As we already know, we can connect VoIP phones or softphones to an Asterisk server in order to make phone calls over the internet. But what if you want to make calls over the Internet using a desk/home phone connected to a classic phone line (PSTN – like Romtelecom)? For this we need a telephone adapter (ATA) or an analog card, which connects the digital network (Asterisk) with the analog network (analog telephone line). More information on connecting an analog line to a server Asterisk can be found in the article “About VoIP” .
Next, we describe how to configure such equipment in order to functionally integrate an analog telephone with an Asterisk server. We use a PAP-2T Linksys phone adapter and an Asterisk server based on v220.127.116.11 trixbox CE distribution.
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The Linksys PAP-2T Phone Adapter has a port RJ45 (Internet), two doors RJ11 (Line 1 andLine 2) and a power port.
Setup / InstallationConnect the local Ethernet network cable to the port RJ45 labeled of “Internet”. Plug analog phone into port RJ11 marked “Line 1”. Turn on the power supply and connect it to the adapter. The green lights on the case should now flash intermittently.
Analog configuration extension
In order to be able to make calls from the analog phone, connected to the VoIP adapter, you need to assign (on the Asterisk server) a corresponding extension to the analog phone. The configuration steps are:
- If no available extension exists, set one on the Asterisk server
- Find the IP address of the adapter the phone is connected to and, using a web browser, access the graphical configuration interface
- Extension-specific data is used (username and password) as well as specific configuration data for the corresponding VoIP protocol (SIP or IAX).
Phone adapter configuration
Basic phone adapter settings PAP-2T Linksys are made from the analog phone connected to the port “Line 1”. Afterwards, all other settings will be in the adapter's web administration interface.
To access the adapter configuration menu, press “*” 4 times to the analog phone. The configuration commands are:
|****||access the menu|
|100 #||Check DHCP – the headset hears whether this service is active or not|
|101 #||Set DHCP -> to disable press “0”; to allow press“1”|
|110 #||Check the IP address|
|111 #||Set IP address -> Enter IP address using phone keys. use “*” to point. if DHCP is activated, the message“Invalid option” is played.|
|120 #||Check Netmask|
|121 #||Set Netmask -> – Enter the netmask using the phone keys. use “*” to point. if DHCP is activated, the message “Invalid option” is played.|
|130 #||Check gateway|
|131 #||Set Gateway -> Enter the gateway address using the phone keys. use “*” to point. if DHCP is activated, the message“Invalid option” is played.|
web administration interface
After retrieving the address IP (110 # in the setup menu) assigned to the phone adapter PAP-2T, use a browser to connect to the administration interface. The default setting does not require password guarantees for both regular (User) or administrator (admin)bills. Select profile of administrator and choose the option “Advanced view”.
Configure “Line 1” port
- Choose section “Line 1” (or “Line 2” depending on the port used). In our case, the phone was connected to the port “Line 1”.
- Check the option “Activate Line” is defined as “Yes”
- fill in the “proxy” and parameters of “Outbound Proxy” with server address Asterisk.
- fill in the “Display Name”, “User ID” and parameters of “Auth ID” with the user-specific values of that extension. The extension used for the tests was previously configured on the server Asterisk.
- Fill in the parameter “password” with the extension password
- Replace the existing configuration for the “Dial Plan” parameter with values (*x |.**X |..X). These values allow dial destinations starting with “*”. In our case, these destinations can be Asterisk predefined short codes (eg “*43” – the short code for applying echo). Also, the part of “x”. It is a wildcard that allows you to dial anywhere, be it an internal extension or an outgoing destination.
- Save settings using button “Save Settings”.
The adapter is now configured. If the “110” extension is called, the analog phone connected to the “line 1” phone adapter port will ring.
The same procedure is used to install an additional analog telephone (if necessary) in the second port of the telephone adapter (“Line 2”). The second extension corresponding to the additional telephone (eg “111”.) Must be defined on the server Asterisk.