How to invoke Dial to PJSIP endpoint without R option which causes undesired behavior - double ringing tone. My voip operator is already providing a ringing tone (I assume as early media) and asterisk is generating it’s own on top of that.
The R option is documented as giving priority to early media. Neither r nor R are set by default. I suspect your ringback is actually originating in the calling device, because the callee is indicating 180 Ringing, not 183 Progress. You may need to call the Progress application explicitly.
And if this is not the case, then you need to actually show a Asterisk console output including SIP trace.
asterisk > VoIP_operator: SIP: INVITE sip:xxxxxxxxxxx@neofon.tp.pl:5060 SIP/2.0
VoIP_operator > asterisk: SIP: SIP/2.0 100 Trying
VoIP_operator > asterisk: SIP: SIP/2.0 183 Session Progress
asterisk > VoIP_operator: SIP: PRACK sip:VoIP_operator:5060 SIP/2.0
VoIP_operator > asterisk: SIP: SIP/2.0 200 OK
VoIP_operator > asterisk: SIP: SIP/2.0 183 Session Progress
VoIP_operator > asterisk: SIP: SIP/2.0 180 Ringing
It looks like the phone starts generating it’s own ringing tone after last SIP message. And it’s doing this together with early media audio. Can I do something about it?
That’s between Asterisk and the VoIP provider.
What about between Asterisk and the phone? By default we won’t send a 180 after 183, you have to enable it in PJSIP explicitly.
On the other side is a Cisco phone using chan_sccp which may have bugs.
-- SEP64D989696A2F: (pbx_indicate) start indicate 'AST_CONTROL_RINGING: Remote end is ringing' (3) condition on channel SCCP/132-00000216 (readStat:2, writeState:2, rtp:yes, default res:inband signaling (-1))
-- SEP64D989696A2F: Finish to indicate channel state:RINGOUT on call:SCCP/132-00000216, lineInstance:1. New channel state:RINGOUT
-- SEP64D989696A2F: Set asterisk state Ring (4) for call 534
There’s an option in sccp.conf that might have something to do with it:
earlyrtp = progress ; valid options: none, offhook, immediate, dial, ringout and progress.
; The audio stream will be open in the progress and connected state by default. Immediate forces overlap dialing.
chan_sccp is third party code.