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Feature                                                           
Outgoing CallsYes
Incoming CallsYes
CLIP incomingYes
CLIP outgoingYes
Call ForwardingNo*
Call TransferYes
Call WaitingYes
DTMFYes
Anonymous CallYes
A-number forwardingYes
Faxsee chart below

*= Forwarding to internal extensions is okay. Forwarding to external parties: signalling OK, no speechpath.

 

 

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Table of Contents

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Example of Deutsche Telefon Standard Trunk gateway settings 1/2.

 

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

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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 P-Asserted-Identity (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.

T.38 support

In order to have the best fax support on the FXS ports, you have to deactivate the T.38 support option. A fallback to G.711 will be used instead.

Advanced Parameter

No advanced parameters necessary.

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

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