Voice
Voice Overview
You can make calls over the Internet using VoIP technology. For this, you first need to set up a SIP account with a SIP service provider.
Use this chapter to:
• Connect an analog phone to the Zyxel Device.
• Configure settings such as speed dial.
• Configure network settings to optimize the voice quality of your phone calls.
What You Can Do in this Chapter
These screens allow you to configure your Zyxel Device to make phone calls over the Internet and your regular phone line, and to set up the phone you connect to the Zyxel Device.
• Use the SIP Account screen to set up information about your SIP account, control which SIP accounts the phones connected to the Zyxel Device use, and configure audio settings such as volume levels for the phones connected to the Zyxel Device (SIP Account).
• Use the SIP Service Provider screen to configure the SIP server information, and the numbers for certain phone functions (SIP Service Provider).
• Use the SIP TLS Common screen to change the default TLS local port if you need to, and select a local certificate for the SIP server to verify the Zyxel Device.(SIP TLS Common).
• Use the Phone screens to change settings that depend on which region of the world the Zyxel Device is in (Phone).
• Use the Call Rule screen to set up shortcuts for dialing frequently-used (VoIP) phone numbers (Call Rule).
You do not necessarily need to use all these screens to set up your account. In fact, if your service provider did not supply information on a particular field in a screen, it is usually best to leave it at its default setting.
What You Need to Know About VoIP
VoIP
VoIP stands for Voice over IP. IP is the Internet Protocol, which is the message-carrying standard the Internet runs on. So, Voice over IP is the sending of voice signals (speech) over the Internet (or another network that uses the Internet Protocol).
SIP
SIP stands for Session Initiation Protocol. SIP is a signaling standard that lets one network device (like a computer or the Zyxel Device) send messages to another. In VoIP, these messages are about phone calls over the network. For example, when you dial a number on your Zyxel Device, it sends a SIP message over the network asking the other device (the number you dialed) to take part in the call. To access this screen, click VoIP > SIP.
SIP Accounts
A SIP account is a type of VoIP account. It is an arrangement with a service provider that lets you make phone calls over the Internet. When you set the Zyxel Device to use your SIP account to make calls, the Zyxel Device is able to send all the information about the phone call to your service provider on the Internet.
Strictly speaking, you do not need a SIP account. It is possible for one SIP device (like the Zyxel Device) to call another without involving a SIP service provider. However, the networking difficulties involved in doing this make it tremendously impractical under normal circumstances. Your SIP account provider removes these difficulties by taking care of the call routing and setup – figuring out how to get your call to the right place in a way that you and the other person can talk to one another.
SIP Address
A SIP address is a URI (Uniform Resource Identifier) that resembles an email address, using the format: user@domain. It uniquely identifies a telephone extension over a VoIP system. A SIP address of 123-45-67@voip-provider.net tells a client to connect to voip-provider.net and request a connection to 123-45-67. While VoIP can only send voice messages over the Internet, SIP (though strictly speaking is a type of VoIP) can send voice, data, video, and other media. VoIP phones also need to be connected to a computer to function, whereas SIP phones only need to be connected to a modem.
Before You Begin
• Before you can use these screens, you need to have a VoIP account already set up. If you do not have one yet, you can sign up with a VoIP service provider over the Internet.
• You should have the information your VoIP service provider gave you ready, before you start to configure the Zyxel Device.
SIP Account
You can make calls over the Internet using VoIP technology. For this, you first need to set up a SIP account with a SIP service provider. The Zyxel Device uses a SIP account to make outgoing VoIP calls, and to check if an incoming call’s destination number matches your SIP account’s VoIP number. In order to make and receive VoIP calls, you need to enable and configure a SIP account, and then map it to a phone port. The SIP account contains information that allows your Zyxel Device to connect to your VoIP service provider.
To access this screen, click VoIP > SIP > SIP Account.
VoIP > SIP > SIP Account
The following table describes the labels in this screen.
Label | DESCRIPTION |
|---|---|
Add New Account | Click this to configure a SIP account. |
# | This is the index number of the entry. |
Enable | This shows whether the SIP account is activated or not. A yellow bulb signifies that this SIP account is activated. A gray bulb signifies that this SIP account is activated. |
SIP Account | This shows the name of the SIP account. |
Service Provider | This shows the name of the SIP service provider. |
Account Number | This shows the SIP number. |
Modify | Click the Modify icon to configure the SIP account. |
Add or Edit SIP Account
Use this screen to configure a SIP account and map it to a phone port in the Phone Device screen. To access this screen, click the Add New Account button or click the Edit icon of an entry in the VoIP > SIP > SIP Account screen.
You do not necessarily need to use all these fields to set up your account.VoIP > SIP > SIP Account > Add Account or Edit
VoIP > SIP > SIP Account > Add Account or Edit (Call Features)
VoIPThe following table describes the labels in this screen.
Label | DESCRIPTION |
|---|---|
SIP Account Selection | |
SIP Account Selection | This field displays ChangeMe if you are creating a new SIP account or the SIP account you are modifying. |
SIP Service Provider Association | |
SIP Account Associated with | Select the SIP service provider profile to use for the SIP account you are configuring in this screen. You should already have configured a SIP service provider profile in the SIP Service Provider screen. This field is read-only when you are modifying an existing SIP account. |
General | |
Enable SIP Account | Select this if you want the Zyxel Device to use this account. Clear it if you do not want the Zyxel Device to use this account. |
SIP Account Number | Enter your SIP number. In the full SIP URI, this is the part before the @ symbol. You can use up to 127 printable characters and spaces. |
Authentication | |
Username | Enter the user name for registering this SIP account, exactly as it was given to you. You can use up to 95 alphanumeric (0-9, a-z, A-Z), printable special characters and spaces. |
Password | Enter the password for registering this SIP account, exactly as it was given to you. You can use up to 95 alphanumeric (0-9, a-z, A-Z), printable special characters and spaces. |
URL Type | |
URL Type | Select whether or not to include the SIP service domain name when the Zyxel Device sends the SIP number. SIP – include the SIP service domain name. TEL – do not include the SIP service domain name. |
Voice Features | |
Primary/Secondary/Third Compression Type | Select the type of voice coder or decoder (codec) that you want the Zyxel Device to use. G.711 provides higher voice quality but requires more bandwidth (64 kbps). • G.729 provides good sound quality and reduces the required bandwidth to 8 kbps. • G.711a is typically used in Europe. • G.711u is typically used in North America and Japan. • G.726-24 operates at 24 kbps. • G.726-32 operates at 32 kbps. • G.722 operates at 6.3 kbps or 5.3 kbps. When two SIP devices start a SIP session, they must agree on a codec. Select the Zyxel Device’s first choice for voice coder or decoder. Select the Zyxel Device’s second choice for voice coder or decoder. Select None if you only want the Zyxel Device to accept the first choice. Select the Zyxel Device’s third choice for voice coder or decoder. Select None if you only want the Zyxel Device to accept the first or second choice. |
Speaking Volume Control | Select the loudness that the Zyxel Device uses for speech that it sends to the peer device. Choices are Minimum, Middle, and Maximum. |
Listening Volume Control | Select the loudness that the Zyxel Device uses for speech that it receives from the peer device. Choices are Minimum, Middle, and Maximum. |
Enable G. 168 (Echo Cancellation) | Select this if you want to eliminate the echo caused by the sound of your voice reverberating in the telephone receiver while you talk. |
Enable VAD (Voice Active Detector) | Select this if the Zyxel Device should stop transmitting when you are not speaking. This reduces the bandwidth the Zyxel Device uses. |
Call Features | |
Send Caller ID | Select this if you want to send identification when you make VoIP phone calls. Clear this if you do not want to send identification. |
Enable Call Transfer | Select this to enable call transfer on the Zyxel Device. This allows you to transfer an incoming call (that you have answered) to another phone. |
Enable Call Waiting | Select this to enable call waiting on the Zyxel Device. This allows you to place a call on hold while you answer another incoming call on the same telephone (directory) number. |
Call Waiting Reject Timer | Specify a time of seconds that the Zyxel Device waits before rejecting the second call if you do not answer it. |
Enable Unconditional Forward | Select this if you want the Zyxel Device to forward all incoming calls to the specified phone number. Specify the phone number in the To Number field on the right. |
Enable Busy Forward | Select this if you want the Zyxel Device to forward incoming calls to the specified phone number if the phone port is busy. Specify the phone number in the To Number field on the right. If you have call waiting, the incoming call is forwarded to the specified phone number if you reject or ignore the second incoming call. |
Enable No Answer Forward | Select this if you want the Zyxel Device to forward incoming calls to the specified phone number if the call is unanswered. (See No Answer Time.) Specify the phone number in the To Number field on the right. |
No Answer Time | This field is used by the Active No Answer Forward feature. Enter the number of seconds the Zyxel Device should wait for you to answer an incoming call before it considers the call unanswered. |
Enable Do Not Disturb (DND) | Select this to turn the do not disturb feature on. This has the Zyxel Device reject all calls destined to the phone line. |
Active Incoming Anonymous Call Block | Select this to have the phone not ring for incoming calls with caller ID deactivated. |
Enable MWI | Select this if you want to hear a waiting (beeping) dial tone on your phone when you have at least one voice message. Your VoIP service provider must support this feature. |
MWI Subscribe Expiration Time | Keep the default value of this field unless your VoIP service provider tells you to change it. Enter the number of seconds the SIP server should provide the message waiting service each time the Zyxel Device subscribes to the service. Before this time passes, the Zyxel Device automatically subscribes again. |
Hot Line / Warm Line Number | Select this to enable the hot line or warm line feature on the Zyxel Device. |
Hot Line | Select this to have the Zyxel Device dial the specified hot line number immediately when you pick up the telephone. |
Warm Line | Select this to have the Zyxel Device dial the specified warm line number after you pick up the telephone and do not press any keys on the keypad for a period of time. |
Hot Line / Warm Line Number | Enter the number of the hot line or warm line that you want the Zyxel Device to dial. |
Warm Line Timer | Enter a number of seconds that the Zyxel Device waits before dialing the warm line number if you pick up the telephone and do not press any keys on the keypad. |
Enable Missed Call Email Notification | Select this option to have the Zyxel Device email you a notification when there is a missed call. |
Mail Account | Select a mail account for the email address specified below. If you select None here, email notifications will not be sent through email. You must have configured a mail account already in the Email Notification screen. |
Send Notification to e-mail | Notifications are sent to the email address specified in this field. If this field is left blank, notifications will not be sent through email. |
Missed Call e-mail Title | Type a title that you want to be in the subject line of the email notifications that the Zyxel Device sends. |
Early Media | Select this if you want people to hear a customized recording when they call you. |
IVR Play Index | Select the tone you want people to hear when they call you. This field is configurable only when you select Early Media. See Technical Reference for information on how to record these tones. |
Music On Hold (MOH) | Select this to play a customized recording when you put people on hold. |
IVR Play Index | Select the tone to play when you put someone on hold. This field is configurable only when you select Music on Hold, See Technical Reference for information on how to record these tones. |
OK | Click this to save your changes. |
Cancel | Click this to exit this screen without saving. |
SIP Service Provider
Use this screen to view the SIP service provider information on the Zyxel Device. A SIP provider offers Internet call services using VoIP technology. You may need to consult your SIP service provider for the following settings.
To access this screen, click VoIP > SIP > SIP Service Provider.
VoIP > SIP > SIP Service Provider
The following table describes the labels in this screen.
Label | DESCRIPTION |
|---|---|
Add New Provider | Click this button to add a new SIP service provider. |
# | This is the index number of the entry. |
SIP Service Provider Name | This shows the name of the SIP service provider. |
SIP Proxy Server Address | This shows the IP address or domain name of the SIP server. |
REGISTER Server Address | This shows the IP address or domain name of the SIP register server. |
SIP Service Domain | Enter the SIP service domain name. In the full SIP URI, this is the part after the @symbol. You can use up to 127 printable ASCII Extended set characters. |
Modify | Click the Edit icon to configure the SIP service provider. Click the Delete icon to delete this SIP service provider from the Zyxel Device. |
Provider Entry Add or Edit
Use this screen to configure the SIP server information, the numbers for certain phone functions and dialing plan for a SIP service provider.
Click the Modify icon next to a profile of SIP service provider settings in the VoIP > SIP > SIP Service Provider to open the following screen.
Click this (
) to see all the fields in the screen. You do not necessarily need to use all these fields to set up your account. Click again to see and configure only the fields needed for this feature.VoIP > SIP > SIP Service Provider: Add New Provider or Edit
The following table describes the labels in this screen.
Label | DESCRIPTION | |||
|---|---|---|---|---|
IP Version Policy | ||||
Policy Selection | Select the policy or strategy for how you want the VoIP application to choose which interface IP address to bind to. Multiple interface IP addresses may be available through the Bound Interface Name. • Select IPv4 Only to bind the IPv4 interface and send both SIP and RTP packets using an IPv4 address. • Select IPv6 Only to bind the IPv6 interface and send both SIP and RTP packets using an IPv6 address. • Select IPv4 First to use an IPv4 address over an IPv6 address. If an IPv4 address is not available on the WAN interface, the Zyxel Device uses an IPv6 address instead. • Select IPv6 First to use an IPv6 address over an IPv4 address. If an IPv6 address is not available on the WAN interface, the Zyxel Device uses an IPv4 address instead. • Select IPv4 Prefer to use an IPv4 address over an IPv6 address. If an IPv4 address is not available on the WAN interface, the Zyxel Device uses an IPv6 address instead. The Zyxel Device changes to an IPv4 address whenever an IPv4 address is available. • Select IPv6 Prefer to use an IPv6 address over an IPv4 address. If an IPv6 address is not available on the WAN interface, the Zyxel Device uses an IPv4 address instead. The Zyxel Device changes to an IPv6 address whenever an IPv6 address is available. Note 1: If you are not sure which policy to use, or if VoIP still does not work, please contact your ISP or VoIP service provider. Note 2: If you select IPv4 First, IPv6 First, IPv4 Prefer or IPv6 Prefer, the second SIP service provider profile will not work. | |||
SIP Service Provider Selection | ||||
Service Provider Selection | This field displays ADD_NEW if you are creating a new SIP service provider profile or the SIP service provider profile name you are modifying. | |||
General | ||||
SIP Service Provider | Select this if you want the Zyxel Device to use this SIP service provider profile. Clear it if you do not want the Zyxel Device to use this SIP service provider profile. | |||
SIP Service Provider Name | Enter the name of your SIP service provider profile. You can use your VoIP service provider’s name as a reference. | |||
SIP Local Port | Enter the Zyxel Device’s listening port number, if your VoIP service provider gave you one. Otherwise, keep the default value. | |||
SIP Proxy Server Address | Enter the IP address or domain name of the SIP server provided by your VoIP service provider. You can use up to 95 printable characters except [ " ], [ ` ], [ ' ], [ < ], [ > ], [ ^ ], [ $ ], [ | ], [ & ], or [ ; ]. It does not matter whether the SIP server is a proxy, redirect or register server. | |||
SIP Proxy Server Port | Enter the SIP server’s listening port number, if your VoIP service provider gave you one. Otherwise, keep the default value. | |||
SIP REGISTRAR Server Address | Enter the IP address or domain name of the SIP register server, if your VoIP service provider gave you one. Otherwise, enter the same address you entered in the SIP Server Address field. You can use up to 95 printable characters except [ " ], [ ` ], [ ' ], [ < ], [ > ], [ ^ ], [ $ ], [ | ], [ & ], or [ ; ]. | |||
SIP REGISTRAR Server Port | Enter the SIP register server’s listening port number, if your VoIP service provider gave you one. Otherwise, enter the same port number you entered in the SIP Server Port field. | |||
SIP Service Domain | Enter the SIP service domain name. In the full SIP URI, this is the part after the @ symbol. You can use up to 127 printable characters except [ " ], [ ` ], [ ' ], [ < ], [ > ], [ ^ ], [ $ ], [ | ], [ & ], or [ ; ]. | |||
RFC Support | ||||
PRACK (RFC 3262, Require: 100rel) | During a call session, there are two types of SIP responses used – final and provisional. Final responses convey the result of a request and require a confirmation response. Provisional responses only convey the request processing progress and does not require a confirmation response, and are therefore considered unreliable. RFC 3262 defines a mechanism to provide reliable transmission of SIP provisional response messages, which convey information on the processing progress of the request. This uses the option tag 100rel and the Provisional Response ACKnowledgement (PRACK) method. Which is, the Zyxel Device includes a SIP Require header field with the option tag 100rel in all INVITE requests. When the Zyxel Device receives a SIP response message indicating that the phone it called is ringing, the Zyxel Device sends a PRACK message to have both sides confirm the message is received. Select this to have the caller require the option tag 100rel to send provisional responses reliably. | |||
VoIP IOP Flags – Select VoIP inter-operability settings. | ||||
Replace dial digit ‘#’ to ‘%23’ in SIP messages | Replace a dial digit “#” with “%23” in the INVITE messages. | |||
Remove the ‘Route’ header in SIP messages | Remove the ‘Route’ header in SIP packets. | |||
Bound Interface Name | ||||
Bound Interface Name | If you select AnyWAN, the Zyxel Device automatically activates the VoIP service when any WAN connection is up. If you select MultiWAN, you also need to select the pre-configured WAN connections. The VoIP service is activated only when one of the selected WAN connections is up. | |||
Outbound Proxy | ||||
Outbound Proxy Address | Enter the IP address or domain name of the SIP outbound proxy server if your VoIP service provider has a SIP outbound server to handle voice calls. This allows the Zyxel Device to work with any type of NAT router and eliminates the need for STUN or a SIP ALG. Turn off any SIP ALG on a NAT router in front of the Zyxel Device to keep it from re-translating the IP address (since this is already handled by the outbound proxy server). | |||
Outbound Proxy Port | Enter the SIP outbound proxy server’s listening port, if your VoIP service provider gave you one. Otherwise, keep the default value. | |||
Use DHCP Option 120 first | Select this to have the Zyxel Device use DHCP Option 120 first. | |||
RTP Port Range | ||||
Start/End Port | Enter the listening port numbers for RTP traffic, if your VoIP service provider gave you this information. Otherwise, keep the default values. To enter one port number, enter the port number in the Start Port and End Port fields. To enter a range of ports, • enter the port number at the beginning of the range in the Start Port field. • enter the port number at the end of the range in the End Port field. | |||
DTMF Mode | Control how the Zyxel Device handles the tones that your telephone makes when you push its buttons. You should use the same mode your VoIP service provider uses. RFC2833 – send the DTMF tones in RTP packets. PCM – send the DTMF tones in the voice data stream. This method works best when you are using a codec that does not use compression (like G.711). Codecs that use compression (like G.729 and G.726) can distort the tones. SIP INFO – send the DTMF tones in SIP messages. | |||
Transport Type | ||||
Transport Type | Select the protocol used to transport the SIP packets. For UDP and TCP, see the Service appendix for more information on the example services and the required protocol and port number. | |||
Ignore Direct IP | Select Enable to have the connected devices accept SIP requests only from the SIP proxy/register server specified above. SIP requests sent from other IP addresses will be ignored. | |||
QoS Tag | ||||
SIP DSCP Mark Setting | Enter the DSCP (DiffServ Code Point) number for SIP message transmissions. The Zyxel Device creates Class of Service (CoS) priority tags with this number to SIP traffic that it transmits. | |||
RTP DSCP Mark Setting | Enter the DSCP (DiffServ Code Point) number for RTP voice transmissions. The Zyxel Device creates Class of Service (CoS) priority tags with this number to RTP traffic that it transmits. | |||
Timer Setting | ||||
SIP Register Expiration Duration | Enter the number of seconds your SIP account is registered with the SIP register server before it is deleted. The Zyxel Device automatically tries to re-register your SIP account when one-half of this time has passed (The SIP register server might have a different expiration). | |||
SIP Register Fall Re-try timer | Enter the number of seconds the Zyxel Device waits before it tries again to register the SIP account, if the first try failed or if there is no response. | |||
Session Expires [SE] | Enter the number of seconds the Zyxel Device lets a SIP session remain idle (without traffic) before it automatically disconnects the session. | |||
Min-SE | Enter the minimum number of seconds the Zyxel Device lets a SIP session remain idle (without traffic) before it automatically disconnects the session. When two SIP devices start a SIP session, they must agree on an expiration time for idle sessions. This field is the shortest expiration time that the Zyxel Device accepts. | |||
Dialing Interval Selection | ||||
Dialing Interval Selection | Enter the number of seconds the Zyxel Device should wait after you stop dialing numbers before it makes the phone call. The value depends on how quickly you dial phone numbers. | |||
SIP Server Location DNS Method | Select the method that the Zyxel Device used to query the ISP’s DNS server for SIP server address. The Zyxel Device will use the query result to locate the SIP server for phone service registration. Select BASIC to have the Zyxel Device query the DNS server for a DNS A record that contains the IP address of the SIP server. Select SRV to have the Zyxel Device query the DNS server for a DNS Service (SRV) record. The SRV record is a list of all available SIP servers information that the DNS server maintains. The Zyxel Device will then use the SRV record to perform A query to get the SIP server IP. This is useful if your primary SIP server experiences difficulties, making it hard for your IP phone users to make SIP calls. Select NAPTR to have the Zyxel Device query the DNS server for DNS Name Authority Pointer (NAPTR) records in order to find the available services (transport protocols) supported by the SIP server. The Zyxel Device will then perform an SRV or A query to get the SIP server information. | |||
OK | Click this to save your changes. | |||
Cancel | Click this to exit this screen without saving. | |||
SIP TLS Common
Use this screen to:
• Change the default TLS local port.
• Select a local certificate for the SIP server to verify the Zyxel Device.
To activate SIP TLS Common, select TLS in Transport Type in the SIP Service Provider screen.To access this screen, click VoIP > SIP > SIP TLS Common.
VoIP > SIP > SIP TLS Common
The following table describes the labels in this screen.
label | description |
|---|---|
TLS Local Port | Port 5061 is typically used for SIP over TLS. Enter the Zyxel Device’s TLS local port number if your VoIP service provider gave you one. Otherwise, keep the default value. |
Local Certificate | This is the certificate the SIP server uses to verify the Zyxel Device. Go to Certificate > Local Certificate and import a Zyxel Device certificate that the SIP server can use to verify the Zyxel Device, if required. Then select the certificate you imported in this field. |
Verify Server Certificate | Click to enable this if you want the Zyxel Device to verify the certificate from the SIP server. If required or if your VoIP service provider gave you a certificate, import the dedicated CA in Certificate > Trusted CA in order for the Zyxel Device to authenticate the SIP server. |
Phone
Use these screens to configure SIP numbers and regions for IP phones that are connected to the Zyxel Device.
Phone Device
Use this screen to view detailed information on phones used for Internet phone calls (SIP). You can define which phones will ring when a specific SIP address receives an incoming call, and which SIP address will be used when an outgoing call is made with a specific phone.
To access this screen, click VoIP > Phone > Phone Device.
VoIP > Phone > Phone Device
Each field is described in the following table.
Label | description |
|---|---|
# | This displays the index number of the phone device. |
Phone ID | This field displays the name of a phone port on the Zyxel Device. |
Internal Number | This field displays the internal call prefix of a phone port on the Zyxel Device. |
Incoming SIP Number | This field displays the SIP address that you use to receive calls on this phone port. |
Outgoing SIP Number | This field displays the SIP address that you use to make calls on this phone port. |
Modify | Click the Edit icon to configure the SIP account. |
Phone Device Edit
Use this screen to control which SIP account and PSTN line each phone uses. Click an Edit icon in VoIP > Phone > Phone Device to open the following screen.
VoIP > Phone > Phone Device > Edit
Each field is described in the following table.
label | description |
|---|---|
SIP Account to Make Outgoing Call | Select the SIP account you want to use when making outgoing calls with the analog phone connected to this phone port. |
SIP Account(s) to Receive Incoming Call | Select a SIP account if you want to receive phone calls for the selected SIP account on this phone port. If you select more than one SIP account for incoming calls, there is no way to distinguish between them when you receive phone calls. If you do not select a source for incoming calls, you cannot receive any calls on this phone port. |
Immediate Dial Enable | Select this if you want to use the pound key (#) to tell the Zyxel Device to make the phone call immediately, instead of waiting for the number of second you selected in the Dialog Interval Selection field of the VoIP > SIP > SIP Service Provider > Add New Provider or Edit screen. If you select this, dial the phone number, and then press the pound key. The Zyxel Device makes the call immediately instead of waiting. You can still wait, if you want. |
Cancel | Click Cancel to exit this screen without saving |
OK | Click OK to save your changes. |
Phone Region
Use this screen to configure settings that depend on which region of the world the Zyxel Device is in. Selecting the region where the device is physically located improves the quality of phone calls.
To access this screen, click VoIP > Phone > Region.
VoIP > Phone > Region
The following table describes the labels in this screen.
LABEL | DESCRIPTION |
|---|---|
Region Setting | Select the place in which the Zyxel Device is located. |
Call Service Mode | Select the mode for supplementary phone services (call hold, call waiting, call transfer and three-way conference calls) that your VoIP service provider supports. • Europe Type – use supplementary phone services in European mode. • USA Type – use supplementary phone services American mode. You might have to subscribe to these services to use them. Contact your VoIP service provider. |
Apply | Click this to save your changes and to apply them to the Zyxel Device. |
Cancel | Click this to set every field in this screen to its last-saved value. |
You need to reboot the Zyxel Device after changing the region settings for it to take effect.Call Rule
Use this screen to add, edit, or remove speed-dial numbers for outgoing calls. Speed dial provides shortcuts for dialing frequently-used (VoIP) phone numbers. You also have to create speed-dial entries if you want to call SIP numbers that contain letters. Once you have configured a speed dial rule, you can use a shortcut (the speed dial number, #01 for example) on your phone's keypad to call the phone number. To access this screen, click VoIP > Call Rule.
VoIP > Call Rule
The following table describes the labels in this screen.
LABEL | DESCRIPTION |
|---|---|
Keys | This field displays the speed-dial number you should dial to use this entry. |
Number | Enter the SIP number you want the Zyxel Device to call when you dial the speed-dial number. |
Description | Enter a short description to identify the party you call when you dial the speed-dial number. You can use up to 127 printable characters except [ " ], [ ` ], [ ' ], [ < ], [ > ], [ ^ ], [ $ ], [ | ], [ & ], or [ ; ]. Spaces are allowed. |
Clear All Speed Dials | Click this button to remove all speed dials saved. |
Apply | Click this to save your changes and to apply them to the Zyxel Device. |
Cancel | Click this to set every field in this screen to its last-saved value. |
Call History
The Zyxel Device logs calls from or to your SIP addresses. This screen allows you to view a summary of received, dialed and missed calls and a call history list. You can also view detailed information on each outgoing and incoming call.
To access this screen, click VoIP > Call History.
VoIP > Call History
Each field is described in the following table.
Label | Description |
|---|---|
Clear | Click this button to remove all entries from the call history list. |
Refresh | Click this button to renew the call history list. |
Summary | |
Date | This is the date when the calls were made. |
Total Calls | This displays the total number of calls from or to your SIP addresses that day. |
Outgoing Calls | This displays how many calls originated from you that day. |
Incoming Calls | This displays how many calls you received that day. |
Missing Calls | This displays how many incoming calls were not answered that day. |
Total Duration (hh:mm:ss) | This displays how long all calls lasted that day. |
Classify | Select the type of the calls. The call types are: All, Incoming, Outgoing and Missed. |
Type | This displays the type of the calls. |
Date/Time | This displays the date and time when the calls were made. |
Peer Number | This displays the SIP address that called you or you called. |
Phone Number | This displays the phone number of the call. |
Duration (hh:mm:ss) | This displays how long the call lasted. |
Delete | Click the Delete icon to remove the call history. |
Technical Reference
This section contains background material relevant to the VoIP screens.
VoIP
VoIP is the sending of voice signals over Internet Protocol. This allows you to make phone calls and send faxes over the Internet at a fraction of the cost of using the traditional circuit-switched telephone network. You can also use servers to run telephone service applications like PBX services and voice mail. Internet Telephony Service Provider (ITSP) companies provide VoIP service.
Circuit-switched telephone networks require 64 kilobits per second (Kbps) in each direction to handle a telephone call. VoIP can use advanced voice coding techniques with compression to reduce the required bandwidth.
SIP
The Session Initiation Protocol (SIP) is an application-layer control (signaling) protocol that handles the setting up, altering and tearing down of voice and multimedia sessions over the Internet.
SIP signaling is separate from the media for which it handles sessions. The media that is exchanged during the session can use a different path from that of the signaling. SIP handles telephone calls and can interface with traditional circuit-switched telephone networks.
SIP Identities
A SIP account uses an identity (sometimes referred to as a SIP address). A complete SIP identity is called a SIP URI (Uniform Resource Identifier). A SIP account's URI identifies the SIP account in a way similar to the way an email address identifies an email account. The format of a SIP identity is SIP-Number@SIP-Service-Domain.
SIP Number
The SIP number is the part of the SIP URI that comes before the “@” symbol. A SIP number can use letters like in an email address (johndoe@your-ITSP.com for example) or numbers like a telephone number (1122334455@VoIP-provider.com for example).
SIP Service Domain
The SIP service domain of the VoIP service provider is the domain name in a SIP URI. For example, if the SIP address is 1122334455@VoIP-provider.com, then “VoIP-provider.com” is the SIP service domain.
SIP Registration
Each Zyxel Device is an individual SIP User Agent (UA). To provide voice service, it has a public IP address for SIP and RTP protocols to communicate with other servers.
A SIP user agent has to register with the SIP registrar and must provide information about the users it represents, as well as its current IP address (for the routing of incoming SIP requests). After successful registration, the SIP server knows that the users (identified by their dedicated SIP URIs) are represented by the UA, and knows the IP address to which the SIP requests and responses should be sent.
Registration is initiated by the User Agent Client (UAC) running in the VoIP gateway (the Zyxel Device). The gateway must be configured with information letting it know where to send the REGISTER message, as well as the relevant user and authorization data.
A SIP registration has a limited lifespan. The User Agent Client must renew its registration within this lifespan. If it does not do so, the registration data will be deleted from the SIP registrar's database and the connection will be broken.
The Zyxel Device attempts to register all enabled subscriber ports when it is switched on. When you enable a subscriber port that was previously disabled, the Zyxel Device attempts to register the port immediately.
Authorization Requirements
SIP registrations (and subsequent SIP requests) require a username and password for authorization. These credentials are validated through a challenge / response system using the HTTP digest mechanism (as detailed in RFC 3261, "SIP: Session Initiation Protocol").
SIP Servers
SIP is a client-server protocol. A SIP client is an application program or device that sends SIP requests. A SIP server responds to the SIP requests.
When you use SIP to make a VoIP call, it originates at a client and terminates at a server. A SIP client could be a computer or a SIP phone. One device can act as both a SIP client and a SIP server.
SIP User Agent
A SIP user agent can make and receive VoIP telephone calls. This means that SIP can be used for peer-to-peer communications even though it is a client-server protocol. In the following figure, either A or B can act as a SIP user agent client to initiate a call. A and B can also both act as SIP user agents to receive the call.
SIP User Agent
SIP Proxy Server
A SIP proxy server receives requests from clients and forwards them to another server.
In the following example, you want to use client device A to call someone who is using client device C.
1 The client device (A in the figure) sends a call invitation to the SIP proxy server (B).
2 The SIP proxy server forwards the call invitation to C.
SIP Proxy Server
SIP Redirect Server
A SIP redirect server accepts SIP requests, translates the destination address to an IP address and sends the translated IP address back to the device that sent the request. Then the client device that originally sent the request can send requests to the IP address that it received back from the redirect server. Redirect servers do not initiate SIP requests.
In the following example, you want to use client device A to call someone who is using client device C.
1 Client device A sends a call invitation for C to the SIP redirect server (B).
2 The SIP redirect server sends the invitation back to A with C’s IP address (or domain name).
3 Client device A then sends the call invitation to client device C.
SIP Redirect Server
SIP Register Server
A SIP register server maintains a database of SIP identity-to-IP address (or domain name) mapping. The register server checks your user name and password when you register.
RTP
When you make a VoIP call using SIP, the RTP (Real time Transport Protocol) is used to handle voice data transfer. See RFC 1889 for details on RTP.
Pulse Code Modulation
Pulse Code Modulation (PCM) measures analog signal amplitudes at regular time intervals and converts them into bits.
SIP Call Progression
The following figure displays the basic steps in the setup and tear down of a SIP call. A calls B.
A | B | |
|---|---|---|
1. INVITE |  | |
 | 2. Ringing | |
 | 3. OK | |
4. ACK |  | |
5.Dialogue (voice traffic) | ||
6. BYE |  | |
 | 7. OK |
1 A sends a SIP INVITE request to B. This message is an invitation for B to participate in a SIP telephone call.
2 B sends a response indicating that the telephone is ringing.
3 B sends an OK response after the call is answered.
4 A then sends an ACK message to acknowledge that B has answered the call.
5 Now A and B exchange voice media (talk).
6 After talking, A hangs up and sends a BYE request.
7 B replies with an OK response confirming receipt of the BYE request and the call is terminated.
SIP Call Progression Through Proxy Servers
Usually, the SIP UAC sets up a phone call by sending a request to the SIP proxy server. Then, the proxy server looks up the destination to which the call should be forwarded (according to the URI requested by the SIP UAC). The request may be forwarded to more than one proxy server before arriving at its destination.
The response to the request goes to all the proxy servers through which the request passed, in reverse sequence. Once the session is set up, session traffic is sent between the UAs directly, bypassing all the proxy servers in between.
The following figure shows the SIP and session traffic flow between the user agents (UA 1 and UA 2) and the proxy servers (this example shows two proxy servers, PROXY 1 and PROXY 2).
SIP Call Through Proxy Servers
The following table shows the SIP call progression.
UA 1 | Proxy 1 | Proxy 2 | UA 2 | |||
|---|---|---|---|---|---|---|
Invite |  | |||||
Invite |  | |||||
 | 100 Trying | Invite |  | |||
 | 100 Trying | |||||
 | 180 Ringing | |||||
 | 180 Ringing | |||||
 | 180 Ringing | |||||
 | 200 OK | |||||
 | 200 OK | |||||
 | 200 OK | |||||
ACK |  | |||||
RTP |  | RTP | ||||
 | BYE | |||||
200 OK |  | |||||
1 User Agent 1 sends a SIP INVITE request to Proxy 1. This message is an invitation to User Agent 2 to participate in a SIP telephone call. Proxy 1 sends a response indicating that it is trying to complete the request.
2 Proxy 1 sends a SIP INVITE request to Proxy 2. Proxy 2 sends a response indicating that it is trying to complete the request.
3 Proxy 2 sends a SIP INVITE request to User Agent 2.
4 User Agent 2 sends a response back to Proxy 2 indicating that the phone is ringing. The response is relayed back to User Agent 1 through Proxy 1.
5 User Agent 2 sends an OK response to Proxy 2 after the call is answered. This is also relayed back to User Agent 1 through Proxy 1.
6 User Agent 1 and User Agent 2 exchange RTP packets containing voice data directly, without involving the proxies.
7 When User Agent 2 hangs up, he sends a BYE request.
8 User Agent 1 replies with an OK response confirming receipt of the BYE request, and the call is terminated.
Voice Coding
A codec (coder/decoder) codes analog voice signals into digital signals and decodes the digital signals back into analog voice signals. The Zyxel Device supports the following codecs.
• G.711 is a Pulse Code Modulation (PCM) waveform codec. PCM measures analog signal amplitudes at regular time intervals and converts them into digital samples. G.711 provides very good sound quality but requires 64 kbps of bandwidth.
• G.726 is an Adaptive Differential PCM (ADPCM) waveform codec that uses a lower bitrate than standard PCM conversion. ADPCM converts analog audio into digital signals based on the difference between each audio sample and a prediction based on previous samples. The more similar the audio sample is to the prediction, the less space needed to describe it. G.726 operates at 16, 24, 32 or 40 kbps.
• G.729 is an Analysis-by-Synthesis (AbS) hybrid waveform codec that uses a filter based on information about how the human vocal tract produces sounds. G.729 provides good sound quality and reduces the required bandwidth to 8 kbps.
Voice Activity Detection/Silence Suppression
Voice Activity Detection (VAD) detects whether or not speech is present. This lets the Zyxel Device reduce the bandwidth that a call uses by not transmitting “silent packets” when you are not speaking.
Comfort Noise Generation
When using VAD, the Zyxel Device generates comfort noise when the other party is not speaking. The comfort noise lets you know that the line is still connected as total silence could easily be mistaken for a lost connection.
Echo Cancellation
G.168 is an ITU-T standard for eliminating the echo caused by the sound of your voice reverberating in the telephone receiver while you talk.
MWI (Message Waiting Indication)
Message Waiting Indication (MWI) enables your phone to give you a message–waiting (beeping) dial tone when you have a voice message(s). Your VoIP service provider must have a messaging system that sends message waiting status SIP packets as defined in RFC 3842.
Custom Tones (IVR)
Interactive Voice Response (IVR) is a feature that allows you to use your telephone to interact with the Zyxel Device. The Zyxel Device allows you to record custom tones for the Early Media and Music On Hold functions. The same recordings apply to both the caller ringing and on hold tones.
label | Description |
|---|---|
Total Time for All Tones | 900 seconds for all custom tones combined |
Maximum Time per Individual Tone | 180 seconds |
Total Number of Tones Recordable | 5 You can record up to 5 different custom tones but the total time must be 900 seconds or less. |
Recording Custom Tones
Use the following steps if you would like to create new tones or change your tones:
1 Pick up the phone and press “****” on your phone’s keypad and wait for the message that says you are in the configuration menu.
2 Press a number from 1101 – 1105 on your phone followed by the “#” key.
3 Play your desired music or voice recording into the receiver’s mouthpiece. Press the “#” key.
4 You can continue to add, listen to, or delete tones, or you can hang up the receiver when you are done.
Listening to Custom Tones
Do the following to listen to a custom tone:
1 Pick up the phone and press “****” on your phone’s keypad and wait for the message that says you are in the configuration menu.
2 Press a number from 1201 – 1208 followed by the “#” key to listen to the tone.
3 You can continue to add, listen to, or delete tones, or you can hang up the receiver when you are done.
Deleting Custom Tones
Do the following to delete a custom tone:
1 Pick up the phone and press “****” on your phone’s keypad and wait for the message that says you are in the configuration menu.
2 Press a number from 1301 – 1308 followed by the “#” key to delete the tone of your choice. Press 14 followed by the “#” key if you wish to clear all your custom tones.
You can continue to add, listen to, or delete tones, or you can hang up the receiver when you are done.
Quality of Service (QoS)
Quality of Service (QoS) refers to both a network's ability to deliver data with minimum delay, and the networking methods used to provide bandwidth for real-time multimedia applications.
Type of Service (ToS)
Network traffic can be classified by setting the ToS (Type of Service) values at the data source (for example, at the Zyxel Device) so a server can decide the best method of delivery, that is the least cost, fastest route and so on.
DiffServ
DiffServ is a Class of Service (CoS) model that marks packets so that they receive specific per-hop treatment at DiffServ-compliant network devices along the route based on the application types and traffic flow. Packets are marked with DiffServ Code Points (DSCP) indicating the level of service desired. This allows the intermediary DiffServ-compliant network devices to handle the packets differently depending on the code points without the need to negotiate paths or remember state information for every flow. In addition, applications do not have to request a particular service or give advanced notice of where the traffic is going.1
DSCP and Per-Hop Behavior
DiffServ defines a new DS (Differentiated Services) field to replace the Type of Service (TOS) field in the IP header. The DS field contains a 2-bit unused field and a 6-bit DSCP field, which can define up to 64 service levels. The following figure illustrates the DS field.
DSCP is backward compatible with the three precedence bits in the ToS octet so that non-DiffServ compliant, ToS-enabled network device will not conflict with the DSCP mapping.
DiffServ: Differentiated Service Field
DSCP (6-bit) | Unused (2-bit) | |
The DSCP value determines the forwarding behavior, the PHB (Per-Hop Behavior), that each packet gets across the DiffServ network. Based on the marking rule, different kinds of traffic can be marked for different priorities of forwarding. Resources can then be allocated according to the DSCP values and the configured policies.
Phone Services Overview
Supplementary services such as call hold, call waiting, and call transfer are generally available from your VoIP service provider. The Zyxel Device supports the following services:
• Call Return
• Call Hold
• Call Waiting
• Making a Second Call
• Call Transfer
• Call Forwarding
• Three-Way Conference
• Internal Calls
• Call Park and Pickup
• Do not Disturb
• IVR
• Call Completion
• CCBS
• Outgoing SIP
To take full advantage of the supplementary phone services available through the Zyxel Device's phone ports, you may need to subscribe to the services from your VoIP service provider.The Flash Key
Flashing means to press the hook for a short period of time (a few hundred milliseconds) before releasing it. On newer telephones, there should be a "flash" key (button) that generates the signal electronically. If the flash key is not available, you can tap (press and immediately release) the hook by hand to achieve the same effect. However, using the flash key is preferred since the timing is much more precise. With manual tapping, if the duration is too long, it may be interpreted as hanging up by the Zyxel Device.
You can invoke all the supplementary services by using the flash key.
Europe Type Supplementary Phone Services
This section describes how to use supplementary phone services with the Europe Type Call Service Mode. Commands for supplementary services are listed in the table below.
After pressing the flash key, if you do not issue the sub-command before the default sub-command timeout (2 seconds) expires or issue an invalid sub-command, the current operation will be aborted.
Command | Sub-command | Description |
|---|---|---|
Flash | Put a current call on hold to place a second call. Switch back to the call (if there is no second call). | |
Flash | 0 | Drop the call presently on hold or reject an incoming call which is waiting to be answered. |
Flash | 1 | Disconnect the current phone connection and answer the incoming call or resume with caller presently on hold. |
Flash | 2 | 1. Switch back and forth between two calls. 2. Put a current call on hold to answer an incoming call. 3. Separate the current three-way conference call into two individual calls (one is on-line, the other is on hold). |
Flash | 3 | Create three-way conference connection. |
Flash | *98# | Transfer the call to another phone. |
European Call Hold
Call hold allows you to put a call (A) on hold by pressing the flash key.
If you have another call, press the flash key and then “2” to switch back and forth between caller A and B by putting either one on hold.
Press the flash key and then “0” to disconnect the call presently on hold and keep the current call on line.
Press the flash key and then “1” to disconnect the current call and resume the call on hold.
If you hang up the phone but a caller is still on hold, there will be a remind ring.
European Call Waiting
This allows you to place a call on hold while you answer another incoming call on the same telephone (directory) number.
If there is a second call to a telephone number, you will hear a call waiting tone. Take one of the following actions.
• Reject the second call.
Press the flash key and then press “0”.
• Disconnect the first call and answer the second call.
Either press the flash key and press “1”, or just hang up the phone and then answer the phone after it rings.
• Put the first call on hold and answer the second call.
Press the flash key and then “2”.
European Call Transfer
Do the following to transfer an incoming call (that you have answered) to another phone.
1 Press the flash key to put the caller on hold.
2 When you hear the dial tone, dial “*98#” followed by the number to which you want to transfer the call.
3 After you hear the ring signal or the second party answers it, hang up the phone.
European Three-Way Conference
Use the following steps to make three-way conference calls.
1 When you are on the phone talking to someone, press the flash key to put the caller on hold and get a dial tone.
2 Dial a phone number directly to make another call.
3 When the second call is answered, press the flash key and press “3” to create a three-way conversation.
4 Hang up the phone to drop the connection.
5 If you want to separate the activated three-way conference into two individual connections (one is on-line, the other is on hold), press the flash key and press “2”.
USA Type Supplementary Services
This section describes how to use supplementary phone services with the USA Type Call Service Mode. Commands for supplementary services are listed in the table below.
After pressing the flash key, if you do not issue the sub-command before the default sub-command timeout (2 seconds) expires or issue an invalid sub-command, the current operation will be aborted.
Command | Sub-command | Description |
|---|---|---|
Flash | Put a current call on hold to place a second call. After the second call is successful, press the flash key again to have a three-way conference call. Put a current call on hold to answer an incoming call. | |
Flash | *98# | Transfer the call to another phone. |
USA Call Hold
Call hold allows you to put a call (A) on hold by pressing the flash key.
If you have another call, press the flash key to switch back and forth between caller A and B by putting either one on hold.
If you hang up the phone but a caller is still on hold, there will be a remind ring.
USA Call Waiting
This allows you to place a call on hold while you answer another incoming call on the same telephone (directory) number.
If there is a second call to your telephone number, you will hear a call waiting tone.
Press the flash key to put the first call on hold and answer the second call.
USA Call Transfer
Do the following to transfer an incoming call (that you have answered) to another phone.
1 Press the flash key to put the caller on hold.
2 When you hear the dial tone, dial “*98#” followed by the number to which you want to transfer the call.
3 After you hear the ring signal or the second party answers it, hang up the phone.
USA Three-Way Conference
Use the following steps to make three-way conference calls.
1 When you are on the phone talking to someone (party A), press the flash key to put the caller on hold and get a dial tone.
2 Dial a phone number directly to make another call (to party B).
3 When party B answers the second call, press the flash key to create a three-way conversation.
4 Hang up the phone to drop the connection.
5 If you want to separate the activated three-way conference into two individual connections (with party A on-line and party B on hold), press the flash key.
6 If you want to go back to the three-way conversation, press the flash key again.
7 If you want to separate the activated three-way conference into two individual connections again, press the flash key. This time the party B is on-line and party A is on hold.
Phone Functions Summary
The following table shows the key combinations you can enter on your phone’s keypad to use certain features.
ACTION | FUNCTION | DESCRIPTION |
|---|---|---|
*98# | Call transfer | Transfer a call to another phone. See Europe Type Supplementary Phone Services (Europe type) and USA Type Supplementary Services (USA type). |
*66# | Call return | Place a call to the last person who called you. |
*95# | Enable Do Not Disturb | Use these to set your phone not to ring when someone calls you, or to turn this function off. |
#95# | Disable Do Not Disturb | |
*41# | Enable Call Waiting | Use these to allow you to put a call on hold when you are answering another, or to turn this function off. |
#41# | Disable Call Waiting | |
**** | IVR | Use these to set up Interactive Voice Response (IVR). IVR allows you to record custom caller ringing tones (the sound a caller hears before you pick up the phone) and on hold tones (the sound someone hears when you put their call on hold). |
#### | Internal Call | Call the phone(s) connected to the Zyxel Device. |
*82 | One Shot Caller Display Call | Activate or deactivate caller ID for the next call only. |
*67 | One Shot Caller Hidden Call |
1 The Zyxel Device does not support DiffServ at the time of writing.