Configure the connection between freeswitch and the PSTN Telephone System
I. Environment Introduction
Currently, we use analog phones to connect to telecom operators and purchase their auxiliary telephone switchboard, barely achieving the company's basic calls. However, it is expensive and difficult to maintain the customer's telephone switchboard system and upgrade and add new applications.
As the scale expands, it is difficult to continue using the original telephone system. Based on the above problems, we decided to adopt the VoIP telephone system in the new business center. The server uses freeswitch, and the phones use Sip/IP phone or analog phone (using Voice Gateway for conversion ), through the Voice Gateway connection, it can be used with the original telephone system on the premise of normal use.
II. Specific settings
Prerequisites:
① Two telephone lines: one link to the telecom operator and the other link to the original telephone system
② Popular Voice Gateway settings: IP: 192.168.11.170, User name: admin/administrator password: hx4
③ Freeswitch server, static IP
1> SIP call analog phone
① Register as a sip account on the analog phone interface of the Gateway: set the SIP account according to the number in the FS.
② Set the gateway Route IP 1019 route FXS 1
Note: All phone numbers that call 1019 are directly transferred to the analog phone number 1.
Now, the phone number can be used as a phone number.
2> call mobile phone/external phone via SIP Phone
① Register the phone number with the FS through the gateway (see step 1)
② Gateway RoutingFxo x Route IP 192.168.11.44: 5080
③ Add processing to dialplan/public. xml of FS
<Extension name = "did">
<Conditionfield = "destination_number" expression = "^ 02962826892 $">
<Actionapplication = "bridge" Data = "User/1001"/>
<! -- Actionapplication = "info" Data = ""/> -->
<! -- Actionapplication = "IVR" Data = "welcome"/> -->
</Condition>
</Extension>
Note: This setting calls the 1001 account directly when an external line comes in. Later, you need to use IVR here and press the key to transfer the call in the dial.
3> existing phone switchboard
This function is set in the same way as that in 2>. Only gateway settings are different. For PSTN access, a telephone number is required. This function must be connected to a number in the current telephone switchboard as the access number.
The following section shows my specific settings and records for later viewing. If you have the same requirements, refer.
Connection between freeswitch and PSTN