Hosted Telephony - Platform Browser Extensions

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A browser extension which converts telephone numbers into clickable links to call using the VoiceHost desktop softphone application

How does it work?

This extension recognizes phone numbers on web pages and converts them into clickable links.

This is done by passing the phone number to the configured system protocol handler and from there to the application which registered this protocol handler. Just like an URL starting with “http”, a link can also start with other protocol specifiers, e.g. “tel”, “sip” or “callto”.

Phone numbers on a web page recognized by the extension and highlighted with an optional icon. When you click on a number. By clicking on this menu item the phone number is passed to the dial-pad of the desktop application “as is”.

Download the Extension for your browser:
Chrome Browser ExtensionFirefox Add-on Microsoft Edge Extension

 

 

 


 

PfSense VoIP Configuration

  • Knowledge Base
How to configure pfSense firewall for VoIP

pfSense is a free and open source firewall and router that also features unified threat management, load balancing, multi WAN, and more.

Configure Ports

Configure your SIP and RTP ports. SIP port is the default 5060 and RTP is between 10000 and 65335.

Configure the WAN IP Address

Asterisk Example - Also be sure to specify "externip" or "externhost" in sip.conf. externhost configured to a dyndns.org account that resolves to my WAN ip address.

Configure NAT

Asterisk Example - Make sure you have "nat=yes" and "canreinvite=yes" in sip.conf

Configure your local network

Make sure you have localnet setup to correspond with your local network in sip.conf. You can use the RFC1918 method or CIDR method.

localnet=192.168.1.0/24
Configure your SIP context

In your SIP provider's context in sip.conf, make sure you have "outboundproxy=192.168.1.1", replacing 192.168.1.1 with whatever your pfSense running siproxd ip address is.

[sipconvergence]
type=peer
user=phone
host=SEE VOICEHOST CONTROL PANEL FOR DETAILS
outboundproxy=192.168.1.1
fromdomain=SEE VOICEHOST CONTROL PANEL FOR DETAILS
fromuser=<censored>
secret=<censored>
username=<censored>
insecure=very
context=ivr
authname=<censored>
canreinvite=yes

Please note that if you don't use a PBX like Aterisk and use a softphone to connect, you will use your pfSense ip address for the proxy instead of sip.sipconvergence.co.uk

Configure pfSense firewall/nat rules
RTP

Add a NAT rule for RTP. This is essential or you will have no audio or one way audio in your calls. Also change the NAT IP to whatever your Asterisk server is and change the description to something that makes sense for you.

Interface: WAN
Protocol: UDP
External port range: From: 10000
External port range: To: 65335
NAT IP: 192.168.1.50
Local Port: 10000
Description: Hosted PBX - RTP
Enable Auto-add a firewall rule to permit traffic through this NAT rule
SIP

Add a NAT rule for SIP. This is essential or you won't be able to receive calls and you may have trouble registering with your SIP provider. Also change the NAT IP to whatever your Asterisk server is and change the description to something that makes sense for you.

Interface: WAN
Protocol: UDP
External port range: From: 5060
External port range: To: 5060
NAT IP: 192.168.1.50
Local Port: 6000
Description: Hosted PBX - SIP
Enable Auto-add a firewall rule to permit traffic through this NAT rule
The SIP Proxy siproxd
Install siproxd

Go to the pfSense web UI and going to System -> Packages. Hit the "+" button to the right of siproxd and let pfSense install the SIP proxy.

Configure siproxd

Go back to the main pfSense web UI page then go to Services -> siproxd. It may be under Services -> SIP Proxy as well. siproxd configured, be sure to change your "Outbound Proxy Hostname" to the hostname or IP (IP in my case) to your sip provider. Options not specified, leave blank or default.

Inbound Interface: LAN
Outbound Interface: WAN
Enable RTP Proxy: Enable
RTP Port Range (lower): 7070
RTP Port Range (upper): 7080
Outbound Proxy Hostname: xx.xx.xx.xx
Summary

Basically when you make a call your asterisk box will talk to the SIP proxy, the SIP proxy will then talk to your VoIP provider. When you receive a call your VoIP provider will talk directly with your asterisk box (this is important for setting "externip" or "externhost" in sip.conf).

QoS (Traffic Shaping) Traffic shaping can be enabled to allow n simultaneous 64kbps calls to happen and guarantee bandwidth. Please refer to http://doc.pfsense.org/index.php/Traffic_Shaping_Guide for traffic shaping help.

SIP Error Codes & SIP Trunk Troubleshooting

  • Knowledge Base
Outbound calls error with "all circuits busy" or "congestion":

This is the default configuration of Asterisk regardless of the actual error generated (which is infuriating when you are trying to diagnose the real problem) unless PBX is updated to send back the real error rather than the changed error. This error most commonly occurs when the call is not authenticating properly, at which point check the above in the SIP trunk configuration (If Asterisk, swap username= for defaultuser= to see if this solves the issue. Just because a trunk is showing as registered does not mean it will authenticate correctly.

Outbound calls fail with SIP error 488 (Not Accepted Here) or I-SUP errors 58 (bearer capability not available) or 88 (incompatible destination):

Check the codecs allowed in the SIP trunk configuration above, VoiceHost only supports: alaw, ulaw, gsm
If a codec is defined in Asterisk that is not one of the above, or is offering a differing sample rate or interval rate (e.g. 8000/20i - 8000Hz at 20ms) cannot interwork with 16000/30i - 16000Hz at 30ms) the call will fail and the codecs in the SIP trunk configuration need to be aligned to use one of the above codecs.

Inbound calls fail with SIP error 408 (Request Timeout):

Check the inbound number is mapped in the system correctly, if necessary the SIP trunk on the portal can be configured to strip the plus, e.g. if Asterisk is configured to use plus somewhere else. Check the trunk is registered. Ascertain how long the 408 error took to come back if it was immediate the trunk is usually unregistered if it took a few seconds the number is usually not mapped correctly.

Calls fail with SIP error 503, I-SUP errors 34 or 38:

If our platform replies back with 503 it usually means the gateway trying to process the call can't due to "issues", or the customer has hit their Calls-Per-Second (CPS) limit and is sending too many calls at once. Sometimes the error is passed back from IP Exchange through VoiceHost to the customer's system, at which point the call will usually hunt to another route to try and place the call.
 

Cause code (ISUP)SIP EquivalentDefinition
1404 Not FoundUnallocated (unassigned) number
2404 Not foundno route to network
3404 Not foundno route to destination
16BYE or CANCEL (*)normal call clearing
17486 Busy hereuser busy
18408 Request Timeoutno user responding
19480 Temporarily unavailableno answer from the user
20480 Temporarily unavailablesubscriber absent
21403 Forbidden (+)call rejected
22410 Gonenumber changed (w/o diagnostic)
22301 Moved Permanentlynumber changed (w/ diagnostic)
23410 Goneredirection to new destination
26404 Not Found (=)non-selected user clearing
27502 Bad Gatewaydestination out of order
28484 Address incompleteaddress incomplete
29501 Not implementedfacility rejected
31480 Temporarily unavailablenormal unspecified
34503 Service unavailableno circuit available
38503 Service unavailablenetwork out of order
41503 Service unavailabletemporary failure
42503 Service unavailableswitching equipment congestion
47503 Service unavailableresource unavailable
55403 Forbiddenincoming calls barred within CUG
57403 Forbiddenbearer capability not authorized
58503 Service unavailablebearer capability not presently
65488 Not Acceptable Herebearer capability not implemented
70488 Not Acceptable Hereonly restricted digital avail
79501 Not implementedservice or option not implemented
87403 Forbiddenuser not member of CUG
88503 Service unavailableincompatible destination
102504 Gateway timeoutrecovery of timer expiry
111500 Server internal errorprotocol error
127500 Server internal errorinterworking unspecified

Broadband Connection Fault Checklist

  • Knowledge Base
Initial Broadband fault checks for VoiceHost ADSL and FTTC connections
  1. Check the router is set to an 'Always on' connection and not 'On demand'.
  2. If you have ADSL try changing the ADSL Micro Filter, the most common cause of intermittent connections is a faulty filter. If you have FTTC please skip this step.
  3. Please ensure that you change the RJ-11 lead between the microfilter/FTTC faceplate and the router/modem.
  4. Make sure your router is connected to the BT Master Socket and no telephone extension leads are used between the wall and the router. Only use the supplied modem cable directly into the BT master socket.
  5. You can also try disconnecting any additional devices connected to the phone line such as fax machines, Sky Box, Red Care alarm, Credit Card terminal/Paying Device, telephone extension leads, etc. to avoid any possible interferences coming from these devices.
  6.  Swap the router out for a replacement.
  7. Noises on the telephone line can cause disconnections in the broadband signal. In order to identify if this is the case please try a Quiet Line Test.
  8. Connect only a phone, preferably a corded one, directly to the phone socket and dial 17070. It is recommended that you disconnect all devices from the line, such as ADSL routers, phones, faxes, credit card terminals, Sky Boxes and alarm systems.
    Once prompted, select option 2, and then observe the line for any cracklings, noises, interferences or clicks.
    If you do hear noises on the line, please contact the line provider and inform them that your line is experiencing high noise on the line and this is affecting your broadband signal.
    If you are still experiencing disconnects after carrying out the above checks please contact the support department to carry out further fault diagnostics on the line.

NOTE: It may require an engineer visit to resolve the issue, therefore it is important to carry out the above checks to rule out any equipment faults on site. Any engineer visits that do not find a fault within the provider network are chargeable.

3CX - SIP Trunk Guide

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System Preparation

Before configuring the SIP trunk it is required to go through the following checklist and make changes where necessary:

Further setup information can be found in our Academy:  3CX Academy Basic Course

3CX Version

Some providers gained support and compatibility with 3CX on a specific product version. It is advisable to always run the latest version of 3CX to ensure ongoing compatibility.

Minimum 3CX Version: 3CX Phone System 16.0

Provider Capabilities

Below is a short overview of the provider's capabilities and services and whether they’re supported or not:

  • CLNS (Clip No Screening): No
  • Catch All Routing: Yes
  • Fax in T38: Yes
  • CLIR (Number Suppression): Yes
  • DTMF via RFC 2833: Yes
  • Outbound Codec Order: G711A, G711U, G722, GSM, Opus
  • NAT Support: Yes
Configuring the Trunk with 3CX

The general instructions outlining how to add a new SIP Trunk to your 3CX installation can be found  here .

Adding the Trunk

Go to  “SIP Trunks”  and select  “Add SIP Trunk”

  • Select Country: UK
  • Select Provider in your Country: VoiceHost
  • Main trunk number: This will have been provided to you by VoiceHost. You must enter the number in the E164 number format (e.g. +44123456789)
  • Press OK

Under the  “General”  tab in the  “Authentication”  section, enter your Authentication ID and Password as well as the registrar address (these will be supplied to you by VoiceHost).

Adding Additional DIDs

To associate all other DIDs/Numbers you have in your VoiceHost account with 3CX, go to the Management Console → SIP Trunks, double-click on your VoiceHost Trunk and go to the  “DIDs”  tab

Here you should already see 1 entry; that is the Main Trunk number you have set. Add all other DIDs/Numbers you have to the list in the E164 number format (e.g. +44123456789) and press OK.

Creating Inbound Rules

Now that you have associated all your DIDs/Numbers with your SIP Trunk in 3CX, you can create Inbound Rules to set where calls will be routed when those numbers are called. Instructions on how to create Inbound Rules can be found  here .

Number Format
General

When configuring VoiceHost SIP Trunks in 3CX, all numbers should be entered in the E.164 number format (e.g. +44123456789), otherwise, call routing will fail.

Outbound Caller ID

VoiceHost trunks do not support Clip No Screening which means you can only present numbers that are associated with your account as Outbound Caller ID.

Outbound Rules

When configuring your Outbound Rules, numbers can be dialled in all valid number formats. More information about how to create Outbound Rules and how they work can be found here

Enabling TLS and SRTP

Under the 'General' tab please update the host to the one shown in the VoiceHost portal. This changes for TLS and SRTP so it will only be changed once enabled.
Under Options, please also upload the linked PEM under the option for the trunks 'TLS Root'.
Ensure SRTP is enabled.
Ensure TLS is set and the transport.
Ensure that host port is set to 'Auto-Detect'

Root Certificate: download here (You will need to rename to .pem)

Hosted Platform Short Codes

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Cloud Platform Vertical Service Codes (Telephone Short Codes)
Action
Dial
Emergency Services999 or 112
Call a group of phones (as defined in the portal under call groups)*<group number>
Intercept/Pickup group call*0#<pickup group ID>
Intercept/Pickup extension call**<seat/extension number>
Call another extension (internal only)<seat/extension number>
Speaking clock (on-platform)123
Dial Welcome Message1234
Withhold number prefix (per call)141<telephone number>
Last Call Identified (DDI calls and Group calls only)1471
Record a custom prompt (e.g. IVR greeting, Queue greeting)151 (Record your prompt)
               |_   1 - Accept the recording
               |_2 - Listen back to the recorded prompt
               |_3 - Re-record the prompt
Call Monitoring (Call Whisper), listen into another seat and optionally whisper to them. (passwords defined in the portal)154, <seat number>,<password>
  |_1 - Listen to the call
  |_2 - Whisper to extension
Dial Echo Test (used for latency diagnostics)160
Time Profile Night Mode (Toggles Active/Inactive destination)*1#<time profile number>
Page extension (one-way audio)*2#<seat/extension number>
Page group (one-way audio)*3#<call group>
Intercom (two-way audio)*4#<seat/extension number>
Wake-up call reminder (Create and Delete)*5#<enter 24H time>
Call Parking1900 <parking reference read back> (Parks the current call)
<dial parking reference given when parking> (retrieves a given parked call)
Call Recording#1 (mute call recording)
#2 (unmute call recording)
Extension Call Intercept/Pickup**<seat number>
Dynamic Call Queue agent login (extensions jumping in/out of queues)120*<queue number> (Login to a call queue)
121*<queue number>  (Logout of a call queue)
Voicemail
Access Voicemail Externally (mailbox & password required from the portal)0843 557 4 557
Access Extension Voicemail (only accessible from the extension itself)1571
Access Shared Voicemail (accessible from any extension within the account)1572
Voicemail Menu0 - Mailbox options
               |_        1 - Record unavailable greeting (rings out)
               |_2 - Record busy greeting (only works if handset sends a busy signal back to platform, disable call waiting)
               |_3 - Record name
               |_4 - Record temporary greeting
               |_5 - Change mailbox password
1 - Listen to old messages (messages previously listened to)
2 - Change folders (Work, Friends, Family)
3 - Advanced options
               |_1 - Call back sender
               |_2 - Move message to another folder (Work, Friends, Family)
4 - Play the previous message (if exists)
5 - Repeat the current message
6 - Play the next message (if exists)
7 - Delete or Restore a recently deleted message
8 - Forward to another users extension
9 - Save Message
* - Help (Repeats the menu options)
# - Exit
Conferencing
Access Conferencing Service Externally0843 557 5 575
Call or transfer into the conferencing facility155, <room>#, <PIN or admin PIN>#, <state name>#
Conference Room Short Codes* - Conferencing Menu
               |_   1 - mute and unmute
               |_2 - Lock and unlock the room - admin only
               |_3 - Kick the last joined user - admin only
               |_4 and 6 - Conference room volume up/down
               |_7 and 9 - users volume up/down
               |_8 - Exit the conference

SIP ALG and why it should be disabled on most routers

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What is SIP ALG?

SIP ALG stands for Application Layer Gateway and is common in all many commercial routers. Its purpose is to prevent some of the problems caused by router firewalls by inspecting VoIP traffic (packets) and if necessary modifying it.

Many routers have SIP ALG turned on by default.

There are various solutions for SIP clients behind NAT, some of them in the client side (STUN, TURN, ICE), others are in the server side (Proxy RTP as RtpProxy, MediaProxy).

Generally speaking, ALG works typically in the client side LAN router or gateway. In some scenarios, some client-side solutions are not valid, for example, STUN with symmetrical NAT router. If the SIP proxy doesn't provide a server-side NAT solution, then an ALG solution could have a place.

An ALG understands the protocol used by the specific applications that it supports (in this case SIP) and does a protocol packet-inspection of traffic through it. A NAT router with a built-in SIP ALG can re-write information within the SIP messages (SIP headers and SDP body) making signalling and audio traffic between the client behind NAT and the SIP endpoint possible.

How can it affect VoIP?

Even though SIP ALG is intended to assist users who have phones on private IP addresses (Class C 192.168.X.X), in many cases it is implemented poorly and actually causes more problems than it solves. SIP ALG modifies SIP packets in unexpected ways, corrupting them and making them unreadable. This can give you unexpected behaviour, such as phones not registering and incoming calls failing.

Therefore if you are experiencing problems we recommend that you check your router settings and turn SIP ALG off if it is enabled.

  • Lack of incoming calls: When a UA is switched on it sends a REGISTER request to the proxy in order to be localisable and receive any incoming calls. This REGISTER is modified by the ALG feature (if not the user wouldn't be reachable by the proxy since it indicated a private IP in REGISTER "Contact" header). Common routers just maintain the UDP "connection" open for a while (30-60 seconds) so after that time the port forwarding is ended and incoming packets are discarded by the router. Many SIP proxies maintain the UDP keepalive by sending OPTIONS or NOTIFY messages to the UA, but they just do it when the UA has been detected as NAT'd during the registration. A SIP ALG router rewrites the REGISTER request to the proxy doesn't detect the NAT and doesn't maintain the keepalive (so incoming calls will be not possible).
  • Breaking SIP signalling: Many of the actual common routers with inbuilt SIP ALG modify SIP headers and the SDP body incorrectly, breaking SIP and making communication just impossible. Some of them do a whole replacing by searching a private address in all SIP headers and body and replacing them with the router public mapped address (for example, replacing the private address if it appears in "Call-ID" header, which makes no sense at all). Many SIP ALG routers corrupt the SIP message when writing into it (i.e. missed semi-colon ";" in header parameters). Writing incorrect port values greater than 65536 is also common in many of these routers.
  • Disallows server-side solutions: Even if you don't need a client-side NAT solution (your SIP proxy gives you a server NAT solution), if your router has SIP ALG enabled that breaks SIP signalling, it will make communication with your proxy impossible.
I have disabled SIP ALG but I'm still experiencing problems...

If you are still having problems after disabling SIP ALG, please check your firewall configuration.

I can't disable SIP-ALG? How to Circumnavigate any networking vendors broken implementation of SIP ALG
  • Enable TLS on SIP Endpoints, VoiceHost supports TLS which masks SIP signalling from the prying eyes of ALG functionality.
  • Enable IPv6 on SIP Endpoints. Practically this is not a realistic option for users requiring mobility but for static locations, this does remove the requirement (Must be supported by your ISP). Most Internet providers do not fully support pure IPv6
  • Change you Router Obviously a last resort if all else fails.
How do I turn off SIP ALG?
Most home/residential routers have a web interface. Typically this is 192.168.1.1 but you just check your default gateway by typing ipconfig in Windows command prompt or ifconfig on Linux systems from any connected device on the same LAN.
If your router does not have a web interface you will most likely need a Telnet client to login.
If you don't have a telnet client installed we recommend Smartty (smartty.sysprogs.com)
Connect in telnet to the IPv4 address of your gateway and hit enter again.
Asus Routers

Disable the option SIP Passthrough under Advanced Settings / WAN -> NAT Passthrough.
If your router doesn't have this option SIP ALG may be disabled via Telnet.

nvram get nf_sip 
(It should return a "1")

nvram set nf_sip=0 
nvram commit
Reboot

AVM Fritz!Box
SIP ALG cannot be disabled. (See above on how to get around this)
Barracuda Firewalls
Go to Firewall > Firewall Rules > Custom FirewallAccess Rules
Click the "Disabled" check box next to any rules named LAN-2-INTERNET-SIP and INTERNET-2-LAN-SIP
This disables SIP ALG.
Billion
Navigate to the web interface
-> Select Configuration
-> Select NAT
-> Select ALG
-> Disable SIP ALG
BT (Homehubs)
SIP ALG cannot be disabled in the settings of BT HomeHubs but can be disabled with BT Business Hub versions 3 and higher.
Cisco RV Range
(RV082, RV016, RV042, RV042G, RV325)
-> Go to System Summary and ensure that the firmware is up to date (1.1.1.06 or later).
-> f needed, update firmware by going to System Management > Firmware Upgrade.
-> Go to Firewall > General.
-> Ensure that Firewall and Remote Management are enabled (checked).
-> Ensure that the following are disabled (unchecked):
-> SPI (Stateful Packet Inspection)
-> DoS (Denial of Service)
-> Block WAN Request
-> SIP ALG
-> Click Save.
-> Browse to IPADDRESS/f_general_hidden.htm.
-> Set UDP Timeout to 300 seconds.
-> Go to Firewall > Access Rules.
-> Whitelist VoiceHost IP ranges
Save all changes.
D-Link
In 'Advanced' settings --> 'Application Level Gateway (ALG) Configuration' un-tick the 'SIP' option.
DD-WRT
No ALG function available - Consider using a public STUN server
DrayTek

DrayTek Vigor 2760 devices, the option can be found in the regular interface at Network -> NAT -> ALG.

If your device does not have a web interface then you'll need a telnet client.

You will be prompted to provide a username and/or password. These are the same credentials used to access the router's web interface.

Afterwards, type in these commands:

sys sip_alg 0
sys commit

On Draytek Vigor2750 and Vigor2130 please use these commands instead:

kmodule_ctl nf_nat_sip disable
kmodule_ctl nf_conntrack_sip disable

EE

Huawei E5330

Navigate to the web interface
Click Settings.
Enter the required username and password, then click Log In. 
Note: The default username and password is admin.
Click the Security dropdown.
Click SIP ALG Settings.
Untick the Enable SIP ALG box.
Click Apply.

Fortinet

Fortigate:

Disabling the SIP ALG in a VoIP profile
SIP is enabled by default in a VoIP profile. If you are just using the VoIP profile for SCCP you can use the following command to disable SIP in the VoIP profile.

config voip profile
edit VoIP_Pro_2
config sip
set status disable
end

Huawei

The SIP ALG setting is usually found in the Security menu.

  1. Vodafone / Huawei (HHG2500)
  2. TalkTalk / Huawei (HG633) 
  3. EE / Huawei (E5330)
Juniper

Type the following into the CLI
To check if currently enabled or disabled run show security alg status | match sip
To disable run:

configure
set security alg sip disable
commit

Linksys:
Check for a SIP ALG option in the Administration tab under 'Advanced'.
You should also disable the SPI Firewall option.
Mikrotik
Disable SIP Helper.
Netgear

Look for a 'SIP ALG' checkbox in 'WAN' settings.

Under 'NAT Filtering' uncheck the option 'SIP ALG'
Port Scan and DoS Protection should also be disabled.
Disable STUN in VoIP phone's settings.

openwrt
No ALG feature - Consider using a public STUN server
PfSense
https://www.voicehost.co.uk/help/pfsense-voip-configuration
SonicWALL Firewall
Under the VoIP tab, the option 'Enable Consistent NAT' should be enabled and 'Enable SIP Transformations' unchecked.
Detailed instructions can be found here: https://www.voicehost.co.uk/help/sonicwall-configuration
Speedtouch

To disable SIP ALG you need to telnet into your Speedtouch router and type the following:

-> connection unbind application=SIP port=5060
-> saveall

TalkTalk

2017/18 See Huawei (HG633)

  1. Navigate to the web interface
  2. Select 'Port Forwarding' from the menu
  3. Uncheck SIP-ALG from the ALG section at the bottom of the page.
Technicolor / Thompson
TG588 TG589 TG582 DWA0120
Open Command Prompt – “Start” → “Run” → type “cmd” and press “Enter”.
In Command Prompt, type “telnet 192.168.1.254” and press enter. 192.168.1.254 is the default IP address of the router. If you are running on Windows 7/8/8.1/10, you might need to install the telnet client from “Control Panel” → “Programs and Features” → “Turn Windows features on and off”.
The default username is “Administrator”, and there is no default password, leave blank.
Type “connection unbind application=SIP port=5060” and press “Enter”.
Type “ saveall ” and press “Enter”.
Type “exit” and press “Enter” to exit the telnet session.
Tomato
Depending on the version of Tomato, SIP ALG can be found under Advanced then Conntrack/Netfilter in the Tracking/NAT Helpers section. If you find SIP checked then SIP ALG is enabled. Uncheck it to disable it.
TP-Link
Navigate to your routers web interface.
The default username is admin and the default password is admin.
On the left, click on Advanced Setup and then click on NAT and then click on ALG.
Uncheck the box by SIP Enabled. (Some TP firmware shows this as SIP Transformations which is the same thing).
Click Save/Apply.
UBEE Gateways
Go to Advanced > Options.
Disable (uncheck) SIP.
Disable (uncheck) RTSP.
Click Apply.
Ubiquiti

Use the configuration tree if supported: system -> conntrack -> modules -> sip -> disable

Alternatively, you can SSH into the device and run the following commands:

configure
set system conntrack modules sip disable
commit
save
exit

Virgin SuperHub
SIP ALG cannot be disabled in the settings of SuperHubs.
Please see our workarounds at the top of the page.
Vodafone
2018 Onwards - See Huawei (HHG2500)
Vyatta / Brocade:

Type the following into the CLI

configure
set system conntrack modules sip disable
commit
save
exit

Watchguard Firewall
Detailed instructions can be found here: https://www.voicehost.co.uk/help/watchguard-firewall-sip-configuration
ZyXEL

Under Network or Advanced -> ALG un-tick the options Enable SIP ALG and Enable SIP Transformations.
Telnet commands must be used to disable SIP ALG with some other Zyxel routers.

  1. Telnet into the router.
  2. Select menu items 24 then 8.
  3. To display the current SIP ALG status run the following command:
  4. ip nat service sip active
  5. To turn off SIP ALG:
  6. ip nat service sip active 0
ZyXEL (ZyWALL USG Routers)
Go to Settings > Configuration > Network > ALG.
Disable SIP ALG.
Turn ON Enable SIP Transformations.
Turn OFF Enable Configure SIP Inactivity Timeout.

2n Helios Verso IP Door Entry Configuration for SIP

  • Knowledge Base

Broadband Network General Settings

  • Knowledge Base
Broadband - General configuration settings for the VoiceHost Broadband Network
 ADSL2+ SoADSLFTTC SoGEA & G.FastFTTP
Line typeAnnex A + M: Analogue/Raw Copper (PSTN)VDSL2: Analogue/Raw Copper (PSTN)Full Fibre
EncapsulationPPPoAPPPoEPPPoE
MultiplexingVC-Mux
IPv4Static x 1 included (see below if >1 required)
IPv6/64 Enabled by default
ATMVPI: 0
VCI: 38		N/AN/A
VLANN/A101 for routers with built-in VDSL2 modemsN/A
MTU1492
AuthenticationVOICEHOST PROVIDED
DNS - Domain Name Servers
 NameIPv4 addressIPv6 address
PrimaryVoiceHost (Private)xx.xxx.xx.xxxxxxx:xx:xxx:xxxx
SecondaryVoiceHost (Private)xx.xxx.xx.xxxxxxx:xx:xxx:xxxx
PrimaryCloudflare (Public)1.1.1.12606:4700:4700::1111
SecondaryCloudflare (Public)1.0.0.12606:4700:4700::1001
PrimaryGoogle DNS (Public)8.8.8.82001:4860:4860::8888
SecondaryGoogle DNS (Public)8.8.4.42001:4860:4860::8844
Unmanaged SMTP relay for sending Email:

Relay SMTP access is granted only for email sent using VoiceHost internet connections and does not require a username or password, this is an unmanaged service and faults regarding email failure are not supported.

The relay domain addresses are:

  • relay.ukdsl.co
  • relay.voicehostdsl.com
  • relay.newbreedbb.co.uk
Speed Test Servers

Speedtest tools calculate a snapshot of the connection speed to our network AS31472:

  • Download - The maximum currently available bandwidth downstream
  • Upload - The maximum currently available bandwidth upstream
  • Ping - <150 ms is preferred for QoS
  • Jitter - <30 ms is preferred for QoS

You should ensure that no other devices are using the connection during the test and you are connected directly into the primary router.

Reverse DNS and SPF:

Reverse DNS is IP address to domain name mapping - the opposite of forward (normal) DNS which maps domain names to IP addresses. Please contact support if you require reverse DNS as you may require this in order to send emails and have them accepted by other networks.

Sender Policy Framework (SPF) is an email validation system designed to prevent email spam by detecting email spoofing, a common vulnerability, by verifying sender IP addresses. If your domain does not have an SPF record, you will also need to add this as some recipient domains may reject messages from your users because they cannot validate that the messages come from an authorised mail server.

Additional IPv4 addressing and IPv6:

Subject to approval based on RIPE guidelines and RFC2050 (Section 2.1), VoiceHost can offer additional static IPv4 subnets to its broadband customers on all products.
Please contact support for further details.

IPv6 is disabled by default but can be enable IPv6 via your account control panel.

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