====== Telephone ports ======
^ Your Internet Gate has no Telephone (FXS) ports. ^
| The below information is not applicable to it. |
^ Your Internet Gate has no Line (FXO) port. ^
| The below information is not applicable to it. |
The Internet Gate has local telephone ports (FXS) allowing connection of analogue telephones, and telephone line port (FXO) allowing connection to a telephone landline.
===== Telephone ports (FXS) =====
Telephones connected to the PH ports can be used as SIP phones.
You either control phone behaviour using the [[:web GUI:Telephone ports page]] or [[dial codes]].
===== Landline port (FXO) =====
The LINE port can be attached to an ordinary telephone line. Then phones connected to the PH ports **as well as all SIP phones on LAN** can use that telephone line to dial out.
==== Outgoing calls ====
Normally, a telephone number dialed on a SIP client is sent to the VoIP service provider for handling. However, SIP calls can also be forced to the ordinary telephone network through the LINE port by:
=== Prefix ===
The dial prefix is %%**%% by default, but can be changed on the [[:web GUI:Telephone Ports page]]. Dialing %%**%%[telephone number] on any telephone connected to the PH ports or on any SIP client on the LAN, will call that telephone number through the LINE port.
=== Virtual domain name ===
The virtual domain name is by default “localgw" but can be changed on the [[:web GUI:Telephone Ports page]]. Any SIP client calling [telephone number]@localgw will dial the stated telephone number through the LINE port ((Some SIP phones may require entering the local IP address of your unit (by default 192.168.0.1) as “outbound proxy" in their configuration for this feature to work. Alternatively you can enable the built-in DNS server of your unit, on the DNS Server web page accessible from the main menu through the Network Configuration web page. If you own a valid second domain name (other than for your SIP server) you can use that domain name instead of localgw.)).
=== Emergency number ===
Emergency numbers are forced to the LINE port as set up on the [[web GUI:SIP Switch]] page.
=== Dial Plan ===
If you purchase the optional SIP Switch feature, the Dial Plan on the [[web GUI:SIP Switch]] page can be used to route telephone calls through the LINE port, in many useful ways.
==== Incoming calls ====
If anyone calls the telephone number of the phone line connected to the LINE port, the call will by default be forwarded to the telephone connected to the PH1 port.
This behavior can be changed on the [[:web GUI:Telephone Ports page]].
You can also set up more advanced functions, like collection of extension digits and directly allow forwarding of incoming calls to certain SIP clients. This works if you have given your SIP clients numeric SIP user names, or if you use the extension number feature of the optional SIP Switch software.