I just got a GrandStream HT802 V2. So far so good. But I read scary reviews about this device. Such as great until it’s not. Recent devices seams to be failing at an alarming rate.
So I’m already seeking for a SMB or even enterprise grade device.
Suggestions are welcome.
Hi @voip4ever,
Yeah. I have my first GrandStream HT701 for about 6–7 years until it started to behave very bad and work when it wanted. I changed it for the HT802. It makes about 2 years now, I think. I do not expect more than 5 more years out of it.
My next step will be to go voip directly using something like the GrandStream DP755 and DP752 devices.
Kinds regards!
I had an OBi202 that was given to me a couple of years ago and three Linkysys PAP2’s of various ages from when I used to set up temporary event communications. When I started using VOIP last month after a ten year absence, I started out with the PAP2’s because they were familiar, even if I had to dust off some cobwebs. The PAP2’s limited abilities, especially registering to off LAN servers quickly became apparent so I switched successfully to the OBi202.I wanted a backup device so I purchased an HT802 but found the configuration to be considerably more difficult; I sent it back and purchased another used OBi202 that turned out to be an OBi302.
If you aren’t limiting yourself to “in production” devices, consider the OBi202/302. They can be found on eBay, Mercari, Facebook for reasonable prices, the configuration is easier IMHO also they are a proven stable device with a long track record and a few well documented quirks. From my research, the only weak link is the power supply, the ATA’s tend to outlive several power supplies. If you want an in production version, look at the HP/Poly 402 that doesn’t appear to be much different than the original OBi devices.
Thanks for your reply. I guess I didn’t know what I was getting into. Used legacy devices are reasonably affordable. But often lack modern security features.
Then the current “enterprise“ grade products are way too expensive to my taste. I’m just a home user after all. So I think I will stick to the HT802v2 for a while. When push comes to shove, I’ll make more use of call forwarding to my cell phone and/or use a softphone app on my cell.
The voip.ms app is not too bad. I’ve installed it on both a iPhone and an Android phone and both seams to work well.
The free VOIP. MS app gets the job done but it certainly isn’t elegant compared with paid apps like Bria or Zoiper. It’s actually a reskin of another free app whose name escapes me.
Thinking back on my time with Ht802, I don’t recall any security features I haven’t implemented with the OBi’s. The only additional security feature I would to see is limiting administrative access to the local subnet. The OBi202/302 have an internal router that has been eliminated in the Poly 402. It dates back to the days when putting the ATA upstream of the router was “just easier”. By eliminating the router nobody uses anymore there is one less thing for bad actors to gain access to and use as a platform to exploit the network.
At VOIP.MS, I’ve had very good luck with the Grandstream HT801V2 and HT802V2.
Below is the instruction sheet that I use to set up either of those models for use with VOIP.MS.
Grandstream HT801V2 (and HT802V2) setup for VOIP.MS
• The info below is specific to the HT801V2 but pretty easy to apply to the HT802V2.
• Before starting, in the VOIP.MS subaccount, set Encrypted SIP Traffic to yes.
Set the HT801 to factory defaults. Connect it to the internet, then dial ***02 to hear its IP address. Use a brower to go to that IP address and log in with:
• username: admin … password: [get it from the Grandstream’s label]
At the prompt, change the password (limited to capital letters, lower-case letters, and digits).
PORT SETTINGS > FXS PORT > GENERAL SETTINGS:
• PRIMARY SIP SERVER: sanjose2.voip.ms [for example]
• OUTBOUND PROXY: [same as entered for PRIMARY SIP SERVER]
• SIP USER ID: [subaccount name]
• SIP AUTHENTICATION PASSWORD: [subaccount password]
• NAT TRAVERSAL: Keep-Alive
At the bottom of the page, click SAVE AND APPLY.
PORT SETTINGS > FXS PORT > SIP SETTINGS:
• SIP TRANSPORT: TLS
• REGISTER EXPIRATION: 5
• ENABLE SIP OPTIONS/NOTIFY KEEP ALIVE: NOTIFY
• SIP OPTIONS/NOTIFY KEEP ALIVE INTERVAL: 20
At the bottom of the page, click SAVE AND APPLY.
PORT SETTINGS > FXS PORT > CODEC SETTINGS:
• SRTP MODE: Enabled and forced
• SRTP KEY LENGTH: AES 128&256 bit
At the bottom of the page, click SAVE AND APPLY.
PORT SETTINGS > FXS PORT > ANALOG SIGNAL LINE CONFIGURATION:
• RING POWER: 55Vrms
At the bottom of the page, click SAVE AND APPLY.
PORT SETTINGS > FXS PORT > CALL SETTINGS:
• NO KEY ENTRY TIMEOUT: 10
• DIAL PLAN: { <=1NPA>[2-9]xxxxxx | 1[2-9]xx[2-9]xxxxxx } . . . . . . . . . . . . . . . . . . . . . . . . but replace “NPA” with your area code . . . . . other options: 011[2-9]x. | *97 | *75xx | 4443 | 911
At the bottom of the page, click SAVE AND APPLY.
PORT SETTINGS > FXS PORT > ADVANCED SETTINGS:
• VALIDATE INCOMING SIP MESSAGE: check the box
• CHECK SIP USER ID FOR INCOMING INVITE: check the box
At the bottom of the page, click SAVE AND APPLY.
PORT SETTINGS > FXS PORT > CALL FEATURE SETTINGS:
• ENABLE LOCAL CALL FEATURES: No
At the bottom of the page, click SAVE AND APPLY.
Reboot the Grandstream.
We have had one for several years with no problems.
@wizardgmb I just got a Poly 402 After getting sick of rebooting the HT802V2 when no longer receiving calls after few days. Looking forward to see the Poly stability. It was twice the price of the Grandstream HT802V2… Relatively easy to setup. Thanks to the OBI example in the Wiki.
I will report later once I have a better idea of the stability.
I have been using Grandstream DP750 for the last few years. The voice is crystal clear due to no analogue conversion is involved.
I didn’t have clarity issues with the Grandstream. Nos longer ringing on incoming calls was the problem. No issues with outgoing calls.
need a bit more information. Did you try the settings by Zeta already? What are your actual settings on both your voip.ms portal and on the ATA?
I will be interested to see your results with the 402 as I have no experience with Poly devices post HP acquisition. My OBi302 has been up for 2 months without a hiccup, it would be 3 be it not for a power outage. This week, upon orders from the boss, I will be relocating it and the base station phone out of the dining room and might even try out the WiFi dongle.
@yzhou I missed @Zeta ’s reply. At first glance it looks darn close to what I have but I can be wrong. I will check and report back.
Thx Zeta btw.
@wizardgmb I will keep you posted. OBi was aquired by Poly then by HP. I’d expect comparable quality OBi vs the Poly. The HP/Poly interface is the same as the OBi’s. Mine is on a UPS so less sensitive to power outage.
Thx @Zeta. Constructive comment: Always suggest to backup the current settings and tell how to do it. Especially if the target audience has an intermittent problem where everything works fine until it doesn’t.
After verification only 2 significant parameters were different, my values are between square brackets [ ] BUT I didn’t do the factory reset. Since enabling TLS and SRTP is a bit more involving than what you suggest.
What I did was:
Could the ring power and local feature interfere with incoming call detection after a while?
As said previously the HT802V2 worked fine all the time for ougoing call. But I was getting incoming calls for days after a reboot, then suddenly no more incoming calls. Calls were still getting thru to the other members of the ring group. Another reboot was fixing it for few more days. Then repeat.
Apparently it is a very common problem with the HT802V2 even more than with the previous HT802V1.
My current setup is for ONE ATA device at the time. But now that I have two ATA devices. I can spend few minutes to setup both (the Grandstream and the Poly) and afford experimenting with the HT802V2 configuration. (Factory reset, your parameters instead of VOIP.MS recommended settings, enabling TLS and sRTP.) and see if I still lose incoming calls after few days of working properly.
EDIT1:
After further investigation. The issue tends to point to TLS sessions getting dropped either by device, and/or my Firewall and/or by the Carrier. The HT802V2 do not have explicit TCP Keep Alive so it doesn’t look good for an easy fix. To answer my own question it is quite unlikely that the Ring Power and/or the Local Call feature will improve the TLS stability. I will now dig a little in my firewall config and see if I can extend the TLS sessions.
@Zeta any advice on improving TLS stability will be welcome.
EDIT2:
It was suggested to bypass the NAT in the router in order to protect the TLS sessions. I didn’t feel comfortable to put the ATA device in the DMZ and fully exposed it to the internet. My Modem does not have port forwarding but it got Virtual Servers. So I configured the ATA ip address as a virtual server for port 5061 (default TLS). I still get NAT protection but it should protect the TLS sessions from been terminated by the modem. Let see if the incoming calls on the HT802V2 now last longer.
Otherwise I could probably end up with the same issue with Poly ATA 402… Cause at the end the problem might not be the device itself but how TLS sessions are handled by the multiple hops across the path of the packets (ATA - Firewall - Modem/Router - Internet - Carrier)
The easiest way to test is to disable TLS and see if I lose incoming calls after few days.
I signed up with voip.ms must have been back in 2016 and I moved 4 people including myself off of Bell POTS to voip.ms and at the time I bought a bunch of Cisco SPA-112 dual-line ATA’s for that, they were about $50 each, and they’re all still in use and working fine. Those Cisco ATA’s have been EOL for about 5 years now, but I bet you can still probably buy some (used) off ebay.
They used to have good reviews back in their time. By current I meant still supported. What I find strange is the popularity of IP phones is still on the rise but there is less and less ATA devices. I guess it is due to the larger availability of pure IP devices (no analog interface required)