Re: [Freeswitch-users] No Ringing tone when call is forwarded to PSTN
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): PSTN(A)INVITE===FS PSTN(A)===TRYING===FS FS===INVITE==PSTN(B) FS==TRYING===PSTN(B) FS==RINGING==PSTN(B) PSTN(A)==PROGRESS===FS FS===OK==PSTN(B) FSACKPSTN(B) PSTN(A)===OKFS 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/06322xxx...@10.11.12.15) State XCHANGE_MEDIA 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/0171...@10.11.12.15:5060 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 Peter 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 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] No Ringing tone when call is forwarded to PSTN
Should I open a JIRA for this? Best regards Peter Peter P GMX schrieb: Hello, we have the following scenario: A PSTN call (A) is coming in to Freeswitch through a Patton gateway. For the called FS user, call forwarding has been enabled to another PSTN extension (B) . Result: The calling party does not hear any ringing tone. Here an Abstract of the SIP protocol we traced (PSTN(A) and PSTN(B) is in fact the same Patton Gateway): PSTN(A)INVITE===FS PSTN(A)===TRYING===FS FS===INVITE==PSTN(B) FS==TRYING===PSTN(B) FS==RINGING==PSTN(B) PSTN(A)==PROGRESS===FS FS===OK==PSTN(B) FSACKPSTN(B) PSTN(A)===OKFS PSTN(A)ACK==FS I would expect that FS answers RINGING back to PSTN(A). Instead it only answers SESSION PROGRESS. When PSTN(B) answers, they can hear each other, but there was no ringing tone to PSTN(A) before. Are there any hints to overcome this, besides playing early media to PSTN(A)? Best regards Peter ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] No Ringing tone when call is forwarded to PSTN
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 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] No Ringing tone when call is forwarded to PSTN
Hello, we have the following scenario: A PSTN call (A) is coming in to Freeswitch through a Patton gateway. For the called FS user, call forwarding has been enabled to another PSTN extension (B) . Result: The calling party does not hear any ringing tone. Here an Abstract of the SIP protocol we traced (PSTN(A) and PSTN(B) is in fact the same Patton Gateway): PSTN(A)INVITE===FS PSTN(A)===TRYING===FS FS===INVITE==PSTN(B) FS==TRYING===PSTN(B) FS==RINGING==PSTN(B) PSTN(A)==PROGRESS===FS FS===OK==PSTN(B) FSACKPSTN(B) PSTN(A)===OKFS PSTN(A)ACK==FS I would expect that FS answers RINGING back to PSTN(A). Instead it only answers SESSION PROGRESS. When PSTN(B) answers, they can hear each other, but there was no ringing tone to PSTN(A) before. Are there any hints to overcome this, besides playing early media to PSTN(A)? Best regards Peter ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
