Handling SIP 180 ringback to freetdm?
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?
https://wiki.freeswitch.org/wiki/Variable_ringback On Mar 19, 2014, at 4:01 PM, Pete Ashdown <pashdown@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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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@xmission.com <mailto:pashdown@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?
look at the trace live and see if it's a delay in us sending ring back indication or a delay in us receiving it? On Mar 20, 2014, at 11:43 PM, Pete Ashdown <pashdown@xmission.com> wrote: 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@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? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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
participants (2)
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Michael Jerris
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Pete Ashdown