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.