Delay in voice from FXS to FXO on HT813


I have set up 2 x HT813 back to back via a dumb switch.
With a POTs phone connected to the FXS and on the other ATA connected to our equipment via its FXO port.
All set up using the document ‘Peering HT813 with HT8XX’.
None of the ATAs is using a SIP server, just straight back to back with no registration.
FXS ID 222 and FXO ID 333.
FXS set up to auto dial 333 immediately.

When a ring enters the FXO, it calls the FXS no issues. No delay in audio on initial connection.
When the POTs phone is lifted to initiate the call, reaches the FXO and my equipment detects the FXO is off hook and the two ATAs are commnicating immediately . BUT, every time I make a call from the FXS to FXO there is about a 4-5 second delay before the FXS transmits its voice to the FXO.
My equipment that is connected to the FXO line is a simple FXS analogue line, that has no delay.
I have also tested with no auto dial and manually dial 333 from the POTs, same result with a delay in the voice.
Both ATAs running latest FW
I have seen many questions about this with finger pointing over networks and VoIP servers, but this is as simple as it gets.

Has anyone out there got to the bottom of this, as I need this resolved as my copious supply of the old Linksys SPA3102 has finally exhausted, the only other reliable FXO device is the AudioCodes that are quite expensive.



You should do a network capture (SIP file) at both HTs. Start the captures, make the call and then start a count immediately (1, 2, 3, etc.) upon answer. Allow the other end to answer whereupon they should be able to hear you and should be able to indicate what number they first heard. Hang-up and stop the captures.

You can examine the capture to see if the messaging and voice match that of what was physically heard and if at both ends. You can also see how the messaging played out. However, if you have not already done so, change all the codecs in both HTs to PCMA first to avoid the possibility of any fragmented packets.


Getting a network capture is like pulling teeth with my work PC that’s locked down.
Anyway, with further testing
When the FXS initiates the call, I can hear the DTMF tones from the auto dial on the FXO side immediately, then silence both ways till after 4-5 seconds.
I have now also tested both of these HT813 as a FXS to FXS using 2 POTs phones with auto dial both ways.
There is no delay in voice once answered either direction, and no matter what one initiated the call.
Hope this may give some more insight to who knows these units a bit more intimately than what I do.


The HT supports the capture, look for SIP file.