The FXS and FXO ports in SurfinBird FW GW models, Intertex

Transcription

1 1 (12) The FXS and FXO ports in SurfinBird FW GW models, Intertex Learn how to setup the gateway functionalities in the Internet Gate This guide describes how to setup the FXO port for PSTN gateway functionality and the FXS port for connecting a plain old analogue phone *. This guide assumes that you already have your Internet Gate up and running. If you are having problems in getting your Internet Gate to work please refer to the user guides at Make sure to always have your Internet Gate updated with the latest firmware. You will find information on how to upgrade your Internet Gate firmware at Contents 1 INTRODUCTION THE FXS PORT REGISTERED AT AN OPERATOR S SIP SERVER To call the FXS phone USING THE INTERNET GATE AS THE SIP SERVER To call the FXS phone THE FXS CALL CONTROL MAKE CALLS FROM THE WEB INTERFACE THE FXO PORT SIMPLE OUTGOING CALLS VIA THE FXO PORT (WITHOUT THE SIP SWITCH) USING THE FXO PORT WITH THE SIP SWITCH DIAL PLAN USING THE FXO PORT REMOTELY Registered user Using a domain TELEPHONE FUNCTIONS WAY CALLING CALL WAITING / HOLD / ALTERNATION CALL TRANSFER (ATTENDED) / HOLD / ALTERNATION BLIND CALL TRANSFER AND OTHER FUNCTIONS...11 * The GW models have one FXS and one FXO port, the GW2 models have two FXS and one FXO port and the PH2 models have two FXS and no FXO ports.

2 2 (12) 1 Introduction The Internet Gate offers flexible ways to configure the FXS (GW, GW2 and PH2 models) and FXO port (GW and GW2 models). The FXS port enables you to make/receive IP telephony calls via SIP from an ordinary analogue phone. You can even make/receive calls to/from the PSTN using an ITSP account (see for a list of ITSP providers) or the built in FXO port from your analogue phone. The FXO port is a full-fledged gateway port enabling you to make outgoing and receive incoming calls to/from the PSTN. The FXO functionality can even be used remotely, i.e. you can receive and make calls even though the Internet Gate is placed at another physical location. FXS functionality: Make outgoing / receive incoming SIP calls Make outgoing / receive incoming PSTN calls via ITSP or FXO port Make/receive calls to/from SIP hard/soft phones on the LAN to/from the Internet FXO functionality: PSTN backup Emergency calls (911, 112 etc.) Receive incoming / make outgoing calls Can be used remotely

3 3 (12) 2 The FXS port You can use your FXS port in two basic ways: registered at an operator s/itsp s SIP server, or registered at your Internet Gate, using the Internet Gate as the SIP server. 2.1 Registered at an operator s SIP server If you will register your client at an operator s/itsp s SIP server, do the following: 1. Surf in to your Internet Gate ( and click on Telephone Ports. 2. Below Phone Port (FXS), enter the SIP user information (SIP address, user id and password) provided by your ITSP. 3. Don t forget to save your settings. 4. The green led (PH) should be lit on the Internet Gate display and you should be able to make/receive PSTN phone calls To call the FXS phone If you have registered your FXS phone directly with an ITSP account there are two ways to reach the FXS phone.

4 4 (12) Call via ITSP If your FXS phone has successfully registered with the ITSP account (check under SIP Users from the main menu), just dial the PSTN number given by your ITSP. Call via SIP Most ITSPs enables you to be reached also on a SIP address, usually username@<ip address> or username@<domain>. Some ITSPs however, has blocked this feature, leaving you with the sole option to be contacted via the PSTN number. If the Global SIP communication feature is marked Yes at the ITSP enables you to be contacted on a SIP address. To be certain, ask your ITSP.

5 5 (12) 2.2 Using the Internet Gate as the SIP server If the Internet Gate will be used as SIP server, you can easily make and receive SIP phone calls. You can also make PSTN calls in two ways: (i) ether by using the SIP Switch software addition (read more at and and have the SIP Switch registered with one or several ITSP accounts, or (ii) by using the local FXO port (read more in section 3 below). If you wish to use the Internet Gate as SIP server - regardless of if or how you wish to access the PSTN network - you should setup user accounts on it. The Internet Gate supports SIP server registrations both from the LAN and the WAN (e.g. Internet). By default the Internet Gate will grant all SIP server registrations from the LAN without any authentication. In order to control who is allowed and who is disallowed to register, users from the WAN (e.g. Internet) must have a pre-configured account registered on the Internet Gate. Our recommendation is that you setup user accounts on the Internet Gate for all users whether LAN users or WAN users - since this gives you administration control and give the users maximum flexibility. Also, if you have the SIP Switch software upgrade installed the setup of user accounts makes it easy to assign a short extension number to each user. Please refer to for more info on how to set up user accounts with the SIP Switch. To register your FXS port on your Internet Gate: 1. Connect your analogue phone to your FXS port. 2. Surf in to your Internet Gate ( and click on Telephone Ports.

6 6 (12) 3. Below Phone Port (FXS), enter preferred SIP address, e.g. phone (without any domain such user id and password). The information will automatically be updated in your SIP Switch. 4. Don t forget to save your settings To call the FXS phone Depending on your setup there are many ways to reach your analogue phones connected to the FXS port. Below are some examples. Receive calls via SIP By default your FXS phone is given the SIP address phone *. If the phone is registered (check under SIP Users from the main menu), you can always reach the FXS phone by dialling the SIP address phone@<your public IP address>, e.g. phone@ If you have pointed a domain address, e.g. company.com at your Internet Gate, you can reach the FXS phone on the SIP address phone@company.com. It is possible to call the FXS phone both from the LAN and the WAN (e.g. the Internet). Receive calls via FXO By default incoming calls on the FXO port will be forwarded to your phone connected to your FXS port. Read more on the FXO port below in section 3. The FXO port. Receive calls via ITSP If you use the Internet Gate as SIP Server and want to reach / be reached from the PSTN you must setup outgoing Dialplan and SIP Accounts and Incoming Call Processing in the SIP Switch page, please check The FXS Call Control make calls from the web interface The Internet Gate provides a possibility to make calls directly from the GUI. In other words, you can initiate phone calls (SIP or PSTN calls) via the Internet Gate s web interface. 1. Surf into your Internet Gate ( 2. Go to Telephone Ports * If you have a GW2 or PH2 model with two FXS ports, by default your phones will be named phone1 and phone2.

7 7 (12) 3. Enter a PSTN phone number (via FXO, e.g. **12345 or via ITSP) or a SIP address (e.g. userxx@company.com) in the Enter number or URI field below Call Control. 4. Click Dial to initiate the call. 5. When your FXS phone starts ringing, just pick up the receiver. Other functions that you can access from this page are Transfer, Reject, Hang up, Other Call (i.e. new call in parallel). 3 The FXO port Also the FXO interface is flexible and enables you to utilize it in many ways. Below you will find numerous scenarios describing how the FXO port can be used. 3.1 Simple outgoing calls via the FXO port (without the SIP Switch) ** These settings should be applied if you are not using the SIP Switch software addition: 1. Connect your telephone line cable to the FXO port. 2. Surf in to your Internet Gate ( and click on Telephone Ports. 3. Below Telephone Line Port (FXO), enter localgw in the Domain name field and optionally your telephone number (which is used as display name for incoming calls) in the Telephone number field. 4. Enter ** (default) in the Prefix for dialling through this port and remember to save your setting. 5. Now, if you wish to call the PSTN number , just dial ** from your FXS phone or SIP hard/soft phone. 3.2 Using the FXO port with the SIP Switch Dial Plan If you use the SIP Switch software addition:

8 8 (12) 1. Connect your telephone line cable to the FXO port. 2. Surf in to your Internet Gate ( and click on Telephone Ports. 3. Below Telephone Line Port (FXO), enter localgw in the Domain name field and (optionally) your telephone number (which is used as display name for incoming calls) in the Telephone number field. 4. Save your settings and exit to the main menu. 5. Click on the SIP Switch menu item 6. Below Outgoing call number processing you can enter localgw in the Forward to field in order to forward calls to the PSTN via the FXO port. NOTE: If you cannot see the Action or the Forward to field, just click the Show User Actions button. Example 1: If you wish to forward emergency calls such as 911 or 112 to the PSTN via the local FXO port, enter 911 or 112 in the Head field, set Tail to Any, Minimum Tail to <blank>, Action to Force and enter localgw in the Forward to field. Example 2: If you wish to forward all outgoing calls to the local FXO port, enter localgw in the Default PSTN Gateway field. 7. Save your settings. 8. Now, depending on the SIP Switch setting you entered, you should be able to make/receive PSTN calls via your FXO port.

9 9 (12) 3.3 Using the FXO port remotely It is possible to make/receive calls via a FXO port on a remote Internet Gate, i.e. where the Internet Gate is placed at another physical location. You can do this in (at least) two ways see below Registered user The first way to utilize a remote FXO port is to register a SIP hard/soft phone on that specific Internet Gate, i.e. using the Internet Gate as SIP server. Remember that the Internet Gate by default needs to authenticate users from the WAN (e.g. the Internet), so make sure to setup user accounts on your Internet Gate (please refer to on how to do this) Using a domain Another way to access a remote FXO port is to setup a domain name with the FXO port. (A) Static domain If you have a static domain, just point it at your Internet Gate and: 1. Connect your telephone line cable to the FXO port. 2. Surf in to your Internet Gate ( and click on Telephone Ports. 3. Below Telephone Line Port (FXO), enter your domain name, e.g. myinternetgate.com, in the Domain name field and (optionally) your telephone number (which is used as display name for incoming calls) in the Telephone number field. 4. Save your settings. 5. You should now be able to make phone calls from any SIP hard/soft phone on the LAN or the Internet on the address <number>@domain>, e.g @myinternetgate.com. NOTE: In order to gain control on who can use the FXO port, it is recommended to enter Authentication: User ID and Password for the FXO port. Otherwise anyone on the Internet could just dial any number in the world from your FXO port at your expense by just dialling e.g. <number>@myinternetgate.com. If you setup a User Id and Password, your SIP soft/hard phone must enclose User Id and Password in the SIP INVITE messages.

10 10 (12) (B) Dynamic Domain If you have a dynamic domain from Tekea or DynDNS, do the following: 1. Connect your telephone line cable to the FXO port. 2. Surf in to your Internet Gate ( and click on Telephone Ports. 3. Below Telephone Line Port (FXO), enter your domain name, e.g. myinternetgate.com, in the Domain name field and (optionally) your telephone number (which is used as display name for incoming calls) in the Telephone number field. 4. Save your settings. 5. Click on Network under Configurations, and then Dynamic DNS. 6. Enter Hostname (e.g. myinternetgate.com), Account type (select Dynamic DNS) and your User name and Password. 7. Save your settings 8. You should now be able to make phone calls from any SIP hard/soft phone on the LAN or the Internet on the address <number>@domain>, e.g @myinternetgate.com. NOTE: In order to gain control on who can use the FXO port, it is recommended to enter Authentication: User ID and Password for the FXO port. Otherwise anyone on the Internet could just dial any number in the world from your FXO port at your expense by just dialling e.g. <number>@myinternetgate.com. If you setup a User Id and Password, your SIP soft/hard phone must enclose User Id and Password in the SIP INVITE messages.

11 11 (12) 4 Telephone functions These FXS telephone functions are included in firmware release 3.16 and later +. 3-way Conference. Call Waiting Call Hold/Alternation. Call Transfer (blind and attended). In the Advanced Telephone Port GUI you can enable/disable: Call Waiting (on by default) Display Caller ID during Call Waiting (Caller ID2, off by default) You can just use hook-flash * to invoke most functions, but get much better flexibility by using the #n commands (which works independently of how you reached a two-calls state) way calling To initiate a 3-way conference call: 1. Place the first call, wait for an answer 2. Hook-flash, listen for 2nd dialtone 3. Dial second call, wait for an answer 4. Hook-flash to bind the calls (R) 4.2 Call waiting / Hold / Alternation 1. You are on the phone 2. You hear the call-waiting alert 3. Hook-flash back and forth between the calls (R) 4.3 Call transfer (attended) / Hold / Alternation 1. You have received an incoming call 2. Hook-flash, listen for 2nd dialtone 3. Dial second call, wait for an answer 4. Hook-flash to transfer the call (R) (disonnecting yourself) 4.4 Blind Call transfer and other functions In addition, independently how you got two calls going - you can always use: #0 hangup waiting or parked call #1 hangup active call (connect waiting or parked) + These functions are only applicable in the USA. More countries are coming in future firmware releases. * The hook-flash is sometimes referred to as the R-button.

12 12 (12) #2 alternate beween the two calls #3 connect to 3-way onference #4 or #90 transfer current call to the other (disconnect yourself). This is equal to attended transfer) These keystrokes are of course much more general and gives you more flexibility. Use either - you don't have to select in advance! To do Blind Call transfer, simply: 1. You have a call 2. Hook-flash, listen for 2nd dialtone 3. Dial second call using prefix #90, i.e. #90 (disonnecting myself)