Help for Telephone Ports

The telephone ports allow you to:
FXS: Connect a telephone 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 "Apply" after your settings!

Phone Port (FXS)

Connect a telephone to the FXS port to make ordinary or SIP calls. It is possible to connect up to 5 phones to each FXS port.

SIP address - the address the phone is registered as. SIP calls made to this address will be sent to the phone. Usually you enter your own SIP address here.
Display name - (optional) descriptive user name sent to other SIP phones when you make calls. Usually you enter your own name here.
Authentication/Password - (optional) if the SIP server you are registered to demands user name and password to register you can enter them here. Otherwise leave empty.

Call Control
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.

Dial - Calls the number or SIP-URI entered in the text box. If the phone is on hook a single pickup-ring will sound, if othe side answers before you pickup intensive ringing will sound, if the other side is busy a short busy like ring sequence will sound. If a call is ongoing when this is pressed it will be put on hold and a new call strarted.
Transfer - Transfer the incoming or current call to the number or SIP-URI entered in the text box, so called blind transfer. Typical sequence: answer an incoming call, enter new destination number, press Transfer and then hangup.

Reject - Deny an incoming call, alternatively a parked call is ended. Is typically used when phone is ringing or when call waiting tones are heard.
Hang up - End the current call or cancel calling, a call on hold or waiting is activated.
Other Call - Put the current call on hold and switch to another on hold or waiting call.
Conference - Join the current call and one on hold, waiting or being called in a conference.
Transfer - Transfer the current call to one put on hold, so called attended transfer. Typical sequence: answer an incoming call, enter new destination number, press Dial, talk to the calle, then press Transfer and finally hangup.

Call Waiting - When in a call and if this is checked you will hear tones and be able to answer to the incoming call, putting the ongoing call on hold.
Do not disturb - When this is checked all incoming calls will be denied and the phone won't ring.
Call Anonymously - When this is checked your Caller ID is blocked by setting SIP from-address to anonymous@anonymous.invalid and display name to Anonymous

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.
Receiving calls:
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.

Outgoing calls
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.

Incoming calls
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.

No 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.

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. The telephone line will be connected and ringback tones etc. will be generated by this unit.
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:
Telephone Ports

Phone Port (FXS)/Telephone Line Port (FXO)

Ring signal - Choose type of ring signal to use.
Ring function - This setting can be used to make the FXS port ring even when other SIP units are called. Useful to get stronger or alternative ring signals on incoming calls, e.g. for deaf or hearing impaired persons by connecting special telephony equipment to FXS port to generate strong sound, flash or vibration. See also "Ring signal regardless of offered media" below.
Ring function SIP URI:s - When the setting "When internal SIP unit listed below is ringing" is chosen above, this field is used to specify SIP URI:s that should make the FXS port ring.
Ringing timeout - How many seconds the phone shall be ringing if no one answers, 0 means region default (typically 2 minutes).

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. Normal is 4 seconds.
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. NOTE! 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.
Use of µ-law and A-law companding. A-law and µ-law are two companding algorithms that compress and expand high resolution samples to and from the 8 bits which is used at transmission. PSTN networks in USA and Japan use µ-law companding, most other countries use A-law companding. Most SIP equipment support both variants for the G.711 codec (PCMU and PCMA). This setting controls how µ-law and A-law is prioritized and selected. When a call goes through a PSTN Gateway it is desireable that the correct companding is used so conversion can be avoided. Early media is used by gateways to pass on PSTN signalling tones, this is used to automatically adapt and is the default setting.

Desired voice packet time - Sets the desired amount of voice in each RTP voice packet transmitted. Depending on active codec the closest feasible value will be used.
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 in decibel. Values greater than zero (+) increases the volume and values less than zero (-) decreases the volume.
Volume - Line output gain and line input gain in decibel. Values greater than zero (+) increases the volume and values less than zero (-) decreases the volume.
Silence suppression - Select "On" if you want to allow reduction of audio packet transmission during silence periods. If the remote end also allows this, fewer and smaller comfort noise packets, or codec specific silence packets, are sent instead of regular audio packets.
Echo cancellation - Select "On" if echo cancellation shall be active.
Fax over IP protocol (T.38) - This device supports T.38 over UDPTL with SIP signalling. T.38 is a protocol to allow reliable transmission of fax over IP; If it is deactivated, this device will try to transmitt faxes using G.711 codec. Fax over G.711 is less reliable if the network connection is not very good, but deactivating T.38 may be appropriate if special interworking problems occur with other equipment not properly supporting T.38. The first menu selection for T.38 allows you to choose if you want to deactivate T.38, if T.38 should be initiated by this device only if this device is on the side of the calling fax device (default), or if T.38 should be initiated by this device regardless if it is on the side of the calling oder called fax device. Calls will always start as usual voice calls using a voice codec, and a switch to T.38 will be initiated by this device after detecting audio tones indicating that a fax device is attached, considering the menu selection just mentioned. If the remote side initiates T.38, this device will accept this unless T.38 is deactivated. The second menu selection for T.38 allows you to choose the T.38 TCF-method. TCF means training check flag, which is a training sequence executed by the fax at the very beginning of a fax call. TCF transferred means that this training sequence signal will be transferred end-to-end (over IP), TCF local means that this is only done locally. TCF local saves some IP-bandwidth, but the standard for T.38 using UDPTL recommends TCF transferred. Furthermore the second menu selection allows you to limit the maximum fax speed to values between 14400 and 2400 bit/second. Selecting a lower maximum speed will result in less bandwidth being used by fax calls, but the transmission of the fax will take longer then.
Caller ID mode - Select if or what type of Caller ID you want to send to the phone.
Caller ID number presentation - Select what information is sent to the number line of the Caller ID display on the phone. Mode default means it is set by the type of Caller ID selected. Different information may be sent depending on what is available. When "Number" is selected special codes for unknown or restricted numbers can also be sent.
Caller ID name presentation - Select what information is sent to the text line of the Caller ID display on the phone (not possible with DTMF). Mode default means it is set by the type of Caller ID selected. Different information may be sent depending on what is available.
Caller ID maximum number length - Limit the amount of characters sent to the phone Caller ID display (to avoid display problems on various phones), empty or 0 means default for profile which typically is unlimited.
Caller ID maximum name length - Limit the amount of characters sent to the phone Caller ID display (to avoid display problems on various phones), empty or 0 means default for profile which typically is unlimited.
Display Caller ID during Call Waiting - If a call comes in while talking and call waiting is enabled a so called Caller ID type 2 can be sent to the phone if this is enabled (not possible with DTMF Caller ID).
Receive Caller ID - Select if or what type of Caller ID that will be received from PSTN.

Disable playing of early media and generate ringback locally - Before a call is connected early media may be sent containing ringback signalling or other information, this option disables playing of the early media and generates ringback tones locally instead.
Ring signal regardless of offered media - Incomming calls without media or unknown media types are normally rejected immediately, this option enables the phone to ring even if conversation won't be possible.
Disable FXS port SIP registration - See RFC 3261
Disable registration of SIP media capability (RFC 3840) - Whether the SIP UA(s) of the FXS port(s) should register it's media capability according to RFC 3840. If this is enabled the SIP UA(s) will register that they support audio and list the SIP methods supported.
SIP "Q-value" for registration - See RFC 3261
SIP "expires" value for registration - See RFC 3261

Audio signal processing - For calls between a FXS and PSTN using the FXO port it is possible to short-cut or do full encoding and decoding of the audio. For calls between non-FXS SIP clients and PSTN using the FXO port the audio is always fully encoded and decoded. This hardware has one software DSP (Digital Signal Processor) per FXS and one for FXO and can always do full encoding and decoding of the audio. Short-cut audio processing between FXS and FXO only use one DSP and does only little processing of the audio. Full encoding and decoding of the audio between FXS and FXO use two DSPs (one per port). Full encoding and decoding of the audio between a non-FXS SIP client and FXO use one DSP. Using short-cut audio processing limit the amount of advanced telephony features (such as for instance 3-way calling), but instead it require less DSP resources. This setting controls how full or short-cut audio processing shall be selected and how DSPs are allocated.

Full encoding och decoding. Always do full encoding and decoding of the audio.
Short-cut with FXS1. FXO will always use FXS 1 DSP, resulting in short-cut if FXS 1 is used, full encoding and decoding if FXS 2 is used.
Short-cut preferred. Prioritize to short-cut, if an FXS is involved in the call FXO will share DSP with that. Non-FXS calls use full encoding and decoding.
Short-cut only. Only use short-cut meaning that only FXS are allowed to use the FXO.

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.