Sending PSTN calls to my VoIP phone/s

#1
Hiya,

Got a bit of a question here. I'd like to roll my existing BT line in to my VoIP set up so that if a call comes in on the usual number, it is passed to the VoIP phones and can be answered, rather than me having to search for the analogue cordless phone.

From what I can tell there are three ways to try and do this, but I wondered if anyone had any better ideas or comments.

The choices are:

1. Port the number to Voipfone and have it as a geographic number
2. Set up an asterisk pbx and have it as an fxo line
3. Use a Sipura 3000/Grandstream Handytone gateway with an FXO connection.

Option 1 won't work as its the physical line my ADSL is on, and if i try and port it, the line itself will probably end up cut off!

Option 2 would work, but needs investment (PC, FXO card, time...)

Option 3 wouild work, but I can't really tell from what I've seen on the 'net if it will do what I want, IE, forward the PSTN call to the nominated IP phone/s

Your advice would be appreciated.

#2
Well option 1 may well be possible in the very near future with what is refered to as 'Naked ADSL' - basically this allows you to have a line without a number just for ADSL - NO DIAL TONE - as things stand at present POTS lines have to have a number allocated to them - so in theory you could port your BT number and have BT allocate the line a new number - and you would have an outgoing POTS PSTN line for bakup.

Option 2 - there are more sofisticated VOIP/PSTN gateways out there which do a lot of fancy routing:
vegastream
mediatrix
epegi
try a google for voip gateways

Option 3
Your best bet if the line is domestic - just stick a DEC phone on the ATA and it will give you failover to pstn should your inet go down and the option to dial over pstn should you need it - and your phone will ring as normal when people call your pstn number
Not so good for business use as DEC/normal phones do not have the features of VOIP phones
Regards,

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#3
I have a client in a similar position to yourself and I made the mistake of picking option 3 using a Grandstream HandyTone 488. It worked fine except for the fact that it was unable to detect a PSTN disconnect. Consequently, it was unable to clear down the PSTN line until the VOIP user hangs up the phone. Although the latest firmware that I downloaded into the 488 had better PSTN disconnect detection, I tried my best but could not get it to work (incidently, the company I bought the device from, couldnt get it to work either).

You may have better luck with the Sipura, but there seems to be a lot of people struggling with it, based on posts in forums, to make me believe that it is not problem free.

I was able to live with the PSTN disconnect problem for a while because, somehow, the call would eventually be terminated from the VOIP side. However, someone complained to VoipFone that the voicemail system was disconnecting the call after 30 seconds and they fixed the bug. Now, the voicemail system will happily record for days without hanging up the line. I eventually pulled the plug on the 488 when I got a 3 hour voicemail sent to me by email. Please VoipFone, allow me to switch off voicemail on some extensions, if no, please allow me to specify the number of rings before voicemail kicks in.

Incidently, if anybody knows of a cast-iron way of getting the 488 to detect PSTN disconnect, I am all ears. I am tempted to try the Sipura, but I would love to hear from anyone that has got it to reliably detect PSTN disconnects.

#5
voipfone wrote:Well option 1 may well be possible in the very near future with what is refered to as 'Naked ADSL' - basically this allows you to have a line without a number just for ADSL - NO DIAL TONE - as things stand at present POTS lines have to have a number allocated to them - so in theory you could port your BT number and have BT allocate the line a new number - and you would have an outgoing POTS PSTN line for bakup.
Do you have any idea when this might happen? Extensive Googling suggests this has been talked about for at least a couple of years, and I'm aware that the Ofcom report says its a good idea. I should imagine BT are desperately trying to avoid being forced to offer NDSL.

Is there any definite news on this? I'll be on the phone to BT cancelling my POTS service within nanoseconds of it being an option... :wink:

#7
Not sure if what I want is the same here or not. Can't tell if the 'simple' (i.e., not Asterisk) ATA solutions do what I'm after.

I would like my current inbound POTS calls to convert to VOIP and then divert to a Voipfone number that can forward freely to any or all extensions throughout our office, my house, etc.. Do the ATA's mentioned do this, or do they only convert the inbound POTS calls to locally attached IP phones?

#8
arcascomp wrote:I would like my current inbound POTS calls to convert to VOIP and then divert to a Voipfone number that can forward freely to any or all extensions throughout our office, my house, etc.. Do the ATA's mentioned do this, or do they only convert the inbound POTS calls to locally attached IP phones?
A Fritz! box can do what you need, and much more. See my review at Voipuser's review pages.

#9
Fritz looks a bit OTT for my simple needs. I only have one analogue POTS line incoming and that I want to route to all of my disperse VOIP extensions.

Thanks for the pointer though.

PSTN to VoIP bridging

#10
What you are looking for is called PSTN to VoIP briding.

The Zoom ATA 5800 allows PSTN to VoIP bridging. Simply put you call your PSTN line from outside, this gets routed to the VoIP connection, you then get another dial tone, and now you can make calls using you VoIP account as though you are at home using the ATA.

#11
Interested in the Zoom but bridging is not quite it either. I have an established customer base that know my pots inbound number. It being a NTL:Telewest number Voipfone can't yet take it over. Therefore I need to have my existing customers to dial my telewest pots number and end up ringing all VOIP extensions in group without knowing that they have been rerouted.

i.e. there should be no IVR or anything that, straight dial the number hear ringing, get an answer from one of us. Can these devices do this?

I know asterisk can do this, but I like the concept of automatic power/VOIP service unavailability failover back to PSTN if I must.

#12
Hiya,

I looked at this long and hard and none of the ATA's i've seen forward calls in the way you desire.

The only possibility, and I've yet to test it, is to use the inherent *72 "divert all calls" feature in the Grandstream ATA. IE: Set a "divert" on the analogue FXS port to forward calls to a SIP number - which would be the main number on your VoIPfone account..

I will have to test (unless anyone has already tried).

#13
If I understand you correctly, you should be able to do this with a Grandstream 488. It will allow you to specify a VOIP account to which any calls received via the PSTN connection should be diverted. It will also allow you to have the connected phone ring a specified number of times before the call is automatically diverted.

So, if you tell the device to divert to <account>@voipfone.co.uk and you have your Voipfone account configured to ring the desired extensions (using call group), then, when the caller dials your NTL number, you will hear your VOIP connected phone ring and they will get straight through to you without being aware of the fact that the call has been diverted. I have tried this and I know it works!

I have been told (by the company I bought the Grandstream from) that you can also do this with the Sipura 3000, which, in my humble opinion, is a much better ATA than the Grandstream.

(The one problem I still have with the Grandstream is: I cant reliably get it to detect PSTN disconnect after the call has been diverted to VOIP).

Is this what you are after or have I got the wrong end of the stick?

#14
You have the stick well and truely grasped at the right end Jaheli! Thanks for the reply. I think I should just go for it now and buy a Sipura, although the Zoom 5800 is very cheap! Anyone able to confirm if the Zoom has similar forwarding capabilities as the 488/S3000? and any comments on quality/reliability would be nice too.

Cheers,

Craig.

#16
So, I recieved my shiny Sipura (Linksys) 3000 today and I had a go with the config!!! :shock:

Rabbit in front of the headlights here!!! :shock:

All I want it to do is detect an incoming call on PSTN, ring my voipfone account (main number will be fine) and when a voipfone extension (or answering service) takes the call, then (and only then) answer the ringing PSTN line and connect the two.

Anyone been here, done this?? Prepared to spill the beans?

Voipfone guys - any pointers??

This box seems to be able to configure SOOOO much!!


Thanks,

Craig.

#17
Okay, I've got what i need working! :D

Now I have a minor issuette to resolve. Every now and then the SPA 3000 loses registration. When this happens the box seems to ignore the reregister every 60 seconds as set in the config and shows times of 700+ secs before next attempt!

I can't find out where this has been set and it's driving me mad. The line it's handling is used as a daily business line so I really need it to behave a bit better.

Any pointers would be good.

Thanks

Craig.

#18
After much playing with the sipura devices I have found that if you change the "Reg Retry Long Intvl" to 120 it will re-register after 2 minutes rather than hanging on for so long. It seems that if ADSL or a server error occurs it causes the sipura to hang on before retrying, ignoring the standard registration interval.
Hope this solves your problem
John

solution -- Sipura 3000

#20
I just managed to solve this

Please see this thread : http://voxilla.com/PNphpBB2-viewtopic-t-8587.html

I have two outstanding issues with this solution though, that hopefully VoipFone Tech Support can help with .....

1. I would like to adjust the time it takes for the voipfone voicemail box to answer
2. I do not get the originating caller's Caller ID in the email from VoipFone.

Issue 2 is possibly a problem caused by BT my lovely PTSN provider.

Issue 1 though is a nuisance .....

Currently, when an incoming PTSN call is switched to Line 2 (voip line on the SPA) it rings for 30 seconds before the VM answers.

What I would like to happen is this:
1. PTSN call comes in, it rings through to Line 2 (voip). After 20 seconds, if there is no answer, the SPA answers the call and routes it to VoipFone where the Voicemail answers immediately.

2. If a normal Voip call comes in, it rings Line 1, after 20 seconds VM picks up.

I can set all of that up on the SPA, except, I think, the answer delay.

What I currently have to do is this :
1. PTSN call comes in the SPA answers the call immediately and rings Line 2. After 30 secs voicemail picks up.

The effect of this is that the caller starts paying immediately, and pays to hear 30 seconds of ring tone before they can leave a message.

Could VoipFone's VM be configured to answer myUsderId --> myUsderId calls (routing PTSN to VOIP) immediately, while answering OtherID --> myUserId calls in say 20 secs?

Thanks for any suggestions.

#22
I agree with Jaheli on this one.... The grandstream 488 is a waste of time, it sits and hangs onto the pstn line for ages before clearing.... the spa 3000 is a much better option (with a bit of tweaking and patience) and you can config it to act as a hot line and automatically dial either the main voipfone account details or pbx number if you are using that option.. However saying all that, look at how many incoming pstn calls you have and think about using BT call divert as a possible solution until you have advised all your BT pstn callers of your new VOIP number... I know you have to pay for the facility and the diverted part of the call, but also think about how much the spa 3000 or grandstream 488 are... its all swings and roundabouts.

Dave.

#23
Please try RPA-2E1S1O from BroadTel. It works fine as a FXO gateway with my asterisk. Connection was cleanly dropped in a second after either end hung up. It also supports TONE selection including dial tone, ring, tone, ring back tone.... and so on. It even supports registration to 3 SIP proxies. I think it is the best buy that I can find in the market so far. You can find it from www.broad-tel.com.

David

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