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 (22.214.171.124 for HT503, 126.96.36.199 for HT813).
Thank you for your help.