to a PSTN destination.
However the Ringing event is not passed to PSTN(A):

>                      FS===INVITE==>PSTN(B)
>                      FS<==TRYING===PSTN(B)
>                      FS<==RINGING==PSTN(B)
>                      FS<===OK======PSTN(B)
>                      FS====ACK====>PSTN(B)
> PSTN(A)<===OK========FS
> PSTN(A)====ACK======>FS

But then I stumbled over the following SOFIA LOOPBACK entry in the logs:
2009-12-21 12:47:00.404145 [DEBUG] switch_core_state_machine.c:351
(sofia/external/06322xxxxxxxxxx at State
2009-12-21 12:47:00.404145 [DEBUG] mod_sofia.c:469 SOFIA LOOPBACK
2009-12-21 12:47:00.404145 [DEBUG] sofia.c:3669 Channel
sofia/external/0171xxxxxxx at skipping state [early][183]

So I modified the dialplan to temporarily use another Patton GW for
outgoing calls, et voilà, I receive a ringing tone at PSTN(A). So I
think this is because Freeswitch thinks this is a loopback, because
incoming and outgoing gateway is the same.

But I due to other restrictions we need the call to pass through the
same Patton Gateway to PSTN(B) as we received it from PSTN(A).
Is there a chance to tell Freeswitch to not consider this call as a
loopback scenario?

Best regards

Brian West schrieb:
> That depends if the call is answered and then you transfer it, you will HAVE to set the transfer_ringback variable you can't send a 180 to the thing or a progress and make it generate the ringback.  You MUST do it yourself.
> You also fail to mention if the progress is a 180 or a 183 with sdp and media... or even better a 180 with sdp and media (silly sip people what were you thinking) either way... set the transfer_ringback variable.
> /b
> On Dec 18, 2009, at 4:00 AM, Peter P GMX wrote:
>> Should I open a JIRA for this?
>> Best regards
>> Peter
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