Occasional lurker, first-time poster here. I’m running into an issue where I can’t get my newly purchased HT801 to make or receive calls. Here’s what I’ve tried so far:
Verified DID routing to ensure the subaccount is using the correct XXXXXX_username.
Enabled TLS as recommended
Set Caller ID Transmission to “Multiple Data Message Format.”
Triple-checked all settings against the official ATA configuration guide.
Tested by replicating my working Polycom setup on the HT801 (to rule out a subaccount problem).
Unfortunately, nothing has worked so far. The Polycom on my main line works fine, so this seems specific to the HT801.
It’s been 60+ hours since I opened a support ticket without a response (#T8ZSSV). I’d ideally prefer official support for the service I’m paying for, but in the meantime, I’m hoping someone here can point me in the right direction.
I’ve attached screenshots of my current setup for reference. Any guidance would be greatly appreciated.
(And to any staff monitoring the forum — please flag this to management. Long support delays aren’t helping customers stick around.)
If you enable TLS on the Grandstream, check you also enabled SRTP (encryption for RTP).
Also check you enabled encryption on voip.ms.
On Voip.ms you can’t have one encrypted and not the other.
The server can’t tell that there is an SRTP issue until you try and make or receive a call - so you will show as registered even if there is SRTP is disabled.
I don’t have a HT-80x online at the moment, but IIRC check the FXS port for SRTP setting.
Remove the prefix … it may not be causing a problem, but it isn’t doing you any good.
Also check the vocoders … I have mine set to G729 first and PCMU for all other choices. Any vocoder not supported by voip.ms is also not doing you any good.
Thanks for the kindness to assist. As you recommended, I adjusted Choice 1 to be G729 and removed the string from dial plan prefix with no change. Still fast busy signal on outbound and no ring other than to my Polycom on inbound calls. To note the string and Vocoder’s are the ones presented in the ATA guide. Grandstream HandyTone 802 - HT802 - VoIP.ms Wiki.
Your manage sub-accounts screenshot is showing UDP protocol not TLS and the Registration Status isn’t showing a lock icon - suggesting that you may have missed the Encrypted SIP Traffic setting for the sub-account.
Hi, I was already been done prior to my posting and achieved no difference so I reverted back as other forums mentioned to revert to the most basic functionality when troubleshooting.
Figured to give it one more go so you don’t think I’m being dismissive of your assistance (I’m grateful). Here are the screenshots. Unfortunately, even configured for TLS still not working.
Thanks for the update. It does look that the SRTP options are set correctly in the new screenshots and I can’t spot anything obvious.
Did it work on UDP with encryption / SRTP disabled?
Do you have SIP ALG enabled on your router (disable this if you have as it is often a cause of problems - although with TLS it shouldn’t interfere with the traffic.)
Any reason to believe port 10000 UDP is blocked by your router/ISP? (any reason not to use default RTP port? Any ports being forwarded on your router?)
Not sure what level of technical knowledge to assume: it’s possible to send debug output to a syslog server elsewhere on the network (Advanced options) which would probably tell you what was happening - but that’s another rabbit hole you may not wish to go down. (If you do, there’s a command line syslog server I wrote at Releases · m-z-b/syslogqd · GitHub - Needs to be enabled on Windows firewall)
If all else fails, I’d start again from scratch. i.e. Factory reset the Grandstream, use a new sub-account, and try a default setup without encryption, following the voip.ms wiki pages. It’s possible there’s something obvious that has been accidentally set which neither of us have spotted. (And watch out for copy/paste including extra spaces when copying passwords)
Thanks for the help. Like I said the service works fine with my polycom handset so that should theoretically rule out any issue with SIP ALG issues. I did check the router and SIP ALG is turned off. At present there is no port forwarding enabled on my router and I changed the local rtp port to the default setting of 5006. Still no dice. I haven’t conducted a factory reset yet so that’s last item on my list - I’ll report back. After that if it’s no longer working I’m plum out of ideas as well.
Partially resolved. A reset of the Grandstream HT801 and a new sub account and everything is working fine now - success. However the polycom will no longer ring for incoming calls. Tried changing routing to from main to the sub account - no dice. Switched back same deal. Any ideas?
Is the Polycom connected to a different sub-account than the 801?
I powered up an old phone last night as a test and forgot that the account information was the same as the phone it replaced (I plugged it in to test PoE on a switch, not to use as a phone). I was making test calls from another device to the replacement phone that kept randomly failing … then I realized I had two devices trying to share the login. Life was better when I disconnected the “test” phone.
Yes the Polycom is configured on the main and the other cordless phones are on the HT801 on a subaccount. Can call out fine from the polycom just can’t receive incoming calls any longer. I thought may it was on DND or something but when I did a test call I picked up the phone while the others on the HT801 were still ringing and got a dial tone. This time it’s likely something to do with the main did setting but I can’t figure it out.
P.S. You may not know so disregard but is this normal for their support not to respond back after 5 days? No calls or emails and I’ve written them back twice requesting an update. Made sure to check Junk and whitelisted the domain name - zilch.
Solved. Support wrote back today and said the issue with the Polycom not ringing was I didn’t create a ring group. Once I enabled it everything is working.
There is a place in the voip.ms portal where one can “View Tickets” to see if there are any replies from voip.ms that were missed. I find the portal is a better way to track my tickets than relying on email. (I have not had any issues serious enough to do a live chat or phone call.)
