'suse the OT, but there's a pretty good chance someone here will know:
Thinking of getting a VoIP service via my ISP (Andrews+Arnold) - and intend to keep my POTS line for emergencies.
I have zero experience of VoIP... Can multiple VoIP clients be registered to the same account whereby all ring at once for an incoming call?
I'm thinking a Siemens Gigaset A580(?) at home (that handles the POTS line too) *and* the Sipgate app on my and SWMBO's mobile phones.
It would be truley awesome to have a mobile that can double as a house phone and take calls to "one true number" wherever I am, as long as I have 3G or WIFI access.
A&A have a hunting feature where a faling call goes to a fallback number, but that's not really what I'm after.
And yes, I will also have a crappy dumb handset on the POTS line for when the power fails and takes out both my broadband and the cell tower, like it did a few months back.
Same ISP as me. I have the POTS line wired into the VoIP box which makes life simpler - and weekend landline calls are free too!
Not sure you can. But the AAISP control pages allow you do do that anyway.
Oh, OK.
I have an Asterisk box set up so that the SIP client on the mobile registers direct with it. It becomes just another extension on the house phone system which works fine. But that may be overkill.
Yes, although the service provider could prevent this, or simply not be able to handle multiple registrations for the same user.
A&A allow multiple separate destinations to be called in parallel, and without actually registering if they have a fixed IP address. However, I do it by bouncing one call through my own SIP server (SER) to multiple destinations.
Yes, not sure of the services required from A&A, but from Gradwell you require one (or more) inbound SIP numbers and then a SIP trunk for outbound calls (alternatively you can use IAX2 to avoid NAT headaches, though we avoid that by using a bridging firewall rather than a routing one).
Hmm - that does sound interesting. I like Bob's suggestion that random VoIP clients can be treated as house extensions which is exactly the model I'm after.
It might also be an idea to buy 2 new numbers - one for "work" and one for home - I'm always giving out my mobile to random reps then having to decide who to ignore on a day off.
No-NAT at home is easy (have lots of IPs) but of course mobile in 3G/WIFI mode is most likely to be on a natted connection, or heavily firewalled.
Reckon I should buy a number and start experimenting! My servers at home are not up 100% of the time (more like down 2 days in a year) but A&A's fallback option could be used to fall back to my POTS number, so that should be OK.
Seems to work well. I have yet to see if the wifi at work firewalls it off - I hope not, because I have a poor 3G signal in my office!
Yup, although I ended up getting the rather more expensive Digium card (instead of an X100P clone) - I tried the latter and never got it to work properly. (it may not have helped in that I'm hosting it all on FreeBSD, not Linux!)
Not sure of the terminology, but mine registers (for multiple numbers) with the AAISP server. That can handle incoming and outgoing calls on all numbers with no problem. House extensions are via two Linksys ATAs (SPA2000 and SPA8000) which register with the Asterisk box.
The phone also registers with the Asterisk box and appears as another extension - so can be made to ring when any incoming call is received (actually it only rings if the general house number, or my personal number, is called).
I've got a router (Billion Bipac 7404VGP: recommended) that has a two POTS sockets for VOIP and use sipgate to handle the interface to the PSTN network.
that feeds a PABX.. solving the multiple ring conundrum.
I do not think it is possible to have a one to many number forwarding service though. Other than using a PABX.
If you use SIPGATE apps on the mobile phones, then I guess they will each have a separate account and separate number.
And presumably a further sipgate account for the Gigaset.
I suspect that what you want could be better achieved somehow with either a PABX or a Linux server running a VOIP based PABX style software.
VOIP with SIPGATE is essentially a call forwarding and interfacing service, not a one to many call routing service like a PABX is.
This box may well do what you want.
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looks like it can handle one or more VOIP 'trunks' as well as real POTS trunks, and connection to many local analogue or VOIP 'phones'
If you have local mobile sets that do VOIP over wifi.
However, that means registering the phones in wifi mode with the router as a sip server.
So the won't then work remotely in 3G, unless you do some fancy work to make the router publicly visible on the internet, and have it relay calls that way.. that ultimately seems to be the solution that 'gets you what you want' but it's quite complex and has potential security implications.
So in essence what you then have (nd I think its the minimal you need) is the following
1/. a SIP gateway like SIPGATE that handles incoming calls, and routes them to your own SIP gateway.
2/. A second private SIP client/server that acts as a PABX and routes incoming calls to a selection of (private extension style) SIP numbers.
3/. SIP apps on a mobile phone that allows that phone to connect to your private server either via wifi, or via 3G, depending where you are. This means you need a fixed IP address probably.
4/. If you want to plug a DECT system into that, you can. It's just another extension. But why bother if you carry your mobile?
Note that if your 3G data costs are cheaper than phone calls you probably never would use the phone over a the mobile network at all, unless only in a 1G or 2G area.
Other handy features are that you could set up as many 'extension' accounts on the router as you like, and allow free calls between - say - you and your family, and allow them low cost SIP calls on your number from wherever they are..that has a VOIP client access.
They're certainly cheap enough but I've yet to find people who accept the default 033 33 numbers without comment. I don't know if they offer more generally accepted ones now. You can also impose time profiles on the number to stop calls after a certain time but I'm not sure if diverting to a 'sod off we're closed' message is possible.
They do have a STUN server if that helps but it is not advertised.
Control panel is a bit clunky but it works.
Gigaset is dead easy to set up.
There's a good knowledge base at uk.telecom.voip , quiet there at the moment but I have had good help there in the past and I think the regulars still lurk.
I'm pleased with the Linksys stuff, and have two SPA boxes. You can configure them via pre-generated XML files, which saves a lot of fiddling (especially with the SPA-8000 and 8 lines). I've taken a punt of one of the very cheap PAP2T boxes from eBay, too.
The Digium clones are quite good these days - and not much more than an ATA. e.g. I've used the OpenVox cards for some years now with good results - especially when you combine it with the OSLEC echo canceller.
Some SIP providers do cater for this - one account, many devices and all devices ring on an inbound call. It all depends on the software (or hardware) at the ITSPs end. Asterisk doesn't support this natively, but it can be front-ended with something that does.
I thought Sipgate now supported that - I could be wrong though.
That may be, but all I could find at the time was the X100P. And it had no FreeBSD driver. It probably would have been easier to use an ATA, but the Digium card has worked flawlessly and I do have expansion options on it (and even the option of a hardware echo canceller, at a price!)
I think the thing to do it try it - I can use our 2 iPhones as guinea pigs for a few pounds to A&A for a number+SIP service.
I don't need to handle the POTS line at all, other than stick a dumb phone on on it for emergencies. I'm happy to advise a change of number to all and sundry for the convenience of a "mobile" household number.
Should A&A not be able to handle multiple clients, do I need Asterisk - or is there a lighter gateway I should be looking at for linux? I'll let A&A handle the voicemail2email (they offer this) so that's another feature I don't need at my end...
Initial tests with the Bria iPhone app and sipgate.co.uk as the VoIP provider show some promise (because I could sign up for a trial sipgate account in 2 minutes on a B/H Monday).
SWMBO's iphone needs a firmware flash to take the VoIP apps, so that can wait.
Seems the next problem to solve is the evil of NAT...
I am prepared to request yet another range of public IPs for VoIP use (and will probably get a /29 without much trouble) - but this does not solve the problem when I am in someone else's zone (eg 3G, or at work).
Bria works fine when the app is open and actively yapping with the sipgate servers. It does not work in backgrounded "PUSH" mode - which I am almost certain is a NAT related problem.
Seem some stuff about STUN and ICE services, so off to check out what that's all about...
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