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I have created a freeswitch learning exchange group, 45211986. welcome to join.
Did stands for direct dial-in. It is a concept proposed by the operator. With this number, the operator can call to the VoIP system and configure a gateway by calling freeswitch, specify the carrier's access number and password. When a call is made to this number, the carrier will send the number to freeswitch.
For example, if I use a landline call number 84366666 and want to route the call to freeswitch, I need to configure the did gateway and add the dial plan.
Add Gateway:
Add an XML file to the/CONF/sip_profiles/external directory and set realm, username, and password. You can specify whether to register on the gateway as needed.(2) configure the dial plan
Add a dialing scheme:
In the dial-up plan, you need to specify the processing logic of the corresponding did number after the incoming call. You can flexibly customize the processing logic as needed, such as entering meetings, faxes, and queues.
Modify public. XML in \ conf \ dialplan and add the following content:
<Extension name = "sipprovider"> <condition field = "destination_number" expression = "^ 84366666 $"> <action application = "transfer" Data ="84366666XML default "/> </condition> </extension>