This document mainly describes the detailed steps of interfacing the service board with Asterisk.
The following steps are configured to enable two-way calls between the phone and the gateway.
•Outbound: from asterisk SIP extension 3002 to the gateway via trunk 1008.
•Inbound: calls from outside lines to the gateway, via SIP trunk 1008 to asterisk, and then through asterisk to the 3002 SIP extension.
In the following steps, the parameters in the table are required configurations, other parameters can be configured according to your needs.
Step 1. Create a SIP trunk in the Asterisk server
Please add the following lines to sip.conf to create SIP trunks (1008) and SIP (3002).
[1008]
username=1008
secret=12345678
host=dynamic
port=5060
type=friend
context = from-gsm
[3002]
username=3002
secret=3002
host=dynamic
port=5060
type=friend
context = from-internal
After editing, save and reload the SIP configuration file.
Step 2. Edit the dial-up rules in asterisk server
[from-internal]
exten => _x.,1,Dial(sip/1008/${EXTEN})
exten => _x.,n,hangup()
[from-gsm]
exten =>_x.,1,Dial(sip/3002)
exten =>_x.,n,Hangup()
Step 3. Set the network parameters in the Web
Login to the network in your browser, and click "Network->LAN Settings" to set your network parameters.
Step 4. Create SIP endpoints in the Web
Please select "VOIP-> VOIP Settings->Add New SIP" to set up SIP trunk. The following figure shows the detailed information about how to set up.
Parameter Name |
Value |
Trunk Name |
1008 |
Name |
1008 |
Trunk Password |
12345678 |
Registration Type |
Client |
IP |
172.16.1.51 |
Port |
5160 |
Transmission Protocol |
UDP |
NAT |
Yes |
Step 5. Set up routing rules in the Web
Click Routing->Call Routing Rules->New Call Routing Rules to set up outbound and inbound routing rules, as follows.
Outbound routing rules：
Parameter Name |
Value |
Route Name |
outbound |
Call from |
1008 |
Call To |
Lte-1 |
Inbound routing rules：
Parameter Name |
Value |
Route Name |
inbound |
Call from |
Lte-1 |
Call To |
中继1008 |
Please save and apply configurations.
Step 6. SIP registration via software
Register SIP (3002) using SIP software such as Xlite, eyeBeam, etc. After completing all the above steps, you can try to make calls and send SMS.
Test call.
Inbound test: call the number of port 1 on the gateway with your cell phone and see if 3002 rings, ringing means the configuration is successful; if it does not ring, please check the configuration
Outbound test: call your own cell phone number in the softphone registered with 3002, if your own phone rings, it means the configuration is successful; if it does not ring, it means the configuration failed, please check the configuration
Parent topic: WTU