OPEN BOX - GRANDSTREAM - HT813 VoIP SIP ATA/Router, 1 x FXS, 1 x FXO, 1 x Fast Ethernet LAN, 1 x Fast E

I am originally from a country where phone calls from the US are over 0.5$ per minute. The local phone call rates in that country are only a couple of US cents per minute if calling a cell phone from a land line (landline to landline is even cheaper). The FXO port on the Grandstream HT813 is what makes it possible to save a lot if calling that country from the US a lot. This unit is a good tradeoff between being easy to setup and having a lot of configuration options. You may spend a bit more time configuring the unit but will have more control over its configuration. The notable features is the encryption support (TLS and SRTP secure connection) and the support of the option to use pulse mode phones on the FXS line (this cannot be done with Cisco SPA112 and SPA122). The only two alternatives to this device in this price range with FXO port worth looking at are the Obi110 and Obi212 but these units do not provide as many lower level configuration options as Grandstream HT813 and the older Obi110 does not seem to support secure connection. There is also a Cisco SPA232D-G1 but it has sound quality echo problems so I wouldn't use it. The one feature of HT813 which deserves a stand alone mention is the option called "Unconditional Call Forward to VoIP" where one can specify a SIP address to forward all incoming calls from PSTN to VoIP over the FXO port. If this option is configured, the SIP phone at the SIP address specified will start ringing after a desired number of rings on the PSTN line upon an incoming call. The SIP phone rings as if it was just another phone on the PSTN line and the person calling the PSTN number has no way of telling this is going on because there is not change in the dial tone etc. This feature is convenient because occasionally my relatives may need to call me in the US from their local cell phones so they would just call their landline in a foreign country to be transferred to the VoIP. I have configured the HT813 so that the forward to my SIP address begins after 10 rings, this way my relatives in a foreign country can still use the landline as usual without me having to answer their local phone calls (as most people wouldn't wait for 10 rings before they hang up). I don't know why but the SIP forward is only active for 3 - 4 rings after that the SIP call gets terminated and the PSTN line just keeps ringing as it used to until after the caller hangs up etc. I couldn't figure out where the 3-4 rings limitation comes from, maybe some settings but I see this more as an advantage because in most cases a call is answered within 4 rings. UPDATE Aug 23, 2022: I could never set up this unit properly so that one could dial different VIOP phone numbers from a foreign country's PSTN line though this device (that is when calling from a cell phone in a foreign country to a landline in a foreign country, said landline being connected to HT813 transferring to a VOIP line). The problem I was having is that the tone dialing would not always work correctly. I ended up setting up the unit so that it would automatically do the "Unconditional Call Forward to VoIP" and call my VOIP line as described above. Since for my application this approach was actually less confusing for my relative abroad to call me. But for some people this would be an issue as one wouldn't be able to dial different VOIP phone numbers if needed. There are a lot of settings in the menu related to how the dialing is interpreted based on the region and PSTN format but I am not familiar with those to determine if the dialing issue may be fixed by adjusting these settings. Also, the 3-4 ring limit mentioned above ".. the SIP forward is only active for 3 - 4 rings ..." is determined by a parameter in the menu and can be adjusted.

La configuration n'est vraiment pas évidente mais après plusieurs heures de debug, tout fonctionne à merveille (y compris les appels entrants et sortants). Utilisé avec FreePBX sur NAS Synology (image docker).

Originale

I've set this up as a FXO receiver and dialer using Asterisk on a home server. I use it to dial out, and also to receive calls using the built in Android sip support on a hunt group with all phones in my house. The default configuration is all set to US settings, and finding the settings for the UK BT lines can be tricky, but one configured it works really well. Still not got the FXS port working yet. Even though the client is connected to Asterisk, it won't ring. I'm not sure if this is a problem or I've configured it incorrectly, but I'm not desperate for this to work anyway. Given this 4 stars due to the US configuration defaults. If there were presets, I could forgive this, but there aren't.

todo ok