Can the HT813 do this?


I am interested in the HT813, but before buying one I would like to know if the following is possible.


The analog phone is currently connected to the fiber modem/router (PLDT Philippines). I guess the port on the modem that the analog phone is connected to isn’t strictly traditional PSTN but an internal ATA for VoIP (which, in turn, would be connected to the PSTN at the other end).

My goals:

  1. To be able to make local calls through the PSTN line from my mobile phone from anywhere in the world.
  2. To receive calls from the PSTN line on my mobile phone if the phone on the FXS port is not picked up after a set number of rings.
  3. Make and receive calls on the PSTN line with the analog phone.

I used to be able to do this with a Sipura/Linksys/Cisco SPA3102. That was 12 years ago, but the call quality was inconsistent. I am hoping technology has improved in the meantime, and would like to attempt this same setup with the HT813. Is this possible? If so, could you please point me to resources/tutorials for the above setup?

Many thanks!


HT813 does not support any APP
to do this, for example you need UCM6301 or another voip server Appliance or Cloud or other application


Thanks for the reply.

HT813 does not support any APP

Wouldn’t any SIP softphone client do?
With the SPA3102 I had free sip accounts set up on the FXS and FXO, and I could call either from a softphone and get a ring and PSTN dial tone, respectively.


in your case the PSTN line is actually a SIP account of which you could ask your provider for the credentials, at which point you can probably use any Sip Client
(the supplier in many cases has a dedicated app)

as I said, other “external” solutions I wrote above

this is a Grandstream forum, so you can find Grandstream solutions


Just to give anyone interested an update, I bought an HT813 and managed to configure it as per the above with all goals achieved, albeit somewhat imperfectly. I signed up with a VoIP provider and configured the HT813 FXS and FXO with the SIP trunks I created with my VoIP account. This provider had a SIP mobile app which I installed on my phone.

I mentioned “imperfectly” above mainly due to the noisy/scratchy tones I’m getting when calling the PSTN line from the mobile app. If anyone has any tips on how to fix this, it would be much appreciated.


NBN FTTP with a PSTN (FXS port from the Modem) over the service? the Noise is usually due to a G729 codec and more than likely an issue but depends on the provider which you do not declare.

Do you hear the Noise if you plug an analogue handset direct into the Modem ?


NBN FTTN. NBN provider is Telstra. VoIP is Crazytel.

No noise when analogue handset is plugged directly into modem’s FXS.

All HT813 settings at default except:

  1. FXS and FXO accounts set up (Crazytel)
  2. Unconditional Call Forward to VOIP set to another Crazytel SIP trunk (works well)
  3. AC Termination in FXO page set to Australia/New Zealand. (Side note: I compared Advanced Settings and FXO pages before and after changing AC Termination from default USA to Aus/NZ, but there was no diff. I’m likely misunderstanding what AC Termination does and have posted a separate question in this forum on this.)

Appreciate suggestions on how to fix noise issue.


Are you using the Netgear V7610 Telstra modem? if not which modem?

Ps. Telstra might be dedicating udp port 5060 for the SIP to analogue service allocated to the internet account. Have you made any allowances for the Crazytel service using the modem given that Telstra might reserve port 5060?


Modem is Sagemcom F@st 5355.

The local SIP port on the HT813’s FXO port is 5062 (the default). I’ve not made any setting changes on the modem or the Crazytel service. I have my doubts that it’s a port issue as I’m able to make and receive two-way calls with my softphone over the PSTN port. It’s just that the audio quality could be better.


Sagemcom F@st 5355 has UPNP turned on usually at default ergo the HT813 is registering. You may need to use STUN so that the RTP ports are opened correctly for the other carrier Crazytel.

Then you will have to train the FXO of the HT813 to the psuedo PSTN from the Sagemcom F@st 5355 to ensure that it is balanced which is why you are hearing noise.

If you have already done that then it might be the telephone line cable from the FXS port of the Sagemcom F@st 5355 to the FXO of the HT813 is receiving earth hum from a power source nearby.


The HT813 is indeed registering and, as I mentioned above, is able to make and receive calls as per my goals, despite FXO audio quality. Pardon my asking, why would I need to check the RTP ports if calls can be made? I wasn’t expecting RTP to affect audio quality. I’m obviously missing some important point.

After a bit more research, I changed AC Termination to impedance based, setting the impedance to “220 + (820 || 115nF)”. I read somewhere that Australia’s standard is “220 + (820 || 120nF)”. I don’t know if the 5nF cap difference matters much. It was previously set to Country Based - Australia/New Zealand. Further (qualitative) testing suggests an improvement, but I could only be imaging things. I will be testing it some more. Voices are still discernible but there is an obvious clipping and slight echo on the dial tone I get when I call the SIP trunk the FXO is registered to. I might try to upload an audio recording. Perhaps there’s a gain setting I can adjust… somewhere…

To train the FXO, what setting(s) should I adjust?

I will try moving the equipment to reduce hum.


The RTP stream which is audio can be affected by the Modem prioritising other ports causing noise to be heard, but what sort of noise sound is really being heard?.

Are you able to record the audio and attach it hear so that it can be determined if it is earth hum from power or actual distortion by other means.

Or can you do a packet capture for a noisy ata call.


Here’s an MP3 recording of the PSTN dial tone from my softphone when I call the SIP trunk on the FXO port. Although the sound is distorted, I can still continue and successfully dial and connect to any landline or mobile through the FXO. (215.2 KB)


Dial tone is generated out of the ATA’s fxs port, so do you still receive the same digital noise out of the call after the call is established ?

Perhaps remove the pstn line from the ATA’s fxo… does the digital noise appear still on the fxs of the ata?


The FXS is just fine. Perhaps you may have misunderstood my setup. Here’s a diagram, if it helps:

Here’s how the problem happens:

With the softphone, I call the FXO port (222@crazytel… “222” is for illustration; in reality, it’s an 8-digit SIP user ID).

The FXO “answers” with a dial tone - as expected - but it’s distorted (recording above). This distorted dial tone is what I’m trying to fix.

At this point, while that dial tone is still active, I can still continue and dial any PSTN or mobile number from the softphone. The call gets established. Audio is ok.


Is the recording of the dial tone noise heard on either a PSTN FXO call and also is it heard via a sip voip call?

If it is yes to both then the ht813 is the problem - factory reset - no programming and see if the issue remains, if it does then you have a faulty unit perhaps.

You can also open a ticket direct t Grandstream else take it back to where it was purchased.


Also duplicate the following for your HT ATA

The following is what I am using for my ATA’s as it is close to what is “normal” to the today’s standards, you might have different ones but let me know if you use or modify these.

Australian Dial Plan

The 1xxx = extension dialling if connected to a UCM with 4 x digit extension numbering - adjust to suit.

Australian Dial Tone (what you hear when you pick up the handpiece to dial)

Australian Ring Tone (system ring cadence - when someone calls you and your phone rings)

Australian Ring Back Tone (the ring you hear when you call another party)

or you can use mobile style Ring Back Tone

Australian Busy Tone


Thank you. I will try these solutions one at a time and test. It will take me a few days but will report back.