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]

REGISTER sip:09a.Z4.bbvoice.vodafone.co.uk SIP/2.0
Via: SIP/2.0/UDP XXX.XX.XX.XXX:5065;rport;branch=XX
From: <sip:voi000XXXXXX @resvoip.vodafone.co.uk>;tag=XX
To: <sip:voi000XXXXXX @resvoip.vodafone.co.uk>
Call-ID: XXX
CSeq: 54665 REGISTER
Contact: <sip:voi000XXXXXX @xxx.XXX.XXX.XXX:5065>
Expires: 3600
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Route: <sip:resvoip.vodafone.co.uk>
Max-Forwards: 70
User-Agent: Asterisk PBX 18.14.0
Content-Length: 0

[cf996]

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@chimpzilla1979 wrote:

Route: <sip:resvoip.vodafone.co.uk>

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This header is the issue. Can you confirm you have the '\;lr\;hide' suffix for all instances of 'outbound_proxy', eg:

outbound_proxy = sip:09a.Z4.bbvoice.vodafone.co.uk\;lr\;hide

[chimpzilla1979]

---

@cf996 wrote:

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@chimpzilla1979 wrote:

Route: <sip:resvoip.vodafone.co.uk>

---

This header is the issue. Can you confirm you have the '\;lr\;hide' suffix for all instances of 'outbound_proxy', eg:

outbound_proxy = sip:09a.Z4.bbvoice.vodafone.co.uk\;lr\;hide

I think so

[acl]
type = acl
deny = 0.0.0.0/0.0.0.0
permit = 127.0.0.1
permit = 192.168.5.0/24
permit = 212.XXX.XX.XXX/24
permit = 195.89.27.0/24

[transport-udp]
type = transport
protocol = udp
tos = af31
bind=0.0.0.0:5065
local_net = 127.0.0.1
local_net = 192.168.5.0/24
external_media_address=212.XXX.XX.XXX
external_signaling_address=212.XXX.XX.XXX

[voi000XXXXXXX]
type = registration
transport = transport-udp
contact_user = voi000XXXXX
client_uri=sip:voi000XXXXXX @resvoip.vodafone.co.uk
server_uri=sip:resvoip.vodafone.co.uk
outbound_proxy = sip:09a.Z4.bbvoice.vodafone.co.uk
outbound_auth = auth_vf

[auth_vf]
type = auth
auth_type = userpass
username= voi000XXXXXXX
password= XXXXX

[voi000XXXXXX]
type = aor
outbound_proxy = sip:09a.Z4.bbvoice.vodafone.co.uk\;lr\;hide
contact = sip:voi000XXXXXX @resvoip.vodafone.co.uk

[voi000XXXXXX]
type = identify
match = 09a.Z4.bbvoice.vodafone.co.uk
endpoint = voi000XXXXXXX

[voi0001XXXXXX]
type = endpoint
transport = transport-udp
context = vf-in
tos_audio = ef
disallow = all
allow = alaw,g729
direct_media=no
from_user = voi000XXXXXX
from_domain = resvoip.vodafone.co.uk
outbound_proxy = sip:09a.Z4.bbvoice.vodafone.co.uk\;lr\;hide
identify_by=username
outbound_auth = auth_vf
aors = voi000XXXXXX

[101]
type=endpoint
transport=transport-udp
context=vf-out
disallow=all
allow=alaw,g729
auth=101
aors=101

[101]
type=auth
auth_type=userpass
password=101
username=101

[101]
type=aor
max_contacts=10

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[chimpzilla1979]

Actually, just looking at what I posted, not the first one!. Just changed that and now it is registering! Thank you for all your help.

[cf996]

You also need to allow UDP source ports 10000-10010 on the Draytek in the same way as 5065, using the Open ports feature.

To test, check bi-directional audio for both outbound and inbound calls.

[chimpzilla1979]

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@cf996 wrote:

You also need to allow UDP source ports 10000-10010 on the Draytek in the same way as 5065, using the Open ports feature.

To test, check bi-directional audio for both outbound and inbound calls.

---

I've been trying to get it to work with Asterisk (not Voda directly) with various Android SIP clients today. The stock Android one that's in Android 11 and lower won't register with asterisk (works fine with a external sip provider).

With Android 12 the baked in sip client seems to be broken on that and on Android 13 it's completely removed (er, thanks google!).

I tried the Grandstream Wave Lite Android SIP client, which does register with Asterisk (Android 13), but often doesn't pick up incoming calls and on incoming calls, you can't hear the incoming audio.

Outgoing is fine.

MixuDroid had similar issues or wouldn't receive a call. Looking at wireshark it looks like it's telling Asterisk it is going to be listening on certain port and then fails to listen on that port/socket.
I don't know of any reason for this from an Android point of view. You can open any UDP/TCP socket that you want, keeping it open whilst the phone is not active is the hard part!
Eventually, I did what I am going to use this for anyway, which is to connect to some old dect phones in my mother's house. So I dug out an old Atcom FXO/FXS unit, punched in the Asterisk server details. Worked first time, audio both ways.

Wish I had done this this morning! I even got one of my old Snom phones out of the garage, but didn't actually plug it in, silly.
I may end up just replacing the Asterisk server with an adapter FXS/FXO unit. When I started this a week ago, I didn't that was possible. I thought Asterisk was maybe the only option.
One thing I will add about the Vodafone SIP details, is that it does only seem to work when connected to their network.  When I try to connect from my home network (Zen/Openreach FTTC) I get no reply from the server.

I've actually been testing all of this out down a VPN tunnel to a Vodafone FTTP connection, which has added some extra complexity.
It is now fully working... I think....
I just need to look at the stability of the connection with Vodafone. Sometimes it seems to take a long time to re-register if the connection has dropped. It has a "407 Proxy Authentication Required" error sometimes too.

[cf996]

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@chimpzilla1979 wrote:

---

[101]
type=endpoint
transport=transport-udp
context=vf-out
disallow=all
allow=alaw,g729
auth=101
aors=101

---

---

Can you try adding "direct_media=no" in this section to see if that helps the one way audio issue?

[chimpzilla1979]

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@cf996 wrote:

---

---

---

Can you try adding "direct_media=no" in this section to see if that helps the one way audio issue?

---

Seems to be missing the incoming audio still with Android clients. Working fine on the ATA.

[cf996]

You may need to try experimenting with the codecs. Maybe try disabling G729 and use only alaw. I've been able to successfully use various Android SIP clients with asterisk.

[ZinGeRs]

Just seen this thread have a HT801 knocking around and have been wanting to use my own router with vodafone due to some annoying niggle with an xbox series x requiring a switch being attached to the vodafone router to give an IP 100% of the time on start up (there is a long forum thread on this) however my folks will not embrace technology and always ring our landline and not mobiles so couldn't swop it out until now it seems...

However vodas so called xperts are less than helpful managed to get my VOIP username, password and server off them. Then realised I need the proxy and they flat refuse to give it me and even said I shouldn't have been given the information that I had been. So I guess my question is can I use one of the proxy servers mentioned? Or is it specific to me?