Grandstream HT813 *Intercom setup*

The intercom system is setup and working on a Grandstream HT813 through the FXS port. When the ext. is dialed to activate the intercom and you speak there is a delay (after you hang up) to when it’s announced. Simply saying it’s not in “real time”… is there a setting or settings to change so this will announce in real time?

This is very unlikely to be endpoint related. As far as the phone is concerned the only thing that the phone sees, which is special, is a custom header saying something like “answer after”. The phone won’t buffer the message.

Are you using Paging Pro? If so,there is an option, valet mode, also referred to as airport paging, which will cause this behaviour. Have you accidentally turned that on?

Actually, as the HT813 is an ATA, it can’t implement answer after, as there is no way of signalling that to the analogue phone. I guess it can ignore answer entirely on one of its lines. The documentation seems to have a mention of a paging feature code, but no real explanation of what it does.

Thanks for the reply… mods or whoever can move this to wherever it needs to be (sorry)

I’m an IT person trying to make this work 100% correctly, so forgive me… not much knowledge in this area. What’s being used is a Bogen TPU100b… it’s plugged into the HT813 (FSX port)… it’s working. I’m just trying to get it to announce over the intercom in real time, if that’s possible. The way it works now is you speak your message into the phone, hang up… then it’s announced over the intercom. Maybe that’s how it works… ?

The Bogen device is a pure amplifier, so won’t delay the audio.

I can’t find any mention of the HT 813 doing anything clever with paging.

I’d therefore have to assume that you have enabled valet paging in FreePBX.

The full log may give a clue, but to be certain of where any delay was, you would probably need to capture the RTP, on both sides, using tcpdump and wireshark, or just the latter, and seeing if there was a delay in that audio.

Reference to Paging Pro Valet options in FreePBX User Guide

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