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Chapter 6: Using the Web-based Utility
The Voice Tab
IP Telephony System
is *72. After the user dials *72, the System will wait for the user to enter a phone number. After the number has
been entered, the System will forward all calls for that phone number.
Vertical Service Announcement Codes
Service Annc Base Number
. Enter the base number for service announcements.
Service Annc Extension Codes
. Enter the extension codes for service announcements.
Outbound Call Codec Selection Codes
Prefer G711u Code
. This is the dialing code that will make this codec the preferred codec for the associated call.
The default is
*017110
.
Force G711u Code
. This is the dialing code that will make this codec the only codec that can be used for the
associated call. The default is
*027110
.
Prefer G711a Code
. This is the dialing code that will make this codec the preferred codec for the associated call.
The default is
*017111
.
Force G711a Code
. This is the dialing code that will make this codec the only codec that can be used for the
associated call. The default is
*027111
.
Prefer G723 Code
. This is the dialing code that will make this codec the preferred codec for the associated call.
The default is
*01723
.
Force G723 Code
. This is the dialing code that will make this codec the only codec that can be used for the
associated call. The default is
*02723
.
Prefer G726r16 Code
. This is the dialing code that will make this codec the preferred codec for the associated
call. The default is
*0172616
.
Force G726r16 Code
. This is the dialing code that will make this codec the only codec that can be used for the
associated call. The default is
*0272616
.
Prefer G726r24 Code
. This is the dialing code that will make this codec the preferred codec for the associated
call. The default is
*0172624
.
Force G726r24 Code
. This is the dialing code that will make this codec the only codec that can be used for the
associated call. The default is
*0272624
.
Figure 6-33: Voice - Regional Screen - Vertical Service
Announcement Codes
Figure 6-34: Voice - Regional Screen - Outbound Call
Codec Selection Codes
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Chapter 6: Using the Web-based Utility
The Voice Tab
IP Telephony System
Prefer G726r32 Code
. This is the dialing code that will make this codec the preferred codec for the associated
call. The default is
*0172632
.
Force G726r32 Code
. This is the dialing code that will make this codec the only codec that can be used for the
associated call. The default is
*0272632
.
Prefer G726r40 Code
. This is the dialing code that will make this codec the preferred codec for the associated
call. The default is
*0172640
.
Force G726r40 Code
. This is the dialing code that will make this codec the only codec that can be used for the
associated call. The default is
*0272640
.
Prefer G729a Code
. This is the dialing code that will make this codec the preferred codec for the associated call.
The default is
*01729
.
Force G729a Code
. This is the dialing code that will make this codec the only codec that can be used for the
associated call. The default is
*02729
.
Miscellaneous
Set Local Date (mm/dd)
. Set the local date (mm stands for months and dd stands for days). The year is optional
and uses two or four digits.
Set Local Time (hh/mm)
. Set the local time (hh stands for hours and mm stands for minutes). Seconds are
optional.
Time Zone
. For caller ID generation, select the number of hours to add to GMT to generate the local time. The
default is
GMT-08:00
.
FXS Port Impedance
. This sets the electrical impedance of the FXS port. Select one of these choices:
600
,
900
,
600+2.16uF
,
900+2.16uF
,
270+750||150nF
,
220+850||120nF
,
220+820||115nF
, or
370+620||310nF
. The
default is
600
.
Daylight Saving Time Rule
. Enter the rule for calculating daylight saving time; it should include the start, end,
and save values. This rule is comprised of three fields. Each field is separated by ; (a semicolon) as shown below.
Optional values inside [ ] (the brackets) are assumed to be 0 if they are not specified. Midnight is represented by
0:0:0 of the given date.
This is the format of the rule: Start = <start-time>; end=<end-time>; save = <save-time>
The <start-time> and <end-time> values specify the start and end dates and times of daylight saving time. Each
value is in this format: <month> /<day> / <weekday>[/HH:[mm[:ss]]]
Figure 6-35: Voice - Regional Screen - Miscellaneous
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Chapter 6: Using the Web-based Utility
The Voice Tab
IP Telephony System
The <save-time> value is the number of hours, minutes, and/or seconds to add to the current time during
daylight saving time. The <save-time> value can be preceded by a negative (-) sign if subtraction is desired
instead of addition. The <save-time> value is in this format: [/[+|-]HH:[mm[:ss]]]
The <month> value equals any value in the range 1-12 (January-December).
The <day> value equals [+|-] any value in the range 1-31.
If <day> is 1, it means the <weekday> on or before the end of the month (in other words the last occurrence of <
weekday> in that month).
The <weekday> value equals any value in the range 1-7 (Monday-Sunday). It can also equal 0.
If the <weekday> value is 0, this means that the date to start or end daylight saving is exactly the date given. In
that case, the <day> value must not be negative.
If the <weekday> value is not 0 and the <day> value is positive, then daylight saving starts or ends on the
<weekday> value on or after the date given.
If the <weekday> value is not 0 and the <day> value is negative, then daylight saving starts or ends on the
<weekday> value on or before the date given.
The abbreviation HH stands for hours (0-23).
The abbreviation mm stands for minutes (0-59).
The abbreviation ss stands for seconds (0-59).
The default Daylight Saving Time Rule is
start=4/1/7;end=10/-1/7;save=1
.
FXS Port Input Gain
. Enter the input gain in dB, up to three decimal places. The range is 6.0 to -infinity. The
default is
-3
.
FXS Port Output Gain
. Enter the output gain in dB, up to three decimal places. The range is 6.0 to -infinity. The
default is
-3
.
DTMF Playback Level
. Enter the local DTMF playback level in dBm, up to one decimal place. The default is
-16
.
DTMF Playback Length
. Enter the local DTMF playback duration in milliseconds. The default is
.1
.
Detect ABCD
. To enable local detection of DTMF ABCD, select
yes
. Otherwise, select
no
. The default is
yes
.
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Chapter 6: Using the Web-based Utility
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IP Telephony System
Playback ABCD
. To enable local playback of OOB DTMF ABCD, select
yes
. Otherwise, select
no
. The default is
yes
.
Caller ID Method
. You have a choice of methods to use for caller ID. Select
Bellcore(N.Amer, China)
for CID,
CIDCW, and VMWI. FSK is sent after the first ring, and there is no polarity reversal or DTAS. Select
DTMF(Finland,Sweden)
for CID only. DTMF is sent after polarity reversal (with no DTAS) and before the first ring.
Select
DTMF(Denmark)
for CID only. DTMF is sent after polarity reversal (with no DTAS) and before the first ring.
Select
ETSI DTMF
for CID only. DTMF is sent after DTAS (with no polarity reversal) and before the first ring. Select
ETSI DTMF With PR
for CID only. DTMF is sent after polarity reversal and DTAS and before the first ring. Select
ETSI DTMF After Ring
for CID only. DTMF is sent after the first ring (with no polarity reversal or DTAS). Select
ETSI FSK
for CID, CIDCW, and VMWI. FSK is sent after DTAS (with no polarity reversal) and before the first ring. It
will wait for ACK from CPE after DTAS for CIDCW. Select
ETSI FSK With PR(UK)
for CID, CIDCW, and VMWI. FSK is
sent after polarity reversal and DTAS and before the first ring. It will wait for ACK from CPE after DTAS for CIDCW.
Polarity reversal is applied only if the equipment is on-hook. The default is
Bellcore(N.Amer, China)
.
Caller ID FSK Standard
. The System supports bell 202 and v.23 standards for caller ID generation. Select the
FSK standard you want to use,
bell 202
or
v.23
. The default is
bell 202
.
Feature Invocation Method
. Select the method you want to use,
Default
or
Sweden default
. The default is
Default
.
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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Chapter 6: Using the Web-based Utility
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IP Telephony System
The Voice - FXS 1/2 Screen
Use the appropriate screen to configure settings for each FXS port, which is called the Phone port on the System.
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 a SIP message. The default
is
0x68
.
SIP CoS Value
. Enter the CoS value for SIP messages. The default is
3
.
RTP ToS/DiffServ Value
. Enter the ToS/DiffServ field value in UDP IP packets carrying RTP data. The default is
0xb8
.
RTP CoS Value
. Enter the CoS value for RTP data. The default is
6
.
Network Jitter Level
. This setting determines how jitter buffer size is adjusted by the System. Jitter buffer size
is adjusted dynamically. The minimum jitter buffer size is 30 milliseconds or (10 milliseconds + current RTP
frame size), whichever is larger, for all jitter level settings. However, the starting jitter buffer size value is larger
for higher jitter levels. This setting controls the rate at which the jitter buffer size is adjusted to reach the
minimum. Select the appropriate setting:
low
,
medium
,
high
,
very high
, or
extremely high
. The default is
high
.
Jitter Buffer Adjustment
. This controls how the jitter buffer should be adjusted. Select the appropriate setting:
up and down
,
up only
,
down only
, or
disable
. The default is
up and down
.
SIP Settings
SIP Port
. Enter the port number of the SIP message listening and transmission port. The default is
5080
.
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
IMPORTANT:
In most cases, you should not change the service settings unless instructed to do
by your ITSP.
Figure 6-36: Voice - FXS 1 Screen - Network Settings
Figure 6-37: Voice - FXS 1 Screen - SIP Settings

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