I've never been a huge fan of Grandstream, but I do prefer using VoIP adapters and analog phones to IP phones, for the simple reason that I have a crapload of analog phones lying around, and VoIP phones are still rather pricey. Plus, in a home situation, if you answer the phone and it's for another family member you can just holler at them that the call's for them, and they can pick it up on the nearest extension, without either you or they needing to dial any special codes, etc. But one of the objections to analog telephone adapters, and the reason some people prefer IP phones, is the lack of wideband audio support in ATA's. Now I know that some cheap imported analog phones limit the bandwidth, but it wasn't always so; as kids we'd hook two old phones (the kind recently retired by Bell or General) in series with a battery and were amazed at how clear the voice sounded compared to calls over the PSTN. But I digress...
Anyway the interesting thing about this new device, the Grandstream HT812 is that is has "Exceptional voice quality with wide-band HD codec" according to their datasheet. The Opus codec is used for HD, which may be an unfortunate choice for Asterisk users, since Wikipedia states that Asterisk lacks builtin Opus support for legal reasons.
Other touted features are:
- Supports 2 SIP profiles and 2 FXS ports
- Strong AES encryption with security certificate per unit
- Automated & secure provisioning options using TR069
- 3-way voice conferencing per port
- Supports T.38 Fax for reliable Fax-over-IP
- Supports dual Gigabit network ports
- High performance NAT router
It does NOT specify whether the "2 SIP profiles and 2 FXS ports" can be mixed and matched, as in an Obihai device. I'm just curious whether anyone has tested one of these, particularly using the wideband codec, though I'm not sure how you would given the lack of support in Asterisk.
My opinion of Grandstream is that they always come so close to getting it right, but always miss it on some small detail. In this case, I don't get why they didn't choose a wideband codec that was more compatible with Asterisk. Am I missing something here?
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