BusinessHome
AA

Help setting up and using VoIP

This page has various hints and tips for setting up VoIP using our servcie. We will add more details over time.


Using voicemail

The voicemail service we offer is very simple. To use it you need to access the control pages for your number and set up a few things.

Voicemail setup
EmailYou have to put an email address for where we send the recording of your voicemail. If you do not, then any message used for voicemail will only be an announcement and no recording will be made.
Recording typeSelect MP3 or OGG format
VMSelect a no answer time or for unlimited time (i.e. only on busy or unavailable).
TargetYou can select that all calls be voicemail to email if you like (i.e no timeout).

Once set up, any calls to your number that pass the no answer time will get a message and a recording is made. The recording is emailed to you if an email address is set up.

If you have a sip phone set up for the number you can call 1571 from the phone and change your outgoing message if you like.

Note: With some numbers you can select voicemail to email goes direct. If you select this then there is a generic outgoing message only and the voicemail is emailed to you in a fixed format. Generally we recommend you select for calls to go via our call server rather than direct.


Call gate

Once in voicemail (recording or announcement only) the user can key a digit. If there is an also ring number set up for the corresponding digit then the call is put through to that number as a divert. This allows you to record an announcement and give the caller various options.

If there is a fail number set up and some also ring numbers then the fail number is used as a default if no divits are keyed. If there is a fail and no also ring then the fail number is used immediately after the announcement.

Using DDI

In some number ranges (e.g. 033) we offer DDI (direct dial in) blocks. These are blocks of 10, 100, 1000 or even 10000 numbers routed to one destination. You could have several blocks.

To manage the blocks each block has a single entry in our control pages with an X where any digit can be. e.g. 0123456789X for a block of 10, or 012345678XX for a block of 100. This allows you to control where the numbers are sent (e.g. SIP, IAX, H.323). Incoming calls to any of the numbers is delivered as per this setting.

To make outgoing calls you can use a SIP or IAX athenticated login using the number (with the X in it) as the login, and the password you set. This is just the same as using single numbers. One difference is that you can set the CLI (calling number) to any of the numbers in the block. If you set the CLI to one of the numbers then that is what is used. If you do not set the CLI or set an invalid one then the number using 0 instead of X is used. If you have a presentation numer then this is used instead when you do not set a CLI. This means you can chose to use any of your numbers, or a presentation number if you have one, on a per call basis from your own phone system.

Billing appears against one billing number (the number with X's in it). The calling number shows on the bill.

We can arrange for individual numbers to have their own entry even when withing a block. In this case you pay for the block and also for the individual numbers you have chosen. The individual numbers can have their own management and billing as normal and are effectively removed from the block. The exception is that they can still be used as a CLI when calling using the block login. This can be useful where most numbers are handled by your phone systems but you need some specific exceptions handled differently.


General VoIP setup

When setting up SIP you need to know your sip number, e.g. 01234567890, that we have allocated to you, and your password. You will have been sent these when you requested the number.

The other bit of information you need is the name of the call server. In different sip devices there are different names. In some cases you may have to enter a proxy server and in some cases a registration server and in some cases both. The server is your sip number followed by .call.me.uk, e.g. 01234567890.call.me.uk

If you have to enter something that looks like an email address, or has sip: at the front it probably needs your number, an @, and then your server, e.g. 01234567890@0123456789.call.me.uk

If you are asked for a realm, just put asterisk


Nokia E61i

The Nokia E61i has build in VoIP client which works well on WiFi. It reportedly works on 3G, and even works on GPRS, but data costs must be considered when using VoIP in this way. Using over WiFi is very practical and works well.

One number? Our VoIP service can be configured with an Also Ring setting to ring a number of other phones - why not set your VoIP number to also ring your mobile number. That way you can give one number that rings your mobile whether you are at the office or not. The VoIP will ring first if in wifi range and registered. If you do answer the mobile call then you are charged for a call to the mobile at our normal rates. You may also want to disable voicemail on your mobile.

SIP settings

The first setting you need is in Menu->Tools->Settings->Connection->SIP settings. You need to use Options to create a new setting from the default template.

SIP Settings
Profile nameAny string you like, e.g. your SIP number.
Service profileIETF
Default access pointSelect an access point, such as a previously set up WiFi AP.
Public user nameYour full sip username, e.g. 01234567890@01234567890.call.me.uk
Use compressionFor WiFi use set to No but you may want to use compression when using mobile data.
RegistrationAlways on
Use securityNo
Proxy server
Proxy server addressYour call server name, e.g. 0123456789.call.me.uk
Realmasterisk
User nameYour sip number, e.g. 0123456789
PasswordYour sip password.
Allow loose routingYes
Transport typeAuto
Port5060
Registrar server
Registrar serv. addr.Your call server name, e.g. 0123456789.call.me.uk
Realmasterisk
User nameYour sip number, e.g. 0123456789
PasswordYour sip password.
Transport typeAuto
Port5060

Internet tel. settings

Having established the sip account, and you can set up several if you like, you can tell the phone which to use for making internet calls. Go to Menu->Tools->Settings->Connection->Internet tel. settings, and use Options to add a new entry.

Internet tel. settings

Default call type

Having set up an internet telephone profile you can set the default calltype to use the sip rather than cellular. Select Menu->Tools->Settings->Call and change Default call type to Internet. When your phone is not sip registered (i.e. out of wifi range) calling a number using the green key will make a cellular call. When it is sip registered it will make a sip call. Using the middle button instead you can select the call type manually.

NameAny string you like, e.g. your SIP number.
SIP profilesSelect which profiles to use.

Trixbox

Configuring a trixbox to use us as a trunk:-


Trunk name = last 6 digits of number

peer details

context=form-pstn
dtmfmode=rfc2833&info84411
fromuser=(username i.e number)
host=81.187.30.110(server host ip)
insecure=very
qualify=yes
secret=(secret i.e password)
type=peer
username=(username i.e number)

user context = the number

user details

context=from-trunk
fromuser=(username i.e number)
secret=(secret i.e password)
type=user
username=(username i.e number)

Username i.e number:secret i.e password@81.187.30.110(host ip)/Username
i.e the number

Note that on our config pages mark as IAX with the same username and password on the right with the endusers TRIXBOX IP. Pres-Digit would need to Be 11 (0...) otherwise the trixbox needs reconfiguring.