storm allows you to route calls to SIP endpoints. The actual SIP endpoints are set up for you by Content Guru engineers, and you then assign them the required extensions or telephone numbers (using Routing > Extensions > Extension Mapping). You can also assign a SIP endpoint a service type, to set up (for example) the required recording options.
storm routes internal calls to a SIP endpoint using the extension dialled.
External calls can be routed via a FLOW script or using access point routing configured in STUDIO.
Unless the SIP endpoint treatment specifies otherwise, the call is placed to the number assigned to it.
Select Routing > Extensions > Add Extensions.
The fields displayed allow you to control the number that is presented to the SIP endpoint, including stripping all or part of the dialled prefix, adding a new prefix, and stripping and adding a suffix to the dialled number. The presentation number can thereby be manipulated in order to present a number to the endpoint in a form that the endpoint recognises.
Prefixes and suffixes are removed and added in the order in which the fields appear. For example, if the prefix '400' is removed from a dialled number and the suffix 456 added, the system will remove 400 before adding 456.
Example:
If the number '400' is defined as a SIP extension prefix, a user wishing to dial a particular SIP endpoint could dial the number '400123'. The prefix 400 denotes that the call is to be routed to the SIP device, and the number 123 is the extension number that will be recognised by the SIP endpoint. In the example graphic above, the number 400 is set as the 'prefix to strip', and a suffix of 456 is the suffix to add. The full destination number presented to the endpoint is thus '123456'. This number can be used by the SIP device to route the call to the correct user. In this way, numbers dialled by UC users can be translated to match the numbering scheme recognised by the SIP device.
A SIP extension range is used to create a batch of extensions to use as SIP extension prefixes. A SIP extension range is used to route calls beginning with any number within the range to a single SIP endpoint. The same prefix and suffix settings can be defined for a range as for a single prefix.
Note: if a SIP extension prefix or range overlaps with a standard extension number, the extension number will be dialled instead of the SIP endpoint. Extension types are checked in the following order: extension, SIP extension prefix, SIP extension range.
To create a SIP extension prefix, select the Create SIP Extension Prefix option, then enter the prefix to use as a SIP extension prefix, and optionally set prefix and suffix settings for the dialled number. Then click Add Extension Range.
To create a SIP extension range, select the Create SIP Extension Range option, then enter the start and end numbers for the extension range. (Ranges can only be created with the same number of digits - for example, between 10 and 99, but not between 10 and 100.) Optionally set prefix and suffix settings for the dialled number. Click Add Extension Range to save your changes. If you have selected the option Add Individual Extensions, all the extensions in the specified range (which do not already exist) will be created.
To create a SIP extension prefix for use with a SIP endpoint treatment, use the SIP Extension Prefix field to enter the destination number.
For information on the other options in the Treatment Type drop-down list see the section Reference: UC/CONTACT Menu Items.
For the meanings of the settings on the screen, refer to the on-screen field descriptions and the following notes:
Field |
Notes |
SIP Endpoint |
Select the actual SIP endpoint. |
Change Onward Destination |
By default, storm will use the telephone number assigned to the selected SIP endpoint. To use a different telephone number, enter it here. |
Treatment Timeout (seconds) |
This field allows you to specify the duration (in seconds) for which the treatment will attempt to route the call using these primary instructions. If the call remains unanswered after this period, the call will be routed using its overflow instructions. If no overflow instructions are configured, the timeout will be ignored. |
CLI Presentation |
This field defines what is displayed to the SIP endpoint when a call is received. |
Note: only the recording mode and options are relevant for SIP endpoints.