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Chapter 6: Using the Web-based Utility
The Voice Tab
IP Telephony System
The Voice - Line 1/2/3/4 Screen
Use the appropriate screen to configure settings for each external Internet phone line.
Line Enable
. To enable this line for service, select
yes
. Otherwise, select
no
. The default is
yes
.
Network Settings
SIP ToS/DiffServ Value
. Enter the TOS/DiffServ field value in UDP IP packets carrying SIP messages. The default
is
0x68
.
SIP CoS Value
. Enter the CoS value for SIP messages. The default is
3
.
SIP Settings
SIP Port
. Enter the port number of the SIP message listening and transmission port. The default is
5060
.
SIP 100REL Enable
. To enable the support of 100REL SIP extension for reliable transmission of provisional
responses (18x) and use of PRACK requests, select
yes
. Otherwise, select
no
. The default is
no
.
Auth Resync-Reboot
. If this feature is enabled, the System will authenticate the sender when it receives the
NOTIFY resync reboot (RFC 2617) message. To use this feature, select
yes
. Otherwise, select
no
. The default
is
yes
.
SIP Proxy-Require
. The SIP proxy can support a specific extension or behavior when it sees this header from the
user agent. If this field is configured and the proxy does not support it, then it will respond with the message,
“unsupported.” Enter the appropriate header in the field provided.
SIP Remote-Party-ID
. To use the Remote-Party-ID header instead of the From header, select
yes
. Otherwise,
select
no
. The default is
yes
.
SIP Debug Option
. SIP messages are received at or sent from the proxy listen port. This feature controls which
SIP messages to log. Select
none
for no logging. Select
1-line
to log the start-line only for all messages. Select
1-line excl. OPT
to log the start-line only for all messages except OPTIONS requests/responses. Select
1-line
excl. NTFY
to log the start-line only for all messages except NOTIFY requests/responses. Select
1-line excl. REG
to log the start-line only for all messages except REGISTER requests/responses. Select
1-line excl.
OPT|NTFY|REG
to log the start-line only for all messages except OPTIONS, NOTIFY, and REGISTER
IMPORTANT:
In most cases, you should not change the service settings unless instructed to do
by your ITSP.
Figure 6-44: Voice - Line 1 Screen - Network Settings
Figure 6-45: Voice - Line 1 Screen - SIP Settings
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Chapter 6: Using the Web-based Utility
The Voice Tab
IP Telephony System
requests/responses. Select
full
to log all SIP messages in full text. Select
full excl. OPT
to log all SIP messages
in full text except OPTIONS requests/responses. Select
full excl. NTFY
to log all SIP messages in full text except
NOTIFY requests/responses. Select
full excl. REG
to log all SIP messages in full text except REGISTER
requests/responses. Select
full excl. OPT|NTFY|REG
to log all SIP messages in full text except for OPTIONS,
NOTIFY, and REGISTER requests/responses. The default is
none
.
Restrict Source IP
. If Lines 1 and 2 use the same SIP Port value and the Restrict Source IP feature is enabled,
then the proxy IP address for Lines 1 and 2 is treated as an acceptable IP address for both lines. To enable the
Restrict Source IP feature, select
yes
. Otherwise, select
no
. The default is
no
.
Referor Bye Delay
. This controls when the System sends BYE to terminate stale call legs upon completion of call
transfers. Multiple delay settings (Referor, Refer Target, Referee, and Refer-To Target) are configured on this
screen. For the Referor Bye Delay, enter the appropriate period of time in seconds. The default is
4
.
Refer Target Bye Delay
. For the Refer Target Bye Delay, enter the appropriate period of time in seconds. The
default is
0
.
Referee Bye Delay
. For the Referee Bye Delay, enter the appropriate period of time in seconds. The default is
0
.
Refer-To Target Contact
. To contact the refer-to target, select
yes
. Otherwise, select
no
. The default is
no
.
Subscriber Information
Display Name
. Enter the display name for caller ID.
User ID
. Enter the extension number for this line.
Password
. Enter the password for this line.
Use Auth ID
. To use the authentication ID and password for SIP authentication, select
yes
. Otherwise, select
no
to use the user ID and password. The default is
no
.
Auth ID
. Enter the authentication ID here.
Call Capacity
. Select the maximum number of calls allowed on this line. (The System does not distinguish
between incoming and outgoing calls when it determines its call capacity.)
Contact List
. This is a list of clients that System should alert when there is an incoming call on this line. Each rule
is also known as a hunt group. The default method to ring a group is to ring all the members simultaneously,
unless a hunt rule is specified. The default contact list is
aa
(auto-attendant).
Figure 6-46: Voice - Line 1 Screen - Subscriber
Information
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Chapter 6: Using the Web-based Utility
The Voice Tab
IP Telephony System
When you create this rule, follow this format:
rule[|rule[|rule[...]]]
The most specific rules should be placed first.
Each rule should be in this format: [
did
:]
ext
[,
ext
[,
ext
[...]]][,name=
gname
][,hunt=
hrule
][,cfwd=
target
]
The term did indicates an embedded Direct Inward Dialing (DID) number. If it is not specified, then the rule applies
to any DID number.
The term ext indicates the client extension number pattern. It accepts * and ? wildcards as well as %xx escaped
characters.
The term name is a name for the call group.
If a hrule is specified, then the listed clients are contacted sequentially (also known as hunting); otherwise, they
will ring simultaneously. When you create this hrule, follow this format:
hunt=<
algo
>;<
interval
>;<
max
>
The term <
algo
> determines the order to ring the clients. It can be one of the following:
restart
or
re=
so it always starts from the beginning of the list
next
or
ne=
so it starts from the next on the list to the last client that rings
random
or
ra=
so the order is random for each call
The term <
interval
> is the time, in seconds, to ring each client.
The term <
max
> is the total time, in seconds, to hunt before the call is rejected or forwarded to voicemail. If
<
max
> is less than <
interval
>, it is interpreted as the number of cycles to go through the hunt group before
call hunting stops. If <
max
> is 0, call hunting will continue indefinitely until the caller hangs up or someone
answers the call.
If necessary, the call is forwarded to a user ID, called the target, in the hrule. If the target is a voicemail mailbox,
then the target starts with
vm
. For example, the target vm3456 will have calls forwarded to voicemail with the
mailbox ID 3456.
For example, the Contact List is 501,502,hunt=ne,4,1;cfwd=aa. This means that 501 will ring first for four
seconds. If 501 does not pick up or is already on another call, then 502 will ring for four seconds. This cycle will
repeat once before the call hunting stops. Then the call goes to the auto-attendant.
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Chapter 6: Using the Web-based Utility
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IP Telephony System
Cfwd No Ans Delay
. Enter the delay, in seconds, before the call forwarding of no-answer calls feature is
triggered. The default is
20
.
Dial Plan
Dial Plan
. Enter the dial plan script for this line. Refer to “Appendix C: Dial Plan and Auto-Attendant Scripting for
Advanced Users” for more details. The default is
(<9:>xx.)
.
NAT Settings
NAT Mapping Enable
. To use externally mapped IP addresses and SIP/RTP ports in SIP messages, select
yes
.
Otherwise, select
no
. The default is
no
.
NAT Keep Alive Enable
. To send the configured NAT keep alive message periodically, select
yes
. Otherwise,
select
no
. The default is
no
.
NAT Keep Alive Msg
. Enter the keep alive message that should be sent periodically to maintain the current NAT
mapping. If the value is $NOTIFY, then a NOTIFY message is sent. If the value is $REGISTER, then a REGISTER
message without contact is sent. The default is
$NOTIFY
.
NAT Keep Alive Dest
. Enter the destination that should receive NAT keep alive messages. If the value is $PROXY,
then the messages will be sent to the current or outbound proxy. The default is
$PROXY
.
EXT SIP Port
. Enter the port number of the external port to substitute for the actual SIP port of the System in all
outgoing SIP messages.
Proxy and Registration
Proxy
. Enter the SIP proxy server for all outbound requests.
Use Outbound Proxy
. To use the outbound proxy, select
yes
. Otherwise, select
no
. The default is
no
.
Outbound Proxy
. Enter the SIP outbound proxy server, where all outbound requests are sent for the first hop.
Use OB Proxy In Dialog
. To force SIP requests to be sent to the outbound proxy within a dialog, select
yes
.
Otherwise, select
no
. The default is
yes
.
Register
. To require periodic registration with the proxy server, select
yes
. Otherwise, select
no
. The default
is
yes
.
Make Call Without Reg
. To allow outbound calls to be made without successful registration by the System,
select
yes
. Otherwise, select
no
. The default is
no
.
Figure 6-48: Voice - Line 1 Screen - NAT Settings
Figure 6-49: Voice - Line 1 Screen - Proxy and
Registration
Figure 6-47: Voice - Line 1 Screen - Dial Plan
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Chapter 6: Using the Web-based Utility
The Voice Tab
IP Telephony System
Register Expires
. This is the expiration value, in seconds, of a REGISTER request. The System will periodically
renew registration shortly before the current registration has expired. The default is
3600
.
Ans Call Without Reg
. To allow inbound calls to be answered without successful registration by the System,
select
yes
. Otherwise, select
no
. The default is
no
.
Use DNS SRV
. To use the DNS SRV lookup for proxy and outbound proxy, select
yes
. Otherwise, select
no
. The
default is
no
.
DNS SRV Auto Prefix
. To have the System automatically prepend the proxy or outbound proxy name with
_sip._udp when it is performing a DNS SRV lookup on that name, select
yes
. Otherwise, select
no
. The default
is
no
.
Proxy Fallback Intvl
. This sets the delay, in seconds, after which the System will retry from the highest priority
proxy (or outbound proxy) servers after it has failed over to a lower priority server. It will work only if the primary
and backup proxy server list is provided to the System via the DNS SRV record lookup on the server name. The
default is
3600
.
Proxy Redundancy Method
. The System will create an internal list of proxies returned in DNS SRV records. You
have a choice of two modes. Select
Normal
mode if you want this list to contain proxies organized by weight and
priority. Select
Based on SRV Port
mode if you want the System to use Normal mode first and then inspect the
port number based on the first proxy’s port on the list. The default is
Normal
.
Mailbox Subscribe URL
. Enter the URL that should receive the SUBSCRIBE messages, so the System will receive
voicemail status notification for all mailboxes on this line.
Mailbox Deposit URL
. Enter the URL that the System will contact when clients and external callers need to
deposit voicemail in any of the mailboxes on this line.
Mailbox Manage URL
. Enter the URL that the IP Telephony will contact when it needs to check voicemail for any
of the mailboxes on this line.
Mailbox Status
. Displayed here is the status for all mailboxes on this line. The status is automatically updated
when the System receives voicemail status notification from the ITSP. The information is shown in this format:
[
mailboxID
:number of new messages/number of old messages[,
mailboxID
:number of new messages/number of
old messages[,
mailboxID
:number of new messages/number of old messages[,...]]]]
When you have finished making changes, click the
Submit All Changes
button to save the changes, or click the
Undo All Changes
button to undo your changes.

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