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Page 26 / 65 Scroll up to view Page 21 - 25
ADSL2+ Wireless N300 Modem Router with VoIP User Guide
YML9WMAXXN
26
www.netcomm.com.au
Voice
This chapter first describes the various options for configuration of the SIP voice service.
It then provides detailed instructions for making
telephone calls using VoIP (Voice over IP) or PSTN (Public Switched Telephone Network) services
5.1 SIP
Session Initiation Protocol (SIP) is a peer-to-peer protocol used for Internet conferencing, telephony, events notification, presence and
instant messaging.
SIP is designed to address the functions of signalling and session management within a packet telephony network.
Signalling allows call
information to be carried across network boundaries.
Session management provides the ability to control the attributes of an end-to-end
call.
The SIP standard defines the following agents/servers:
1.
User Agents (UA) - SIP phone clients (hardware or software)
2.
Proxy Server – relays data between UA and external servers
3.
Registrar Server - a server that accepts register requests from UA
4.
Redirect Server – provides an address lookup service to UA
NOTE:
The SIP standard is set by the Internet Engineering Task Force (IETF).
The following subsections present Basic, Advanced and Debug SIP screens.
Each screen provides various options for customizing the SIP
configuration.
5.1.1
SIP BASIC
This screen contains basic SIP configuration settings.
Once settings are configured click
Save/Apply
to begin using the service.
NOTE:
Consult the tables that follow for detailed field descriptions
Interface name
Choose the WAN interface
Locale Selection
Sets tone, ring type and physical characteristics for specific countries
Preferred codecs
Choose G.711U, G.711A, G.726 or G.729
Preferred ptime
The time period used to digitally sample the analog voice signal.
The default is 20 ms.
Use SIP proxy
Enable the SIP proxy by selecting the checkbox and setting proxy parameters.
SIP Proxy
Input IP address or domain name of the SIP proxy server, used for VOIP service.
SIP Proxy port
This value is set by your VoIP provider and is normally port 5060.
Registration Expire Timeout
The time period the user would like the registration to be valid for the Registrar/ Proxy Server.
The
default is 300 seconds.
SIP domain name
Provided by your VoIP provider.
Page 27 / 65
YML9WMAXXN
ADSL2+ Wireless N300 Modem Router with VoIP User Guide
www.netcomm.com.au
27
NetComm Gateway
TM
Series - ADSL2+ Wireless N300 Modem Router with VoIP
•
A proxy is an intermediary program that acts as both a server and a client for the purpose of making requests on behalf of other
clients.
Requests are serviced internally or transferred to other servers.
A proxy interprets and, if necessary, rewrites a request
message before forwarding it
Use SIP outbound
proxy
Select if required by your VoIP provider.
Enter SIP Outbound proxy IP and port.
Line 1 & 2
Ports FXS1 & FXS2
Disabled
Ticking the checkbox disables the line
Extension
The line extension number
Display Name
The caller ID display name
Authentication Name
The authentication username for the Registrar/Proxy, given by VOIP provider.
Authentication
Password
The authentication password for the Registrar/proxy, given by VOIP provider.
5.1.2
SIP ADVANCED
This screen contains advanced SIP configuration settings.
Once settings are configured click
Save/Apply
to begin using the service.
Line 1 & 2
Ports FXS1 & FXS2
Forwarding number
Enter the forwarding phone number
Call forwarding when
busy
Tick the checkbox to enable this option
Forwarding all calls
Tick the checkbox to enable this option
Call forwarding if no
answer
Tick the checkbox to enable this option
Call waiting (default:
enabled)
Tick the checkbox to enable this option
NOTE:
These options can also be set using telephone keypad commands, as described in the call command list of section 8.2 Telephone Calls
Enable MWI
Subscription
Enable or disable Message-Waiting Indicator (MWI) for FXS Phones with this checkbox.
Enable T.38 support
(default: enabled)
Enable or disable T.38 Fax mode support with this checkbox.
You can plug a fax machine into either phone port
to send or receive faxes.
Functionality depends upon FAX support by your VoIP service provider.
Max Digits
Sets the maximum number of digits for a phone number.
Emergency Setting
Multiple emergency numbers can be set using the “|” character (shift + backslash). For example, to set 911 and
114 as emergency numbers, enter “911|114”.
Please Note:
These numbers must be changed to correspond to the emergency numbers that are used in your
location.
Dtmf Relay Setting
Set the special use of RTP packets to transmit digit events.
SIP Transport
Protocol
Set the special use of SIP protocol to transmit digit events.
Incoming PSTN Call
Routing
If PSTN route rule is Auto, an incoming PSTN call will ring an idle phone, either Phone1 or Phone2 (if Phone1 is
busy).
If PSTN route rule is Line1 or Line2, an incoming PSTN call will attempt to ring only the assigned phone line
(FXS1 or FXS2).
Enable SIP tag matching
Select if required by your VoIP provider.
(e.g. disable with Vonage service.)
Enable Music Server
Enable/disable the Music Server.
Enter the Music Server IP address and port.
5.1.3
SIP DEBUG
Page 28 / 65
ADSL2+ Wireless N300 Modem Router with VoIP User Guide
YML9WMAXXN
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www.netcomm.com.au
This screen contains SIP configuration settings used for debugging.
Once settings are configured click
Save/Apply
to begin using the service.
5.2 Telephone Calls
To make a call, simply dial the number.
The dial plan (i.e. the dialled digits) is normally customized for each installation.
The default dial plan
allows for dialling of 4-digit extensions or direct IP addresses.
Shorter extension numbers (e.g. 3-digits) can be dialled by completing the
dial string with a final #.
When a Call Server (SIP Proxy Server) is configured into the system, the dialled digits are translated and routed by the Call Server to the
correct destination as registered with the Call Server.
If no Call Server is configured, calls can still be made using 4-digit extensions, rather than using full IP addresses.
The originator translates
the dialled-digits to a destination device as follows:
First Digit:
Line identifier (for multi-line gateways)
Remaining digits:
Host number part of an IP address.
The Network number part is considered to be the same
as the caller’s IP address.
For example, if a caller at address 10.136.64.33/24 dials “2023”, the call will be placed to the second line at address 10.136.64.23.
All
devices have to be on the same Class C subnet (24-bit subnet mask).
To dial an IP address directly, dial the IP address digits using * on the keypad as the dot.
Complete the address with a final * or #.
When
using IP address dialling it is not possible to specify which line at a gateway is called, so the gateway always routes IP-address dialled calls
to the first line.
Network busy tone (fast busy) will be played for unknown or unreachable destinations.
To answer a call, pick up the phone or press the
hands free button.
Page 29 / 65
YML9WMAXXN
ADSL2+ Wireless N300 Modem Router with VoIP User Guide
www.netcomm.com.au
29
NetComm Gateway
TM
Series - ADSL2+ Wireless N300 Modem Router with VoIP
CALL COMMAND LIST
Caller ID
The Call Manager delivers Caller ID when placing calls.
The caller ID is transmitted to the analog line for CLASS recognition.
Call Hold
To put a call on hold, press flash then hang up (optional).
To return to the original call, press flash or pick up the phone.
The phone will
issue a short ring burst every 30 seconds or so while on-hook to remind you that a call is on hold.
Call Transfer
•
To transfer a call, press flash then dial the new number.
•
To transfer immediately, hang up (blind transfer).
•
To transfer with consultation, wait for the party to answer, consult, and hang up.
•
To abort the transfer (if the third party does not answer); press flash to return to the original call.
Conference Calling
To turn a two-party call into a three-party conference call, press flash and dial the third party.
Wait for the party to answer, then press flash.
To drop the third party and return to a two-party call, press flash again.
To drop yourself out of the conference, hang up.
The call will be
transferred (so that the other two parties remain connected to each other).
NOTE:
In conference mode, the conference initiator performs the audio bridge/mixing function – there are only two voice streams established.
Call Waiting
If call waiting is enabled on a line, and you hear the call waiting tone during a call, press flash to answer the second call.
The first call is
automatically placed on hold.
To switch between calls, press flash again.
•
To disable the call waiting feature, dial *60.
•
To enable the call waiting feature, dial *61.
NOTE:
Call forward feature settings (Busy or All) take priority over the call-waiting feature.
The call-waiting feature is ignored on new incoming calls if there is already a call on
hold or in conference.
Call Forward Number
•
To set the call forward number, dial *74 then the number.
(Note that this does not actually enable forwarding; to do so, select the
call forward action as described below.)
•
To disable all call forwarding features, dial *70
Call Forward
No Answer
•
To enable call forward on no answer, dial *71. Incoming calls will be forwarded if unanswered for 18 seconds.
Call Forward Busy
•
To enable call forward if busy, dial *72. Incoming calls will forward immediately if the phone is off-hook.
Call Forward All
•
To enable call forward for all calls, dial *73.
•
To disable the “forward all calls” feature, dial *75. Settings for Call Forward Busy or
No Answer are not modified.
Call Return
•
To place a call to the last known incoming caller (unanswered or not), dial *69.
Redial
•
To redial the last outgoing number, dial *68.
Page 30 / 65
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