Problem with HT813 FXO port


Hello everybody,
I’m running an Asterisk server with a HT503 ATA. I use it to filter out unwanted calls from the landline (based on a white and black-lists and some dialplan logic) and to connect fixed handsets as well as softphones (conventional smart-phones with a SIP client app). The setup has been running fine for several months. Unfortunately the HT503 broke down and I replaced it with a more recent HT813. The 2 devices seem to be quite the same thing but I was unable to get a working FXO port with the HT813. In the mean time I also got a new HT503 from the after-sales service, which works like a charm.
To investigate why the HT813 FXO port and what exactly does not work, and for lack of specialized tools, I mounted both the HT503 and the HT813 “top to tail”, that is:

  • the HT503 “phone” port (FXS) connected to the HT813 FXO
  • the HT813 FXS port connected to the HT503 “line” port (FXO)
    I also modified the Asterisk sip.conf file to have all ports connected as SIP clients:
  • HT503 FXS on port 5060 (default)
  • HT503 FXO on port 5062 (default)
  • HT813 FXS on port 5064
  • HT813 FXO on port 5066
    I also modified the dialplan (extensions.conf file) to process incoming calls (unconditionnal call forward to VoIP parameter on Basic Settings page):
  • extension 2000 for HT503 FXO
  • extension 4000 for HT813 FXO
    These extensions dial the corresponding FXS port (on the same box) as well as the available softphones.
    There are also extensions to call a given softphone, the FXs ports, and to call an external number through one of the FXO port.
    With that set-up, I was able to mimic:
  • an incoming call to a FXO port, initiated from the other ATA’s FXS port.
  • a outgoing call to a FXO port, dispatched according to the dialed number sent as a DTMF sequence (in this case, a softphone) to the other ATA’s FXS port.
    Every call is monitored with Asterisk console.
    The results are as follows:
  • A call initiated from the HT813 FXS, forwarded to the HT503 FXO port (incoming call) rings and the call can be answered correctly.
  • A call initiated from the HT503 FXO (outgoing call), forwarded to the HT813 FXS, selects the right extension, makes it ring and the call can be answered correctly.
  • A call initiated from the HT503 FXS, forwarded to the HT813 FXO port (incoming) rings but none of the connected softphones does. However, if the SLIC setting of the HT503 FXS is set to USA instead of France (where I live and which is the setting I normally use), the phones ring and the call can be answered correctly.
  • A call initiated from the HT813 FXO (outgoing call) is immediately cancelled with a 403 (forbidden) response, whatever the SLIC setting (USA or France) of the HT503 FXS port.
    I also tried to retrieve the logs from the HT813 to an external syslog server, but they are quite cryptic for someone who does not know the inner workings of the beast.
    This definitely shows that the HT813 has a malfunctionning FXO port, which very probably stems from a firmware bug.
    I clarify below the various settings I used (they are the same for both ATAs), only for settings different form their default value:
  • Basic Settings:
    • Unconditional Call Forward to VOIP: see above, same SIP server, but different extension and port for both ATAs
    • Time Zone
  • Advanced Settings:
    • System Ring Cadence and Ring Tones set for France (1500/3500)
  • FXS Port:
    • SIP Account Active (see above)
    • Polarity Reversal: Yes (my PSTN uses it, does not change anything to problems mentionned above)
  • FXO Port:
    • SIP account active (see above)
    • Enable PSTN Disconnect Tone Detection: Yes (with adequate tone)
    • Enable Polarity Reversal: Yes
    • AC Termination Model: Country based, France
    • Number of Rings: 3
    • PSTN Ring Thru FXS: No
    • Stage Method (1/2): 1
      Firmware versions used: the latest available ( for HT503, for HT813).
      Thank you for your help.


Sorry, but this seems overly complicated to determine the issue with why you have not been able to replace a 503 with an 813.

A 403 error is because there is something in the INVITE that the associated SIP server does not like and is therefore refusing to service the request.

You indicated - “HT813 FXO port (incoming) rings but none of the connected softphones does”, This would seem to indicate that if the call is seen at the Asterisk console monitoring the call, that the call reached the PBX and that the issue with why the softphones did not ring is at the PBX level. I guess I am not certain if the observation of getting a ring is being seen from within the HT status page or if the result of the INVITE reaching the console and seeing the ring there visually, but the softphone not ringing audibly.
This really needs a wireshark capture to determine why the call did not progress to your desire.

As I decipher the post, the issues appear to be:

  1. The HT813 FXO not passing the call to the softphones
  2. The HT813 FXO getting a 403 when a call is placed directly from the SIP server to same.

I am not saying that there are not some issues, but rather that it is difficult to follow when the scenario you are using is not really the one you intend to use.

I have a 813 that I just got in and I will play with it today to see if I can get the function to work on both the FXS and FXO ports. Will let you know.


Thank you for your reply.
I must clarify my wording about “A call initiated from the HT503 FXS, forwarded to the HT813 FXO port (incoming) rings but none of the connected softphones does”. What is actually ringing is the calling phone (ring back tone). Nothing gets out of the FXO and reaches Asterisk, unless, as I said, the SLIC setting of the (HT503) FXS is set to USA instead of France.
I know my setup is a bit complicated, but when I initially mounted the original setup, I did not have to fiddle much with the HT503 settings nor the Asterisk configuration to get it working. In fact, before getting an ATA, I had a working Asterisk setup with softphones used as intercom.
So I was quite disappointed when I discovered that my HT813 was “deaf and dumb” with respect to the PSTN. As I don’t have an oscilloscope to watch what is exchanged on the phone line ahead of the FXO port, I came up with that setup.
As it is still up and running, if you want me to try things that I didn’t think of, just tell me.


Hello jyf0008, something new ?


Hello Ludovic,
Unfortunately, nothing new (I have not investigated any further either). I restored my original setup with a replaced HT503. Problem is in standby…