HELLO & WELCOME

[dcowan]

Hi,
I'm not getting the Caller ID (HT813) with the above settings. I got this to replace the SPA3000 (I was also never able to get Caller ID working on that). Has anything changed?

Thank you!

[WelshPaul]

@dcowan On both SIP and PSTN or just PSTN?

Who supplies your PSTN line? Some telco's require a CLI subscription. If CLI is enabled on the line, have you tried using a different telephone? After all, it could be a fault with the phone and not the HT813 or PSTN line.

[andrewclark55]

Thanks Paul for taking time to post all that info.

I have entered all of it with my new HT801 on a new Sipgate account connected to a Draytek Vigor 2862ac router but without success after trying setting various other suggestions from the Grandstream forum, including updating the firmware and setting up an entry for a STUN server.

Symptoms are I can dial out OK and complete calls but incoming calls don't ring. Initially I was still able to answer my own test calls made from my BT line although it didn't ring but now, after trying various settings changes I find that calling the VOIP number just results in hearing one ring on the calling phone followed by the unobtainable tone.

Can anyone throw some light on this issue or point me to a configuration file that will actually work with Sipgate (not the screenshot they give as I tried that and already had to make changes to what they said which seems to be for the US market).

[VoipIT]

I doubt it's the Grandstream ATA at fault, sounds to me like the Vigor 2862ac is the cause. A Google search resulted in the following hit: https://community.spiceworks.com/topic/ ... n-dial-out

[andrewclark55]

You were right. It seems that the HT was fine but both the telephones I was using are not fully compatible. One that did work was the Binatone SpeakEasy 6 so I have bought another one off eBay.

Thanks for responding.

[BrianG61UK]

I doubt it's the Grandstream ATA at fault, sounds to me like the Vigor 2862ac is the cause. A Google search resulted in the following hit: https://community.spiceworks.com/topic/ ... n-dial-out

I'm told it's worth checking your router for a SIP ALG, SIP application layer gateway, or any SIP settings that might reffer to SIP ALG.

Although enabling SIP ALG is intended to help SIP work right through NAT it doesn't always work, sometimes modern implementation of SIP are better at working around limitations of NAT without SIP ALG.

So if you have a SIP ALG setting in your router try changing it either from OFF to ON or ON to OFF, both might help.
If both work leave SIP ALG off, it's a slight security risk leaving it on if you don't need it.