[Sipdroid] Build a local sipdroid Development Environment

Http://www.cnblogs.com/lovers/articles/2624695.html

1. Create an asterisk virtual SIP server.

(1) Establish a client-server connection directly through a crossover line

At this time, you need to prepare a crossover network cable.

At the same time, the IP addresses of the two computers are allocated to the same network segment, for example, 192.168.0.1 and 192.168.0.2.

Then use the same netmask.

At the same time, choose Control Panel> Administrative Tools> Local Security Policies> Local Policies> User permission assignment> deny access to the computer from the network, and delete the gest; choose Control Panel> Administrative Tools> Local Security Settings> Local Policies> Security Options> network access: sharing and security modes of the local account. Select 'classic: the local user authenticates with his/her own identity.

However, you also need to disable the firewall if you want to ping both hosts.

(2) establish a client-server connection through the LAN

Note that because Asterisk is installed on the vmwarevm and the main operating system is windows, if other computers want to remotely connect to the asterisk server, select bridging mode and DHCP in network settings, in this way, a lan ip address independent from the host is obtained.

2. Register Extension)

Extension: ext trunk: Port

You can use asterisk-> Add extension to add an extension. You can specify displayname, extension number, and extension password.

None of the above.

In the beginning, we thought we wanted to add trunk. In fact, we didn't need to set the trunk for the SIP network-based IP Phone.

The client can use sipdroid-android, X-lite, and ekiga. However, I found that When configuring the SIP account for ekiga, there will always be a problem: Registration failed. So I finally chose free X-Lite to set up the SIP account, mainly pay attention to the following parameters:

(1) username indicates the extension number, not the displayname (this is dispensable)

(2) passward refers to the extension password.

(3) authentication user name indicates the extension number.

(4) domain refers to the IP address of the asterisk local server. Note that it is not the configured IP address of the HTTP Server Admin.

If you want to set up a voice call, you need to set it. This is relatively simple, but don't forget, otherwise you cannot make a voice call.

Problem: the video call has not been set successfully.

The setting method in sipdroid is similar. Note that you must select the WLAN and 3G modes in the set call mode.

To view the current real-time extension status of the system, you need to go to the Panel. Panel needs security code, which needs to be modified in Asterisk command mode. In

Modify the default foppassword value in/etc/amportal. conf. You can view the dialing status of some real-time extensions.