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A SIP URI is the SIP addressing schema to call another person via SIP. In other words, a SIP URI is a user's SIP phone number. The SIP URI resembles an e-mail address and is written in the following format: x@y:port (x=Username, y=host|domain|IP)

Example: [email protected]

A general description of SIP addressing is at wikipedia:SIP address. The addresses, which use the same user@domain... format as e-mail addresses, allow an individual Internet telephony user to be reached directly online without passing via the public switched telephone network or incurring the corresponding tolls.

A SIP address may be used as a destination to which to forward a DID number, as a target for an individual speed dial entry (*75xx) in a user address book or as a means to transfer incoming calls into your extensions or numbers from outside Internet servers.

  1. One is to send calls to an external SIP URI, via your DID number,
  2. A second option is to receive calls via SIP URI, we can achieve this using our DID number or an internal extension from a sub account.
  3. The third option is to use a Virtual number.
Please note SIP URI is only functional through an external source, it is not intended to be used internally with


Send calls to an external SIP URI address

You can route incoming calls to your DID numbers using a SIP URI address from other companies. You will need to create a string like youraccount@yourip to which you can route the DID(s).

Make sure the other company or provider supports the use of SIP URI

Creating a new SIP URI

To make a SIP URI that can be used with multiple DIDs, use the {DID} tag. The {DID} expression will be replaced automatically by the DID number as it is listed in the "Manage DID section".


[email protected]
[email protected]:5080
[email protected]

Dialing out a SIP URI or using a SIP URI as forwarding does not generate cost (Outgoing call).

two newer options for SIP URIs

  • CallerID Override: Permits you to override the callerID that will be received on the receiving end of the SIP URI.
  • CallerID E164: Your CallerID will become E164 compliant and thus show on the receiving end as +12123262233 (+(countrycode)(areacode)(number)).

Extended SIP URI Format (BETA)

In addition to the standard format outlined above, an Extended SIP URI Format can also be used. This Format contains additional options as indicated below:


To specify any of the additional parameters of the Extended SIP URI Format, you must introduce one or more colon (:) characters between the username value and the @ character. Parameters are optional and read from left to right. To supply a particular parameter, it is necessary to precede it with the appropriate number of colons to its left. For example, to specify a username, password and transport, the SIP URI would be as follows:


Specifying the username, password and authname would be done as follows:


The additional parameters of the Extended SIP URI Format are described below:

password: This is the plain text password to be used when authentication is required by the destination endpoint.

md5secret: Alternatively an md5 representation of the password can also be used instead of the plain text version when authentication is required by the destination endpoint.

authname: An optional authentication name can also be supplied as a parameter, which will be used instead of the username.

transport: A specific transport type can be specified for the outbound connection. Valid values for this parameter are 'tcp' for the TCP transport, 'tls' for TLS encrypted signalling, as well as 'udp' for UDP transport (the default).

Enabling TLS on SIP URI

If you wish to proceed on having TLS enabled on your SIP URI, open a ticket from your customer portal or email us at [email protected] by providing the following details:

  • the extended SIP URI format that you have created.
  • The server(s) you will be using with this SIP URI.

Creating a phone book entry

A SIP URI may be associated with a phone book or speed dial entry in the same manner as any other telephone number.

See Phone book#Create a Phone Book Entry with a SIP URI.

Pb entry sipuri.jpg

SIP URI: Here you can either select Use Existing or Create New to assign the SIP URI to your Phone Book entry.

This replaces alphanumeric addresses (such as sip:[email protected]) with numeric abbreviations (such as *7501) which can be easily dialled from IP phones that only offer a numeric keypad.

Codec Negotation

By default when you route your incoming calls to an external SIP URI address, the system sends the INVITE allowing all supported codecs (ulaw, g729a and GSM). In that case if you want to use a specific codec (from the supported ones) you need to restrict that in your end. For instance, if you are using an Asterisk/PBX System and only wish to use ulaw codec, you will need to make sure to have the following settings in the trunk:


Receiving incoming calls from a SIP URI

Using your DID number

You can receive SIP URI calls using the following format {Number}, this can be used with your local US and Canada numbers, so they can be reached from outside.


This format of a SIP address must follow this SIP URI scheme as a means to reach subscribers.

Another variant, also valid, is to specify the specific server on which your DID is registered, ie:

sip:[email protected]

Please note that the option to dial is more reliable than using the server as you won't have to specify the Point of Presence. Using the server instead, you will have to dial the correct server that the number is using.

Using your sub account internal extension

When you assign an internal extension for a sub account, it can also be used as an external SIP URI. For example, if your extension is 2, you could be reached directly via SIP from another network with a URI like this: [email protected]

(Replace by the server you are registered to, 100000 by your account ID and the 2 by your internal extension).

Important: no call flow or filtering can be applied to calls make to the external SIP URI. Calls will immediately ring the device registered to this sub-account.


Using a Virtual number

Virtual SIP numbers are similar to standard DID numbers. The major difference is that virtual SIP numbers are not accessible via "PSTN". They can only be reached via "SIP URI" over internet. For example, if you have a DID number with another provider and they support SIP URI Forwarding, you could forward your number to a virtual number at just like if it was one of our numbers.

All virtual numbers consist of the following digits: 11 + Accountcode + 3 digits of your choice for a total of 11 digits. The final uri will be that number followed by the @ sign at one of our server. If you intend to send the calls to a phone or adapter, you'll need to point it to the proper server.

Example SIP URI: [email protected]


SIP URI using the Reseller Interface

The feature is available for your client through the Reseller interface. You must enable this feature in your package in order to give them the ability to leverage this.

Note that the DID must be linked to your client. (Reseller > Manage Client's accounts > Click on Manage client where your client.)

Go under the navigation bar on [Reseller] then click on [Manage Rates & Packages]

SIPURI Reseller 1.png

Click on the Edit button to edit your package, or click on [Create a new package] to create a new one.

SIPURI Reseller 2.png

Go under the [Reseller System Configuration] Tab, and on the section "Type of configuration" select: [Package Configuration],

SIPURI Reseller 3.png

Then scroll down and find the feature "SIP URI", and enable it.

SIPURI Reseller 4.png

1) To add a SIP URI entry for your client, or to help your client adding one. Go under the [Services] at the left navigation bar, then on [SIP URI]

SIPURI Add.png

2) Click on the tab Add New SIP URI and enter the URI and a Description for the new entry. The description field will help the search bar or searchability.

SIPURI Add2.png

3) Click on the [Save SIP URI] button.

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