I’m trying to set up call forwarding from PSTN FXO line to VoIP. FXO to FXS calls work fine. Incoming VoIP calls work fine.
However forwarding from PSTN to VoIP doesn’t work. I use Linphone SIP server. I tried TLS and UDP transport modes, STUN server is enabled. SIP profile is configured in FXS (shouldn’t matter if in FXS or FXO, right?) Including syslog:
2023-02-23T13:26:06.242071+01:00 HT813 [c0: 74:ad:8e:76:de] [1.0.15.7] GS_ATA: USER.DEBUG ATACtrl::processFxoIncomingRing, Unconditional Forward to VoIP
2023-02-23T13:26:06.242462+01:00 HT813 [c0: 74:ad:8e:76:de] [1.0.15.7] GS_ATA: USER.DEBUG ATACtrl::processFxoIncomingRing, URI <sip:<REDACTED>@sip.linphone.org:5060>
2023-02-23T13:26:06.248442+01:00 HT813 [c0: 74:ad:8e:76:de] [1.0.15.7] GS_ATA: USER.DEBUG EventManager::registerEventListener: listener Call
2023-02-23T13:26:06.248779+01:00 HT813 [c0: 74:ad:8e:76:de] [1.0.15.7] GS_ATA: USER.DEBUG Call(2)::init, DialPlan callFeatureFlag = 0x0
2023-02-23T13:26:06.252280+01:00 HT813 [c0: 74:ad:8e:76:de] [1.0.15.7] GS_ATA: USER.INFO Call(2)::Call, Creating Call object 2 at port 1:0 with digits <sip:<REDACTED>@sip.linphone.org:5060>
2023-02-23T13:26:06.252587+01:00 HT813 [c0: 74:ad:8e:76:de] [1.0.15.7] GS_ATA: USER.DEBUG EventManager::run: Dispatching event 80 (PVALUE_CHANGED) on port -1:-1
2023-02-23T13:26:06.266280+01:00 HT813 [c0: 74:ad:8e:76:de] [1.0.15.7] GS_ATA: USER.INFO Call(2)::run, Dialing <sip:<REDACTED>@sip.linphone.org:5060>, disableCW =1
2023-02-23T13:26:06.267059+01:00 HT813 [c0: 74:ad:8e:76:de] [1.0.15.7] GS_ATA: USER.DEBUG ATACtrl::call, TR-104 call_cnt_out_att fxs:1
2023-02-23T13:26:06.282486+01:00 HT813 [c0: 74:ad:8e:76:de] [1.0.15.7] GS_ATA: USER.DEBUG ATACtrl::call, uri <sip:<REDACTED>@sip.linphone.org:5060>, isURI=1
2023-02-23T13:26:06.284221+01:00 HT813 [c0: 74:ad:8e:76:de] [1.0.15.7] GS_ATA: USER.DEBUG ATACtrl::call, caller number = <REDACTED>
2023-02-23T13:26:06.285194+01:00 HT813 [c0: 74:ad:8e:76:de] [1.0.15.7] GS_ATA: USER.DEBUG ATACtrl::call on port 1:0, status = 2(CALL_DIALED), disableLEC=0 canConf:1
2023-02-23T13:26:06.286518+01:00 HT813 [c0: 74:ad:8e:76:de] [1.0.15.7] GS_ATA: USER.DEBUG RTP::RTP, construct 0x1b4558
2023-02-23T13:26:06.288228+01:00 HT813 [c0: 74:ad:8e:76:de] [1.0.15.7] GS_ATA: USER.DEBUG RTP::openSocket, Bound to RTP port 5014, socket 11
2023-02-23T13:26:06.289765+01:00 HT813 [c0: 74:ad:8e:76:de] [1.0.15.7] GS_ATA: USER.DEBUG RTP::openSocket, Bound to RTCP port 5015, socket 27
2023-02-23T13:26:06.296624+01:00 HT813 [c0: 74:ad:8e:76:de] [1.0.15.7] GS_ATA: USER.ERR RTP::openSocket: Layer 3 DSCP for RTP set to 46
2023-02-23T13:26:06.297577+01:00 HT813 [c0: 74:ad:8e:76:de] [1.0.15.7] GS_ATA: USER.DEBUG ATACtrl::call, ATA haven't make the call due to Signaling issue or call cancelled
2023-02-23T13:26:06.299409+01:00 HT813 [c0: 74:ad:8e:76:de] [1.0.15.7] GS_ATA: USER.DEBUG ATACtrl::call, TR-104 call_cnt_out_fail fxs:1
2023-02-23T13:26:06.314983+01:00 HT813 [c0: 74:ad:8e:76:de] [1.0.15.7] GS_ATA: USER.INFO ATACtrl::call, cannot make the call, statusCode = 0, chan status = CALL_DIALED