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Hi all,
Looking advice about testing VOIP before committing to BT FTTP without landline.
I have an existing BT landline and two analogue phones, one DECT and one corded. I have been looking at Sipgate Basic and the Grandstrean HT-812 as a test base. However I am unsure if I plug both phones into the HT-812, with adapters, will both phones ring with one number? Or does each HT-812 phone port need its own phone number?
Many thanks for all and any replies .
Edited by Adduxi (Sun 26-Jun-22 10:54:01)
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If you have two wired handsets you can hang them both off the same port, simply wired as an extension (ie simply in parallel or just use 2-into-1 adapter) - then they’ll both ring on an incoming call from the single Sipgate sign on. Presumably they’re modern-ish electronic handsets with a modest REN load.
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Pheasant,
Many thanks for the clarification. Yes, both modern phones and they are currently wired as master
and extension. Hope to get this setup within the next couple of weeks and will report back in due course.
Cheers !!
Edited by Adduxi (Sun 26-Jun-22 15:18:35)
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Rather than adding an additional box, how about replacing the existing phones? We replaced our PSTN phones with a Grandstream WP810 and WP820 (sipgate allow multiple registrations). These are more compact and there is scope for improved audio quality using the Opus codec(though not yet in use with sipgate basic).
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Just a note about codecs. To get the benefit of a particular “HD” codec, both parties if using SIP need to be using the same codec otherwise they’ll fall back to a common denominator which will be something like G.711u/a.
If calling/receiving from a ‘regular’ PSTN line then typically there will be no benefit in using a higher quality HD codec as the same thing will happen and they’ll default to a common but lower quality codec.
As more folks migrate to SIP endpoints then that of course ought to improve but they still need agree a common codec on initial handshake, so there’s still a reasonable chance the call quality will drop back to PSTN standard if the order of priority bin the respective codec lists doesn’t align.
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Just a note about codecs. To get the benefit of a particular “HD” codec, both parties if using SIP need to be using the same codec otherwise they’ll fall back to a common denominator which will be something like G.711u/a.
A note about codecs on sipgate, their page on the topic doesn't mention G722, but whenever I call their test numbers my router (which is my ATA) always seems to negotiate it... Anything from PSTN seems to come through as G711a.
Not that my ear can tell the difference between G711a and G722 on my house phone anyhow, I suspect DECTs additional processing of the sound (which I think is G726) will round off any additional frequencies you'd receive from any HD codec!
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DECT supports G722 doesn't it? My phone's seem to use it when the SIP connection negotiates a G722 connection (interestingly the information line 0800 500005 seems to connect as G722, but other fixed lines are G711)
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Nothing surprising. See my post above.
If you’re calling other PSTN lines then it’s really only G.711 / PCM as that’s the been the default codec on switched phone networks for decades.
When you’re calling a test line, the call is staying on-net rather than breaking out onto a third party network, and so a higher quality codec will be used. A similar situation would arise if you called another party on the same SIP provider (and you both had higher priority set on the codec)
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Rather than adding an additional box, how about replacing the existing phones? <snip>
Maybe down the line, as I'm only testing at the moment and most likely only going to move the DECT cordless phone across to VOIP.
Good suggestion however
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Just an update as promised. I've setup the HT812 with my DECT phone and it worked first time!
I was quite surprised really. So, will run with this for a while and if it all works out, will ditch the BT landline. Besides, it's only a handful of people would ever call on that line. Most people now use the mobile.
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