Software Version
Program 1.0.5.9
Boot 1.0.5.2
Core 1.0.5.2
Base 1.0.5.7
CPE 1.0.4.145
There’s no wiki entry for the 812 V2, so I followed the config instructions for the HT-801, (except fax settings). I also use a softphone on my PC, which works just fine. There shouldn’t be any common port conflicts, as I have my softclient configured to use a distant set of SIP ports.
The HT-812 V2 registers fine, and I can make outgoing calls, but can’t receive incoming calls. Google results talked about needing to create a subaccount just for the Grandstream. I did that. Then it starts to get complicated. Another recommendation here on the forum for a Grandstream suggested you add the subaccount for it to your ring group. I can’t seem to figure out how to add the Grandstream subaccount to my ring group. I still can’t receive incoming calls.
Yes, creating subaccount for each device is a best practice: one for your softphone and one for your HT812. SIP forking does work at VoIP.ms but you give up many functionalities and expose yourself to a more complicated troubleshooting. My recommandation: forget about the default account, just leave it unused, always use subaccounts.
Here is what I think you would need (replace 1234567 with your account number):
Create a subaccount for your HT812: 1234567_ht812
Create a subaccount for your softphone: 1234567_softphone
Create a ring group (DID Numbers > Ring Groups)
Add both subaccounts as members. To add a member, click on the field “Type to search member” below the language.
In your DID (DID Numbers > Manage DID(s) > Edit your DID), set the main routing to “Ring group” and choose the ring group you just created.
If you want to have a voicemail answering the call if none of your subaccount picks up, set it in the DID, not the Ring group.
Update your HT812 and Softphone configuration to use the new subaccounts.
When registering your HT812 and Softphone, make sure you are connecting to the SAME server that you choose as the POP for your DID. If you don’t connect to the same server, incoming calls will not route!
Thanks for your reply. I already set up a subaccount for the Grandstream. (The softclient already had one, that appears to work fine). There was already an existing ring group with two subaccounts in it, which included my softlicent and another subaccount for an account I never used. However, I can’t figure out how to add the Grandstream subaccount to the ring group. The Grandstream subaccount doesn’t appear in that menu, only the softlicent subaccount, and the account I previously deleted.
And with the Grandstream subaccount, I don’t see where it gives me the choice to choose which server. It does with my other subaccount, though.
Please ignore my last post. I finally figured out how to add the subaccount to the ring group. So far, it seems to be working. I’ll test further and post back if that changes. Does anyone know if there’s an IP Hostname setting in the 812 V2? I can’t find one, even though one appears in the ATA’s search results. This makes me wonder whether there was one, but they removed it.
After some testing, I’ve discovered that my ATA is deregistering/getting deregistered from the server after a period of time. I don’t know what that period of time is. I thought I set all the keepalive/timeout settings the way they were supposed to be (at least acc. to the HT-801 wiki entry), but I guess something is wrong. Is this likely a keepalive problem? What settings should I be looking at?
Oh, and if I don’t configure the ATA to connect to the same server as the softclient, it doesn’t receive incoming calls.
The ATA and the softclient are configured to use, and appear to be using different SIP and RTP ports, so I don’t think that’s the problem.
The likely glitch: In the settings, the name varies: either “hostname” or “host name”. (That’s Grandstream’s fault, but it’s not a big deal.) Do the search with the space. The setting you’re looking for is probably spelled “host name”, as it is in this HT801 V2 settings screenshot:
Hmm…I figured out part of what’s going on. The Host Name field doesn’t appear unless you select Auto-Configured from the IPv4 dropdown just above. I don’t understand why. I’m no guru, but don’t devices usually let you configure a host name if you’ve chosen to assign a static IP?
And now, the ATA is getting deregistered after some period of time. I’m not sure how much time. What settings should I be checking? I believe I have Keepalives enabled already.
Bump. Does anyone know how to fix my problem with deregistration? VoIP.ms still is taking days or longer to respond to ticket replies, and there’s still no human in the chat function (after 3 straight weeks).
I never have dealt with HT-812 v2, but I had an HT-801 before. I don’t remember needing to fiddling with those network parameters at all. You can probably the adaptor to factory default and follwed the instructions by voip.ms
Thanks. Yes, I configured it 99% according to those settings. The audio quality has been generally good, but it keeps deregistering. I know I probably just have to change one or two settings to change to fix that, but no one seems to know what those settings are. Grandstream, like VoIP.ms is keeping me busy with lots of homework and is slow to respond to my emails, closed after hours, on weekends etc. What should take 1/2 an hour over the phone takes weeks by email.
Anyone else realize the irony of VoIP companies that don’t offer support by phone?
As mentioned before, I’ve been trying to reach a human via live chat for 4 weeks. No one is availabe, at any time of day. Many other people in the forums here, and on social media are reporting the same thing.
On top of that, for weeks, the chat function kept insisting it couldn’t find a record of my account.
Primary SIP Server - the same one you seleted in your VoIP.ms sub account “POP Restriction”
Prefer Primary SIP Server - no
Outbound Proxy - leave blank
Backup Outbound Proxy - unchecked
NAT Traversal - Auto or Keep-Alive
Under SIP Settings:
Register Expiration (minutes) - 5
Re-Register before Expiration (seconds) - 5
Use Random SIP Port - checked
Under Codec Settings:
Set the list of Vocoder Settings
Vocoder Settings(in listed order) - match the codecs you selected in your VoIP.ms sub account “Allowed Codecs” list
In RTP Settings:
Use Random RTP Port - yes
You didn’t say if you are using SIP over UDP/TCP or TLS. I use SIP/TLS and encrypted RTP. This requires you select the appropriate options in the ATA and your sub-account. It also stops prying eye and internet routers from trying to “help” your connection using Application Layer Gateway devices, your own router might have this feature.
I actually have got a reliable SIP connection now for close to 7 days, with both ITSPs. I suspected a router custom setting was the cause. I removed it, and so far, so good. BTW, recommending to use a random SIP port is maybe not a great idea when there’s at least one other VoIP device on the network. I currently have set the two devices to use different SIP ports (and for the moment, different ITSPs). So far, so good.
Agreed, if you have more than one device then you can utilize different fixed SIP source ports. But the likelihood of both or more devices choosing the same SIP source port from ~64000 choices is slim.
