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SIP DID Virtual Numbers
Over 65 Countries Available

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Ideal for SIP & VoIP Telephones
Use with Asterisk & FreePBX

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Caller ID on outbound Calls
From your account page you have the options to set the caller ID number (CID) to be sent with oubound calls.

The options are;

Anonymous: 'anonymous' is sent
Caller ID DID Number: the selected DID number will be sent
Set by user device: a valid CID sent by the user device will be sent.

Note: how exactly a recipient network chooses to display CID is not a metter under our control. For example, some networks will not display a CID unless the CID number originates from the termination network.

There are also some issues caused by what is known as Caller ID Name. CID name is only supported by some networks. We set CID name = 'anonymous'. If CID is 'Set by user device' we will pass the CID name sent.

If you choose 'Set by user device' this is only likely to work if the CID is properly sent and probaly more suited to pbx users.

What is a SIP URI?
URI Means "Uniform Resource Identifier". A SIP URI is used to identify an entity to be addressed via SIP, just like an e-mail address identifies the e-mail recipient. The general format of a SIP URI is user@domain:port

"user" identifies the particular user to address. The user is usualy the extension number, or sip user number, if calls are being sent to an endpoint like a softphone or voip telephone. The user can also be a phone number, if this is supported by the SIP client (sip trunking for pbx).

"domain" can be a fully-qualified domain name or a numeric IP address.

"port" is optional, and if not specified then the standard value of 5060 will be used.
Examples of valid SIP URIs: or user24@

Do you provide a sales invoice
yes, sales invoices and payment receipts are available to download from your account page as pdf files.

CID set by user device - for pbx trunks
Configuration example for Asterisk 1.6 pbx to send CID using sip trunk.

Enable Caller ID 'Set by user device' from your account page.

In your asterisk sip.conf file set something like this:

callerid = Anonymous

register =>

;here we place the sip settings for

;here is our example extension that wants to send a CID

insecure = no
canreinvite = no
regexten = 8000
dtmf = rfc2833
context = sip-phone
host = dynamic
type = friend
username = 8000
secret = 1234
nat = yes

In extensions.conf the CID would be set like this (we are setting 353539431111):

exten => _X.,1,Answer
exten => _X.,2,Set(CALLERID(name)=353539431111)
exten => _X.,3,Set(CALLERID(num)=353539431111)
exten => _X.,4,Dial(SIP/${EXTEN}@mydivert,60,r)
exten => _X.,5,Hangup

If you are using FreePBX here are some tips:

add sendrpid=yes to the trunk's PEER Details on the FreePBX GUI

The CallerID Number format is very strict. It must follow the format "abcdefg"<123456789>. The important bits are:
(a) you must have the string section even if it is an empty string.
(b) you must have the angle brackets
(c) the number itself must contain numberals only and can contain any number of digits, but must not start with a single 0.

Buy numbers for your VoIP service
Buyers can buy DIDs for their VoIP services such as internet telephones or other VoIP related products.

Geographical DID numbers purchased at can also be used with your own SIP provider. Simply update your phone number SIP settings from your MyDivert user account.
All incoming calls will then ring to your chosen SIP without diversion.

How can keep their rates so low?
We always try to offer the best prices possible and constantly aims to negotiate better deals with suppliers to pass this discount on to our customers!

By buying in bulk from major carriers and using state of the art technology is able to significantly undercut other international call providers. In addition, low overheads and a no frills approach enable our customers to benefit even more.

Do you have an API available?
Yes, we have an easy-to-use DID API (Application Programming Interface) that is available to our users at no charge. This API may be integrated into your website or application so as to facilitate the purchasing, configuration and maintenance of DIDs by your own customers. Please contact support for further information.

Is There a Maximum Call Duration per DID?
There is no maximum limit on call duration for the DID numbers.

If you are experiencing cut, or dropped, incoming calls please open a support ticket from your account area.

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Other Questions Asked

CID set by user device - for pbx trunks
How long does my credit last for calls?
Can I use my account when I'm in a foreign country?
Problems receiving incoming calls
Caller ID on outbound Calls
How can keep their rates so low?
What is the monthly phone number charge?
Do you provide a sales invoice
What is the difference between DID and DDI?
Is SIP mapping updated in real time ?
Which codecs do you support?
What is a SIP URI?
Do DID numbers Support Callback?
Can I Pick any DID Number from a list?
Do I Pay Per-Minute Charges?
Do you have an API available?
Do I have Access to Call Records?
Is Porting of Numbers Permitted?
Is There a Maximum Call Duration per DID?
Do you provide wholesale call termination?
How to configure FreePBX
Can I peer my DID number to an IAX address
What IP address do you send calls from
How to call Gizmo with a DID number
Buy numbers for your VoIP service
How to call my Free World Dialup account
Do the DID numbers display Caller ID
What is
Example - forwarding Calls to Hong Kong
What is a Virtual Phone Number?
Do I need internet or any device?
Can I have multiple numbers?
Making outgoing calls
Are the phone numbers individual?
Instant Activation for Phone Numbers
Call forwarding only when unavailable?
What dialing format do you use?
charges for incoming calls
Can I purchase toll free numbers?
discounts for annual payment
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