Landline phone with own router on FTTP

[bruce_miranda]

Just got FTTP and everything is working fine off the VF router. Phone lines are plugged into the VF router, VF router's WiFi is switched off. 3rd party Mesh has been switched to Bridge mode and plugged into the VF's ethernet port.

However I am shocked at how feature poor the VF router is. e.g. There are no Parental controls at all. I know I can get rid of the VF router and plug my own Mesh router into the Openreach ONT, but what about the Landline. 

Are there any 3rd party routers in the market that have a telephone socket at the back to allow the home phone to be plugged in? 

[chimpzilla1979]

---

@cf996 wrote:

---

I initially set up the config using Android apps before switching to the Grandstream devices once the config was finalised. Having tested again I confirm there are no issues here using Mizudroid (with default settings) using the provided config (also without direct_media=no). Is your local subnet covered by the 'permit' and 'local_net' in the config? Secondly, did you try calling locally between clients to see if the same issue occurs? I suspect the issue may be on the Draytek side.

---

I couldn't figure out the issue, it could have just been something to do with the VPN I was testing on. I've ditched the Asterisk anyway and now have just the Grandstream HT801 onsite. All working great with the settings you helped Xjacko with.
Thanks again for all your help, I would have given up and transferred the number to a 3rd party voip provider otherwise. 
Hopefully others find this thread helpful too,

[ZinGeRs]

Bit the bullet despite my ongoing speed/contention issues in an evening stuck my router on and attached the HT801 and can state another successful configuration thanks again to all involved.

However will stick my voda box back on until they sort my speeds out

[Jayach]

@ZinGeRs That is amazing. So we now know it is possible. Did you need any extra details over and above what Vodaphone gave you?

[ZinGeRs]

Just need your details from Vodafone and copy the already shared Google drive image of the config and change the relevant information plus your own details and worked first time.

[Luciftian]

Hi there, thank you for all of the info in this thread.
I have just been activated onto FTTP and I am still using the VF router at the moment, but I am trying to ditch it (as I had for my copper fibre plan). 
I have my SIP details (the tobi agent found them for me, I did not have to argue) but I do not have a static IP yet (I will need to request one).  

@cf996  I  have two questions, if you could help me:

1) how did you configure MizuDroid? I just tried it and if fails with "cannot resolve resvoip.vodafone.co.uk".
Could you please share your configuration? Is a static IP required?
2) I need to purchase an ATA box; Grandstream HT801 is £43 and the HT812 is £53, so the price is very close. The HT812 has an extra port and a nat router. Will the HT812 work? @bruce_miranda Did you manage to make it work?

Many thanks in advance!

[cf996]

1) how did you configure MizuDroid? I just tried it and if fails with "cannot resolve resvoip.vodafone.co.uk".
Could you please share your configuration? Is a static IP required?

---

Mizudroid doesn't work. I only used it as an Asterisk client. It can't be used to directly log in.

You don't need a static IP as you can use STUN. Having a static IP makes it easier but it's not required.

2) I need to purchase an ATA box; Grandstream HT801 is £43 and the HT812 is £53, so the price is very close. The HT812 has an extra port and a nat router. Will the HT812 work? @bruce_miranda Did you manage to make it work?

---

Either will work.

[Luciftian]

@cf996 Thank you. 
I  have ordered the HT812 and will have a go after Christmas.
Thank you again.

[Luciftian]

@cf996 Hello! I have my HT-812 (V4.0A, firmware v. 1.0.43.11) connected and configured as per the advice in this thread, with a static IP address (the only setting that is notably different from @Xjacko's configuration is the dial plan; mine is the GrandStream default  { x+ | \+x+ | *x+ | *xx*x+ }) .

The port status is 'registered'.

I have not set up any port forwarding on my router (tp-link Archer VR400, but I have disabled SIP ALG (PPTP pass-through, L2TP Pass-through, IPSec Pass-through, FTP ALG, TFTP ALG and H323 ALG are still enabled).

1) I saw that I should allow "inbound traffic from the subnet below source port 5060 dest port 5065" (I assume the subnet is 195.89.27.0/24 to the HT-812's local IP), but I have never done this before, does this mean I need to set up the following rule on my router's virtual server? I cannot see where I would specify the incoming IP. Apologies for being daft, but I am not familiar with port forwarding on a router.

2) I can receive calls. I can make calls to landlines (national and international).
I cannot make any calls to mobile numbers or non-geographic numbers, like 0800 and 111. I get an engaged tone.
Any ideas what might be the cause?
Here is a screenshot of my profile configuration on HT-812 

Any help is much appreciated 🙂

[cf996]

---

@Luciftian wrote:

2) I can receive calls. I can make calls to landlines (national and international).
I cannot make any calls to mobile numbers or non-geographic numbers, like 0800 and 111. I get an engaged tone.
Any ideas what might be the cause?

---

You need to set "Sip User-Agent" to "Vox 3.0".

---

@Luciftian wrote:

1) I saw that I should allow "inbound traffic from the subnet below source port 5060 dest port 5065" (I assume the subnet is 195.89.27.0/24 to the HT-812's local IP), but I have never done this before, does this mean I need to set up the following rule on my router's virtual server? I cannot see where I would specify the incoming IP. Apologies for being daft, but I am not familiar with port forwarding on a router.

---

As long as the NAT session from the outbound registration doesn't time out, you should be okay. If it does time out, you won't be able to receive calls from that point until the next registration occurs. I see in your config that "Session Expiration" is 180 seconds, so if the NAT timeout is 120 seconds, you'll have a 60 second window where no calls can be received. You could test this by trying placing a call several times, say every 2 minutes and 30 seconds and checking that a ring occurs. If you do face issues, you can change the Session Expiration to 90 seconds.

[Luciftian]

@cf996 Hi, thank you for the suggestions.

I have set the Sip User-Agent" to "Vox 3.0". 

Thanks to this change, calls to  mobile phones which are not directed to a voicemail service are working, but not calls to non-geographic numbers, including voicemail services for mobile phones!
Also, instead of getting an engaged tone immediately, there are ca. 110 seconds of silence followed by an engaged tone.

Any ideas as per why non-geographic calls are still failing?
This is the screengrab of the current settings.
Thank you again for your help.