Own router and digital voice methods

Switching to Zen FTTP next month with digital voice so trying to work out how to get all this plumbed in and set up in advance.

Currently on FTTC with an Asus AC86U (running Merlin) connecting via PPPoE to the openreach modem.
Several wireless repeaters in the house, separate guest networks and VLANs for IoT devices and so on.
Inbound forwarded services are things such as backup DNS, Backup MX, Wireguard and OpenVPNs and so on.

In short, i really cant afford to lose these or reconfigure absolutely everything on the FritzBox so want to use the existing router and infrastructure.

If i didn't want digital voice it would be a simple case of plugging the router in and configuring the PPPoE but the problem is digital voice (dect and analogue port) will be needed.

Im assuming there's no such thing as an adaptor to plug into the existing router to allow the functionality?

Assuming i need to use the FritzBox plugged into the ONT as a first hop then what's the easier way to preserve my current network whilst getting digital voice?

Can it be configured dumb with DHCP and everything turned else off and in the same subnet as my existing LAN so i just change the default gateway on the WAN interface of the existing router?

Would it be better or even work configured as a slave with wifi etc disabled plugged into the existing router?

In short what would be the easiest, least obtrusive way to keep my current LAN setup with FTTP and the digital voice?

Ive got the 7530 manual and going through it but it isnt overly helpful here.

Ideally, if i have to use the FritzBox in the setup, i want it to be as invisible and unobtrusive as possible.

Would "configure as an ip client" be the best option? ( https://en.avm.de/service/knowledge-base/dok/FRITZ-B... )

Im comfortable with the networking but not in how digital voice authenticates and works.


Thanks

You definitely need the Fritzbox in the chain. Zen don't supply the necessary SIP details and they will no doubt be encrypted.

The easiest way is to go Fritzbox > Router.
You would need to disable DHCP on the Fritzbox or put your router in a DMZ but the Fritzbox would still be doing the PPP.
It isn't ideal.

BT and Sky do similar and users have worked out how to put their respective hubs "behind" their own router by tricking the hubs in to thinking they are running the connection.
So it's ONT > Your router > BT/SKY Hub.

The BT hub guide is posted here

I can't find the Sky guide right now but it's done different but with similar outcome.

If you're smart enough and can use wireshark you may be able to work out how to do similar with the Fritzbox.

I use a Granstream HT802 ATA which just plugs into my switch. The telephone wiring also plugs into the same unit. It was all fairly painless, just had to plan and buy the correct additional extension wiring etc.
As for VOIP supplier, I used Sipgate, but as they no longer do a "free" offering, there are now cheaper elsewhere.

In reply to a post by Adduxi:

I use a Granstream HT802 ATA which just plugs into my switch. The telephone wiring also plugs into the same unit. It was all fairly painless, just had to plan and buy the correct additional extension wiring etc.
As for VOIP supplier, I used Sipgate, but as they no longer do a "free" offering, there are now cheaper elsewhere.

He's specifically asking about Zens own VOIP/Digital Voice service.
You can't use an ATA with this.

Sadly that's not an option with the Zen Digital Voice (and i dont want a SIP provider as cant afford to lose the landline number)

Cheers - was hoping i could not have to use the Fritzbox but looks like i can.

I guess the only way to do it is the messy option of Fritzbox with WiFi, DHCP and everything disabled and DMZ to the old router. Fritzbox having a fixed LAN ip in my current subnet.

That or the "IP Slave" the manual mentions. If that works it would be a better option.

Main problem i have is i've never seen the kit in my life and there's a high chance i'll have to talk someone through the setup of this on the phone as i wont be around. Tricky when you've never seen the equipment and going by PDFs as to how it should work.

In reply to a post by String:

Sadly that's not an option with the Zen Digital Voice (and i dont want a SIP provider as cant afford to lose the landline number)

You should be able to transfer your landline number to a SIP provider. However, that will also cease your broadband connection (so you may end up with penalty charges for early termination etc.). Worth looking into as you get towards the end of your broadband contract.

In reply to a post by j0hn83:

He's specifically asking about Zens own VOIP/Digital Voice service.
You can't use an ATA with this.

Ahh, I wasn't aware of that. Oh well, every day is a school day !

In reply to a post by j0hn83:

You definitely need the Fritzbox in the chain. Zen don't supply the necessary SIP details and they will no doubt be encrypted.

You can decode FritzBox data using stuff from https://github.com/PeterPawn/decoder

I've just tried copying the scripts https://github.com/PeterPawn/decoder/tree/master/scr... to a linux box and using the decode_export script on a configuration file exported from a FritzBox.

This worked successfully on an ex-Zen device repurposed for other use with providers (CIX broadband & AAISP VoIP). I can't see that a Zen-provisioned device would be different unless they can lock down what is included in the export, but it really needs someone with Zen Digital Voice to confirm this.

Zen have had a SIP platform for some considerable time and they used to provide connection details, but you were pretty much on your own getting working if there were any issues. Most of the old help / knowledgebase information appears to have gone but there is some in the refreshed help pages https://www.zen.co.uk/help-support/general-sip-settings

If their Digital Voice offering is the same platform underneath it may be that as with other providers they don't want to provide credentials to reduce support queries, etc., but you can configure a third party device having obtained the required information.

To set up Zen DV as just another SIP provider you can set the server address(es) to voip2.zen.co.uk, set your DV phone number as the user ID, and get the voip password here:
Zen customer portal / My Services / General / select Digital Voice / show VoIP password
This works for me on my Gigaset N300A DECT base, connected to the internet, although I usually use the normal Zen procedure, connecting to the Fritz box FON port with the fixed line connection from the N300A.

Edited by cjn (Thu 16-Feb-23 14:21:36)

@cjn

This is the first time, I've seen info on Zen's Digital voice, many just state they'll not give the information, I would be interested in why you stopped using it ?

einsateinagogo

can you not use a Gigaset N300A DECT base and settings as per here

https://www.zen.co.uk/help-support/general-sip-settings

which is what @cjn confirmed, and then you don't have to use the Fritzbox

Edited by einsteinagogo (Sun 19-Feb-23 19:32:22)

@cjn
This is the first time, I've seen info on Zen's Digital voice, many just state they'll not give the information, I would be interested in why you stopped using it ?
einsateinagogo
------------
I didn't stop using it; I started with the 'correct' route via the Fritz!Box and its FON port, but I had a couple of other SIP providers already programmed into the N300A so I just wanted to try Zen DV by the same method.

They both seem to work so it doesn't seem to matter which one to use. One point in favour of the 7530AX/FON is that it gives a simple way to block and divert incoming calls.

The connection info was already established for me in the Fritz!Box, where I could read the server address in plain text, and I found the password by idly browsing my portal.

Edited by cjn (Sun 19-Feb-23 22:09:56)

can you not use a Gigaset N300A DECT base and settings as per here
https://www.zen.co.uk/help-support/general-sip-settings
which is what @cjn confirmed, and then you don't have to use the Fritzbox
Edited by einsteinagogo (Sun 19-Feb-23 19:32:22)
--------
Yes, but this doesn't supply the PW, or what I believe is the correct server address. This info is only found by 'investigation'.

Edited by cjn (Sun 19-Feb-23 22:10:55)

Thanks for the pointers, cjn.

With your help I've managed to get asterisk/pjsip to talk to Digital Voice. For the record my settings are:

[zen]
type = wizard
sends_auth = yes
sends_registrations = yes
remote_hosts = voip2.zen.co.uk
outbound_auth/username = <zen phone number>
outbound_auth/password = <passwd>
endpoint/allow = !all,ulaw,alaw
endpoint/context = default
identify/match = voip.zen.co.uk

Make sure you've opened the SIP and RTP ports as per the zen general sip settings page (incoming SIP is from voip.zen.co.uk, NOT voip2).

Also make sure you've unset all CALLERID parts except for the CALLERID(number), which must be your zen phone number.

Hope this helps someone else.

Tim

Anybody knows what other makes of the phones could possibly work?
Every SIP capable phone?
ie Yealink W76P

sebus

Edited by sebus (Fri 24-Mar-23 07:57:29)

I have W70B with W56H
The unit shows as Registered with Zen using the info from support @ AA

https://support.aa.net.uk/VoIP_Phones_-_Yealink_DECT

Had to register WITHOUT +44 (just the number)

But cannot make/receive any calls

Outgoing phonecalls "Global error" on W56H
Incoming gives an error to the dialing user: "Phone is currently unavailable"

So no idea if something is missing from config, or the line is not yet fully configured on Zen end.

I just find the Zen support being total idiots, so hardly can get a reply from them, especially that they only "support" only own router - that I do not care about)

sebus

Edited by sebus (Sat 08-Apr-23 14:13:22)

I am getting 603 Declined while trying to make a call

No idea why N300A works and Yealink does not

Anybody any idea?

Seb

What CODEC are you using on the yealink? AFAIK AA only support g.711 alaw

Zen also only support g.711Zen SIP settings

Edited by wiggsc00 (Fri 14-Apr-23 12:25:54)

Yes, using PCMA (that what is called in Yealink), I did check the document before

Seb

Hmmm. Have you got caller id populated in the yealink. Might be required. If blank it may well cause this too. Make sure label, display, register name are not blank

I have an n300 and never experienced this issue with aaisp. Not got a yealink although it has Poe which means I could relocate my dect hub, I'm too attached to my n300 which I've had 12 years !

AA would not be a problem (problem is only ZEN)
I have all that is required filled, and the unit is Registered (if I put wrong info it shows Failed registration, so I know the very few details are correct)

I really cannot get from this thread if it is only N300A that somehow works with Zen)

Seb

As per my post above - you need to set Caller ID number to your Zen phone number to call out.

Don't set any other Caller ID details (e.g. Name).

Incoming calls need your firewall opened on 5060 from voip.zen.co.uk (not voip2).

As per link above to AA support page for Yealink

There are ONLY fields for

Register Name Your phone number, +44...
Username Your phone number, +44...

I could not use +44 because registration would fail
Had to use just the number 017…… in both

There is no CallerID field anywhere

I will worry about incoming calls when I can make an outgoing one (and not failing with 603 error)

Seb

In reply to a post by wiggsc00:

What CODEC are you using on the yealink? AFAIK AA only support g.711 alaw

For the record, AA support G.722 just fine - at least AFAIK its working as I'm not sure how to check which is being used in a call but calls seem just as clear as POTS.

Actually they confirmed on this forum that they don't

https://forums.thinkbroadband.com/aaisp/4732262-hd-v...

You may think you're using g.722, but you're not, they only support g.711

"calls seem just as clear as POTS."
That'll be g.711 then, which is the long-standing standard codec used to digitise analogue POTS calls for onward transmission

Yes, don't use international format for your Zen number, just use 01nnnnnnnnn format.

I'm guessing from the AA screenshot that "Display Name" is your "Caller ID Name", so make sure it's blanked out.

Is there anything in the Advanced / Number Assignment / Handset Name tabs that looks like caller id settings?

In reply to a post by Preset2169:

make sure it's blanked out.

And if blanked out doesn't work, try putting just your number in (01nnn format)

EXACTLY what I have (as explained above)

In reply to a post by wiggsc00:

Actually they confirmed on this forum that they don't

https://forums.thinkbroadband.com/aaisp/4732262-hd-v...

You may think you're using g.722, but you're not, they only support g.711

Interesting, as I specifically asked customer support and they gave a different answer but obviously I trust the one you linked more.

Interesting that 711 is still better than any mobile call I've ever made.

Edited by alexatkin (Mon 24-Apr-23 16:16:34)

In reply to a post by alexatkin:

In reply to a post by wiggsc00:

Actually they confirmed on this forum that they don't

https://forums.thinkbroadband.com/aaisp/4732262-hd-v...

You may think you're using g.722, but you're not, they only support g.711

Interesting, as I specifically asked customer support and they gave a different answer but obviously I trust the one you linked more.

Interesting that 711 is still better than any mobile call I've ever made.

Try setting your VoIP adapter to g.722 only. All outbound calls will fail. I know as I tried when I migrated and only g 711 would connect.

Tried again, and indeed, “Display Name” HAD TO be empty
Once re-registered, it works fine

A bit late to the party, I was also moved to Digital Voice with Zen, and using a Synology RT2600ac. We only receive calls on the landline, and I thought I'd repurpose an old smartphone with no SIM and use an Android app.

I first tried MicroSIP on the PC to begin with, and I got it to work. Then, however, of the two Android Apps I tried—MizuDROID and Zoiper—both can register and make calls, but neither can receive calls, and the SIP settings are the same between the PC app and the Android Apps. The PC is on wired, phone on WiFi, but the same IP network, and no router setting that is different for the wired and wireless clients. Any thoughts?

Edit: I cracked it. For those willing to use an app instead of a terminal, the key point is setting the local signalling port to 5060, not just the SIP server port. I figured this one out after using WireShark to trace the calls through MicroSIP and Zoiper—which I also installed on the PC—and saw the requests coming in on that specific port. Zoiper Free version does not allow to specify the local signalling port, either Android or PC, but MizuDROID does. Chicken dinner!

Edited by LoneStranger (Fri 26-May-23 00:55:27)

I thought I would post my lessons learnt as I have just spent bank holiday Monday getting my digital voice to work, mostly I wish I had found this thread earlier. I actually ended up trying out both methods when using an alternative router. (I use a vigor 130 in bridge mode and openbsd (PF) in proxmox in HA mode for my firewall)

1) Connect your Fritzbox to your phone line as the router and just make sure digital voice works. (They appear to have pushed an update to the Frizbox with the voip info in it, search to earily and your find nothing)

2) Export the config from the Frizbox (I used the password "a") then put your old router/firewall setup back

3) So what are my VOIP settings?
Registrar: voip2.zen.co.uk
Proxy: voip2.zen.co.uk
Username: <Your phone number>
Password: See below

4) So whats my password? If you use the Fritzbox you don't need (and I don't mean as a router see later)
I checked my Zen page as suggested by other people in this thread and sadly the info was not there.

So to get it I used the FritzBox Decoder As suggested by tdw42 it compliled out of the box on my linux machine. With the exported config I decoded it with

./decoder decode_export a < ../FRITZ.Box_7530_164.07.50i_29.05.23_1444.export
and searched for passwd in the file there is a voip section.

5) Easy method, you don't to even to part 3 or 4! On your Fritzbox go to wizards -> Configuring the Internet Connection
Select existing connection over lan
when it has reconfigured plug in LAN1 to you existing network.

Logon to the Fritzbox again in Home Network select the network settings and select IP Client-LAN configure a manual ip address or setup your DHCP server to give it a know IP. Skip to 7

6) Other FXS device I had also bought a Grand Stream HA612 for my Digital Voice change over, not knowing how useful the Fritzbox might be. IT Truely horrible interface! Set it up with a static IP or give it a known one via DHCP. Use the details for 3 and 4, tell it is using NAT and tell it your external IP address.

7) Firewall, I could have setup a sip application gateway on my firewall but that seemed complicated. So instead I redirected traffic from voip2.zen.co.uk (212.23.7.235) port 5060 and voice.zen.co.uk (212.23.7.228) port 5060 to the IP address of my FXS gateway/Fritzbox. That way I will not be bothered by people trying to direct call me via voip, I only wanted to keep the phone number as my Dad wanted to keep calling me on it.

8) Outstanding Issues....

Frittzbox client mode - None
GA812 outgoing caller id does not work. Meh not making calls from that line anyway but still only 9/10

9) Bootnote
If I get FTTP I will get a new install (with no phone) as suggested, then port my VDSL to sipgate which will kill my broadband. I didn't know that and thanks for the great advice.

Just to add another option for anyone else checking this thread.

Once you have moved to Zen Digital Voice, they are (currently) quite happy to let you move your number to another provider without impacting your FTTC/FTTP supply/contract.

In reply to a post by oolon:

I checked my Zen page as suggested by other people in this thread and sadly the info was not there.

It's only shown in the "old" portal - My services > General > View Technical Details, then select your Digital Voice order item.

Whole point of Digital Voice is NOT to have voice tied to the broadband
So moving the DV to anybody else will NOT affect existing broadband (nothing will be cancelled)

From a technical standpoint that is correct. It depends on the terms of your contract, some providers make it difficult to unbundle packaged services.

What is the minimum contract term for Zen DV contract if a customer migrates from existing Zen PSTN to Zen DV? 1, 12, 18 months? Is there any way to exit early?

Are Zen only offering Zen DV with 1000 minutes for £6 / month or have they decided to offer a PAYG package to anyone who wouldn't see the benefit of the minutes package?

Will a customer currently contracted to Zen FTTC with a no price increase for life guarantee, who first migrates from Zen PSTN to Zen DV, and then possibly migrates to another VoIP provide in the future, definitely not lose the price guarantee?

Late to the party on this!

You can use the Zen digital voice login on other devices but it will only work if you are behind the static IP address on the Fritz!box

Hi, I'm late to this party, but just trying to wrap my head around things.

Why does the static IP have to be on the Fritzbox ? My router will happily get the same static IP from Zen as the Fritzbox does, and I can feed it with SIP details if I have them, including any NAT traversal stuff.

No need for Fritzbox (that is the purpose of this thread!

sebus

Sorry, I was just preparing an updated summary of my current understanding, but you beat me to it smile. Here's what I currently think the situation is.

If I want to use my own router ( and I do, because I'm using features the FritzBox doesn't support ), and I don't want to port my number to some other provider ( and I don't, because that would loose me the current broadband deal I have, I don't want to pay more, and I can't find a standalone SIP based VoIP deal that I like ), I have three options.

1. Configure my router ( which supports SIP and has phone sockets ) with SIP parameters that allow me to plug my phone into it.

This is the obvious solution but, rather obnoxiously, not supported by Zen even though they have all the info required. However the info in this thread looks like it might allow me to achieve it. I think I can see how to make this work, provided I can get hold of the password. Thanks to all who have helped !

2. Put the FritzBox on the WAN side of my router, and have it make the PPPoE connection.

The public IP then belongs to the FritzBox, and I assume I could just plug my phone in to it. At least, that's what the ( very minimal ) Zen 'documentation' implies. I would also need to plug my existing router into it.

But it's not at all clear to me that there's an easy way to do this which will continue to allow my router to be the gateway, DHCP server, firewall and QOS manager for all my VLAN's and subnets on the LAN side. I don't believe I can just put my router into bridge mode and preserve all this functionality.

3. Put the FritzBox on the LAN side of my router.

This is clearly a possibility as far as AVM/Fritz!OS is concerned - see

https://en.avm.de/service/knowledge-base/dok/FRITZ-B...
and
https://en.avm.de/service/knowledge-base/dok/FRITZ-B...
https://en.avm.de/service/knowledge-base/dok/FRITZ-B...

The 7530 appears to support this 'IP Client' mode. I'm assuming that I need to punch appropriate holes in my main router's firewall, per the info provided here :

https://www.zen.co.uk/help-support/general-sip-settings

There's a post from @oolon somewhere high above that describes what I think is this method.

I'm going to shoot for option 3 here, for the simple reason that it will make my phone cable routing easier if it works smile. Thanks to all.

In reply to a post by EvD:

Are Zen only offering Zen DV with 1000 minutes for £6 / month or have they decided to offer a PAYG package to anyone who wouldn't see the benefit of the minutes package?

TL:DR There is a Zen DV PAYG

I received an email this am from Zen saying that I had been migrated to the new system.
I had not responded to the offer of DV as we very rarely use the landline (for calls). In fact for many years we had more nuisance calls & wrong numbers than genuine calls. We would be happy to ditch it except for the price guarantee & we hope to move later this year so would not want a new contract. Anyway my account in Zen states that I am on DV PAYG. I was paying £34.99 (inc VAT) for Unlimited Fibre 2 FTTC 80Mb/s including phone line with PAYG calls and am on the price guarantee. I received a small refund relating to the change of the last 6 days I had paid up front for and there is an invoice on the system but I have not received yet raised for a slightly smaller amount for charges for 5 days suggesting a free day on the day of potential disruption of transfer as the upshot of it is as expected the equivalent appears to be the same overall cost of same £34.99 (inc VAT) going forward. This means that given the UL Fibre 2 is currently £34 I will be getting the DV PAYG for 99p/month but probably only because of the price guarantee. I suspect if I ditched the DV PAYG I might affect my price guarantee. Also I could not see any reference to DV PAYG on the Zen site. Hope this clarifies things a bit at least.

Are you so attached to whatever phone you have now, that you try hard to keep using it?
Why not just buy a proper IP phone (like Yealink) and just plug it to your network?

No need to have router with SIP port, because that is just network device

It will just work

sebus

Edited by sebus (Fri 30-Jun-23 13:04:50)

In reply to a post by sebus:

Tried again, and indeed, “Display Name” HAD TO be empty
Once re-registered, it works fine

In fact ”Display Name” CAN have same phone number, but not words

Also for Yealink on Zen (because they use different server for Register and Incoming calls) one needs to DISABLE

Accept SIP Trust Server Only

sebus

I presume Zen warned you about the impending migration to Zen DV in order for you to expect downtime and to know that you would likely need to unplug your phone and plug it into the FXS port on the router. Zen don't appear to be advertising a Zen DV PAYG package as of yet. I cannot find anything. Can anyone ask Zen to move them to it or do we have to wait to be contacted?

I got an email at the 4th of May saying the change to Digital Voice (but not any particular package) would be made at some point and that I would not be notified when but they would say when it had been done. They also said I should get in contact if I was considered vulnerable, presumably because of the inoperability of the phone if there was a power cut etc. There was a link to a video saying what to do re the router. The next information I had regarding it was yesterday to say it was complete. I was still getting a dial tone with the phone plugged in the microfilter but did not test call it. I am aware that under some circumstances you can still get a dial tone with no number on the line. I plugged the phone to the back of the router and tested an incoming call which worked. I have had the general marketing emails in between suggesting I get a digital voice package. To confirm my account definitely says I am on DV PAYG.

Thats what I did.

Bought a Grandstream phone, plugged it into my router.

Signed up with Sipgate, ported my landline number across, cancelled the original number they gave me and job done.

And because I have installed Zoiper on my mobile, my landline calls also ring on my mobile.

And how does that relate to Zen (or you were with them and left - for whatever reason?)

sebus

In reply to a post by sebus:

And how does that relate to Zen (or you were with them and left - for whatever reason?)

sebus

Im still with Zen The ISP doesnt matter. I have a VOIP product with a digitial handset on my landline number that costs me nothing a month as I signed up over a year ago.

It costs nothing (to rent I assume)
Because outgoing calls cost something (no matter when you subscribed?)

Irrespective of VoIP provider (or any voice provider for that matter), you’ll pay for outbound calls either via the monthly sub charge for the account (if there is one), an add on calls package or ad hoc per call.

Whichever is the ‘best’ option is very much dependent on one’s personal situation.

Did you ever get this sorted? A lot of people here making claims like 'you need the fritzbox' are talking rubbish. I've been running Zen DV without a fritzbox since day 1 - Zen even provide their SIP server settings for exactly this purpose - they just won't hold your hand through setting it up.

I have been using a Grandstream ATA with Zen's Digital Voice for a while now (very complex but not impossible) and just swapped over to a Yealink VOIP Base + DECT phone combo which was a lot easier to set up.

All the chuff in this thread is nonsense. You can easily set up a third party phone to work with Zen's SIP service - Zen provide your password in the 'classic' account area so no need to go wiresharking anything.

Beyond that settings are very simple, the Yealink doesn't need config beyond user/pass for example.

Much like you I have the ONT connected by PPPoE to my home Router directly and don't use the Fritzbox. I also have a lot of internal networking and didn't want to add the Zen router as a middle useless step. The VOIP/SIP and DECT phone setup to add it on the 'other' side of the router wasn't too hard.

Let me know if you want some tips - obviously my familiarity is with the Yealink system but I think the Gigaset phones are very similar.

Edited by aod1985 (Sat 05-Aug-23 12:12:01)

Has anyone got Zen DV working with Acrobits Softphone or Groundwire? Have got it working on my Mac with Telephone app, but cannot get it working on IOS despite exact same settings.

Boy have I been banging my head against this one. I thought to myself "Digital Voice is just VOIP, right, how hard can it be"... Given there's a massive mandatory national switch you'd think there'd be tonnes of info online about it too, but no, everything's top secret, hush, hush, need to know.

After much back and forth with Zen I managed to get them to agree I *could* continue to use my own equipment, but that I really really shouldn't because then they can't support you. Ok great, that's wonderful, please can I have the connection details now - "oh, we can't do that" - sigh. I just need to know what they are so I can sort it myself, but what little information there is online is contradictory at best. And no Zen, I do not want to use your silly little Fritz!box that is worse in every single way then the rest of my kit, especially when I'd have to upend my entire network to do so. (I did end up having to plug it in for a bit, to make sure Digital Voice was working).

Anyway, what I've discovered is:
- it turns out the SIP password is in the Zen portal, but not in the new Portal. You can only find it if you click through "Return to old Portal" (the old portal being something they're getting rid of pretty soon by all accounts).
- The SIP settings here: https://www.zen.co.uk/help-support/general-sip-setti... do not work, at least not for me. I can make outgoing calls (or to be more precise I can dial out, I've not actually tried answering a call) but can't receive incoming calls.

I'm running OPNsense on my own hardware (in front of a BTOpenreach modem) with the ports open they specify and have tried setting up with Linphone and microSIP to no avail. That's both with the Zen specified settings and modified with a combination with the info AAISP publish: https://support.aa.net.uk/Category:VoIP_Phones.

I'm a bit stuck now, sorely tempted just to flip the number to AAISP just because they actually support you. But that's an extra monthly cost we could really do without.

Edited by drmegalomaniac (Mon 04-Sep-23 21:44:01)

In reply to a post by drmegalomaniac:

with the ports open they specify

With SIP you shouldn't need to open any ports.

In reply to a post by PCJM40:

In reply to a post by drmegalomaniac:

with the ports open they specify

With SIP you shouldn't need to open any ports.

It depends the details of how the SIP client registers with the server.

The Fritz!box when used as the ISP router is by its nature exposed on the incoming connection so it always has access to traffic that would be filtered by a default block inbound.

A SIP client often uses STUN/ICE/TURN and can ask the router to create (IPv4) port forwarding on demand for the signalling and media traffic flows but only if the corresponding service (e.g. UPnP/NAT-PMP/PCP) is enabled on the router.

A SIP client that registers with the provider's server as a NAT proxy (IPv4) can maintain a connection that supports incoming calls though it may need a keepalive to hold the connection open beyond default timeouts.

There are other variations of SIP-like protocols which provide the traversal and/or keepalive features.
Many providers will assume the home router is going to be a problem to be worked around including directions to disable SIP ALG features.

However in the original vanilla type of SIP the incoming calls would not reach a client behind NAT without the co-operation of the router/firewall, or a separate SIP gateway (such as a session border controller in an enterprise context).

Edited by prlzx (Mon 04-Sep-23 23:07:07)

In reply to a post by prlzx:

A SIP client that registers with the provider's server as a NAT proxy (IPv4) can maintain a connection that supports incoming calls though it may need a keepalive to hold the connection open beyond default timeouts.

There are other variations of SIP-like protocols which provide the traversal and/or keepalive features.

Thank you for the detailed explanation, I was only thinking of enabling NAT Traversal and Keep alive pings on the client when I said there was no need for ports opening on the router. I haven't come across the other SIP types that require router config changes so thanks for that extra info.

No probs - it was somewhat worse in the early days particularly if you had more than one extension on your home or office network and one-way voice during a call, calls longer than 30 min or 1 hour going silent, or outgoing calls only were all common symptoms when deploying.

It's still something that needs troubleshooting to a lesser extent even today.

I recently received an email from zen about digital voice and I noticed that on "https://www.zen.co.uk/voice-hub-details" it says you can use your own hardware, you need to phone zen support on the day activation for your credentials but they wont give you support on setting up your own hardware.

So I ordered digital voice and on the day activation I phoned support and sure enough they was happy to give me the details. I'm using a Gigaset n300a with the following settings -

http://www.mustyweb.co.uk/voip/n300a.png


I'm using pfsense with the following firewall / nat rules
http://www.mustyweb.co.uk/voip/nat.png
http://www.mustyweb.co.uk/voip/wan.png

Incoming and outgoing calls work great

I hope this helps

Oh blimey, credit to Zen then. They've actually changed the process quite a bit. They said they would after I complained, and they did say I wasn't the only one. Kudos.

That all looks very interesting, thanks, I'll have a look through later

What settings did you use on Mizu- whatever I try it doesn’t work for incoming calls? Does it have to be on your local Zen connection or can you roam on to say 4g?

A few things I learnt from the Zen digital voice migration:

1) Zen use what I call traditional SIP - i.e. connecting to a fixed port on a fixed IP address (whereas many other VoIP providers use the SIP registration initiated by the client to give the IP address and port). Therefore it only works 'on net', you cannot use from other connections.

2) For those Zen users with a static IP address range, rather than a single address, it seems that by default they assume you will be using the highest IP address in the usable range (but seemingly tech support can change it to one of the others). This meant initially I couldn't receive inbound calls, as they were trying to connect to a different IP address than I was using.

3) They use voip2.zen.co.uk for outbound calls, but inbound calls will come from voip.zen.co.uk

4) You need a static port mapping from UDP 5060 on the chosen IP address to your VoIP device (nb: it can only seemingly be one device, not multiple)

5) Firewall open UDP inbound from 62.3.88.0 to 62.3.88.31 and 212.23.7.228 for all UDP ports > 1024. In my case I've left ALG for SIP disabled.

5) Username is your phone number in domestic format (i.e. starting with 0 not +44) and password can be found in the old customer portal

6) Most other settings are generic defaults.

7) Codecs:
PCMU / G.711 a-law - Sample rate 20ms
PCMA / G.711 -law - Sample rate 20ms

8) Other settings:
SIP Transport: UDP
NAT traversal: No
DTMF: RFC2833
SLIC Setting: UK
Caller ID Scheme: SIN227 / BT

Edited by rikur (Fri 29-Sep-23 15:56:18)

I gave up on the apps... Anyone have any luck getting a Yealink W70B (or similar) working with Zen and willing to share their config? Mine just will not register for some reason.

Edited by drmegalomaniac (Sat 30-Sep-23 15:34:09)

So happy to have found this thread.

In reply to a post by rikur:

A few things I learnt from the Zen digital voice migration:

1) Zen use what I call traditional SIP - i.e. connecting to a fixed port on a fixed IP address (whereas many other VoIP providers use the SIP registration initiated by the client to give the IP address and port). Therefore it only works 'on net', you cannot use from other connections.

2) For those Zen users with a static IP address range, rather than a single address, it seems that by default they assume you will be using the highest IP address in the usable range (but seemingly tech support can change it to one of the others). This meant initially I couldn't receive inbound calls, as they were trying to connect to a different IP address than I was using.

Good to know. My initial plan was to connect the fritzbox to the LAN router and port forward from there. The LAN router doesn't run the pppoe connection however. So that wouldn't have worked unless support pointed the voip to router ip minus 1 where the LAN router is and I do all the port forwarding you mention later in your post from there.

Why is this so opaque.

Once the switchover happens, if you don't like or can't use the service, can you immediately go elsewhere without penalty? Is there a test service one can use to make sure things work before switching over?

In reply to a post by bsd99:

My initial plan was to connect the fritzbox to the LAN router and port forward from there. The LAN router doesn't run the pppoe connection however. So that wouldn't have worked unless support pointed the voip to router ip minus 1 where the LAN router is and I do all the port forwarding you mention later in your post from there.

So the FritzBox is terminating the PPPoE connection and you are having double NAT to the LAN? IIRC the FritzBox doesn't support routed public subnets. Not that you would have one for a new connection as Zen stopped handing out free /29s years ago, unless you are paying extra on a business service.

Once the switchover happens, if you don't like or can't use the service, can you immediately go elsewhere without penalty? Is there a test service one can use to make sure things work before switching over?

You have 14 days from ordering to cancel (usual distance selling regulations), if it takes longer than this you have no rights to cancel unless the service does not meet the description when ordered. As they state that the service is provided by the FritzBox, and other setups may work but are not supported you can't use that as an excuse for cancelling. No.

In reply to a post by tdw42:

So the FritzBox is terminating the PPPoE connection and you are having double NAT to the LAN? IIRC the FritzBox doesn't support routed public subnets. Not that you would have one for a new connection as Zen stopped handing out free /29s years ago, unless you are paying extra on a business service.

No, I have another router controlling the pppoe & handing out the rest of the /29 and ipv6 /48. I've had the account for a while so yeah got the /29 for free.

You have 14 days from ordering to cancel (usual distance selling regulations), if it takes longer than this you have no rights to cancel unless the service does not meet the description when ordered. As they state that the service is provided by the FritzBox, and other setups may work but are not supported you can't use that as an excuse for cancelling. No.

that sucks. More straightforward to go with a voip provider that doesn't tie to their own equipment.

What device are you using - followed what you had set using a YEALINK T33 and no calls received, outbound works fine.

Nick

Any chance you can share your Yealink config?

Now got this working on a Yealink T33 - as reported by others only a single device seems to receive calls - multiple cannot. For those wondering - removing the 'proxy' is what has solved this for me despite what the documentation says!

Hi all. VOIP noob here - pls be gentle.

Also very pleased I found this thread as I've been banging my head against the wall for a couple of days on this.
I recently bought a Yealink T46U from Amazon, supposedly 'new'. I was under the niaive impression that I could just plug a SIP phone into an eithernet port on my router and use it woth Zen DV. But no.....

Plugged it in, and went to setup the phone via its web interface to find out that the default uname/pword didn't work. Turned out the phone is (quite likely) a refurb as it was looking up and overriding uname and pword from Yealink's RPS server. After a bit of back and forth with Yealink support, got the MAC address deleted from their RPS so now I can configure it.

I called Zen and got my uname (landline number, as discussed above) and pword. Note: the pword is no longer available on the portal, both old and new versions.

So now I have access to the settings page on the phone. However the phone is still not registering.
The only params I have entered are the uname, pword, and the server name (voip.zen.co.uk). All other params are default (port is already set to 5060). I've left the label, display, and register name blank, as advised earlier in the thread)

I'm almost at the stage where I'm just going to repackage the phone and return it to Amazon as defective (as it's likely a refurb), but this is my last ditch attempt to try and configure it. As others have suggested, i'm not keen on moving to a new SIP provider as I don't want the expense and hassle.

Can anyone tell me whether there are any other settings needed to be changed for the phone to register?

Hi

Have you opened the firewall ports for voip?

Nick

Hi - I haven't - the SIP phone has a ping function and I can ping the zen voip server from the phone. I'm no tech expert but I interpreted that to mean there's no firewall issue getting in the way of the connection.
Shout if that's not correct!
If not - the firewall settings should be in the Frtizbox. I did have a quick look but there's no obvious place in the Fritzbox config screens to turn a firewall off.

I've spoken to zen a couple of times; I want them to tell me what their server logs are saying about why they're rejecting my registration request.
If I can't do this with my Zen services, the next step will be to request an unbundling of my VOIP and my broadband (which I am happy with) as there are plenty of other SIP providers who can help here.

Hi,

If you look at the Zen VOIP SIP setup guide it tells you what ports need to be open, worth checking you have the inbound ports open too, not just outbound. Not sure where that is in Fritz box as I don’t use mine.

Dear All,

My Zen Broadband & Digital Voice service commenced 2 Jun 23 and I was able to get Digital Voice working with the supplied FRITZ!Box 7530. Due to various limitations is FRITZ!OS, I migrated to OpenWRT and bought a Grandstream HT801 to support SIP Digital Voice.

I wasn't able to get the HT801 working, even after lengthy interaction with Zen Customer Support (excellent staff, let down by counter-productive management attitude). So I bought a Siemens Gigaset GO Box 100 (same hardware as N300A). With quite some fiddling, I was able to get this to work.

A month later, I tried the HT801 again and it worked. Have asked Customer Support but still don't know why. The only feature that I haven't yet got working on the HT801 is Caller Line ID.

I have also tried the excellent MizuDroid SIP app on my Google Pixel Android phone. It works at home, but not out of the house, backing up @rikur in his "traditional SIP" hypothesis.

Happy to share my detialed config notes with anyone. I have invested quite some time in this.

Found this thread. I have the Frit!Box working behind my router but cannot get a Cisco ATA192 to work. It's soooo complicated - so many config options, really regretting my decision to buy. Anyway at least it has decent diagnostics. I have tried a bunch of things and got it to register a few times but no longer and no dial tone, the outbound log shows this:


Feb 18 19:27:10 CISCO-PHONE local1.debug vsock: REGISTER sip:voip2.zen.co.uk SIP/2.0 Via: SIP/2.0/UDP 192.168.1.134:5060;branch=z9hG4bK-4eb0e6f3 From: \"014********\" ;tag=e212926f3caab3a3o0 To: \"014********\" Call-ID: [email protected] CSeq: 1688 REGISTER Max-Forwards: 70 Contact: \"014********\" ;expires=3600 User-Agent: 014******** P-Station-Name: ;mac=14a2a0aabf6c; display=\"\"; sn=FCH25492AHD Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces, sdp-anat

Really my question is what should I be aiming for with this string? Anyone got the registration URL for the Fritz!Box? Just about to give up. Anyone got an ATA192 working with Zen?

Have you asked Zen?

I've asked Zen for VOIP details - and I have what they have given me, but those don't quite match what is needed from the ATA 192

In case anyone else ends up here, I have solved this for the Cisco ATA192. What I did:

- Install the latest firmware 11-2-4MPP0001-115 Aug 9 2023 - not sure if this was necessary, but here for completeness
- Reset all parameters to default including voice
- Then made *only* the following changes:

Network Setup -> Bridge
IPv4 Settings -> Static IP etc
Quick Setup -> Line1 -> Proxy -> voip2.zen.co.uk
Quick Setup -> Line1 -> User ID -> <uk landline number incl area code>
Quick Setup -> Line1 -> Password -> <password from zen>
Voice -> Provisioning -> Provision enable -> no
Voice -> SIP -> NAT -> Ext IP -> Fixed IP from Zen
Voice -> SIP -> RTP -> Port Min -> 7078
Voice -> SIP -> RTP -> Port Max -> 7097
Voice -> SIP -> SIP -> TCP Port Max -> 5060

I then punched port 5060 inbound in my router to go to the fixed IP above
It then all worked!

Hi all - complete noob here, but found this thread when trying to resolve a frankly stupid situation with our provider.

We've just gone FTTP with IDNet and opted to go with their UBoss Basic service. Rather than go with their Yealink option I ordered myself a Gigaset N300AIP. Landline number was ported and then they requested the mac address of the device. They haven't been able to get it setup and now are effectively beyond the point of helping as it's not a supported device.

They've sent through a connection guide from Zen DV, who I guess the service is actually with. All this tells me is the following:

Domain Proxy address: sbc.insmartcloud.com
Registrar Server Address: as above
Proxy Server Port: 5060

I've got the Codecs as well.

But crucially no authentication info and they've said that they're unable to provide that as "the device is setup via the MAC address".

When I found this thread last night I was hopeful that with the info in here I'd be able to sort it myself, but without a password I'm stuffed I guess?

Not sure what to do right now. They've offered to send out an adapter for free (I assume they mean ATA) but our old phones are a bit knackered and I've got the new Gigaset base station and DECT phone sat here that I can see being a pain to return.

In reply to a post by Swervin_:

Hi all - complete noob here, but found this thread when trying to resolve a frankly stupid situation with our provider.

We've just gone FTTP with IDNet and opted to go with their UBoss Basic service. Rather than go with their Yealink option I ordered myself a Gigaset N300AIP. Landline number was ported and then they requested the mac address of the device. They haven't been able to get it setup and now are effectively beyond the point of helping as it's not a supported device.

They've sent through a connection guide from Zen DV, who I guess the service is actually with. All this tells me is the following:

Domain Proxy address: sbc.insmartcloud.com
Registrar Server Address: as above
Proxy Server Port: 5060

I've got the Codecs as well.

But crucially no authentication info and they've said that they're unable to provide that as "the device is setup via the MAC address".

When I found this thread last night I was hopeful that with the info in here I'd be able to sort it myself, but without a password I'm stuffed I guess?

Not sure what to do right now. They've offered to send out an adapter for free (I assume they mean ATA) but our old phones are a bit knackered and I've got the new Gigaset base station and DECT phone sat here that I can see being a pain to return.

The MAC address is a red herring I suspect. With the Yealinks they are set up to use Yealinks auto configuration, so the phone calls home to Yealink with the MAC address, Yealink will see that has been configured on their system by Zen, and then send the necessary setup files.

This is never going to happen with a different device regardless of what the MAC address is or isn't.

Basically you are not going to get anywhere now without having the credentials.

In reply to a post by E300:

The MAC address is a red herring I suspect. With the Yealinks they are set up to use Yealinks auto configuration, so the phone calls home to Yealink with the MAC address, Yealink will see that has been configured on their system by Zen, and then send the necessary setup files.

This is never going to happen with a different device regardless of what the MAC address is or isn't.

Basically you are not going to get anywhere now without having the credentials.

Thanks E300. Well, I phoned Zen in the end and they confirmed that I should be able to get the relevant details via our provider. Credit to the guy at Zen as it wasn't even the right depratment for them, but he was incredibly helpful. I've been sent a "username" and "password", and have since tried multiple combinations using both that info and the BYOD info they sent, and tried with the details listed above using the voip2 Zen address etc, but still to no avail.

So I've requested they confirm with Zen exactly what the provided username and password relates to and how that differs with what I've seen online of people using the telephone number as the username etc.

At the moment I'm not hopeful of a positive resolution and can see me looking around for another VOIP provider.

ETA: I'm also wondering if I need to change any firewall settings in my RT-AX59U router, but not sure what needs to be set.

Edited by Swervin_ (Tue 27-Feb-24 14:23:42)

In reply to a post by Swervin_:

ETA: I'm also wondering if I need to change any firewall settings in my RT-AX59U router, but not sure what needs to be set.

I have never needed to change a broadband router to allow VOIP through it. The N300 starts the connection by registering with one or more VOIP ISPs.

I have also not added any codecs.

For Sipgate I use the following settings:

Connection name or Number: BV Sipgate

Authentication name: 1234567
Authentication password: ABCDEDG
Username: 1234567
Display name: Sipgate


Hide advanced settings

DNS SRV Lookup Yes

Domain: Sipgate.co.uk
Proxy server address: Sipgate.co.uk
Proxy server port: (Blank)
Registration server: Sipgate.co.uk
Registration server port: (Blank)
Registration refresh time: 60 sec

STUN enabled: Yes
STUN server address: stun.sipgate.net
STUN server port: 3478
STUN refresh time: 240 sec
NAT refresh time: 20 sec
Outbound proxy mode: Automatic
Use DHCP Option 120 "SIP Server" Not selected
Outbound server address: (Blank)
Outbound proxy port: (Blank)
Select Network Protocol: Automatic

OK, I'm pretty much at the point of giving up with this.

I've disabled the SIP Passthrough on the router. All settings match those provided, albeit I don't have any info regarding whether STUN is enabled or not. I've gone with not on that basis.

Only one other thing I've noticed - when I paste in the SIP password and hit "set", when I go back into the settings the number of *** in the SIP password box is much shorter than the number of characters in the password I've entered. Not sure if this is just a quirk of the N300 or something is actually amiss.

Not used an N300 but in most circumstances would expect the number of *s to be the length of the password.

Do you use a password manager? I have found with some webpages (usually with 2-step authentication) if I just hit enter at the end of the password/addition question response it works but if I click the Next button the password mangler replaces what I typed with somethingh else before submitting (I suspect it tries to insert the same password as for the previous page).. In your case it might be submitting the N300 password rather than the SIP password you have entered.

In reply to a post by Swervin_:

Only one other thing I've noticed - when I paste in the SIP password and hit "set", when I go back into the settings the number of *** in the SIP password box is much shorter than the number of characters in the password I've entered. Not sure if this is just a quirk of the N300 or something is actually amiss.

This is commonplace in password entry, to stop anyone looking over your shoulder and guessing the password from the number of characters.

I have outgoing calls! Was provided with a couple more setup details by IDNet, which at first didn't seem to work. Came back an hour later and the N300 was showing the connection as registered.

However, if I try calling the number it comes up as invalid number. So there's work to be done yet on figuring out the incoming calls.

I'm guessing that this issue is now going to be firewall related, so will re-read a few of the posts above as I can see some have been tweaking firewall settings

Does anyone have any bright ideas as to what to start looking at next?

In reply to a post by Swervin_:

I'm guessing that this issue is now going to be firewall related, so will re-read a few of the posts above as I can see some have been tweaking firewall settings

Does anyone have any bright ideas as to what to start looking at next?

It's fully working finally. Turns out I just needed to make a proper call from it, rather than test dialling my mobile. Can now make and receive calls.

Post deleted by Spudgun

Post deleted by sebus

Edited by sebus (Fri 29-Mar-24 11:18:16)

Post deleted by sebus

I know this is an old thread, but if anyone is still interested, it is fully possible to use an own router with Zen over CityFibre -- provided your own router is a good one. My solution with a Ubiquiti EdgeRouter is described here..

https://community.ui.com/questions/EdgeRouter-as-pri...

As many people are struggling with using the Fritz!Box with Zen in IP Client mode behind another device doing NAT, after many years of absence, I have re-registered to explain what is going on for those that have failed to get it to work.

Zen's Digital Voice (VOIP) service, assumes that the customer will be connecting the provided Fritz!box router directly to DSL or FTTP. This would mean the Fritz!box would have the public IP address on its WAN port, making it easy for it to send and receive calls using the public IP address.

If, like me, you have your own firewall, (in my case OPNsense), or another router, instead of the Fritz!Box connected to the internet. Some additional setup is required for successful incoming calls. In my case I have the Fritz!Box connected to my OPNsense firewall to provide VOIP telephone service and act as a DECT base station. You don't necessarily have to use the Fritz!Box, you can use other VOIP devices or a SIP PBX like Astersk if you prefer.

The first thing, as has been mentioned a few times is, ports must be forwarded from Zen’s VOIP servers to IP address of the Fritz!box, using the information provided by Zen in their help document general-sip-settings While this will allow the incoming call to be initiated, for some people, depending on the type of NAT you have, this may be enough calls will drop out after approximately 30 seconds.

In addition to the inbound port forwarding rules, you will also need to make sure that you configure your outbound NAT, so that connections from the Fritz!Box have static ports. On PFSense & OPNsense this can be done as follows...

1. First go to the Firewall -> NAT setting menu, click on outbound. Set the NAT mode to ‘Hybrid outbound NAT rule generation’.

2. Add an outgoing rule for the IP address of the Fritz!Box...

Interface: WAN
TCP/IP Version: IPv4
Source address: Fritz!box IP address
Source port: UDP/*
Destination address: any
Destination port: any
Translation/target: Interface address
Static Port: yes

Save and apply this rule.

Why this second step is impotant

There are different types of NAT available, but most dynamically generate new port numbers on out going connections, i.e port address translation as well as address translation. While this works for many things, SIP is one of the few protocols where this is a problem. SIP expects the source port to be the same as the destination, in this example port 5060. When the outgoing port is changed from 5060 to a random port by your firewall's NAT, the connection will fail.

The failure is due to the call sett-up being a 3-way process...

1. The incoming call sends an INVITE to your public IP address on the SIP port 5060, this is forwarded to your Fritz!Box according to your port forwarding rule, the phone starts ringing!
2. The Fritz!Box sends an OK response so the call can be set-up, but this needs to have a source port of 5060, not a random port.
3. For the call to be successful, the server will need to send an ACK response to Fritz!Box, to acknowledge the OK sent. Herein lies the problem, the OK response will be sent on the same port that the OK was received. Using the common dynamic port mapping on NAT, this will be a random port. As there is no corresponding port forwarding rule matching the random port, to receive this connection on, the ACK response will never be received an the call dropped.

Setting a static port on the outgoing NAT, ensures that the outgoing port remains 5060, now when Zen tries to send the ACK to your public IP address, you have a port forwarding rule that forwards requests to port 5060 to your Fritz!box, the ACK is therefore received and the call stays up.

Hope this helps some people.

PS, my old forum username from many years ago that got deleted through non-use was 'Going_Digital'

Edited by Tandy (Sun 05-May-24 10:05:31)

Re: Own router and digital voice methods

Thanks very much - I've filed that for useful future reference because Zen aren't broadcasting and attitude of great helpfulness on the subject.

In reply to a post by rikur:

A few things I learnt from the Zen digital voice migration:

1) Zen use what I call traditional SIP - i.e. connecting to a fixed port on a fixed IP address (whereas many other VoIP providers use the SIP registration initiated by the client to give the IP address and port). Therefore it only works 'on net', you cannot use from other connections.

I realise this is an old post but it’s the only place I’ve seen this mentioned and wondered if anyone else can confirm that Zen Digital Voice does not function, for example, when the Zen SIP settings are entered into a soft phone app such as Groundwire as installed on a smartphone which is being used away from home.

Thanks.

I don't suppose you got your W70B working with Zen?

Many thanks!

Probably no help to the original poster, but I've just migrated to a Fibre connection with Digital Voice.
I also require to retain my own routing set-up so I've configure that to connect to the ONT to allow control of the PPpoE session.

The Fritzbox which is most definitely require for seamless integration to the supplied SIP offering is now a network client.

To enable the SIP client to register on the Zen network, I just needed to forward the usual SIP ports to the Fritzbox.

All is working well.

I can make incoming and out going calls - but there's no audio... so, kinda, but it's totally useless.

Thanks @splidbob

I also have a mesh setup with Asus AX92u nodes and would like to keep it that way.

If I understood correctly, you connected the ONT to your original router - I assume to a LAN port? - and just setup the authentication for the Zen FFTP connection on that original router. Then have your Fritzbox as a client of that original router (connected by ethernet?). Then setup SIP port redirection from the original router to the Fritzbox client?

Thx

Many thanks for this explaination, took awhile but I managed to create the appropriate Outbound NAT Rule on my UniFI Gateway and now inbound calls don't drop after 30 secs. Cheers

Edited by sudman (Sun 02-Mar-25 13:45:16)

Thanks to everyone who posted here - just had my FTTP with DV from Zen installed and thought I'd share my findings with anyone who's still trying to get this working.

FYI - Using my own Cisco Router and Firewall to terminate the broadband, and this worked perfectly fine without any issues. I also have a business contract with a /29 range of IPs.

Confirmed VOIP working with FritzBox directly into ONT.

Called Zen support and they gave out the basic details very readily (Username/Password/Registrar).

Failed to get things working with their help, and ended up giving up - shame really as they spent a long time trying to help, but just defaulted to they only support their box connected to the ONT.

After a long evening yesterday, and long morning today I eventually got things working.

I used the FritzBox as a basic IP Client which picked up a DHCP from my network.
For the phone settings I used the following:

Telephony Provider: Other Provider
Telephone number for registration: Full number with area code
Internal telephone number in the FRITZ!Box: Number without area code (tried with full number and it failed - not sure why)
Username: Full number with area code
Password: As given by support
Registrar: voip2.zen.co.uk
Under Line settings/Location settings check area code is correct

Left everything else as default.

The bigger issue was with NAT. Had to make sure the FritzBox was NAT'd to the highest useable IP address in the /29 range. I also had to ensure that TCP/UDP 5060 was forwarded back to FritzBox (in the end I gave it sole use of that IP as I have 5 others for other services - IE full NAT, not PAT).

Once I'd done that, I could make calls in and out without any issue.

Hope that helps someone else!

Cheers.

So near, yet so far.....

Despite my initial success, it appears all isn't great - had a few minor issues, and some of my family can't call me.

Looks like i'm back to the drawing board...

I've just been on the phone to Zen about this issue and the use of 3rd party VOIP devices or adapters is not supported. If I migrate my phone and BB service to them I would have to keep my telephone service with them for the term of the contract before I could move it to a 3rd party VOIP provider which is a real pain. So I would be stuck with the FritzBox for Digital voice. I use my own Router so would potentially have to live with double NAT. Still researching this but not looking too promising at the moment.

Anyone successfully put their Friztbox on the LAN side with digital voice operational

In reply to a post by mrwrighty:

I've just been on the phone to Zen about this issue and the use of 3rd party VOIP devices or adapters is not supported. If I migrate my phone and BB service to them I would have to keep my telephone service with them for the term of the contract before I could move it to a 3rd party VOIP provider which is a real pain. So I would be stuck with the FritzBox for Digital voice. I use my own Router so would potentially have to live with double NAT. Still researching this but not looking too promising at the moment.

By "not supported" do they mean that their "help" will not assist if there are any queries or that it will not work? If the former, nothing to stop you going ahead as you wish but it is up to you to resolve any problems.

Half the battle in supporting 3rd party VOIP hardware such as a grandstream would be getting the VOIP settings and credentials from ZEN. If this isn't forthcoming then I'm stuck. The actual configuration doesn't faze me, just the availability of such information to make it work.

In reply to a post by mrwrighty:

Half the battle in supporting 3rd party VOIP hardware such as a grandstream would be getting the VOIP settings and credentials from ZEN. If this isn't forthcoming then I'm stuck. The actual configuration doesn't faze me, just the availability of such information to make it work.

Then don't get VoIP from Zen. Superficially it looks simpler to take telephony from your ISP, but it is a world of complication when you want to change ISP. There are plenty of VoIP service providers out there and provided you avoid CGNAT ISPs, changing ISP does not affect your telephone service.

I can’t separate them. TalkTalk will not let me separate them as it’s linked to my FTTP account, moving my phone line from TalkTalk will cancel my FTTP service, yes true even though the landline is still on copper. ZEN will take over the FTTP and my landline, but I have to keep both with them for the duration of the 18 months contract as confirmed by a phone call from their sales team.

In reply to a post by mrwrighty:

I can’t separate them. TalkTalk will not let me separate them as it’s linked to my FTTP account, moving my phone line from TalkTalk will cancel my FTTP service, yes true even though the landline is still on copper. ZEN will take over the FTTP and my landline, but I have to keep both with them for the duration of the 18 months contract as confirmed by a phone call from their sales team.

You can break that logjam by ordering a new FTTP service (not a take over of an existing service) from Zen without any phone element then once you are happy that is working you get a VoIP provider to take over your phone service. The VoIP provider will trigger the cessation of your TalkTalk service but you already have the Zen FTTP up and running. The downside is one or two months of double cost while you run TalkTalk and Zen in parallel.

Then I have 2 fibre connection to the house, the old one will never be removed and 2 x ONT installations, yes an option, but not one I will be taking.

An ISP who knows what they are doing could request a replacement ONT with four ports so you only have a slightly larger ONT but still a single box on wall / power supply / fibre connection to the premises.

In reply to a post by tdw42:

An ISP who knows what they are doing could request a replacement ONT with four ports so you only have a slightly larger ONT but still a single box on wall / power supply / fibre connection to the premises.

There was a time when some ISPs had an issue with providing/taking over a FTTP service on anything other than the first port, does anyone know if that issue has been resolved now?

Hi DazColl,
Did you ever resolve this?