Signalling Type: FXS Kewlstart on FXO Port

We have a Sangoma PBX and our FXO ports show correctly in DAHDI Config under FXO settings, but in the Asterisk cli they show as:

Signalling Type: FXS Kewlstart under:

freepbx*CLI> dahdi show channel 2

Which is am FXO port. Am i missing something.

We looked into this as trying to use that port we cannot call into the PBX at all. And the PBX doesn’t not seem to pick up the call. But the line works as we plugged an analog phone into it and it rings if called. But not if plugged into the PBX.


The total output of Channel 2 is:

freepbx*CLI> dahdi show channel 2
Channel: 2
File Descriptor: 30
Span: 1
Dialing: no
Context: from-analog
Caller ID:
Calling TON: 0
Caller ID subaddress:
Caller ID name:
Mailbox: none
Destroy: 0
InAlarm: 0
Signalling Type: FXS Kewlstart
Radio: 0
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: no
Busy Detection: yes
Busy Count: 10
Busy Pattern: 0,0,0,0
TDD: no
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: ulaw
Fax Handled: no
Pulse phone: no
HW Gains (RX/TX): Disabled/Disabled
SW Gains (RX/TX): 0.00/0.00
Dynamic Range Compression (RX/TX): 0.00/0.00
DND: no
Echo Cancellation:
128 taps
(unless TDM bridged) currently OFF
Wait for dialtone: 0ms
Actual Confinfo: Num/0, Mode/0x0000
Actual Confmute: No
Hookstate (FXS only): Offhook

FXO ports should have FXS signalling, that is normal.


ok thanks, do you have any idea why the analog line wont dial into the PBX when its live and tested with a phone.

Are there any “show” commands or conf’s i can look into to diagnose the issue.

I’ve tried sending the Dahdi to an individual DID that goes to one extension i know works and I dont get anything. Nor will it work in a ring group.

So i know the line works, but not sure if its the card or a setting, the rest of the ports work fine it seems.


The way you are describing your troubleshooting is really confusing me. An FXO can only be plugged into the PSTN. You can’t use a phone on it at all.

So, if we assume you have a phone jack on the wall and dial the number, it will ring a phone, right? When you plug it into the system and watch the /var/log/asterisk/full file, you should see the phone “ringing” in the logs.

After that, you need to set up your inbound routes so that when the call comes in on that port/group, the call goes somewhere.

So, step 1 is to make sure the PBX is reading the ringing. If that’s working, step 2 is to set up a DAHDI trunk to answer the call and use an “any/any” route to pass the call into the PBX.

I just tested the incoming line with analog phone to make sure the actual line was working, which it is.

I then plugged that line into the PBX. We have a couple inbetween so it converts it to the smaller plug. This has also been tested and found working.

So which the working line in the PBX all dahdi and incoming routes set up correctly. We still do not get called from that line on the pbx.

I turned on:

asterisk -rd

For debugging and get the below log when placing a call. I get a lot more than that, but not sure what else is relevant in the log.

[2019-04-22 16:24:24] DEBUG[2545]: chan_dahdi.c:11670 do_monitor: Monitor doohicky got event Polarity Reversal on channel 2
[2019-04-22 16:24:24] DEBUG[2545]: sig_analog.c:3668 analog_handle_init_event: channel (2) - signaling (5) - event (ANALOG_EVENT_POLARITY)

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