HT813 - Outbound VoIP calls don't work

#1

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
#2

There is an FXO and FXS port on the HT and they are independent in operation to one another, so yes, the profiles for both need to be set up. The FXS would typically be an extension to the PBX whereas the FXO would be a trunk.

#3

OK, so I disabled FXS SIP profile and configured FXO SIP profile. After disabling Caller ID, the forwarding now works. However the caller now receives “hold the line” music while hearing my voice at the same time, I don’t hear this music. Any tips to troubleshoot this?

#4

That is not the HT. The HT has no ability to announce or offer MOH (music on hold). You need to look to the PBX or whatever the HT is connected to.

#5

OK, the MOH was caused by the Linphone client, not PBX. Everything works now. Thanks.