*= Forwarding to internal extensions is okay. Forwarding to external parties: signalling OK, but speechpath only after 16 seconds!!!

Example of Deutsche Telefon Standard Trunk gateway group settings.

Outbound caller ID

As just the last digit of the trunk number block is changing, you can select just the last digit (in brackets) and put it into the variable $1. As the provider wants the signalling number in the format +(country-code)(city-code)(number), you have to enter this number into the next line (here: +49891234567$1, where the $1 represents the changing part).

The provider will take care about the number representation to the called party. He will present e.g. 08912345678 to the calling party.

Asserted Identity

These fields are used specially for external forwardings. Some providers need special settings in order to present the correct number at the receiving party. For Deutsche Telefon Standard Trunk, you can the entries from the outbound caller ID fields e.g. ^\d\d(\d) and +49891234567$1

Inbound DIDs

For incoming calls, you can use the regular expression, generated by the setup assistant. In your case you would have to adjust the city-code and the pilot-number, according to your line-settings.

^(?:(?:(?:0049|\+49|49|0)?89)?12345)?(.*)

This expression cuts all possibly available country- and city-codes and the pilot-number from the incoming number and only the extension is remaining (here: 678). This will be forwarded to the Inbound Routes.

Inbound caller ID

For incoming calls, we will add an additional 0 in front of the external number of the caller, in order to use the callback-feature of the phone (e.g. 0089987654321). During the call-setup the additional 0 will be automatically removed.

Gateways

This field will be empty when you create this Gateway group. It will show later the assigned SIP gateways.

Example of Deutsche Telefon Standard Trunk gateway settings 1/3.

Example of Deutsche Telefon Standard Trunk gateway settings 2/3.

Example of Deutsche Telefon Standard Trunk gateway settings 3/3.

Registrar

You can use here the DNS-name of the server.

Dial command

The dial command is the command which is used in the asterisk software. The term {number:1}  means, that at the dialled number (e.g. 0089987654321) the first digit is removed. If you don't use a line access code (in most cases '0' is used), you have to remove the ':1' ! ! ! In case you want to use the prefix-fields from the outbound rules, please add '{prefix} before '{number:1} .

Group

Select here the previously created gateway group.

Update remote party ID (CLIP)

To display the correct extension at external parties, you have to use the P-Asserted-Identity (PAI) setting. If this is not selected, only the head-number (e.g. 0891234567-0) will be presented, no matter which extension is used for dialing.

Advanced Parameter

You have to enter 3 different terms into the advanced settings: 'fromuser=08912345678',  'fromdomain=sip.dtst.de' and  'outboundproxy=sip.dtst.de' . Enter these terms separately into in the value-field and press the '+'-sign.

Underneath you will see then a preview of the entry in the sip.conf file.

Example of Deutsche Telefon Standard Trunk Inbound routes.

Rule

Enter here a name for the according rule.

Date / Weekdays / Time

With these settings you can configure a time-controlled routing to different targets.

Pattern / Target

In the pattern you define which part of the incoming number is used to forward the call to the according extension.

In our example we receive already the correct extension from the gateway group. Therefore no further number-manipulation is necessary.

When the PBX finds an according extension it will route the call to it.

But you can add here exceptions from this rule, for example for internal fax users or waiting queues, etc.

Please have in mind the order of these rules, as the system is using First Match!!!