|
IPX hardware and software are completely and solely designed and developed by Orionox; Asterisk is open source software running Linux. The two completely different products have two things in common. Both provide IP PBX solutions for SMB customers; both support standard SIP protocols. It is possible for both systems to work together when needed.
IPX as Server, Asterisk as a cluster of clients
The IPX can be configured as a VoIP/SIP server when it works with Asterisk. On the IPX, you need to configure one or multiple SIP accounts, i.e. IP extensions under "basic voice > extensions > IP extensions". The extension number is the SIP account id. It should follow the extension format on the IPX. For example:
IP Extension Number: 120
Password: 112200
User Name: Tom Claude
Concurrency: 10
In Asterisk system, you need to configure the according SIP account using Asterisk script or some GUI interfaces, one for each IPX IP extension.
Add a new SIP trunk
a.Call id/user id: 120
b.Trunk name: 120
c.Peer script:
Host = the IPX's LAN or WAN IP address
Insecure = very
Port = 8060 (the default on IPX)
Secret = 112200 (the password of the IP extension)
Type = peer
User name = 120
d.Register script:
Userid:password@host:port/userid
120:112200@192.168.1.100:8060/120
Set outbound route
Set the defined trunk to be a default outbound route with a selected digit, e.g. 9. In this way, you can dial 9-101 to reach extension 101 on the IPX.
Allow anonymous inbound SIP calls in the security settings
Add inbound route
Set the SIP trunk 120 to DID 205 of Asterisk extension as its destination. Now, calls from IPX to extension 120 will ring Asterisk extension 205.
Asterisk as a SIP server and IPX as a cluster of clients
Asterisk can be configured as a server and the IPX can have one client to form a trunk to the Asterisk server.
In Asterisk, add SIP extensions. Each extension has extension number, display name, and secret. For example:
Extension number: 205
Display name: Jack Gordon
Secret: 2255
In the IPX, add a new VoIP account, which is the SIP trunk, for each Asterisk extension. For example:
User name: 205 (SIP account id)
Authorization User Name: 205
Extension number: 205
Password: 2255
Register Server: Asterisk system IP address
Signaling Port: 5060 (if 5060 is selected by Asterisk)
Set DID to an extension 105
Now, from Asterisk, calls to extension 205 will ring IPX extension 105. From the IPX, call 9-211 will ring the Asterisk extension 211. If the IPX is set to ATA dialing method, dialing "9" is not needed. If you create one VoIP account for each IPX extension with proper DID settings, all IPX extension and Asterisk extensions can call each other directly.
Back
|
|
|
