The telephone ports allow you to:
FXS: Connect a local telephone or headset to this unit, and effectively convert it into an IP-phone
FXO: Connect this unit to the global telephone network, and be able to make and receive telephone calls
NOTE! Press "Save" after your settings!
You can connect a headset or (using an adapter cable) a telephone to the EXP port to be able to make ordinary as well as SIP telephone calls.
Normally the unit automatically detects the type of device connected, or you can explicitly specify it manually.
NOTE: You are allowed to connect only one telephone to this device!
SIP address - the address the phone/headset is registered as. SIP calls made to this address will be sent to the phone/headset. Usually you enter your own SIP address here.
Display name - (optional) descriptive user name sent to other SIP phones when phone/headset is used to make SIP calls. Usually you enter your own name here.
Authentication/Password - (optional) if the SIP server you are registered to demands that you enter user name and password to register you can enter them here. Otherwise leave empty.
To make a call using the phone lift its receiver and either dial the number you want to call on the phone's keypad, or enter the number or SIP URI you want to call and click Dial on web page. Also see "Dialling timeout" in advanced settings.
To receive a call using the phone lift its receiver when it rings. NOTE: The ringing is not performed by the phone but by this unit.
Telephone Line Port (FXO)
By connecting this unit to the global telephone network it can act as a gateway between the ordinary telephone system and IP telephony.
You can use your SIP phones/SIP clients to call ordinary phones, and answer incoming calls.
To make calls:
On your SIP phone you can either enter the telephone number you want to call as a SIP address within a special domain, or preceded by a special prefix.
If anyone calls the phone number of the line this unit is connected to he/she will be able to enter extension numbers that select which SIP phone to route the call to.
Domain name - a virtual SIP domain used to place calls to the ordinary global telephone network. Dial using <telephone number>@<domain name>. Thus for example if the domain name is "localgw" then on your SIP phone/SIP client you enter "5551234@localgw" if you want to make an ordinary telephone call to phone number 555 1234. You can set up a real domain name for the gateway in DNS pointing to an interface of the SIPGT. Enter that domain name here and calls through the FXO port can then be made from any authorized SIP client (even unknown to the built in SIP-server). All SIP-traffic matching the assigned domain name will be taken care of by the FXO port so make sure it is unique.
Telephone number - (optional) phone number of the connected line. Used as a caller ID on incoming calls when this unit forwards the call to a SIP phone/SIP client (if caller ID not already present).
Display name - (optional) display name of the connected line. Used as a caller ID on incoming calls when this unit forwards the call to a SIP phone/SIP client (if caller ID not already present).
Authentication/Password - (optional) if the destination of the call requires authentication, enter the user ID and password for the FXO port here.
Select if you want to enable/disable calls out to the telephone network. Uncheck this if you only want to receive calls and disable outgoing calls on your network.
Allowed to call out: - Specifies who is allowed to make outgoing calls. "Inside" refers to clients on the LAN and "Outside" refers to all other clients.
Await dial tone after initially dialled - Most PBX requires that you enter a digit before you get an outside line (usually 0 or 9). If the FXO port is connected to a PBX, enter that required digit. Leave blank if direct line.
Prefix for dialling through this port - a special prefix used to indicate call is destined to the ordinary global telephone network. If "**" then dial using **<telephone number>. Thus for example if the prefix is ** then on your IP phone you enter **5551234 if you want to make an ordinary telephone call to phone number 555 1234.
NOTE: If you have purchased the SIP Switch functionality, its dial plan will override this setting.
You can forward calls coming from the telephone network to one specific SIP phone/SIP client, or allow people ringing you to select among the SIP clients using extension numbers.
Select if you want to enable/disable incoming calls. Uncheck this if you only want to use FXO for outgoing calls and/or want to use other equipment for incoming calls.
Forward incoming calls to - SIP address to forward incoming calls to. This is the SIP client that receives the call as soon as it starts ringing (at the incoming callers side). When field is empty, see next setting.
during - Select when to interrupt the forwarding and proceed to the next action (collect setting below) e.g. if you select 3, the call will be directly forwarded and then after 3 ring signals the incoming caller will hear a dial tone (if collect, see settings below, is selected). If the "forward incoming..." field is empty the call is not forwarded anywhere but further actions are not taken until after e.g. 3 number of signals.
Options that needs additional explanation
- 0 Turns off forwarding of incoming calls. Instead further actions are taken immediately (dial tone).
- Infinite The forwarding will never be interrupted i.e. further actions will not be taken.
Collect - Select how many digits to collect when the incoming caller gets a dial tone. The call is immediately forwarded to the corresponding extension number when the amount of digits is reached.
Options that needs additional explanation
- No The incoming caller will never hear a dial tone i.e. the call is immediately forwarded to the SIP address in the "Forward to" field.
- Any When the incoming caller has dialled a number and no more digits are entered within 4 seconds (can be changed in the advanced settings), the call will be forwarded. Select this if you don't want to restrict to a certain amount of digits. The '#' can be used to complete a number immediately.
SIP server - Leave empty if you want the dialled extension to be handled to the built in SIP server, otherwise the forward address will be: <extension>@<SIP server>.
Try forwarding to collected extension for - Select how many seconds to try to reach a collected extension number i.e. the corresponding SIP address. When timer elapses (SIP address cannot be reached) the call is forwarded to the SIP address in the "forward to" field. Infinite means never give up i.e. it is up to the incoming caller to end the call he/she believes there is no answer.
Forward to - SIP address to forward incoming calls to if the collected extension number could not be found or if the timer elapses (or directly if no extension digits is selected). If field is empty the incoming caller will get a tone telling that destination is unreachable.
Read more online:
Advanced Settings for Telephone Ports
Phone Port/Telephone Line Port (FXO)
Ring signal - Choose type of ring signal to use.
Dialling timeout - If no digits are detected within this amount of seconds the number will be considered as complete. If you don't want to wait for this timeout, press '#' to state that the number is complete.
Extension dialling timeout - If you select to collect any digits for an incoming caller (see collect setting) you can change the timeout for when a number is considered to be complete when no further digits are detected.
Maximum call duration - Number of seconds until a call will be terminated (empty or 0 turns this feature off).
Disconnect call if silent for - Select if or after how long time you want automatic disconnection when silence on both sides is detected. NB! Does not work if silence suppression is disabled or if silence suppression is not supported by the other side.
Disconnect call if no RTP receved for - Select if or after how long time you want automatic disconnection when no RTP voice packets are received from the remote side.
Voice codecs used - At call set-up, codec lists are exchanged between the SIP clients (using SDP protocol). These settings affect how this is done.
- Codec prioritizations. Select your codec list based on your quality and bandwidth limitation.
- Adapt to remote side codec prioritization. The remote part may have other codec prioritizations. Mark checkbox if you want to adapt to those priorities instead of your own. For highest compatibility, keep this checked.
- Select a single codec for incoming calls. Mark checkbox if you only want to reply with a single codec as option. For highest compatibility, keep this checked.
DTMF transmission - Select if you want to send DTMF tones in the audio stream or detect and remove the DTMF tones from the audio stream and instead send them as separate telephony event packets. Mainly you should use the out-of-band setting. This requires that the remote part supports telephony events (which is detected at call set-up). If in-band is selected this unit will not reveal its out-of-band capabilities at call set-up.
Volume - Speaker output gain and microphone input gain. Values greater than zero (+) increases the volume and values less than zero (-) decreases the volume. Allowed range is -30 to +30 decibel.
Volume - Line output gain and line input gain. Values greater than zero (+) increases the volume and values less than zero (-) decreases the volume. Allowed range is -30 to +30 decibel.
Silence suppression - Select what to do when detecting absence of audio (i.e. no one speaks), or turn this feature off. It may be disturbing when there is complete silence, then you may want to send comfort noise or follow the codec standard. To lower effort on the network resources you may want to not send data at all during audio absence.
Echo cancellation - Select if or what type of echo cancellation you want.
Caller ID mode - Select if or what type of Caller ID you want.
Caller ID presentation - Select what information is sent to the caller ID display. First item in the option is sent to number line and the "plus" item is sent to the name line. Different information may be sent depending on what is available.
Check line free before using it - Select if you want to detect if line is idle before dialling. Uncheck this if idle detection doesn't work e.g. FXO thinks that line is busy even if it isn't.
Use voltage reversal on telephone line to detect on/off hook - Detects if the other side hangs up (only works if it is supported by the telephone system you are connected to).
Check dial tone before dialling - Select if you want to detect dial tone before dialling. Uncheck this if dial tone detection doesn't work e.g. phone system sends no or a strange dial tone.
Log calls in system log - Select if you want to log when calls start and finish.
Debug enable - Select if you want to enable debugging. Mainly for support matters.
Read more online:
Advanced Settings for Telephone Ports