HT503 and Dukane MCS350

JL1 · 2017-07-26 16:54:52 UTC

I’m attempting to setup my HT503 to interface with a Dukane MCS350 paging/intercom system.
Firmware version 1.0.15

Prior to making any changes - the old PBX could successfully initiate a “page” to the Dukane (actually a two way/intercom call). The Dukane was connected to the trunk module block.
Via buttset tone can be heard, though it’s not dial-tone. Connecting directly to the paging wiring, a buttset can dial paging/intercom codes as soon as it goes off hook. (The Dukane provides a tone)

When connecting to the FXO port, I’m registered, but “busy” as though it isn’t detecting any tone.

Has anyone had any experience/luck configuring the FXO port to accept/use paging system voltage?

Bnorm · 2020-08-11 03:01:23 UTC

I also tried to make this work with a gxw4104. I was told by someone it’s a voltage issue. The old Dukane system uses some odd voltage that the newer gateways do not support. Did you have any luck getting it to work?

lpneblett · 2020-08-11 08:51:13 UTC

Please define to which module you made an attempt to connect.

microchipmatt · 2020-08-29 19:39:26 UTC

@lpneblett, I am trying to do the same thing, the only difference, is that I am using the updated HT513, with the FXO port. I registered the extension to the FXO port (FreePBX), when I dial the extension, it rings once, and then it goes to busy, I was hoping to get the boards DTFM TONE. I am hooked up to TIP and Ring on the MCS350 board, and then I have a RJ-11 Male end on it, which is in the HT513 FXO port. I Think the board I have is this one:
101F542 Telapex Telephone Interface Card

I can also get the numbers off the board, and also take a picture if need be, but I would LOVE to resolve this. Basically I want to Dial the extension, get the tone from the board on the FXO port, Dial *01, and have the Dukane page all the speakers. If I hook up an analog phone directly to the board, on it’s FXO port as a test, I can page the DUKANE system. Any help would be MUCH appreciated.

lpneblett · 2020-08-31 12:28:54 UTC

Use the FXS port of the HT connected to the FXO of the paging system.

microchipmatt · 2020-08-31 16:53:29 UTC

Well, Technically the paging system does NOT have a port, it just has two wires on the board, TIP and RING, do you just mean take the TIP and ring wires from the board to the RJ-11 end, and move that to the FXS port?

microchipmatt · 2020-08-31 17:09:16 UTC

Okay so I’ve been down this road before, thats why I am confident this card is an FXS, and needs an FXO port on the HT813, hense why I bought this box. When I hook up to the FXS port instead on the HT813, The FXS port is ALWAYS Flashing, and showing as OFF HOOK…

lpneblett · 2020-08-31 20:54:44 UTC

So now you are confident it is FXS?

microchipmatt · 2020-08-31 22:12:47 UTC

The 101F542 Telapex Telephone Interface Card is acting as an FXS putting out 15V on the TIP and RING. I have the TIP and RING wires going to an RJ11 end, and have that RJ11 end plugged into the FXO port on the HT813, as I believe it should be. Does that make sense?

microchipmatt · 2020-08-31 22:14:12 UTC

Sorry, to clarify, I know the 101F542 Telapex Telephone Interface Card is FXS on the TIP and RING, because I hook up the ANALOG PHONES FXO port, to the 101F542 Telapex Telephone Interface cards TIP and RING, and when I do that, I Get a dial tone and can dial *01 and make a page/annoncement. Thats how I know the 101F542 Telapex Telephone Interface Card is FXS.

However I did notice that the 101F542 Telapex Telephone Interface Card TIP and RING is only putting out 15V is that a problem for the HT813?

microchipmatt · 2020-09-01 03:46:05 UTC

Okay, I measured the voltage coming off of the TIP and RING of the 101F542 Telapex Telephone Interface Card, and it is only 15V, could this be my issue? I have been told anything less than 20V to the HT513, could be seen as busy.

lpneblett · 2020-09-01 10:56:22 UTC

A typical loop start line will have 48V. The phone may have a mechanical hook switch which is why it may work with the board as is, whereas the HT is electronic and must sense the presence of the line. You can try and disable the loop current settings and see if that makes a difference or get the document for the board and see if there are jumpers you can manipulate to alter the voltage.

microchipmatt · 2020-10-15 21:06:38 UTC

this is what i see from the debug log:
HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]SIPStack(1)::receiveMessage:(914)INVITE sip:*01@10.23.240.44:5062 SIP/2.0 Via: SIP/2.0/UDP 10.23.240.50:5060;branch=z9hG4bK1418095469;rport From: <s

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]ip:6666@10.23.240.44:5062>;tag=272931157 To: <sip:*01@10.23.240.44:5062> Call-ID: 1416009218-5060-3@BA.CD.CEA.FA CSeq: 30 INVITE Contact: <sip:666

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]6@10.23.240.50:5060> Max-Forwards: 70 User-Agent: Grandstream WP820 1.0.5.6 Privacy: none P-Preferred-Identity: <sip:6666@10.23.240.44:5062> Supp

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]orted: replaces, path, timer, eventlist Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Type: appl

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]ication/sdp Accept: application/sdp, application/dtmf-relay Content-Length: 260 v=0 o=6666 8000 8000 IN IP4 10.23.240.50 s=SIP Call c=IN IP4

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1] 10.23.240.50 t=0 0 m=audio 50040 RTP/AVP 0 8 9 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtp

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]map:101 telephone-event/8000 a=fmtp:101 0-15

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]SIPStack(1)::cb_rcvreq: Received SIP request INVITE

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]SIPDialog(3)::SIPDialog, create a SIPDialog, id = 3

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]SIPDialog, set SDP media to remote 10.23.240.50:50040, state=1 sdp=0 new sdp=0x185a08

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]SIPDialog, set SDP media, sdp set to 0x185a08

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]EventManager::run: Dispatching event 57 (SIG_REMOTE_CONNECT) on port -1:-1

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]SIPStack(1)::cb_snd_message, host=10.23.240.50, original port=5060

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]SIPStack(1)::snd_message: Present IP Addr:10.23.240.50

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]Call::incCallCount, callCountIn =3

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]SIPStack(1)::snd_message:(420)SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.23.240.50:5060;branch=z9hG4bK1418095469;rport=5060 From: <sip:6666@10.23.240.44

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]:5062>;tag=272931157 To: <sip:*01@10.23.240.44:5062> Call-ID: 1416009218-5060-3@BA.CD.CEA.FA CSeq: 30 INVITE Supported: replaces, path, timer, eve

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]ntlist User-Agent: Grandstream HT813 1.0.9.1 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Length: 0

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]SIPStack(1)::run: Active call dialogs: 1

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]Call::incCallCount, callCountTotal =3

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]SigCtrl::processSigRemoteConnect, TR-104 call_cnt_in_rcv fxs:1

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]SigCtrl::processSigRemoteConnect, FXO Port 1 , channel status = CALL_IDLE/CALL_IDLE, special feature=100

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]SigCtrl::SigRemoteConnect, FXO outgoing Call join existing call, at port 1 ringChannel 0

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]EventManager::run: Dispatching event 40 (FXO_OUTGOING_CALL_INITIATED) on port 1:0

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]ATACtrl::processFxoOutgoingCallInitiated on port 1:0, status = CALL_IDLE/CALL_IDLE, sigReferred:0

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]Nuvoton::getFxoLineStatus, Ch(1) Hook status is .

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]Nuvoton::getFxoLineStatus, Ch(1) Batt status is .

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]Nuvoton::getFxoLineStatus, Ch(1) Ring status is

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]Nuvoton::getFxoLineStatus, Ch(1) Pol status is .

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]Nuvoton::getFxoLineStatus, Ch(1) Apoh status is .

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]Nuvoton::getFxoLineStatus, Ch(1) Line Voltage is . V, Line current is 0 mA

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]Nuvoton::isFxoPortAvailable, FXO Port Busy

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]ATACtrl::processFxoOutgoingCallInitiated, FXO Port 1:0 Not Available

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]EventManager::run: Dispatching event 20 (CALL_FAILED) on port 1:0

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]SigCtrl::processCallFailed() on port 1:0, status CALL_IDLE

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]SIPStack(1)::cb_snd_message, host=10.23.240.50, original port=5060

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]SIPStack(1)::snd_message: Present IP Addr:10.23.240.50

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]SIPStack(1)::snd_message:(438)SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 10.23.240.50:5060;branch=z9hG4bK1418095469;rport=5060 From: <sip:6666@10.23.240

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1].44:5062>;tag=272931157 To: <sip:*01@10.23.240.44:5062>;tag=1467371504 Call-ID: 1416009218-5060-3@BA.CD.CEA.FA CSeq: 30 INVITE Supported: replaces

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1], path, timer, eventlist User-Agent: Grandstream HT813 1.0.9.1 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Con

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]tent-Length: 0

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]SIPStack(1)::run: Active call dialogs: 1

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]ATACtrl::processCallFailed on port 1:0, status = CALL_IDLE/CALL_IDLE stCode:486 canConf:0

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]ATACtrl::processCallFailed, FXO Port 1:0 On-hook sigReferred:0

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]ATACtrl::cleanupTimer, port:1 ch:0, isOffHook:0

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]ATACtrl::resetFxoRingCnt, !!!reset ring counter to 0

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]ATACtrl::resetCidInfo, @@@@@reset Cid Info!!! port 1

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]EventManager::run: Dispatching event 80 (PVALUE_CHANGED) on port -1:-1

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]SIPStack(1)::receiveMessage:(280)ACK sip:*01@10.23.240.44:5062 SIP/2.0 Via: SIP/2.0/UDP 10.23.240.50:5060;branch=z9hG4bK1418095469;rport From: <sip:

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]6666@10.23.240.44:5062>;tag=272931157 To: <sip:*01@10.23.240.44:5062>;tag=1467371504 Call-ID: 1416009218-5060-3@BA.CD.CEA.FA CSeq: 30 ACK Content-

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]Length: 0

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]SIPStack(1)::cb_rcvreq: Received SIP request ACK

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]SIPStack(1)::run: Active call dialogs: 1

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]SIPStack(1)::cb_ist_kill_transaction: Kill IST transaction 4

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]SIPStack(1)::run: Active call dialogs: 1

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]TR069 INFO [1.0.1.141] Memory Info::free{10780}/total{22616} rate{47%} stat:Normal

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]SIPStack(1)::receiveMessage:(915)INVITE sip:*01@10.23.240.44:5062 SIP/2.0 Via: SIP/2.0/UDP 10.23.240.50:5060;branch=z9hG4bK2067769156;rport From: <s

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]ip:6666@10.23.240.44:5062>;tag=1096490936 To: <sip:*01@10.23.240.44:5062> Call-ID: 1872728286-5060-4@BA.CD.CEA.FA CSeq: 40 INVITE Contact: <sip:66

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]66@10.23.240.50:5060> Max-Forwards: 70 User-Agent: Grandstream WP820 1.0.5.6 Privacy: none P-Preferred-Identity: <sip:6666@10.23.240.44:5062> Sup

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]ported: replaces, path, timer, eventlist Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Type: app

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]lication/sdp Accept: application/sdp, application/dtmf-relay Content-Length: 260 v=0 o=6666 8000 8000 IN IP4 10.23.240.50 s=SIP Call c=IN IP

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]4 10.23.240.50 t=0 0 m=audio 50040 RTP/AVP 0 8 9 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rt

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]pmap:101 telephone-event/8000 a=fmtp:101 0-15

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]SIPStack(1)::cb_rcvreq: Received SIP request INVITE

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]SIPDialog(4)::SIPDialog, create a SIPDialog, id = 4

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]SIPDialog, set SDP media to remote 10.23.240.50:50040, state=1 sdp=0 new sdp=0x1885a8

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]SIPDialog, set SDP media, sdp set to 0x1885a8

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]EventManager::run: Dispatching event 57 (SIG_REMOTE_CONNECT) on port -1:-1

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]Call::incCallCount, callCountIn =4

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]SIPStack(1)::cb_snd_message, host=10.23.240.50, original port=5060

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]SIPStack(1)::snd_message: Present IP Addr:10.23.240.50

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]SIPStack(1)::snd_message:(421)SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.23.240.50:5060;branch=z9hG4bK2067769156;rport=5060 From: <sip:6666@10.23.240.44

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]:5062>;tag=1096490936 To: <sip:*01@10.23.240.44:5062> Call-ID: 1872728286-5060-4@BA.CD.CEA.FA CSeq: 40 INVITE Supported: replaces, path, timer, ev

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]entlist User-Agent: Grandstream HT813 1.0.9.1 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Length: 0

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]Call::incCallCount, callCountTotal =4

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]SigCtrl::processSigRemoteConnect, TR-104 call_cnt_in_rcv fxs:1

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]SIPStack(1)::run: Deleting call dialog (3)

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]SIPDialog::~SIPDialog: Cleaning in-dialog in-transaction

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]SIPDialog(3)::~SIPDialog: Cleaning in-dialog out-transaction

HT813 [00:0b:82:e9:e6:4d] [1.0.9.1]SIPDialog(3)::~SIPDialog, Deleting SDP in the dialog