- Download Zoiper Here
- The only option required from this screen to make the softphone work is the Accounts button. All the other options are personal to your own requirements from the phone and will rarely be needed to get Zoiper working with your VoIP account.
- After selecting Accounts you then need to add an account by tapping the + symbol highlighted above, then choose SIP account and enter your extension credentials:
- Account name: This can be set as you wish and is merely for reference. If you’re configuring multiple accounts then it makes sense to set this to something relevant.
- Domain: As per your VoiceHost control panel.
- User name: As per your VoiceHost control panel.
- Password: This is the password relevant to your ext/seat
- Network Settings: In this option is another setting titled Refresh which should be set to 60.
- With all the above done, come out of the Network Settings screen so your back at the SIP Account screen, scroll to the top and tap Register.
- Your Zoiper softphone should now be registered.
Hosted Telephony - Platform Browser Extensions
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”.
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PfSense VoIP Configuration
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
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 Equivalent | Definition |
|---|---|---|
| 1 | 404 Not Found | Unallocated (unassigned) number |
| 2 | 404 Not found | no route to network |
| 3 | 404 Not found | no route to destination |
| 16 | BYE or CANCEL (*) | normal call clearing |
| 17 | 486 Busy here | user busy |
| 18 | 408 Request Timeout | no user responding |
| 19 | 480 Temporarily unavailable | no answer from the user |
| 20 | 480 Temporarily unavailable | subscriber absent |
| 21 | 403 Forbidden (+) | call rejected |
| 22 | 410 Gone | number changed (w/o diagnostic) |
| 22 | 301 Moved Permanently | number changed (w/ diagnostic) |
| 23 | 410 Gone | redirection to new destination |
| 26 | 404 Not Found (=) | non-selected user clearing |
| 27 | 502 Bad Gateway | destination out of order |
| 28 | 484 Address incomplete | address incomplete |
| 29 | 501 Not implemented | facility rejected |
| 31 | 480 Temporarily unavailable | normal unspecified |
| 34 | 503 Service unavailable | no circuit available |
| 38 | 503 Service unavailable | network out of order |
| 41 | 503 Service unavailable | temporary failure |
| 42 | 503 Service unavailable | switching equipment congestion |
| 47 | 503 Service unavailable | resource unavailable |
| 55 | 403 Forbidden | incoming calls barred within CUG |
| 57 | 403 Forbidden | bearer capability not authorized |
| 58 | 503 Service unavailable | bearer capability not presently |
| 65 | 488 Not Acceptable Here | bearer capability not implemented |
| 70 | 488 Not Acceptable Here | only restricted digital avail |
| 79 | 501 Not implemented | service or option not implemented |
| 87 | 403 Forbidden | user not member of CUG |
| 88 | 503 Service unavailable | incompatible destination |
| 102 | 504 Gateway timeout | recovery of timer expiry |
| 111 | 500 Server internal error | protocol error |
| 127 | 500 Server internal error | interworking unspecified |
Broadband Connection Fault Checklist
Initial Broadband fault checks for VoiceHost ADSL and FTTC connections
- Check the router is set to an 'Always on' connection and not 'On demand'.
- 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.
- Please ensure that you change the RJ-11 lead between the microfilter/FTTC faceplate and the router/modem.
- 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.
- 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.
- Swap the router out for a replacement.
- 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.
- 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
System Preparation
Before configuring the SIP trunk it is required to go through the following checklist and make changes where necessary:
- NAT (when used) created to 3CX
( https://www.3cx.com/docs/ports/ ) - Firewall Checker passed
( https://www.3cx.com/docs/troubleshooting-firewall-checker/ ) - Firewall SIP ALG checked and if present disabled
( https://www.3cx.com/docs/manual/firewall-router-configuration/ )
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
Action | Dial | |
| Emergency Services | 999 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 Message | 1234 | |
| 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 Parking | 1900 <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 Menu | 0 - 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 Externally | 0843 557 5 575 | |
| Call or transfer into the conferencing facility | 155, <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
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.
| 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. nvram get nf_sip nvram set nf_sip=0 |
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 | -> 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. Afterwards, type in these commands:
On Draytek Vigor2750 and Vigor2130 please use these commands instead:
|
EE | Huawei E5330 Navigate to the web interface |
Fortinet | Fortigate: Disabling the SIP ALG in a VoIP profile
|
Huawei | The SIP ALG setting is usually found in the Security menu.
|
Juniper | Type the following into the CLI
|
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' |
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 |
TalkTalk | 2017/18 See Huawei (HG633)
|
Technicolor / ThompsonTG588 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:
|
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
|
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.
|
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
The 2N IP Verso is a security intercom that is highly scaled thanks to its modular design. The Helios Verso can not only be easily integrated into your current camera and monitoring system but thanks to programmable scripts, the whole system can be used as a security component to protect the building.
Broadband Network General Settings
Broadband - General configuration settings for the VoiceHost Broadband Network
| ADSL2+ SoADSL | FTTC SoGEA & G.Fast | FTTP | |
| Line type | Annex A + M: Analogue/Raw Copper (PSTN) | VDSL2: Analogue/Raw Copper (PSTN) | Full Fibre |
| Encapsulation | PPPoA | PPPoE | PPPoE |
| Multiplexing | VC-Mux | ||
| IPv4 | Static x 1 included (see below if >1 required) | ||
| IPv6 | /64 Enabled by default | ||
| ATM | VPI: 0 VCI: 38 | N/A | N/A |
| VLAN | N/A | 101 for routers with built-in VDSL2 modems | N/A |
| MTU | 1492 | ||
| Authentication | VOICEHOST PROVIDED | ||
DNS - Domain Name Servers
| Name | IPv4 address | IPv6 address | |
| Primary | VoiceHost (Private) | xx.xxx.xx.xxx | xxxx:xx:xxx:xxxx |
| Secondary | VoiceHost (Private) | xx.xxx.xx.xxx | xxxx:xx:xxx:xxxx |
| Primary | Cloudflare (Public) | 1.1.1.1 | 2606:4700:4700::1111 |
| Secondary | Cloudflare (Public) | 1.0.0.1 | 2606:4700:4700::1001 |
| Primary | Google DNS (Public) | 8.8.8.8 | 2001:4860:4860::8888 |
| Secondary | Google DNS (Public) | 8.8.4.4 | 2001:4860:4860::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.