Knowledge base Technical details
Note: We are deploying a new call server, and so some of the features described are subject to subtle changes.
SIP/VoIP technical details
Our telephone services are designed to be simple to use whether you have a single phone at home, or a multi site office with phones on desks, mobile phones, and hunt groups. The services have a range of features which we are constantly improving. This page details how the features work. Mostly the features are controlled by our control pages and changes take effect immediately.
The system is designed around phone numbers just like traditional phones. A login to our phone service is a phone number. Internally we use international format starting + for phone numbers, e.g. +443333400200. However, most parts of the system allow you to use a national format number, e.g. 03333400200.
This means any VoIP equipment you use with our service has to have a real phone number. In simple cases that is just your phone number but in the case of a typical office it means every phone has a direct dial number even if it is also part of a hunt group of some sort. The number does not have to be visible to other people, i.e. calls could appear to come from a main office number for example.
Some features need numbers to work, such as hunt groups, which may not have a phone themselves. We charge for each number you have per month whether you have a phone connected or not. You can however reserve numbers which you can activate when needed, and these are a lower cost while they are reserved. Some really nice numbers cost more.
You can connect any suitable equipment to our service. This can be simple SIP handsets, or complete phone systems. Our service makes it simple to run a virtual phone system (centrex) by simply having SIP handsets, allowing internal calls between handsets for free and using simple short number dialling (like extension numbers).
Not all equipment supports all features in the same way. Whilst we will investigate any issues and ensure we meet the standards, we can't guarantee that specific equipment will work. We recommend SNOM handsets as we use them in our own offices and they work well.
Networking, firewalls, and protocols
We support SIP/2.0 protocol using UDP for call control. With the 'Voiceless' call server we proxy the SDP/media between customers and carriers. Our carriers and our equipment all support G.711 a-law and you should ensure you configure your equipment to support this.
Our call servers support IPv6 and enables IPv6 device to still talk to IPv4 devices as we proxy the media.
Please see our wiki for help with configuring your Firewall
Outgoing features are those that relate to making calls from a SIP handset or similar equipment.
- SIP registration: You can set up a password to allow SIP handsets to connect to your number and make calls.
- Emergency: Calling 999 or 112 connects to emergency services. This takes priority over centrex or other dialling. These calls are not recorded.
- Non-Emergency: Calling 101 or 111 connects to non-emergency police and NHS services respectively. This only works if you do not have 3 digit centrex set. The police may not be able to automatically identify your location if called through this service.
- Local dialling: We can set an area code which is applied if you dial a local number.
- Centrex: We can set a number of digits for short dialling, e.g. 3 digits. If you dial 3 digits then you are dialling the same as your number with the last 3 digits replaced by what you dialled. This is generally used where a company has a block of numbers and other phones in the company are in the same block so dialling a short code gets those phones. This is like dialling an internal extension number on an office phone system. Where the block you have is not the full 100, or 1,000 numbers, then you could dial someone completely different using short codes - there is nothing that restricts the short dialling to just your block.
- Cost limit: We normally restrict calls that could cost more than 20p+VAT for a one minute call. This limit is only 2p+VAT for international calls. Ask sales or support if you need to increase these limits.
- Recording: You can ask for outgoing calls to be recorded and emailed to you. The recording is stereo (one person each side) and can be WAV, MP3 or OGG format. We can even encrypt the recording.
- Presentation number: You can set a presentation number to be used as the calling number when you make calls. This is not used when making centrex calls (short numbers). You can also override this and send your number by prefixing 1470. You have to demonstrate you own the number you wish to use, and there is a set-up fee.
- 2nd Presentation number: We can set a 2nd number which your equipment is allowed to send and we pass on as your number. You have to demonstrate you own the number you wish to use.
- Withhold number: You can prefix 141 to withhold your number when making calls. Bear in mind some people reject such calls automatically.
Incoming features relate to what happens when someone calls your number. You can have numbers that do not have a phone connected, and are used to ring multiple phones.
- ACR: You can set your number to reject calls where the calling number is withheld. The caller gets a suitable message and are not charged for the call.
- Profile: You can set time of day (hour by hour and days of week) where your number is to work or not work. When out of hours you can set an alternative number to ring.
- SIP registration: You can set up a password to allow SIP handsets to connect to your number. When your number is called, registered SIP handsets are called. The system allows up to 10 simultaneous registered handsets per number. In general, if you want multiple phones to ring at once we recommend setting up separate numbers for each and set up a group ring.
- Deliver to you: You can set a hostname, username, and password for a SIP system to which we will try and pass the call.
- Group: You can set up a number of additional numbers that are rung at the same time. This is how hunt groups are set up. Each of the numbers can be set to delay ringing so you can make a cascade of phones that ring.
- Divert on fail: A number to ring if your phone (SIP registered) fails for some reason.
- Recording: You can ask for incoming calls to be recorded and emailed to you. The recording is stereo (one person each side) and can be WAV, MP3 or OGG format. We can even encrypt the recording.
- Voicemail: You can set up voicemail, to apply on no answer or busy, and can configure how long to ring before no answer. The recording is emailed to you.
- SIP address: Incoming calls can be made from the internet to sip:firstname.lastname@example.org and delivered just like a normal incoming call. The CLI is not trusted for such calls, so are sent to your SIP handset with a ? on the front, and not passed on if the call is diverted.
Note: ACR and Profile apply before considering registered SIP phones and group rings, etc.
Note: An incoming call can be trying to call group numbers, SIP registered phones, and delivery endpoints all at the same time. The first to answer gets the call.
Diverts and transfers
It is possible to have a SIP handset set to divert or transfer calls. This is treated as a call from your phone.
Diverts and redirects and group ringing work as if dialled from your number, allowing you to use centrex (short number), local, and so on. Note the mobile 9 prefix is a feature related to the SIM and not the number, so cannot be used in diverts and transfers.
Calls made using your VoIP login details are charged. Calls are only charged if answered. Calls are charged based on the number called and duration of call (to the second). Different rates apply for peak (9am to 6pm Monday to Friday), off peak, and weekend. We publish the dialling code to rate name, and the cost of each named rate. Not all numbers can be called (e.g. you cannot call premium rate numbers).
Where other numbers are made to ring as a result of a call to your number, such as group ring, out of hours number, failed number ring, divert, or transfer, then they are charged as if called from your number.
Where systems such as SIP2SIM have your VoIP login details and make calls, they are charged in the same way. These are separate to any charges for SIP2SIM itself (i.e. mobile leg of the call).
There are a couple of special cases that are worth noting.
- If your number is a freephone number, e.g. 0800, then you have a charge for any incoming calls to that number. This can depend on whether the call is from a landline, mobile, or payphone.
- Calling another A&A geographic number is normally free of charge.
Calls are charged from when they are answered. Recorded messages from the telephone network are not normally charged. Some systems answer then play a message which is a chargeable call. Most phones can indicate if the call has been answered or not ("ringing", or "call" or similar on the display).
Credit control and warnings
We provide real time usage details and call costs via the control pages. However, some charges may take some time before they are available.
We can provide warnings of unexpected high usage via email. See VoIP security for more details.
- As the system uses phone numbers the domain part is not relevant for registration and incoming SIP calls, but we recommend using the full E.164 number as the username part and aa.org.uk as the hostname, e.g. sip:+email@example.com. We also accept calls to tel: URIs.
- Whilst the system currently uses UDP only, it may be made to use TCP as per the standards if this is required in the future. UDP is preferred.
Call Data Records
Call and text charges are invoiced in arrears. The invoice has a line item for each billable number showing total charges. The PDF attachment includes a printable itemised bill for each billable number. The invoice also has an XML attachment which includes the full CDR for all calls so that you can process these automatically. If you do not have XML billing, ask accounts to enable it for you.
Some calls are just incoming to a phone or outgoing from a phone, but it is possible for a call to come in, and ring multiple other numbers, some of which are A&A numbers that ring multiple numbers, etc. In this case a single call can result in multiple CDRs for separate billable numbers. The CDRs log failed calls if they have got as far as actually trying to call a number (i.e. delayed ring group numbers that have not reached the delay time are not logged).
|Billable number||The CDRs are listed for each billable number. This is either the number of the SIP account or a SIM ICCID. The SIM is used where calls/texts are direct to/from the SIM (SIP2SIM) and no number is assigned.|
|Call/UID||A unique reference for the CDR entry and a group reference for records that relate to the same call|
|Call date||This is the date and time the call started.|
|Call type||Incoming||A call to a number that was not diverted or transferred.|
|Outgoing||A call from a number|
|Relay||An incoming call that was diverted or transferred to an outgoing number.|
|Origin||PSTN||A call from the telephone network|
|INTERNET||A call from the internet|
|SIP||A call from a registered SIP user|
|SIM||A call from an A&A mobile|
|SIP2SIM||A call from registered SIP2SIM end-point|
|AA||A call resulting from an A&A outbound/relayed call|
|Destination||PSTN||A call to the telephone network|
|INTERNET||A call to a SIP endpoint on the internet|
|SIP||A call to a registered SIP user|
|SIM||A call to an A&A mobile|
|SIP2SIM||A call to registered SIP2SIM end-point|