Frequently Asked Questions
Please contact Support who will advise the best way to rectify your issue.
Please have your account number(s) ready when you call to assist your support representative.
The registration time for the phone or the trunk is probably set too high, most default on setup to 3600 seconds (1 hour).
3600 seconds registration is fine for phones and trunks that will only make outgoing calls.
The registration time needs to be much lower than that to be able to continuously receive calls because routers and firewalls close ports often under 1 hour.
In the PBX or phone's settings, look for a 'Registration time' or 'Registration period' or similar, and change it to 60 seconds.
If necessary, restart the phone/PBX and retry incoming calls.
If you're still having problems contact Support. Please have your account number(s) ready for the support representative when you call.
Check to see whether the incoming telephone number is working when pointed directly to your extension or trunk.
Also check the registration interval is less than 10 minutes (600 seconds), we recommend 60 seconds as routers may close the port the PBX/handset is using.
If you have Time Profiles or Call Groups or other PBX features in the logic chain after the number, remove them until it works, then add them in one by one until you find which is causing your phone not to ring.
If you find that after this the phone is still not ringing, please contact Support.
Please have the account number(s) and telephone number(s) ready for the support representative.
First thing to try is rebooting the PBX/handset, if it's a softphone completely close it down and reopen it. If that doesn't fix it see below.
There are only a finite number of reasons why a registration error would occur. Some of these errors are detailed below:
- Registration error 401 (unauthorised): check the username and password for the trunk/extension, the IP address restrictions for SIP trunks, and the host/domain/registrar/proxy.
- Registration error 403 (forbidden - bad auth)[rarely occurs]: check the username and the password for the SIP trunk/hosted seat.
- Registration error 404 (not found): check the host/domain/registrar/proxy.
- Registration error 408 (timed out): check all of the above, disable the software firewall (especially Virgin Media) if it's a softphone, then open the ports below on the router.
- SIP: by default UDP port 5060 (handset or PBX may be configured to use something else, if so open that port)
- TLS UDP port 5061
- RTP: UDP ports 10000 to 65335 (handset or PBX normally has a much narrower or more specific port range, open that instead)
- Registration error 503 (service unavailable): check the domain, then the username and password for the SIP trunk/hosted seat.
When copying and pasting details make sure that what you are copying is the exact string required by pasting it in a text file first.
Passwords in some browsers may have a space or tab after them that will be picked up by the device.
If assistance is needed in solving these errors please contact Support. Please have the account number and trunk/extension username ready for the support representative.
You can, this is known as number masquerading or number spoofing. There are strict Ofcom regulations governing this, consequently we have to adhere to them.
Before we can allow you, or your customer, to present the number we need the following:
- The Letter of Authority signed by the end user (please ask Support for a copy of this form);
- Top copy of a recent bill from the current communications provider of the number.
Once we have both those returned we can then allow you to present this number for outgoing calls. The bill is also an Ofcom requirement.
More than likely yes, depending on who the current provider / network operator, and rangeholder / original provider is. Please contact Support for a copy of the porting form.
Most geographic numbers can be ported in.
For non-geographic numbers please check with Support as to the likelihood of porting the number.
0845s and 0870s are normally OK, 0842/3/4 and 0871/2/3 revenue sharing numbers are not guaranteed, particularly with newer providers.
Lead times vary depending on the number(s) and current or prior services, especially if it was ported from the rangeholder into another network operator.
If you want to be absolutely sure please speak to one of our Support team.
IP telephony (VoIP) relies on a solid IP connection by its very nature. If this connection goes down you have no IP phones. If your business relies on the telephones this is a problem.
For this reason, any customer that uses IP telephony and relies on phones for their business should definitely consider and deploy more than one data connection, and avoid mixing VoIP traffic with normal internet.
Choose a data product or products that is appropriate to the customer's needs in terms of simultaneous calls, uptime and redundancy.
As a guide, each concurrent call will use 85Kbits in both directions with the G711 codec and overheads.
For easy calculations and a little bit of headroom allow 100Kbits per concurrent call when planning data requirements. This applies for SIP trunks or hosted seats.
N.B: rough guide for uptimes and redundancy of fixed line data products:
- ADSL and SDSL has a normal target uptime of 99% (not guaranteed as xDSL products do not come with an SLA) with no redundancy on the line.
- FTTC through Openreach has a normal target uptime of 99.99% (not guaranteed as it's a VDSL product with no SLA) with no redundancy on the line.
- EFM and leased lines usually come with a target SLA and redundancy built-in.
- With each of those range of products the price goes up accordingly.
We do provide SIP trunks for compatible PABX systems, Asterisk included. Please consult your account manager for pricing.
Please ensure your SIP trunk connected system is properly secured. You are held liable for all calls dialled with your accounts.
Support are able to provide you with some of the essential details for the SIP trunk to aid setting up your PABX to connect correctly.
However you have to be familiar with the hardware you're connecting as there are many PABX systems out there and we don't know them all!
Support will be happy to try and assist you as far as possible in getting the trunk online.
With any deployment of a PABX intended for SIP, we recommend reading and understanding the following ITSPA best practises document:http://www.itspa.org.uk/ (PDF link)
SIP trunks by default are limited on the number of outbound calls they can make, to reduce the risk of fraudulent calls.
If you need this limit increased beyond the maximum available and configurable in your control panel, please speak to your Account manager and please have the relevant account number to hand.
Did you put the IP address of the PBX in the Direct IP field? If so, it will not allow you to register the trunk!
The "Direct IP" method allows you to receive calls directly to the IP address of your PBX without registering the SIP trunk.
If you are intending to register the trunk with your PBX to make outgoing calls out then leave that field blank and the platform will allow you to register.
If you continue to have problems registering the trunk after this please contact Support who will be happy to assist.
Please have the SIP trunk number to hand and your reseller account number.
Any hardware not purchased directly from us will need to have all of the correct settings to work correctly with the service.
Ideally the device should be provisioned to ensure the firmware version and account settings, dial plan etc are all correct.
Whilst support can try to assist you with this there are no guarantees the equipment will be compatible and it is YOUR responsibility to set up your customers equipment which is not purchased from us.
Please note that a provisioning charge will apply for 3rd party hardware that Support has to configure and provision to an account.
If it can't communicate with the provisioning service, it won't get the necessary details automatically.
Try forcing the DNS settings on the phone to the OpenDNS IP addresses below.
- OpenDNS: 208.67.222.222 and 208.67.220.220
If that doesn't work, please check the user-name and password and re-enter them if necessary for the SIP account within the phone.
If it still can't get the settings please contact Support.
Please give them the account number, extension, device name, as well as the MAC address and/or Serial Number to confirm it is set up correctly for the provisioning service.
Simple - use "Call Groups".
Create a Group in the account, and add the internal (200, 201, 202 etc) and/or NTS Destinations (e.g mobiles) you wish to be rung.
This Group will then appear in the drop-down box in the "Telephone Numbers" section, where you can point the number to the Group.
Please be aware that if you set the failover destination to "None", the call will drop if it gets that far.
This is easily achieved with a Time Profile.
Add a Time Profile to the account and set the period during the day you want it to be active, then set the appropriate destination for active or "open" hours, and inactive or "closed" hours.
Remember to save so the Time Profile can then be selected in the Hosted PBX.
You can also layer the Time Profiles so that more hours are filtered, e.g. lunchtime.
Add the number to divert to as an "NTS Destination", then within a Time Profile set the Inactive Destination (out of hours) to that NTS Dest.
Ensure the Time Profile is made active by pointing a telephone number to it, or including it in the logic chain of the Hosted PBX.
Dial 1571 from the handset and select option 0 for "Mailbox options".
There are separate messages for unavailable and busy/engaged.
Press 1 to record the unavailable message, press 2 to record the busy message.
Follow the prompts through to record the greeting.
The account also has a central mailbox, accessible by dialling 1572 from any phone registered to an extension in the account.
The "Custom Prompts" section allows you to upload just that - custom prompts. Voicemail messages uploaded via this page will not and can not be assigned to a mailbox.
Voicemail messages can be recorded by dialling 1571 (or 1572 for Central Mailbox) then option 0 and follow the spoken prompts.
If you have a custom pre-recorded or professionally recorded voicemail message that you want assigned to a mailbox please contact Support who will be able to advise.
Please include the SIP/hosted account number and extension mailbox.
Navigate in the account to "Custom Prompts" then click "Record Prompts". Follow the instructions on that page to record custom prompts from the handset.
You can also have your own custom prompts recorded and upload them to the account, we recommend uncompressed WAV format for the best quality as the platform will re-encode all uploads.
You can, this is achieved by adding the destination number as an "NTS Destination", and then pointing the incoming telephone number's destination to the NTS Dest.
Hosted PBX features can also be used to filter the calls to the NTS Dest, e.g. Time Profiles and IVR menus.
For the Hosted PBX logic to work as expected, disable voicemail on your diverted mobiles. If the mobile's voicemail answers the call the logic will fail, unless you want the voicemail to answer it.
Please be aware that it takes anywhere between 1 and 6 seconds to dial the mobile number, so allow enough time in the logic for the platform to make the call. If your call disconnects after only a couple of rings increase the ring time in the group from the 20 second default.
Absolutely. Firstly add the mobile numbers in as NTS Destinations, then add those NTS Dests in a new Group.
All you then need to do is point the destination of the incoming telephone number to the Group.
Dial 160 from the handset to initiate a loopback echo test. This is a free call and can be dialled with SIP trunks or hosted seats.
This could be one of several issues, particularly to do with your internet access.
We recommend an internet router with QoS enabled to prioritise VoIP, and unbranded Thomson/Technicolor ADSL routers have this enabled by default.
If that doesn't help or is not an option, try reducing the bandwidth the handset uses by changing the codec to GSM where available.
If that doesn't help, try rebooting the internet router and the handset.
A dedicated VoIP phone typically plugs into your internet router and is designed to make calls via a VoIP network.
An ATA, or Analogue Telephone Adapter, allows you to use an existing analogue telephone on a VoIP network.
This is useful if you are already familiar with the phone you currently have, however ATAs will add another layer of complexity into the mix.
Some phones can connect to your analogue telephone line as well as the internet, for example Siemens cordless IP phones (e.g A580IP, N300IP), allowing you to use one phone for both landline and VoIP calls.
As to which is better, it really depends on the situation you want the IP phone or ATA.
Depending on what functionality you require (call transfers, place on hold, group pickup etc.), dedicated IP phones almost certainly are more suitable.
If you already have a network of phones that lead to a device, an ATA could be a drop-in replacement for that device to enable VoIP functionality on those phones.
We prefer IP phones over ATAs, however the choice is both yours and your customers to make.
In X-Lite, the SIP account user-name and password are those for the seat/extension (typically 200).
Go to Seat Setup and use the user-name and password for that seat/extension.
If you have any further issues please contact Support.
Go to Options -> Advanced -> Network. Untick 'In times of network inactivity...' -> OK.
If your call disconnects or goes silent after 20 seconds (not 30 seconds) this is normally an internet problem, commonly to do with the router used.
For further assistance in solving these problems please contact Support. Please have your hosted SIP account number ready to assist your support representative.
Firstly, within the account there is a section called Conferencing. Within that, you must set the Conference PIN and the Admin PIN.
- On platform: dial 155. On any other UK phone: dial 0843 5575575. You can also add a local geographic/international number into the account and point it to Conferencing in the destination.
- You are prompted to enter your conference number (usually the account number), followed by the # key.
- You are then prompted to enter the conference PIN (you can either enter that or the Admin PIN), followed by the # key.
- You are then prompted to say your name and then press the # key.
- If nobody else is in the conference you will be played some music to tide you over until somebody else joins the conference.
The normal procedure for a conference call is for you to dial in as above, with the Admin PIN, and to give your participants dialling in the following details:
- Conference telephone number
- Conference number
- Conference PIN
Admin users have the ability to eject the last entrant in the conference, and block further entries into the conference. This is accessible by pressing * in the call when dialled in with the Admin PIN.
If you require more conference rooms, or have any questions not answered here about the conference service, please contact Support.
Any of our IP phones that we sell are fully supported with our hosted conferencing service, and they can access this by dialling 155.
A dedicated conference IP phone suitable for a round-table discussion is the snom MeetingPoint, this has excellent call quality and full hosted seat features and IP phone functionality.
Please see the data sheet for this unit and hardware pricing, and contact your account manager if you wish to purchase one.
Please note that whilst 3rd party conference/star phones may be compatible we will not support them if you decide to connect them to our network.
Contact Support with the country and/or area code you're looking for and they will be able to source your required numbers for you.
Please note there is no free choice for international numbers due to supplier limitations.
Please include the SIP/hosted account number in the request, and please allow at least 10 working days for the numbers to be made available for you.
UK and USA geographic numbers are easily sourced, other countries take longer.
If your customer reports that they can't dial a telephone number for whatever reason please follow these steps:
- Obtain the telephone number they're trying to dial (this sounds obvious but you'd be surprised just how hard people make it for you to do your job), without this we cannot raise a fault.
- Get the message that they receive, and the voice (male/female, British/Non-British) or tone they hear or SIP error code, and how long it takes to get to this stage (i.e. immediate, after 30 or 60 seconds ringing).
- Common system voices (in parentheses the remedy) in a distinctive British male voice are:
- "You have dialled the wrong number, please check the number and dial again." (ask them to double-check the number they're dialling is correct)
- "The number is not answering." (this is normally heard after 1 minute's ringing where there's nothing on the other end to pick up, if it is heard immediately please raise a fault as below)
- "That number has been blocked." (check the SIP trunk configuration to see if it will allow that call to be made)
- "Simultaneous call limit reached." (for a SIP trunk check the channel allocation for it, for a hosted seat extension speak to Support)
- "That number is currently blocked at this time." (for the SIP trunk check the timings available for it to call, if it's unticked it can't make outgoing calls)
- If the customer is being told they have dialled the wrong number, please check that the number is in service and accepting calls by dialling it from a PSTN landline or mobile, or alternative network.
- If they receive a message on-platform that is not male-British, ascertain if the call is connecting to a voicemail on the far end or not by checking the account's Call Records.
- For wrong numbers on SIP trunks, please check the dial plan of the PBX to make sure the outbound call is allowed to a) connect to the trunk, and b) correctly dial, i.e. with no prefix.
(One thing to be aware of for wrong numbers for international destinations is that often non-geographic numbers in an international destination can only be called within that country.
BT have the same policy for inbound calls to the UK, you can only call non-geographic (03/05/070/08/09) numbers from inside the UK.
If an international non-geographic destination fails, try calling the number from a landline. If it fails to dial from a landline then obtain a geographic number to call instead.)
If you've completed the above checks and the call is not passing out correctly, please raise a fault with Support including the following information:
- Whether it has been successfully called on-network in the past
- Dates/times of both successful and failed calls
- SIP/Hosted Account number (obligatory)
- Originating telephone number making the call
- Originating SIP trunk ID or Hosted Seat/Extension
- Destination telephone number with fault
- Whether you can successfully dial the destination off-network from an analogue line and/or mobile and/or other SIP provider, if yes what network(s) and dates/times of successful calls
Upon receipt of the fault, you will be allocated a fault ticket reference which should be used in all email correspondence about the fault.
Easy: change the destination of the telephone number to "Fax", then adjust the destination email address in the number configuration if needed.
If you want the "from" email address to be your company's domain, please contact Support with your SIP/hosted account numbers and they will be able to action this for you.
Yes we do. You have to have the corresponding 0845 number in the account, and it has to be pointed to the same destination, i.e. not route differently.
We also have 0343 numbers if you wish to migrate in a similar manner from the corresponding 0843 number.
Please speak to Support about adding an 0345 or 0343 number into the account.
Absolutely, our 999 emergency services access means those with a SIP trunk or hosted seat extension can call the emergency services without restrictions.
Other providers claiming not to provide the access actually can dial 999 but likely can't pass the address.
Our numbers send the emergency services address database with new entries and updates several times a day.
You can update the address details for accounts, SIP trunks, hosted seats and telephone numbers any time of day or night from within the customer's control panel.
For multiple sites with SIP trunks or fixed (i.e. non-nomadic or non-roaming) hosted seats you should have 1 number per site set for 999 address purposes.
e.g. 3 SIP trunks in one account, each on different sites = 3 numbers needed.
Any of our telephone numbers are suitable for 999 address purposes. If you have any questions about this access please speak to Support, and have your SIP/hosted account number ready.
IMPORTANT INFORMATION: The Customer understands and acknowledges that this service allows calls to the emergency services numbers 999 and 112 and that calls to these services may fail if there is a power cut or if the customer's broadband connection fails.
The customer understands and acknowledges that the address provided by the customer will be passed to the Emergency Services and will be used in location finding during a 999 call.
The customer understands and acknowledges that the address provided is the location that the service will be used and that it is the customer's responsibility to notify the service provider of any changes to this information.
Draytek routers, including but not limited to the Vigor 28xx and 29xx series, have broken SIP handling that closes the port the PBX or IP handset is using, regardless of if it's in the DMZ or not.
This means that SIP trunks, or SIP handsets connected to hosted seats, will not stay connected to our platform and may have problems including:
- Dropped registration
- Dropped calls
- Incoming calls don't ring correctly
Draytek's so-called fix for this problem with Vigor 2820 routers is to downgrade the firmware to version 3.3.3, however this might not work for routers bought in the last 12-18 months.
Their SDSL router (Vigor 3100) seems to be unaffected by this, it is only their ethernet WAN and ADSL routers.
Our recommended replacement for a Draytek dual-WAN router is a Cisco RV042.
This takes 2 connections (of type PPPoE, Automatic/Static IP or PPTP), and is suitable for any type of line with a 10/100Mb ethernet interface, e.g. ADSL modem.
It also allows remote management to the router's web configuration, as well as VPN access to the network.
If you need more LAN ports you can connect a switch to it, and Cisco also do models RV082 and RV016. The RV016 is multi-WAN (up to 7 connections) 10/100Mb.
Please note when configuring for ADSL connections the default (automatic) MTU of 1500 bytes will often not work, you are recommended to use at most 1492 bytes.
You can indeed. Where you are unable to input our snom provisioning URL you may unlock any snom phone with any other SIP provider.
Please note this completely resets ALL SETTINGS on the phone, so back up the directory and any other settings before proceeding to unlock.
We must also have the MAC address and new seat account/extension in our provisioning service, in order that the snom phone will take the new settings.
The unlock can be downloaded from: SnomRedirect.zip. This can also be found under the "Documentation" section of the "Home" tab.
Within the download are instructions on how to unlock the snom phone. If you have any queries about how this works or need assistance with this then please contact Support.
Some routers don't need the ports opened on them (e.g. Thomson/Technicolor TG582n or Cisco routers, e.g. RV042), however if required please accept traffic on the following ports:
- UDP port 5060
- The UDP SIP port your PBX or seat's IP phone(s) uses, normally this is 5060 but can be changed (e.g. 6000 or 7000)
- UDP ports for RTP, normally this is 10000 to 20000 however if your PBX / IP phone has a more specific range use that instead
If you need to allow traffic on your firewall specifically to and from our IP addresses for hosted seats and SIP trunks please use:
- 37.157.52.128/25
- 37.157.54.192/28
The recommendation is to register seats and SIP trunks using the URLs and not IP addresses.
Directing incoming calls to an IP address, or SIP termination, is done via a SIP trunk.
For a new customer, create the SIP trunk, and use the Direct IP method. In the boxes input your IP address to terminate calls to.
Inbound numbers can then be mapped for inward calling in the E.164 format (e.g. +442079460000). If the + is not compatible, in the Advanced tab you can toggle to remove the +.
If you need to terminate to multiple IP addresses create a new SIP trunk for each unique IP address.
For termination to a domain name (e.g. for SRV purposes) please contact Support with your Hosted/SIP account number in question.
The Gigaset DECT base units are not a PBX, SIP trunks are meant for a PBX with advanced call handling (e.g. ISDN).
Consequently, SIP trunks are not compatible for Gigaset DECT base units intended to receive calls and SIP trunks will never be compatible in this manner.
For full compatibility with the Hosted PBX, use seats for the Gigaset DECT base units.
If a problem with SIP trunks connected to a Gigaset IP DECT base unit (such as the N300IP or A580IP) is reported to us we cannot and will not support it. Use seats within the Hosted PBX instead.
If required we can allow a strict subnet of IP addresses to register with a SIP trunk, for example if on a leased line there is a /28 range of IPv4 addresses then we can allow any IP address within that allocation.
If they are completely different IP addresses then the answer is no. If you have (e.g.) Plusnet and TalkTalk lines, these will have different allocations of IP addresses.
You should use separate SIP trunks for the outgoing routes and then configure the PBX to route outgoing calls through the SIP trunks as needed.
Inbound fail-over for numbers pointed at SIP trunks is a work-in-progress and is high on our development list.
