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12 Voice Configuration Guide Book
12 Voice Configuration Guide Book
Copyright 2014, Hangzhou H3C Technologies Co., Ltd. and its licensors
Preface
The H3C MSR documentation set includes 14 configuration guides, which describe the software features
for the H3C MSR Series Routers and guide you through the software configuration procedures. These
configuration guides also provide configuration examples to help you apply software features to different
network scenarios.
The Voice Configuration Guide(V7) describes fundamentals and configuration of Voice Entity, Voice
Subscriber, Dial Plan, SIP, H.323, Call Services, and so on.
This preface includes:
Audience
Conventions
Obtaining documentation
Technical support
Documentation feedback
These configuration guides apply to the following models of the H3C MSR series routers:
Model
MSR 2600
MSR 26-30
MSR 3600
MSR 5600
MSR 36-10
MSR 36-20
MSR 36-40
MSR 36-60
MSR3600-28
MSR3600-51
MSR 56-60
MSR 56-80
Audience
This documentation is intended for:
Network planners
Conventions
This section describes the conventions used in this documentation set.
Command conventions
Convention
Description
Boldface
Bold text represents commands and keywords that you enter literally as shown.
Italic
Italic text represents arguments that you replace with actual values.
[]
Square brackets enclose syntax choices (keywords or arguments) that are optional.
{ x | y | ... }
Braces enclose a set of required syntax choices separated by vertical bars, from which
you select one.
[ x | y | ... ]
Square brackets enclose a set of optional syntax choices separated by vertical bars, from
which you select one or none.
{ x | y | ... } *
Asterisk marked braces enclose a set of required syntax choices separated by vertical
bars, from which you select at least one.
[ x | y | ... ] *
Asterisk marked square brackets enclose optional syntax choices separated by vertical
bars, from which you select one choice, multiple choices, or none.
&<1-n>
The argument or keyword and argument combination before the ampersand (&) sign can
be entered 1 to n times.
Convention
Description
Symbols
WARNING
An alert that calls attention to important information that if not understood or followed can
result in personal injury.
CAUTION
An alert that calls attention to important information that if not understood or followed can
result in data loss, data corruption, or damage to hardware or software.
IMPORTANT
NOTE
TIP
Documents
Purposes
Marketing brochures
Installation guide
Hardware specifications
and installation
Software configuration
Operations and
maintenance
Release notes
Obtaining documentation
You can access the most up-to-date H3C product documentation on the World Wide Web
at http://www.h3c.com.
Click the links on the top navigation bar to obtain different categories of product documentation:
[Technical Support & Documents > Technical Documents]Provides hardware installation, software
upgrading, and software feature configuration and maintenance documentation.
[Products & Solutions]Provides information about products and technologies, as well as solutions.
[Technical Support & Documents > Software Download]Provides the documentation released with the
software version.
Technical support
service@h3c.com
http://www.h3c.com
Documentation feedback
You can e-mail your comments about product documentation to info@h3c.com.
Contents
Configuring analog voice interfaces 1
FXS interface 1
FXO interface 1
E&M interface 1
Configuration task list 1
Configuring basic functions 2
Configuring call progress tones 2
Configuring an FXS interface 3
Configuring CID 3
Setting the electrical impedance 4
Configuring the packet loss compensation mode 5
Configuring an FXS interface to send LCFO signals 5
Configuring an FXO interface 6
Configuring CID 6
Configuring busy tone detection 6
Configuring an on-hook delay 9
Configuring the off-hook mode 10
Configuring ring detection parameters 10
Setting the electrical impedance 11
Configuring the packet loss compensation mode 11
Binding an FXS interface to an FXO interface 11
Configuring an E&M interface 12
Configuring the cable type 12
Configuring the signal type 12
Configuring a start mode for E&M signaling 12
Configuring E&M non-signaling mode 15
Enabling E&M control signals pass-through 16
Configuring the output gain of SLIC chip 16
Configuring DTMF 16
Configuring DTMF tone sending 17
Configuring DTMF tone detection 17
Adjusting parameters for voice interfaces 18
Adjusting gains 18
Adjusting timing parameters 18
Configuring the comfortable noise function 19
Configuring echo cancellation 19
Displaying and maintaining analog voice interfaces 21
Analog voice interface configuration examples 21
Two-dial configuration example for the FXO interface 21
FXO interface PLAR configuration example 22
E&M interface configuration example 24
E&M non-signaling mode configuration example 25
FXS&FXO 1:1 binding configuration example 27
Configuring digital voice interfaces 29
E1 and T1 interfaces 29
BSV interfaces 29
Configuration task list 29
Configuring basic parameters for an E1 interface 30
i
Configuring SIP 79
Overview 79
Terminology 79
SIP functions 79
SIP messages 80
SIP configuration task list 80
Configuring SIP UA registration 81
Configuration prerequisites 81
Configuring SIP credentials 82
Enabling a POTS entity to register with the registrar 84
Configuring registrar information 84
Displaying SIP UA registration status 85
Configuring the call destination address for a VoIP entity 85
Configuring the call destination IP address for a VoIP entity 85
Configuring a VoIP entity to obtain the call destination address from a proxy server 85
Configuring the destination domain name and port number for a VoIP entity 86
Configuring extended SIP functions 86
Configuring out-of-band DTMF 86
Configuring periodic refresh of SIP sessions 87
Configuring PSTN cause-to-SIP status mappings 87
Configuring caller privacy 88
Setting the P-Asserted-Identity or P-Preferred-Identity header field 88
Displaying and maintaining SIP 89
SIP UA configuration examples 89
Configuring direct SIP calling 89
Configuring SIP calling through a SIP server 90
Configuring SIP calling through DNS 92
Configuring out-of-band DTMF 93
Configuring SIP trunk 96
Background 96
Features 97
Typical applications 97
Protocols and standards 98
SIP trunk configuration task list 98
Enabling SIP-to-SIP calling 98
Configuring a SIP trunk account 99
Enabling codec transparent transmission 99
Enabling media flow-around 99
Enabling DO-EO conversion 100
iii
iv
FXS interface
A Foreign Exchange Station (FXS) interface connects to a standard telephone, fax machine, or a Private
Branch Exchange (PBX) through an RJ-11 connector and a telephone cable. It provides ring, voltage, and
dial tone based on level changes on the Tip/Ring line. An FXS interface can only connect to an FXO
interface.
FXO interface
A foreign exchange office (FXO) interface connects to a PBX through an RJ-11 connector and a telephone
cable. It provides ring, voltage, and dial tone based on level changes on the Tip/Ring line. An FXO
interface can only connect to an FXS interface.
E&M interface
An ear & mouth or receive & transmit (E&M) interface is a common trunk line that connects to a PBX
through an RJ-48 connector. The E&M interface supports E&M signaling. The E signaling receives signals
from the peer, and the M signaling sends signals to the peer. An E&M interface can only connect to an
E&M interface.
Configuring CID
Setting the electrical impedance
Configuring the packet loss compensation mode
Configuring an FXS interface to send LCFO signals
Tasks at a glance
(Optional.) Configuring an FXO interface
Configuring CID
Configuring busy tone detection
Configuring an on-hook delay
Configuring the off-hook mode
Configuring ring detection parameters
Setting the electrical impedance
Configuring the packet loss compensation mode
Adjusting gains
Adjusting timing parameters
Configuring the comfortable noise function
Configuring echo cancellation
Command
Remarks
1.
system-view
N/A
2.
subscriber-line line-number
N/A
3.
description text
4.
default
N/A
5.
undo shutdown
Step
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
Specify a country:
4.
parameters:
cptone custom { busy-tone |
congestion-tone | dial-tone |
ringback-tone |
special-dial-tone |
waiting-tone } comb freq1
freq2 time1 time2 time3 time4
Configuring CID
Caller identification (CID) enables called terminals to display the calling information, including the
calling number, calling name, date, and time.
CID supports two data formats:
The called telephone displays the calling number if the terminating device is enabled with CID and
can obtain the calling number.
The called telephone displays the character "O" if the terminating device is enabled with CID but
fails to obtain the calling number (for example, the originating device does not send the calling
number).
The called telephone displays the character "P" if CID is disabled on the terminating device.
The FXS interface sends the CID to the called telephone through frequency shift keying (FSK) modulation
between the first and second rings. For the CID to appear on the called telephone, the called user should
pick up the handset after the second ring.
The CID function requires configurations on both FXS and FXO interfaces. For configuration on the FXO
interface, see "Configuring CID."
Configuration guidelines
To ensure correct call time, make sure the router system time transmitted in data-message format
stays synchronous with the local standard time.
For the CID function to operate correctly, keep the cid send command enabled.
Configuration procedure
Step
Command
Remarks
1.
system-view
N/A
2.
subscriber-line line-number
N/A
3.
4.
calling-name text
cid send
5.
6.
7.
cid display
Command
Remarks
system-view
N/A
Step
Enter FXS interface view.
2.
3.
Command
Remarks
subscriber-line line-number
N/A
For discrete packet loss, you can use the general compensation mode to reconstruct lost packets.
For continuous packet loss, you can use the voice gateway-specific compensation mode to
compensate for lost packets.
Command
Remarks
1.
System-view
N/A
2.
subscriber-line line-number
N/A
3.
Command
Remarks
1.
System-view
N/A
2.
subscriber-line line-number
N/A
3.
disconnect lcfo
4.
Configuring CID
The CID function must be configured on both the FXS and FXO interfaces. For information about
configuring this function on the FXS interface, see "Configuring CID."
For the CID function to work correctly, enable both CID receiving and CID sending.
Command
Remarks
1.
system-view
N/A
2.
subscriber-line line-number
N/A
3.
cid receive
Command
Remarks
4.
system-view
N/A
2.
subscriber-line line-number
N/A
3.
cid send
You can configure busy tone detection by customizing busy tone parameters or configuring automatic
busy tone detection. If the tone that the router receives from the PBX matches the busy tone parameters,
the router considers the tone as a busy tone and shuts down the FXO interface.
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
area { europe |
north-america }
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
busytone-detect custom
area-number index argu f1 f2 p1
p2 p3 p4 p5 p6 p7
N/A
By default, the European standard
is used.
4.
area custom
2.
Telephone A first goes on-hook. The PBX plays busy tones to Router A after detecting the on-hook
condition.
3.
Execute the busytone-detect auto command on Router A to detect the busy tone. To make sure the
FXO interface can capture the busy tone sent by the PBX, H3C recommends that you execute this
command two seconds after Telephone A goes on-hook.
4.
The console prompts that busy tone detection is in progress and prompts detection success when
the detection is complete.
5.
Check whether the detected busy tone parameters are valid by repeating steps 1 and 2.
After Telephone A goes on-hook, the PBX plays a busy tone to Router A. If Router A detects the busy
tone, it shuts down the FXO interface.
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
N/A
4.
quit
N/A
5.
subscriber-line line-number
N/A
6.
The default is 2.
busytone-detect period value
Command
Remarks
1.
system-view
N/A
2.
subscriber-line line-number
N/A
3.
send-busytone enable
4.
Command
Remarks
1.
system-view
N/A
2.
subscriber-line line-number
N/A
3.
Configure silence
detection-based automatic
on-hook.
Command
Remarks
1.
system-view
N/A
2.
subscriber-line line-number
N/A
By default, forced on-hook is disabled.
3.
hookoff-time time
Command
Remarks
1.
system-view
N/A
2.
subscriber-line line-number
N/A
3.
busytone-hookon delay-timer
value
Immediate modeUpon receiving a call, the FXO interface goes off-hook and sends a dial tone to
the calling party. Then, the calling party dials the destination number.
Delay modeUpon receiving a call, the FXO interface places a call to the specified private line
number. When the called party picks up the phone, the FXO goes off-hook. This mode needs to
work with the private line auto ring-down (PLAR) function. For more information about PLAR, see
"Configuring dial programs."
Command
Remarks
1.
system-view
N/A
2.
subscriber-line line-number
N/A
3.
hookoff-mode { delay |
immediate }
Command
Remarks
1.
system-view
N/A
2.
subscriber-line line-number
N/A
The default is 10 milliseconds.
4.
10
Command
Remarks
1.
system-view
N/A
2.
subscriber-line line-number
N/A
3.
For discrete packet loss, you can use the general compensation mode to reconstruct lost packets.
For continuous packet loss, you can use the voice gateway-specific compensation mode to
reconstruct lost packets.
Command
Remarks
1.
System-view
N/A
2.
subscriber-line line-number
N/A
3.
Command
Remarks
system-view
N/A
11
Step
Command
Remarks
2.
subscriber-line line-number
N/A
3.
4.
private-line string
5.
Command
Remarks
1.
system-view
N/A
2.
subscriber-line line-number
N/A
3.
Command
Remarks
1.
system-view
N/A
2.
subscriber-line line-number
N/A
3.
type { 1 | 2 | 3 | 5 }
Immediate startAfter off-hook, the originating side waits for a specified period of time to send the
called number to the terminating side. During this period of time, the originating side does not
check whether the terminating side is ready to receive the called number. The terminating side
enters off-hook state after receiving the called number.
Figure 3 Immediate start
Delay startThe originating side goes off-hook and seizes the trunk. After detecting the seizure
signal from the originating side, the terminating side enters the off-hook state and remains in this
state until it is ready to receive the called number. Then, the terminating side enters the on-hook state
and sends a signal to indicate that the line is idle. After receiving the idle signal, the originating
side sends the called number to the terminating side, which relays the call to the user phone.
Figure 4 Delay start
Wink startThe terminating side remains in on-hook state until it receives the seizure signal from
the originating side and then sends a wink to the originating side. After receiving the wink, the
originating side sends the called number to the terminating side, which relays the call to the user
phone.
13
Command
Remarks
1.
system-view
N/A
2.
subscriber-line line-number
N/A
3.
signal immediate
4.
Command
Remarks
1.
system-view
N/A
2.
subscriber-line line-number
N/A
3.
signal delay
4.
5.
Command
Remarks
system-view
N/A
14
Step
Command
Remarks
2.
subscriber-line line-number
N/A
3.
signal wink
4.
5.
6.
Command
Remarks
1.
system-view
N/A
2.
subscriber-line line-number
N/A
3.
signal immediate
4.
5.
private-line string
15
Command
Remarks
1.
system-view
N/A
2.
subscriber-line line-number
N/A
3.
Command
Remarks
1.
system-view
N/A
2.
subscriber-line line-number
N/A
Optional.
3.
slic-gain { 0 | 1 }
Configuring DTMF
Dual tone multi-frequency (DTMF) uses a mixture of a high frequency tone and a lower frequency tone to
represent a key on a keypad. Each column of keys is represented by a high frequency tone and each row
of keys is represented by a low frequency tone. For example, as shown in Figure 7, the digit 1 is
represented by the combination of a pure 697 Hz signal and a pure 1209 Hz signal. Such DTMF tones
have good immunity to interference.
16
1336Hz
1477Hz
1633Hz
697Hz
770Hz
852Hz
941Hz
A DTMF tone must last at least 45 milliseconds. A minimum interval of 23 milliseconds is required
between two DTMF tones to make sure DTMF tones are recognizable. Such requirements are roughly the
same in all countries. For more information, see the ITU-T Recommendation Q.24.
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
4.
Energy detectionCalculates the spectrum shape of input DTMF signals and matches the spectrum
shape with the threshold parameters. A DTMF tone is considered valid if it matches all threshold
parameters.
Sensitivity detectionA higher DTMF detection sensitivity reduces the possibility of missing a true
DTMF tone but increases the possibility of false detection. A lower DTMF detection sensitivity
reduces the possibility of false detection but increases the possibility of missing a true DTMF tone.
Step
Command
Remarks
1.
system-view
N/A
2.
subscriber-line line-number
N/A
3.
4.
Adjusting gains
You can adjust gains to control the amount of volume in the input or output direction.
IMPORTANT:
Gain adjustment might lead to call failures. H3C recommends not adjusting the gain. If necessary, do it
under the guidance of technical personnel.
To adjust gains:
Step
Command
Remarks
1.
system-view
N/A
2.
subscriber-line line-number
N/A
3.
4.
Command
Remarks
1.
system-view
N/A
2.
subscriber-line line-number
N/A
18
Step
3.
4.
5.
Command
Remarks
The default is 10 seconds.
6.
7.
timer hookflash-detect
hookflash-range
8.
Command
Remarks
1.
system-view
N/A
2.
subscriber-line line-number
N/A
3.
cng-on
the echo cancellation delay (the time between when an interface sends out a signal and when to the
interface receives an echo) to 33 ms and the echo cancellation coverage to 16 ms.
Figure 8 Echo
Command
Remarks
1.
system-view
N/A
2.
subscriber-line line-number
N/A
3.
echo-canceler enable
4.
5.
echo-canceler tail-length
milliseconds
Parameters adjusted
Effect
Command
Remarks
system-view
N/A
20
Step
2.
3.
Command
Remarks
voice-setup
N/A
echo-canceler { convergence-rate
value | max-amplitude value |
mix-proportion-ratio value |
talk-threshold value }
Command
Remarks
1.
system-view
N/A
2.
subscriber-line line-number
N/A
3.
nlp-on
Command
Configure the two routers to enable Telephone B to establish a call with Telephone A through two dials.
Figure 9 Network diagram
Eth2/1
1.1.1.1/24
FXS 1/0
Telephone A
0101001
IP network
Eth2/1
2.2.2.2/24
FXO 1/0
Router B
Router A
PBX
Telephone B
07552001
Configuration procedure
1.
On Router A, configure the local number as 0101001 for POTS entity 1001, and bind FXS
interface line1/0 to the POTS entity.
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 1001 pots
[RouterA-voice-dial-entity1001] match-template 0101001
[RouterA-voice-dial-entity1001] line 1/0
2.
Configure Router B:
# Configure the called number as 010 for VoIP entity 010, and configure the destination IP
address as 1.1.1.1.
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 010 voip
[RouterB-voice-dial-entity10] match-template 0101001.
[RouterB-voice-dial-entity10] address sip ip 1.1.1.1
[RouterB-voice-dial-entity10] quit
# Configure the local number as 07552001 for POTS entity 2001, and bind FXO interface
line1/0 to POTS entity 2001.
[RouterB-voice-dial] entity 2001 pots
[RouterB-voice-dial-entity2001] match-template 07552001
[RouterB-voice-dial-entity2001] line 1/0
22
Configure PLAR for the FXO interface of Router B. When the user of Telephone B dials 07552003, the
FXO interface automatically calls Telephone A.
Figure 10 Network diagram
Eth2/1
1.1.1.1/24
FXS 1/0
Telephone A
0101001
IP network
Eth2/1
2.2.2.2/24
FXO 1/0
Router B
Router A
PBX
Telephone B
07552001
Configuration procedure
1.
On Router A, configure the local number as 0101001 for POTS entity 1001, and bind FXS
interface line1/0 to the POTS entity.
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 1001 pots
[RouterA-voice-dial-entity1001] match-template 0101001
[RouterA-voice-dial-entity1001] line 1/0
2.
Configure Router B:
# Configure the called number as 010 for VoIP entity 010, and configure the destination IP
address as 1.1.1.1.
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 010 voip
[RouterB-voice-dial-entity10] match-template 0101001.
[RouterB-voice-dial-entity10] address sip ip 1.1.1.1
[RouterB-voice-dial-entity10] quit
# Configure the local number as 07552001 for POTS entity 2001, and bind FXO interface
line1/0 to the POTS entity.
[RouterB-voice-dial] entity 2001 pots
[RouterB-voice-dial-entity2001] match-template 07552001
[RouterB-voice-dial-entity2001] line 1/0
# Enable the PLAR function and configure the delay off-hook mode for the FXO interface.
[RouterB] subscriber-line 1/0
[RouterB-subscriber-line1/0] private-line 0101001
[RouterB-subscriber-line1/0] hookoff-mode delay
23
FXS 1/0
Telephone A
0101001
IP network
Eth2/1
2.2.2.2/24
Router A
E&M 5/0
Router B
PBX
Telephone B
Configuration procedure
1.
Configure Router A:
# Configure the called number as 0755 for VoIP entity 0755, and configure the destination IP
address as 2.2.2.2.
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 0755 voip
[RouterA-voice-dial-entity755] match-template 07552001
[RouterA-voice-dial-entity755] address sip ip 2.2.2.2
[RouterA-voice-dial-entity755] quit
# Configure the local number as 0101001 for POTS entity 1001, and bind FXS interface line1/0
to the POTS entity.
[RouterA-voice-dial] entity 1001 pots
[RouterA-voice-dial-entity1001] match-template 0101001
[RouterA-voice-dial-entity1001] line 1/0
2.
Configure Router B:
# Configure the called number as 010 for VoIP entity 010, and configure the destination IP
address as 1.1.1.1.
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 010 voip
[RouterB-voice-dial-entity10] match-template 0101001
[RouterB-voice-dial-entity10] address sip ip 1.1.1.1
[RouterB-voice-dial-entity10] quit
# Configure the local number as 07552001 for POTS entity 2001, and bind E&M interface line
5/0 to the POTS entity.
[RouterB-voice-dial] entity 2001 pots
[RouterB-voice-dial-entity2001] match-template 07552001
[RouterB-voice-dial-entity2001] send-number all
[RouterB-voice-dial-entity2001] line 5/0
[RouterB-voice-dial-entity2001] return
24
# Enter the view of E&M interface 5/0. The E&M interface must have the same configuration as the
connected PBX.
<RouterB> system-view
[RouterB] subscriber-line 5/0
# Configure the 4-wire cable type (optional, because the default is the 4-wire cable type).
[RouterB-subscriber-line5/0] cable 4-wire
Configuration procedure
1.
Configure Router A:
# Configure the called number as 2000 for VoIP entity 2000, and configure the destination IP
address as 2.2.2.2.
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 2000 voip
[RouterA-voice-dial-entity2000] match-template 2000
[RouterA-voice-dial-entity2000] address sip ip 2.2.2.2
[RouterA-voice-dial-entity2000] quit
# Configure the local number as 1000 for POTS entity 1000, and bind E&M interface line 5/0 to
the POTS entity.
25
# Configure the immediate start mode (optional, because the default is the immediate start mode).
[RouterA-subscriber-line5/0] signal immediate
2.
Configure Router B:
# Configure the called number as 1000 for VoIP entity 1000, and configure the destination IP
address as 1.1.1.1.
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 1000 voip
[RouterB-voice-dial-entity10] match-template 1000
[RouterB-voice-dial-entity10] address sip ip 1.1.1.1
[RouterB-voice-dial-entity10] quit
# Configure the local number as 2000 for POTS entity 2000, and bind E&M interface line 5/0 to
the POTS entity.
[RouterB-voice-dial] entity 2000 pots
[RouterB-voice-dial-entity2000] match-template 2000
[RouterB-voice-dial-entity2000] line 5/0
[RouterB-voice-dial-entity2000] return
# Configure the immediate start mode (optional, because the default is the immediate start mode).
[RouterB-subscriber-line5/0] signal immediate
26
192.168.0.71/24
FXS 1/0
Telephone A
0101001
IP
192.168.0.76/24
Router B
Router A
FXS 1/0
Telephone B
2101002
FXO 2/0
PSTN
FXO 2/0
Configuration procedure
1.
Configure Router A:
# Configure the called number template as 210. for VoIP entity 210, and configure the
destination IP address as 192.168.0.76.
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 210 voip
[RouterA-voice-dial-entity210] match-template 210....
[RouterA-voice-dial-entity210] address sip ip 192.168.0.76
[RouterA-voice-dial-entity210] quit
# Configure the local number as 0101001 for POTS entity 0101001, and bind the FXS interface
line1/0 to the POTS entity.
[RouterA-voice-dial] entity 0101001 pots
[RouterA-voice-dial-entity101001] match-template 0101001
[RouterA-voice-dial-entity101001] line 1/0
[RouterA-voice-dial-entity101001] quit
# Configure the called number template as .T for POTS entity 211 used for call backup on FXO
interface line 2/0, and configure the permitted calling number as 0101001.
[RouterA-voice-dial] entity 211 pots
[RouterA-voice-dial-entity211] match-template .T
[RouterA-voice-dial-entity211] line 2/0
[RouterA-voice-dial-entity211] send-number all
[RouterA-voice-dial-entity211] caller-permit 0101001
[RouterA-voice-dial-entity211] quit
27
[RouterA-voice-dial] quit
[RouterA-voice] quit
# Enable the PLAR function for the FXO interface line 2/0, and bind the FXS interface line 1/0 to
the FXO interface line 2/0.
[RouterA] subscriber-line 2/0
[RouterA-subscriber-line2/0] private-line 0101001
[RouterA-subscriber-line2/0] hookoff-mode delay bind 1/0
2.
Configure Router B:
# Configure the called number template as 010. for VoIP entity 010, and configure the
destination IP address as 192.168.0.71.
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 010 voip
[RouterB-voice-dial-entity10] match-template 010....
[RouterB-voice-dial-entity10] address sip ip 192.168.0.71
[RouterB-voice-dial-entity10] quit
# Configure the local number as 2101002 for POTS entity 2101002, and bind the FXS interface
line1/0 to the POTS entity.
[RouterB-voice-dial] entity 2101002 pots
[RouterB-voice-dial-entity2101002] match-template 2101002
[RouterB-voice-dial-entity2101002] line 1/0
[RouterB-voice-dial-entity2101002] quit
# Configure the called number template as .T for POTS entity 011 used for call backup on FXO
interface line 2/0, and configure the permitted calling number as 2101002.
[RouterB-voice-dial] entity 011 pots
[RouterB-voice-dial-entity11] match-template .T
[RouterB-voice-dial-entity11] line 2/0
[RouterB-voice-dial-entity11] send-number all
[RouterB-voice-dial-entity11] caller-permit 2101002
[RouterB-voice-dial-entity11] quit
[RouterB-voice-dial] quit
[RouterB-voice] quit
# Enable the PLAR function for the FXO interface line2/0, and bind the FXS interface line1/0 to the
FXO interface line2/0.
[RouterB] subscriber-line 2/0
[RouterB-subscriber-line2/0] private-line 2101002
[RouterB-subscriber-line2/0] hookoff-mode delay bind 1/0
When the IP network operates correctly, Telephone A calls Telephone B through the IP network.
When the IP network is unavailable, Telephone A uses the bound FXO interface to call Telephone
B through the PSTN network.
28
E1 and T1 interfaces
E1 interfaces (also called VE1 interfaces) and T1 interfaces (also called VT1 interfaces) can connect to
the PSTN as trunk interfaces. An E1 interface provides 32 timeslots and 2.048 Mbps bandwidth. A T1
interface provides 24 timeslots and 1.544 Mbps bandwidth. ITU-T E1 is used mainly in Europe and
China. ANSI T1 is used mainly in America, Canada, and Japan.
E1/T1 voice transmission allows a router to provide more communication channels, greatly improving
router utilization.
Figure 14 shows a typical network for E1/T1 interfaces.
Figure 14 Network diagram
BSV interfaces
BRI S/T voice (BSV) interfaces can compress, send, receive, and decompress voice packets. BSV
interfaces connect to PBXs as trunk interfaces. A BSV interface provides two B channels and one D
channel (2B+D) and supports only DSS1 signaling.
29
Tasks at a glance
(Required.) Perform one of the following tasks:
Configuring R2 signaling
{
If the line keyword is specified for all interfaces, the clock on the interface with the lowest number
is used. If the interface goes down, the clock on the interface with the second lowest number is used.
If the line and primary keywords are specified for one interface and the line or internal keyword is
specified for all other interfaces, the clock on that one interface is used.
If the line keyword is specified for one interface and the internal keyword is specified for all other
interfaces, the clock on that one interface is used.
The clock source of only one interface can be set to line primary.
The TDM clock sources on the local and peer devices must match. For example, if you set the clock source
to line for a subsystem on the local device, you must set the clock source to internal on the peer device;
and vice versa.
To configure a TDM clock source for an E1 interface:
30
Step
Command
Remarks
1.
system-view
N/A
2.
controller e1 number
N/A
3.
Command
Remarks
1.
system-view
N/A
2.
controller e1 number
N/A
3.
Configure a description.
description text
4.
5.
6.
7.
idle-code { 7e | ff }
8.
itf type { 7e | ff }
9.
default
N/A
undo shutdown
Command
Remarks
1.
system-view
N/A
2.
controller t1 number
N/A
31
Step
Configure a TDM clock source
for the T1 interface.
3.
Command
Remarks
Command
Remarks
1.
system-view
N/A
2.
controller t1 number
N/A
3.
Configure a description.
description text
4.
frame-format { esf | sf }
5.
6.
7.
itf type { 7e | ff }
8.
9.
default
N/A
undo shutdown
BRIHas the same logical characteristics as an ISDN BRI but can be used for only voice
transmission. You can configure ISDN parameters on a BRI interface.
Digital voice interfaceCan be configured with voice commands. A digital voice interface has two
subinterfaces, which correspond to the two B channels. You cannot configure commands on the
32
subinterfaces, but you can use the display voice subscriber-line line-number.subnumber command
to view the call state of a subinterface.
Command
Remarks
1.
system-view
N/A
2.
N/A
3.
(Optional.) Configure a
description for the interface.
description text
4.
loopback { b1 | b2 | both }
5.
bandwidth bandwidth-value
6.
mtu size
7.
timer-hold seconds
8.
default
N/A
9.
undo shutdown
activate
For more information about these commands, see Interface Command Reference.
Step
Command
Remarks
1.
system-view
N/A
2.
Enter E1 or T1 interface
view.
controller { e1 | t1 } number
N/A
3.
timeslot-set ts-set-number
timeslot-list timeslots-list signal r2
Command
Remarks
1.
system-view
N/A
2.
subscriber-line
number:ts-set-number
N/A
3.
Configuring R2 signaling
ITU-T recommendations Q.400 through Q.490 define the R2 signaling standards. However, the R2
signaling standards implemented in different countries and regions are ITU variants.
In R2 signaling, the calling side serves as the originating PBX, and the called side serves as the
terminating PBX. Signaling sent by the originating PBX is called forward signaling, and signaling sent by
the terminating PBX is called backward signaling, as shown in Figure 15.
Figure 15 R2 signaling elements
34
R2 signaling include two categories: digital line signaling and interregister signaling. Digital line
signaling conveys status information about E1 trunks to describe whether the trunks are seized, released,
or blocked. Interregister signaling transmits and requests calling and called numbers.
Digital line signaling sets the line to be idle or seized according to the state of the trunk. This signaling
is transmitted through timeslot 16. The two transmission directions of each line have four bits (A, B, C and
D) as flag bits, with C and D bits fixed to 01. The forward line signaling adopts af and bf bits, and the
backward line signaling adopts ab and bb bits, as shown in Table 2:
Table 2 Line signaling bit description
Bit
Description
Value = 0
Value = 1
af
Off-hook, seized
On-hook (idle)
bf
Normal
Faulty
ab
Off-hook by called
party
bb
Idle
Seized or blocked
Forward
Backward
af
bf
ab
bb
Idle or release
Seized
Seizure-ack
Answer
Clear-back
Clear-forward
0/1
Blocked
Unblocked
Call establishment.
35
When the circuit is idle, the originating side sends a forward seizure signal (00) to the terminating
side, and the terminating side sends back a seizure acknowledgement signal (11). Then the circuit
is seized, and interregister signaling exchange begins. When the called party picks up the phone,
the terminating side sends a backward answer signal (01). After the originating side recognizes
the received signal, it establishes the call.
Figure 16 Call establishment
36
After the originating side receives a blocking signal 11 from the terminating side when the circuit
is idle or during conversation, the circuit is blocked. In this case, the originating side still sends the
forward signal 10 to indicate that the line is idle. When the terminating side unblocks the circuit,
it sends a backward signal 10 to indicate that the line is idle. The originating side maintains the
forward signal 10 and unblocks the local-end circuit for the next call.
Basic meaning
I-11
I-12
Request refused
I-13
I-14
I-15
Group A backward signalsControl signals used for controlling and acknowledging Group I
forward signals.
Basic Meaning
A-1
A-2
A-3
A-4
A-5
A-6
A-7
A-8
A-9
A-10
A-11
37
Designation
Basic Meaning
A-12
A-13
A-14
A-15
Group II forward signalsIdentify the calling party category. The system looks at the calling party
category to decide whether the calling party can perform forced release or break-in.
Basic Meaning
II-1
II-2
II-3
Maintenance equipment
II-4
II-5
Operator
II-6
Data transmission
II-7
Subscriber (or operator without forward transfer facility), for international use
II-8
II-9
II-10
Group B backward signalsAcknowledge Group II signals and indicate the status of the called
party.
Basic Meaning
B-1
B-2
B-3
B-4
Congestion
B-5
Unallocated number
B-6
B-7
B-8
38
Figure 19 shows the exchange process requesting calling party information, which is typical of R2
interregister signaling.
Figure 19 ITU-T R2 interregister signaling exchange process
Originating side
Calling number: 123
Terminating side
Called number 789
Command
Remarks
1.
system-view
N/A
2.
controller { e1 | t1 } number
N/A
3.
timeslot-set ts-set-number
timeslot-list timeslots-list signal r2
4.
cas ts-set-number
N/A
5.
mode zone-name
[ default-standard ]
39
Command
Remarks
1.
system-view
N/A
2.
controller { e1 | t1 } number
N/A
3.
timeslot-set ts-set-number
timeslot-list timeslots-list signal r2
4.
cas ts-set-number
N/A
The default is dual.
5.
trunk-direction timeslots
timeslots-list { dual | in | out }
6.
Configuring the terminating side to send busy tones to the originating side
Step
Command
Remarks
1.
system-view
N/A
2.
controller { e1 | t1 } number
N/A
3.
timeslot-set ts-set-number
timeslot-list timeslots-list signal r2
4.
cas ts-set-number
N/A
5.
6.
40
MFCThe originating and terminating sides use interregister signaling to transmit and request
number information, including the calling number, line information, and billing. In the exchange
process, the terminating side sends responses to the originating side.
DTMFThe originating side transmits the called number to the terminating side digit by digit. The
terminating side does not send any responses.
Compared with the MFC mode, the DTMF mode has a faster connection speed but transmits a smaller
amount of information.
To enable DTMF to receive and send numbers:
Step
Command
Remarks
1.
system-view
N/A
2.
controller { e1 | t1 } number
N/A
3.
timeslot-set ts-set-number
timeslot-list timeslots-list signal r2
4.
cas ts-set-number
N/A
By default, MFC mode is used.
5.
6.
dtmf enable
terminalAfter receiving the called number, the terminating side must wait for the real state (busy
or idle) of the called party before returning the corresponding interregister information to the
originating side.
segmentAfter receiving the called number, the terminating side directly returns the "called party
idle" interregister signaling, without waiting for the real state of the terminating side. If the called
party is busy, the terminating side plays busy tones to the originating side.
Command
Remarks
1.
system-view
N/A
2.
controller { e1 | t1 } number
N/A
3.
4.
cas ts-set-number
N/A
41
Step
5.
Command
Remarks
Command
Remarks
1.
system-view
N/A
2.
controller { e1 | t1 } number
N/A
3.
By default, no timeslot
set is created.
4.
cas ts-set-number
N/A
5.
N/A
Command
Remarks
1.
system-view
N/A
2.
controller { e1 | t1 } number
N/A
3.
timeslot-set ts-set-number
timeslot-list timeslots-list signal r2
4.
cas ts-set-number
N/A
5.
answer enable
re-answer enable
clear-forward-ack enable
8.
metering enable
9.
seizure-ack enable
6.
7.
Step
Command
Remarks
renew ABCD
reverse ABCD
signals.
Command
Remarks
1.
system-view
N/A
2.
controller { e1 | t1 } number
N/A
3.
timeslot-set ts-set-number
timeslot-list timeslots-list signal r2
4.
cas ts-set-number
N/A
5.
ani { all | ka }
ani-digit number
The default is 1.
group-b enable
final-callednum enable
6.
7.
8.
9.
special-character character
number
43
Step
Command
Remarks
register-value { billingcategory |
callcreate-in-groupa |
callingcategory | congestion |
demand-refused | digit-end |
null-number | req-billingcategory
|
req-callednum-and-switchgroupa
| req-callingcategory |
req-currentcallednum-in-groupc |
req-currentdigit |
req-firstcallednum-in-groupc |
req-firstcallingnum | req-firstdigit
| req-lastfirstdigit |
req-lastseconddigit |
req-lastthirddigit |
req-nextcallednum |
req-nextcallingnum |
req-switch-groupb |
subscriber-abnormal
|subscriber-busy |
subscriber-charge
|subscriber-idle } value
Command
1.
system-view
2.
controller { e1 | t1 } number
3.
44
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
dial-program
N/A
4.
N/A
line line-number:
Command
45
Configuration procedure
1.
Configure Router A:
# Configure the IP address 1.1.1.1/24 for interface GigabitEthernet 0/0.
<RouterA> system-view
[RouterA] interface GigabitEthernet 0/0
[RouterA-GigabitEthernet0/0] ip address 1.1.1.1 255.255.255.0
[RouterA-GigabitEthernet0/0] quit
# Configure the local number 0101001 for POTS entity 1001, and bind the digital voice interface
line 5/0:1 to the POTS entity.
[RouterA-voice-dial] entity 1001 pots
[RouterA-voice-dial-entity1001] match-template 0101001
[RouterA-voice-dial-entity1001] line 5/0:1
[RouterA-voice-dial-entity1001] send-number all
[RouterA-voice-dial-entity1001] quit
# Configure the local number 0101002 for PORTS entity 1002, and bind the digital voice
interface line 5/0:1 to the POTS entity.
[RouterA-voice-dial] entity 1002 pots
[RouterA-voice-dial-entity1002] match-template 0101002
[RouterA-voice-dial-entity1002] line 5/0:1
[RouterA-voice-dial-entity1002] send-number all
[RouterA-voice-dial-entity1002] quit
# Configure the called number template 0755. for VoIP entity 0755, and configure the
destination IP address as 2.2.2.2.
[RouterA-voice-dial] entity 0755 voip
[RouterA-voice-dial-entity755] match-template 0755....
[RouterA-voice-dial-entity755] address sip ip 2.2.2.2
2.
Configure Router B:
# Configure the IP address 2.2.2.2/24 for interface GigabitEthernet 0/0.
<RouterB> system-view
[RouterB] interface GigabitEthernet 0/0
[RouterB-GigabitEthernet0/0] ip address 2.2.2.2 255.255.255.0
46
[RouterB-GigabitEthernet0/0] quit
# Configure the local number 07552001 for POTS entity 2001, and bind the digital voice
interfaces line 5/0:1 to the POTS entity.
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 2001 pots
[RouterB-voice-dial-entity2001] match-template 07552001
[RouterB-voice-dial-entity2001] line 5/0:1
[RouterB-voice-dial-entity2001] send-number all
[RouterB-voice-dial-entity2001] quit
# Configure the local number 07552002 for POTS entity 2002, and bind the digital voice
interfaces line 5/0:1 to the POTS entity.
[RouterB-voice-dial] entity 2002 pots
[RouterB-voice-dial-entity2002] match-template 07552002
[RouterB-voice-dial-entity2002] line 5/0:1
[RouterB-voice-dial-entity2002] send-number all
[RouterB-voice-dial-entity2002] quit
# Configure the called number template 010. for VoIP entity 010, and configure the destination
IP address as 1.1.1.1.
[RouterB-voice-dial] entity 010 voip
[RouterB-voice-dial-entity10] match-template 010....
[RouterB-voice-dial-entity10] address ip 1.1.1.1
47
Configuration procedure
1.
Configure Router A:
# Configure the IP address 1.1.1.1/24 for interface GigabitEthernet 0/0.
<RouterA> system-view
[RouterA] interface GigabitEthernet 0/0
[RouterA-GigabitEthernet0/0] ip address 1.1.1.1 255.255.255.0
[RouterA-GigabitEthernet0/0] quit
# Configure the local number 0101001 for POTS entity 1001, and bind the digital voice
interfaces line 5/0:15 to the POTS entity.
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 1001 pots
[RouterA-voice-dial-entity1001] match-template 0101001
[RouterA-voice-dial-entity1001] line 5/0:15
[RouterA-voice-dial-entity1001] send-number all
[RouterA-voice-dial-entity1001] quit
# Configure the local number 0101002 for POTS entity 1002, and bind the digital voice
interfaces line 5/0:15 to the POTS voice entity.
[RouterA-voice-dial] entity 1002 pots
[RouterA-voice-dial-entity1002] match-template 0101002
[RouterA-voice-dial-entity1002] line 5/0:15
[RouterA-voice-dial-entity1002] send-number all
[RouterA-voice-dial-entity1002] quit
# Configure the called number template 0755. for VoIP entity 0755, and configure the
destination IP address as 2.2.2.2.
[RouterA-voice-dial] entity 0755 voip
[RouterA-voice-dial-entity755] match-template 0755....
[RouterA-voice-dial-entity755] address ip 2.2.2.2
2.
Configure Router B:
# Configure the IP address 2.2.2.2/24 for interface GigabitEthernet 0/0.
<RouterB> system-view
48
# Configure the local number 07552001 for POTS entity 2001, and bind the digital voice
interfaces line 5/0:15 to the POTS entity.
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 2001 pots
[RouterB-voice-dial-entity2001] match-template 07552001
[RouterB-voice-dial-entity2001] line 5/0:15
[RouterB-voice-dial-entity2001] send-number all
[RouterB-voice-dial-entity2001] quit
# Configure the local number 07552002 for POTS entity 2002, and bind the digital voice
interfaces line 5/0:15 to the POTS entity.
[RouterB-voice-dial] entity 2002 pots
[RouterB-voice-dial-entity2002] match-template 07552002
[RouterB-voice-dial-entity2002] line 5/0:15
[RouterB-voice-dial-entity2002] send-number all
[RouterB-voice-dial-entity2002] quit
# Configure the called number template 010. for VoIP entity 010, and configure the destination
IP address as 1.1.1.1.
[RouterB-voice-dial] entity 010 voip
[RouterB-voice-dial-entity10] match-template 010....
[RouterB-voice-dial-entity10] address ip 1.1.1.1
49
Configuration procedure
1.
Configure Router A:
# Configure the IP address 1.1.1.1/24 for interface GigabitEthernet 0/0.
<RouterA> system-view
[RouterA] interface GigabitEthernet 0/0
[RouterA-GigabitEthernet0/0] ip address 1.1.1.1 255.255.255.0
[RouterA-GigabitEthernet0/0] quit
# Configure the local number 0101001 for POTS entity 1001, and bind the digital voice
interfaces line 1/1 to the POTS entity.
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 1001 pots
[RouterA-voice-dial-entity1001] match-template 0101001
[RouterA-voice-dial-entity1001] line 1/1
[RouterA-voice-dial-entity1001] send-number all
[RouterA-voice-dial-entity1001] quit
# Configure the local number 0101002 for POTS entity 1002, and bind the digital voice
interfaces line 1/1 to the POTS entity.
[RouterA-voice-dial] entity 1002 pots
[RouterA-voice-dial-entity1002] match-template 0101002
[RouterA-voice-dial-entity1002] line 1/1
[RouterA-voice-dial-entity1002] send-number all
[RouterA-voice-dial-entity1002] quit
# Configure the called number template 0755. for VoIP entity 0755, and configure the
destination IP address as 2.2.2.2.
[RouterA-voice-dial] entity 0755 voip
[RouterA-voice-dial-entity755] match-template 0755....
[RouterA-voice-dial-entity755] address sip 2.2.2.2
2.
Configure Router B:
# Configure the IP address 2.2.2.2/24 for interface GigabitEthernet 0/0.
<RouterB> system-view
[RouterB] interface GigabitEthernet 0/0
[RouterB-GigabitEthernet0/0] ip address 2.2.2.2 255.255.255.0
[RouterB-GigabitEthernet0/0] quit
50
# Configure the local number 07552001 for POTS entity 2001, and bind the digital voice
interfaces line 1/1 to the POTS entity.
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 2001 pots
[RouterB-voice-dial-entity2001] match-template 07552001
[RouterB-voice-dial-entity2001] line 1/1
[RouterB-voice-dial-entity2001] send-number all
[RouterB-voice-dial-entity2001] quit
# Configure the local number 07552002 for POTS entity 2002, and bind the digital voice
interfaces line 1/1 to the POTS entity.
[RouterB-voice-dial] entity 2002 pots
[RouterB-voice-dial-entity2002] match-template 07552002
[RouterB-voice-dial-entity2002] line 1/1
[RouterB-voice-dial-entity2002] send-number all
[RouterB-voice-dial-entity2002] quit
# Configure the called number template 010. for VoIP entity 010, and configure the destination
IP address as 1.1.1.1.
[RouterB-voice-dial] entity 010 voip
[RouterB-voice-dial-entity10] match-template 010....
[RouterB-voice-dial-entity10] address ip 1.1.1.1
51
POTS entityA local POTS entity connects to a local telephone and maintains local number
information. A trunk POTS entity connects to the PSTN and maintains call destination information.
VoIP entityConnects to the IP side and maintains the called information such as the called
number and call destination.. A VoIP entity can use SIP to make VoIP calls.
Router A
Router B
PSTN
Local POTS
entity
IP
POTS entity
VoIP entity
52
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
dial-program
N/A
4.
5.
description string
By default, no description is
configured.
By default, no number template
is configured for the POTS
entity.
The POTS entity uses the number
template to match phone
numbers. For example, the
match-template 20 command
matches all numbers beginning
with 20.
6.
match-template match-string
7.
line line-number
8.
undo shutdown
Method 2: Create a codec template, assign priorities to codecs in the codec template, and bind the
codec template to a POTS entity.
Two parties must have the same codecs to communicate with each other. You can use the display voice
sip call command (see Voice Command Reference) to view the codec used by two parties.
To configure codecs for a POTS entity (method 1):
Step
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
dial-program
N/A
4.
N/A
53
Step
5.
Command
Remarks
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
5.
quit
N/A
6.
dial-program
N/A
7.
N/A
8.
4.
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
dial-program
N/A
4.
N/A
5.
54
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
dial-program
N/A
4.
N/A
mode:
{
5.
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
dial-program
N/A
4.
N/A
By default, VAD is disabled.
5.
Enable VAD.
55
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
dial-program
N/A
4.
N/A
5.
caller-permit calling-string
Optional.
By default, the voice entity permits all calling
numbers.
Optional.
priority priority-order
7.
substitute { called |
calling } list-number
8.
caller-group { deny |
permit } group-id
9.
send-number
{ digit-number | all |
truncate }
6.
For more information about the above commands, see "Configuring dial programs."
56
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
dial-program
N/A
4.
5.
(Optional.) Configure a
description for the VoIP entity.
description string
By default, no description is
configured.
By default, no number template is
configured for the POTS entity.
6.
match-template match-string
7.
8.
undo shutdown
Method 2: Create a codec template, assign priorities to codecs in the codec template, and bind the
codec template to a VoIP entity.
Two parties must have the same codecs to communicate with each other. You can use the display voice
sip call command (see Voice Command Reference) to view the codec used by two parties.
To configure codecs for a VoIP entity (method 1):
Step
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
dial-program
N/A
4.
N/A
57
Step
5.
Command
Remarks
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
5.
quit
N/A
6.
dial-program
N/A
7.
N/A
8.
4.
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
dial-program
N/A
4.
N/A
58
Step
Command
Remarks
mode:
{
5.
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
dial-program
N/A
4.
N/A
By default, VAD is disabled.
5.
Enable VAD.
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
dial-program
N/A
4.
N/A
5.
caller-permit calling-string
Optional.
59
Step
Command
Remarks
Optional.
priority priority-order
7.
substitute { called |
calling } list-number
8.
caller-group { deny |
permit } group-id
6.
For more information about the above commands, see "Configuring dial programs."
Command
60
Procedure
To configure caller control:
Step
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
dial-program
N/A
4.
N/A
61
Step
Command
Remarks
By default, the voice entity permits
all calling numbers.
5.
caller-permit calling-string
Examples
As shown in Figure 24, configure caller control to permit only the number 1000 to call the number 2000.
Figure 24 Caller control diagram
You can configure caller control either on the calling side or on the called side to meet the requirement.
Method 1: Configure the calling side to permit only the calling number 1000 to call the number
2000.
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 2000 voip
[RouterA-voice-dial-entity2000] match-template 2000
[RouterA-voice-dial-entity2000] address sip ip 1.1.1.2
[RouterA-voice-dial-entity2000] caller-permit 1000$
Method 2: Configure the called side to permit only the calling number 1000 to call the number
2000.
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 2000 pots
[RouterB-voice-dial-entity2000] match-template 2000
[RouterB-voice-dial-entity2000] caller-permit 1000$
[RouterB-voice-dial-entity2000] line 1/0
62
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
dial-program
N/A
4.
subscriber-group group-id
(Optional.) Configure a
description for the subscriber
group.
description text
5.
6.
match-template match-string
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
dial-program
N/A
4.
N/A
5.
63
Command
Remarks
1.
system-view
N/A
2.
subscriber-line line-number
N/A
3.
Configure PLAR.
private-line string
If shortest match is used, the router matches and calls the number 0106688.
If longest match is used, the router matches and calls the number 01066880011.
If shortest match is used, the router immediately matches and calls the number 0106688.
If longest match is used, the router waits until the dial timer expires. Then the router matches the
received number against match templates. If a match is found, the router calls the matching number.
If no match is found, the router releases the call.
You can configure a terminator that identifies the end of a number. Suppose the terminator is the pound
sign #. When a subscriber dials 0106688#, the router immediately matches the number 0106688 with
match templates regardless of whether longest match is used. If a match is found, the router calls the
matching number.
Procedure
To configure a number match mode:
Step
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
dial-program
N/A
4.
number-match { longest |
shortest }
5.
Configure a terminator.
terminator character
By default, no terminator is
configured.
64
Examples
As shown in Figure 25, configure number match modes for calls from Telephone A to Telephones B and
C.
Figure 25 Network diagram
Configure Router A:
# Configure POTS entity 1000.
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 1000 pots
[RouterA-voice-dial-entity1000] match-template 10001234$
[RouterA-voice-dial-entity1000] line 1/0
[RouterA-voice-dial-entity1000] quit
2.
Configure Router B:
# Configure POTS entity 2000 and POTS entity 2001.
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 2000 pots
[RouterB-voice-dial-entity2000] match-template 20001234$
65
After you dial number 20001234 at Telephone A, the number matches VoIP entity 2000 and
Telephone B rings because the device uses shortest match mode by default.
After you dial number 20001234 at Telephone A and waits for a period of time, the number matches
VoIP entity 2000 and Telephone B rings. If you dial 1234 during the waiting period (the dialed number
is 200012341234), the number matches VoIP entity 2001 and Telephone C rings.
Configuring a terminator
# Configure the terminator # on Router A.
[RouterA-voice-dial] terminator #
After you dial 20001234# at Telephone A, the number immediately matches VoIP entity 2000 and
Telephone B rings.
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
dial-program
N/A
4.
N/A
5.
max-conn max-number
66
1.
The router first matches a number against the preferred number substitution rule. If they match, the
router replaces the number based on the preferred rule.
2.
If the match fails, the router matches the number against other number substitution rules in
sequence. Once a rule is matched, the router replaces the number based on the matching rule and
stops matching other number substitution rules.
The router matches the number against global number substitution rules. If a match is found, the
router replaces the number based on the matching rule.
2.
If no match is found, the router selects a matching voice entity, and replaces the number based on
the matching rule on the voice entity.
3.
PSTN
Yes
Global number
substitution rule matched?
Performs number
substitution
No
No
Yes
Number
substitution rule on the voice
entity matched?
No
Yes
Performs number substitution
Call failure
The router matches the number against global number substitution rules. If a match is found, the
router replaces the number based on the matching rule.
67
2.
If no match is found, the router selects a matching voice entity, and matches the number against
rules on the voice entity. If a match is found, the router replaces the number based on the matching
rule.
3.
If no match is found, the router calls the called number if the callee is a local subscribe line, or
initiates a call to the PSTN if the callee is in the PSTN.
Yes
Performs number
substitution
No
No
Voice entity matched?
Yes
No
Yes
PSTN
Call failure
Voice interface
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
dial-program
N/A
4.
number-substitute list-number
5.
68
Step
Command
Remarks
6.
Configure a number
substitution rule.
rule id input-template
output-template [ number-type
input-number-type
output-number-type |
numbering-plan
input-numbering-plan
output-numbering-plan ] *
7.
first-rule id
8.
quit
N/A
9.
substitute { incoming-call |
outgoing-call } { called | calling }
list-number
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
dial-program
N/A
4.
number-substitute list-number
6.
Configure a number
substitution rule.
rule id input-template
output-template [ number-type
input-number-type
output-number-type |
numbering-plan
input-numbering-plan
output-numbering-plan ] *
7.
first-rule id
8.
quit
N/A
9.
N/A
5.
69
Examples
This section provides examples for global number substitution and voice entity number substitution.
Figure 28 Network diagram
# Configure the destination address as 1.1.1.2 and the called number as 1001 for VoIP entity 1001.
[RouterA-voice-dial] entity 1001 voip
[RouterA-voice-dial-entity1001] address sip ip 1.1.1.2
[RouterA-voice-dial-entity1001] match-template 1001
[RouterA-voice-dial-entity1001] quit
# Apply number substitution rule list 1 to the called number of outgoing calls.
[RouterA-voice-dial] substitute outgoing-call called 1
When you use Telephone A to call number 0101001, Router A replaces 0101001 with 1001 and sends
the call to the destination address 1.1.1.2. Then Telephone B rings.
# Configure the destination address as 1.1.1.2 and the called number as 0101001 for VoIP entity 1001.
[RouterA-voice-dial] entity 1001 voip
[RouterA-voice-dial-entity1001] address sip ip 1.1.1.2
[RouterA-voice-dial-entity1001] match-template 0101001
# Apply number substitution rule list 1 to the called number on VoIP entity 1001.
[RouterA-voice-dial-entity1001] substitute called 1
When you use Telephone A to call number 0101001, VoIP entity 1001 replaces 0101001 with 1001 and
sends the call to the destination address 1.1.1.2. Then Telephone B rings.
70
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
dial-program
N/A
4.
N/A
5.
priority priority-order
Sends the least significant digits of a called number (the number of the least significant digits is
specified with the send-number digit-number command).
Sends a truncated called number. When the match-template command configured for a voice
entity contains dots ".", only the digits that match the dots are sent.
Procedure
To configure a number sending mode:
Step
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
dial-program
N/A
4.
N/A
5.
Examples
As shown in Figure 29, Router A and Router B connect to each other through an IP link and an E1 link.
The telephones at two sides call each other through the E1 link. This section provides examples for
configuring number sending modes to achieve different call policies.
71
FXS 3/0
Telephone A
2000
Router A
Eth2/1
1.1.1.1/24
Eth2/1
1.1.1.2/24
E1 1/1
E1 1/1
Telephone B
01001
FXS 3/1
Telephone C
0101001
FXS 3/2
Telephone D
1001
Router B
Configure Router A:
# Configure a timeslot set using R2 signaling for controller E1 1/1.
<RouterA> system-view
[RouterA] controller e1 1/1
[RouterA-E1 1/1] timeslot-set 0 timeslot-list 1-31 signal r2
[RouterA-E1 1/1] quit
[RouterA] voice-setup
[RouterA-voice] dial-program
# Configure a match template to match 010.$ for POTS entity 1001 and bind line 1/1:0 to the
entity.
[RouterA-voice-dial] entity 1001 pots
[RouterA-voice-dial-entity1001] match-template 010....$
[RouterA-voice-dial-entity1001] line 1/1:0
# Configure a local number 2000 for POTS entity 2000, and bind line3/0 to the entity.
[RouterA-voice-dial] entity 2000 pots
[RouterA-voice-dial-entity2000] match-template 2000
[RouterA-voice-dial-entity2000] line 3/0
2.
Configure Router B:
# Configure a timeslot set using R2 signaling for controller E1 1/1.
<RouterB> system-view
[RouterB] controller e1 1/1
[RouterB-E1 1/1] timeslot-set 0 timeslot-list 1-31 signal r2
[RouterB-E1 1/1] quit
# Configure a local number 1001 for POTS entity 1002, and bind line 3/0 to the entity.
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 1002 pots
[RouterB-voice-dial-entity1002] match-template 01001
[RouterB-voice-dial-entity1002] line 3/0
[RouterB-voice-dial-entity1002] quit
# Configure a local number 0101001 for POTS entity 0101001, and bind line 3/1 to the entity.
[RouterB-voice-dial] entity 0101001 pots
72
# Configure a local number 1001 for POTS entity 1001, and bind line 3/2 to the entity.
[RouterB-voice-dial] entity 1001 pots
[RouterB-voice-dial-entity1001] match-template 1001
[RouterB-voice-dial-entity1001] line 3/2
If you use Telephone A to call 0101001, Router A sends the called number 01001, which are the five least
significant digits. Then Telephone B rings.
# Configure Router A to send all the digits of a called number.
[RouterA-voice-dial-entity1001] send-number all
If you use Telephone A to call 0101001, Router A sends the complete called number 0101001. Then
Telephone C rings.
# Configure Router A to send a truncated called number.
[RouterA-voice-dial-entity1001] send-number truncate
If you use Telephone A to call 0101001, Router A sends the called number 1001, which match the dots in
"010.$". Then Telephone D rings.
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
dial-program
N/A
4.
N/A
5.
dial-prefix string
73
M Dept. 3366
M Dept. 6788
Eth2/1
2.2.2.2/24
FXO 1/0
F Dept. 1688
PBX
WAN
Eth2/1
1.1.1.1/24
Router B
FXO 1/0
Router A
PBX
F Dept 1234
S Dept. 6565
S Dept. 2323
Configuration procedure
The following only provides the number substitution configuration for calls from area B to area A.
1.
Configure Router B:
# Create number substitution rule list 21101 and add rules to the list.
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] number-substitute 21101
[RouterB-voice-dial-substitute21101] rule 1 ^0101688$ 0001
[RouterB-voice-dial-substitute21101] rule 2 ^0103366$ 0002
[RouterB-voice-dial-substitute21101] rule 3 ^0102323$ 0003
[RouterB-voice-dial-substitute21101] quit
# Create number substitution rule list 21102 and add rules to the list.
[RouterB-voice-dial] number-substitute 21102
[RouterB-voice-dial-substitute21102] rule 1 ^1688$ 0210001
[RouterB-voice-dial-substitute21102] rule 2 ^3366$ 0210002
[RouterB-voice-dial-substitute21102] rule 3 ^2323$ 0210003
[RouterB-voice-dial-substitute21102] quit
# Configure the match template "010...." to match called numbers, and set the destination IP
address as 1.1.1.1 for VoIP entity 10.
[RouterB-voice-dial] entity 10 voip
[RouterB-voice-dial-entity10] match-template 010....
[RouterB-voice-dial-entity10] address sip ip 1.1.1.1
74
# Apply the number substitution rule list 21101 to called numbers on VoIP entity 10, which will
change the called numbers 0101688, 0103366, and 0102323 into 0001, 0002, and 0003
respectively.
[RouterB-voice-dial-entity10] substitute called 21101
# Apply the number substitution rule list 21102 to calling numbers on VoIP entity 10, which will
change the calling numbers 1688, 3366, and 2323 into 0210001, 0210002, and 0210003
respectively.
[RouterB-voice-dial-entity10] substitute calling 21102
2.
Configure Router A:
# Create number substitution rule list 101 and add rules to the list.
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] number-substitute 101
[RouterA-voice-dial-substitute101] rule 1 ^0001$ 1234
[RouterA-voice-dial-substitute101] rule 2 ^0002$ 6788
[RouterA-voice-dial-substitute101] rule 3 ^0003$ 6565
[RouterA-voice-dial-substitute101] quit
# Create number substitution rule list 102 and add rules to the list.
[RouterA-voice-dial] number-substitute 102
[RouterA-voice-dial-substitute102] dot-match left-right
[RouterA-voice-dial-substitute102] rule 1 ^...0001$ ...1234
[RouterA-voice-dial-substitute102] rule 2 ^...0002$ ...6788
[RouterA-voice-dial-substitute102] rule 3 ^...0003$ ...6565
[RouterA-voice-dial-substitute102] quit
# Apply the number substitution rule list 101 to the called numbers of incoming calls, which will
change the called numbers 0001, 0002, and 0003 into 1234, 6788, and 6565 respectively.
[RouterA-voice-dial] substitute incoming-call called 101
# Apply the number substitution rule list 102 to the calling numbers of incoming calls, which will
change the called numbers 0210001, 0210002, and 0210003 into 0211234, 0216788, and
0216565 respectively.
[RouterA-voice-dial] substitute incoming-call calling 102
# Configure the match template "." for POTS entity 1010, bind line 1/0 to the entity, and
configure the entity to send all the digits of a called number.
[RouterA-voice-dial] entity 1010 pots
[RouterA-voice-dial-entity1010] match-template ....
[RouterA-voice-dial-entity1010] line 1/0
[RouterA-voice-dial-entity1010] send-number all
75
The phone numbers starting with 1100 in area A can only call the phone numbers in area B.
The phone numbers starting with 1200 in area A can call the phone numbers both in area B and
area C.
Area B
110000
Router B
2100
E1 1/0
1100..
Internal
PBX
110099
Router A
IP
120000
Internal
PBX
2200
Public PBX
Area C
Router C
1200..
3100
E1 1/0
120099
Public PBX
Internal
PBX
SIP server
3200
Public PBX
Configuration procedure
This example does not provide SIP server and digital voice interface configurations. For related
information, see "Configuring SIP" and "Configuring voice interfaces."
1.
Configure Router A:
# Configure subscriber group 1 that matches calling numbers starting with 1100 and configure
subscriber group 2 that matches calling numbers starting with 1200.
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] subscriber-group 1
[RouterA-voice-dial-group1] match-template 1100..
[RouterA-voice-dial-group1] quit
[RouterA-voice-dial] subscriber-group 2
[RouterA-voice-dial-group2] match-template 1200..
[RouterA-voice-dial-group2] quit
# Configure VoIP entity 2000 to use the SIP proxy server to find call destination addresses and to
match called numbers in area B.
[RouterA-voice-dial] entity 2000 voip
76
# Configure VoIP entity 3000 to use the SIP proxy server to find call destination addresses and to
match called numbers in area C.
[RouterA-voice-dial] entity 3000 voip
[RouterA-voice-dial-entity3000] address sip proxy
[RouterA-voice-dial-entity3000] match-template 3...
[RouterA-voice-dial-entity3000] quit
# Configure a caller group to permit subscriber groups 1 and 2 on POTS entity 2100 so the phone
numbers starting with 1100 or 1200 in area A can call the phone numbers in area B.
[RouterA-voice-dial] entity 2100 pots
[RouterA-voice-dial-entity2100] line 2/0:15
[RouterA-voice-dial-entity2100] send-number all
[RouterA-voice-dial-entity2100] match-template 2...
[RouterA-voice-dial-entity2100] caller-group permit 1
[RouterA-voice-dial-entity2100] caller-group permit 2
[RouterA-voice-dial-entity2100] quit
# Configure a caller group to permit subscriber group 2 on POTS entity 3100 so the phone
numbers starting with 1200 in area A can call the phone numbers in area C.
[RouterA-voice-dial] entity 3100 pots
[RouterA-voice-dial-entity3100] line 2/0:15
[RouterA-voice-dial-entity3100] send-number all
[RouterA-voice-dial-entity3100] match-template 3...
[RouterA-voice-dial-entity3100] caller-group permit 2
[RouterA-voice-dial] quit
# Configure POTS entity 2100 to match local numbers and bind line 1/0:15 to the entity.
[RouterA-voice-dial] entity 2100 pots
[RouterA-voice-dial-entity2100] line 1/0:15
[RouterA-voice-dial-entity2100] send-number all
[RouterA-voice-dial-entity2100] match-template 1.....
2.
Configure Router B:
# Configure POTS entity 2100 to match local numbers and bind line 1/0:15 to the entity.
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 2100 pots
[RouterB-voice-dial-entity2100] line 1/0:15
[RouterB-voice-dial-entity2100] send-number all
[RouterB-voice-dial-entity2100] match-template 2...
3.
Configure Router C:
# Configure POTS entity 3100 to match local numbers and bind line 1/0:15 to the entity.
<RouterC> system-view
[RouterC] voice-setup
[RouterC-voice] dial-program
[RouterC-voice-dial] entity 3100 pots
[RouterC-voice-dial-entity3100] line 1/0:15
77
78
Configuring SIP
Overview
The Session Initiation Protocol (SIP) is an application layer control protocol that can create, modify, and
terminate multimedia sessions such as voice and video calls over IP networks.
Terminology
User agent
A user agent (UA) is a SIP endpoint such as a phone, a gateway, or a router.
There are two types of UAs: user agent client (UAC) and user agent server (UAS). A UAC sends SIP
requests, and a User Agent Server (UAS) receives SIP requests and returns SIP responses. These roles of
UAC and UAS only last for the duration of a SIP transaction.
Proxy server
A proxy server primarily forwards session requests and responses. It can also provide call control,
accounting, and authorization functions.
Redirect server
A redirect server sends new addresses to UACs so the UAC sends session requests to the new addresses.
Location server
A location server provides UA information to proxy and redirect servers.
Registrar
A registrar receives registrations from UAs and generates UA information. The UA information is stored
on the location server.
SIP functions
SIP supports the following facets of establishing and terminating multimedia communications:
User locationDetermines the end system to be used for communication. SIP can use UA
information on the registrar, or use information provided by DNS or LDAP to locate end systems.
User capabilitiesDetermines the media type and media parameters to be used. In a message
exchange process, each SIP endpoint advertises media information so that all other participants
can learn about its capabilities.
79
SIP messages
SIP is a text-based protocol. A SIP message is either a request from a client to a server, or a response from
a server to a client.
SIP requests include INVITE, ACK, OPTIONS, BYE, CANCEL, and REGISTER.
SIP responses indicate the status of a call or registration. Responses are distinguished by status codes. As
shown in Table 8, each status code is a 3-digit integer, where the first digit defines the class of the
response, and the last two digits describe the response message in more detail.
Table 8 Status codes of responses
Code
Description
Class
100199
Provisional
200299
Success
300399
Redirection
400499
Client error
500599
Server error
600699
Global failure
80
Tasks at a glance
(Optional.) Configuring extended SIP functions
2.
UA
IP network
REGISTER
200 OK
If the registrar needs to authenticate the UA, the UA registers with the registrar as shown in Figure 33:
3.
4.
The registrar returns a 401/407 response, challenging the originator to provide credentials.
5.
6.
Configuration prerequisites
Complete the following tasks before you configure SIP credentials information:
81
Configure a number template on each voice entity with the match-template command and bind
each voice entity to a voice subscriber line with the line command.
Enable the voice entities and voice subscriber lines (in undo shutdown state).
Use the user command in SIP view to configure global SIP credentials.
Use the credentials command to configure SIP credentials for a SIP trunk account. For more
information about SIP trunk, see Configuring SIP trunk.
Use the user command in voice entity view to configure SIP credentials for a voice entity.
The first three bindings each contain a domain name, and the last binding contains no domain name. If
the SIP UA receives a 401/407 response that includes a domain name server2, the SIP UA responds with
the username 1000 and password 1000. If the SIP UA receives a 401/407 response that includes a
domain name server4, the SIP UA responds with the username 1000 and password 3000 because no
credentials binding contains the domain name server4.
Credentials selection for a phone number that exists on multiple voice entities
Upon receiving a 401/407 response for a phone number that exists on multiple voice entities, the SIP UA
considers the phone number belongs to the voice entity with the smallest ID and selects the credentials for
the phone number in the following order:
1.
2.
3.
The SIP UA always uses the matching credentials for the phone number even if a voice entity that has a
higher match priority is added. If no matching credentials are found, the SIP UA fails to register the
phone number.
For example, the registrar maintains the username abcd, password 1234, and domain name abc for the
phone number 1000. The SIP UA has the following settings for the phone number 1000:
POTS entity 1 maintains the username abcd, password 1234, and domain name aaa for the phone
number 1000.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
82
POTS entity 2 maintains the username abcd, password 1234, and domain name abc for the phone
number 1000.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 2 pots
[Sysname-voice-dial-entity2] match-template 1000
[Sysname-voice-dial-entity2] user abcd password simple 1234 realm abc
The SIP trunk account maintains the username abcd, password 1234, and domain name abc for the
phone number 1000.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] sip
[Sysname-voice-sip] credentials number 1000 username abcd password simple 1234 realm
abc
The global SIP credentials information includes the username abcd, password 1234, and domain
name abc.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] sip
[Sysname-voice-sip] user abcd password simple 1234 realm abc
Upon receiving a 401/407 response for the phone number 1000, the SIP UA considers the phone
number belongs to POTS entity 1 and selects the credentials on POTS entity 1 because POTS entity 1 has
a smaller ID, but the credentials fail the authentication. Then the SIP UA selects the credentials configured
with the credentials command and the credentials pass the authentication. The output from the display
voice sip register-status command shows that the phone number 1000 belongs to voice entity 1.
<Sysname> display voice sip register-status
Number
Entity
Registrar Server
Expires Status
-------------------------------------------------------------------------------1000
192.168.4.240:5060
2877
Online
Configuration procedure
To configure global SIP credentials:
Step
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
sip
N/A
4.
83
Step
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
dial-program
N/A
4.
N/A
5.
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
dial-program
N/A
4.
N/A
5.
register-number
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
sip
N/A
4.
84
Command
Configure the VoIP entity to obtain the call destination address from a proxy server.
Configure the destination domain name and port number. Only DNS A records are supported. For
more information about DNS, see Layer 3IP Services Configuration Guide.
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
dial-program
N/A
4.
N/A
5.
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
sip
N/A
4.
5.
quit
85
Step
Command
Remarks
6.
dial-program
N/A
7.
N/A
8.
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
dial-program
N/A
4.
N/A
5.
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
dial-program
N/A
4.
N/A
5.
outband sip
86
Min-SEConveys the minimum session expiration time, which is used to avoid frequent refresh
requests from occupying excessive network bandwidth.
422 responseWhen a UAS or SIP proxy server receives a request in which the Session-Expires
field conveys a value smaller than the local Min-SE value, the UAS or SIP proxy server sends a 422
response that contains the local Min-SE value to notify the requesting party of the minimum session
expiration time.
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
sip
N/A
session refresh
4.
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
sip
N/A
87
Step
Command
Remarks
4.
5.
Configure a SIP
status-to-PSTN cause
mapping.
6.
N/A
PrivacyIf the Privacy header field carries "Privacy: none", the caller ID is displayed. If the Privacy
header field carries "Privacy: id", the caller ID is hidden.
NOTE:
The Remote-Party-ID header field cannot coexist with the P-Preferred-Identity or P-Asserted-Identity
header field.
To configure caller privacy:
Step
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
sip
N/A
4.
privacy
5.
remote-party-id
88
Step
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
sip
N/A
4.
Command
Configuration procedure
1.
Configure Router A:
# Configure an IP address for Ethernet 2/1.
<RouterA> system-view
[RouterA] interface ethernet 2/1
[RouterA-Ethernet2/1] ip address 192.168.2.1 255.255.255.0
[RouterA-Ethernet2/1] quit
# Configure call destination address 192.168.2.2 and number 2222 for VoIP entity 2222.
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 2222 voip
89
# Configure local number 1111 for POTS entity 1111 and bind FXS voice subscriber line 1/0 to
the POTS entity.
[RouterA-voice-dial] entity 1111 pots
[RouterA-voice-dial-entity1111] line 1/0
[RouterA-voice-dial-entity1111] match-template 1111
2.
Configure Router B:
# Configure an IP address for Ethernet 2/1.
<RouterB> system-view
[RouterB] interface ethernet 2/1
[RouterB-Ethernet2/1] ip address 192.168.2.2 255.255.255.0
[RouterB-Ethernet2/1] quit
# Configure local number 2222 for POTS entity 1111 and bind FXS voice subscriber line 1/0 to
the POTS entity.
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 2222 pots
[RouterB-voice-dial-entity2222] line 1/0
[RouterB-voice-dial-entity2222] match-template 2222
[RouterB-voice-dial-entity2222] quit
# Configure call destination address 192.168.2.1 and number 1111 for VoIP entity 1111.
[RouterB-voice-dial]entity 1111 voip
[RouterB-voice-dial-entity1111] address sip ip 192.168.2.1
[RouterB-voice-dial-entity1111] match-template 1111
90
Configuration procedure
1.
Configure Router A:
# Configure an IP address for Ethernet 2/1.
<RouterA> system-view
[RouterA] interface ethernet 2/1
[RouterA-Ethernet2/1] ip address 192.168.2.1 255.255.255.0
[RouterA-Ethernet2/1] quit
# Configure local number 1111 for POTS entity 1111 and bind FXS voice subscriber line 1/0 to
the POTS entity.
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 1111 pots
[RouterA-voice-dial-entity1111] line 1/0
[RouterA-voice-dial-entity1111] match-template 1111
[RouterA-voice-dial-entity1111] quit
# Configure VoIP entity 2222 to get the call destination address from the proxy server and to
match number 2222.
[RouterA-voice-dial] entity 2222 voip
[RouterA-voice-dial-entity2222] address sip proxy
[RouterA-voice-dial-entity2222] match-template 2222
[RouterA-voice-dial-entity2222] quit
2.
Configure Router B:
# Configure an IP address for Ethernet 2/1.
<RouterB> system-view
[RouterB] interface ethernet 2/1
[RouterB-Ethernet2/1] ip address 192.168.2.2 255.255.255.0
[RouterB-Ethernet2/1] quit
91
[RouterB-voice] sip
[RouterB-voice-sip] registrar 1 ip 192.168.2.3
[RouterB-voice-sip] proxy ip 192.168.2.3
[RouterB-voice-sip] quit
# Configure local number 2222 for POTS entity 2222 and bind FXS voice subscriber line 1/0 to
the POTS entity.
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 2222 pots
[RouterB-voice-dial-entity2222] line 1/0
[RouterB-voice-dial-entity2222] match-template 2222
# Configure user name routerb, plaintext password 7890, and domain name server1.
[RouterB-voice-dial-entity2222] user routerB password simple 7890 realm server1
# Configure VoIP entity 2222 to get the call destination address from the proxy server and to
match number 1111.
[RouterB-voice-dial-entity2222] quit
[RouterB-voice-dial] entity 1111 voip
[RouterB-voice-dial-entity1111] address sip proxy
[RouterB-voice-dial-entity1111] match-template 1111
[RouterB-voice-dial-entity1111] quit
Configuration procedure
1.
Configure Router A:
# Configure an IP address for Ethernet 2/1.
<RouterA> system-view
[RouterA] interface ethernet 2/1
[RouterA-Ethernet2/1] ip address 192.168.2.1 255.255.255.0
[RouterA-Ethernet2/1] quit
92
# Configure the VoIP entity 2222 to call the destination through domain name cc.news.com and
to match number 2222.
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 2222 voip
[RouterA-voice-dial-entity2222] address sip dns cc.news.com port 5060
[RouterA-voice-dial-entity2222] match-template 2222
[RouterA-voice-dial-entity2222] quit
# Configure local number 1111 for POTS entity 1111 and bind FXS voice subscriber line 1/0 to
the POTS entity.
[RouterA-voice-dial] entity 1111 pots
[RouterA-voice-dial-entity1111] line 1/0
[RouterA-voice-dial-entity1111] match-template 1111
2.
Configure Router B:
# Configure an IP address for Ethernet 2/1.
<RouterB> system-view
[RouterB] interface ethernet 2/1
[RouterB-Ethernet2/1] ip address 192.168.2.2 255.255.255.0
# Configure local number 2222 for POTS entity 2222 and bind FXS voice subscriber line 1/0 to
the POTS entity.
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 2222 pots
[RouterB-voice-dial-entity2222] line 1/0
[RouterB-voice-dial-entity2222] match-template 2222
[RouterB-voice-dial-entity2222] quit
# Configure the destination address 192.168.2.1 and number 1111 for the VoIP entity 1111.
[RouterB-voice-dial]entity 1111 voip
[RouterB-voice-dial-entity1111] address sip ip 192.168.2.1
[RouterB-voice-dial-entity1111] match-template 1111
93
Configuration procedure
1.
Configure Router A:
# Configure an IP address for Ethernet 2/1.
<RouterA> system-view
[RouterA] interface ethernet 2/1
[RouterA-Ethernet2/1] ip address 192.168.2.1 255.255.255.0
[RouterA-Ethernet2/1] quit
# Configure call destination address 192.168.2.2 and number 2222 for VoIP entity 2222.
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 2222 voip
[RouterA-voice-dial-entity2222] address sip ip 192.168.2.2
[RouterA-voice-dial-entity2222] match-template 2222
# Configure local number 1111 for POTS entity 1111 and bind FXS voice subscriber line 1/0 to
the POTS entity.
[RouterA-voice-dial] entity 1111 pots
[RouterA-voice-dial-entity1111] line 1/0
[RouterA-voice-dial-entity1111] match-template 1111
2.
Configure Router B:
# Configure an IP address for Ethernet 2/1.
<RouterB> system-view
[RouterB] interface ethernet 2/1
[RouterB-Ethernet2/1] ip address 192.168.2.2 255.255.255.0
[RouterB-Ethernet2/1] quit
# Configure call destination address 192.168.2.1 and number 1111 for VoIP entity 1111.
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 1111 voip
[RouterB-voice-dial-entity1111] address sip ip 192.168.2.1
[RouterB-voice-dial-entity1111] match-template 1111
# Configure local number 2222 for POTS entity 2222 and bind FXS voice subscriber line 1/0 to
the POTS entity.
94
3.
95
Background
As shown in Figure 38, in a typical telephone network, a PBX forwards internal calls among enterprise
phones and forwards outbound calls over a PSTN trunk.
Figure 38 Typical telephone network
With the development of IP technology, many enterprises deploy SIP-based IP-PBX networks as shown
in Figure 39. All internal calls are placed using SIP, and external calls are still placed over a PSTN trunk.
The problem is that the enterprises have to maintain both the SIP network and PSTN trunk, which
increases the difficulty of network management.
Figure 39 SIP+PSTN network
As more enterprise IP-PBX networks run SIP and more Internet Telephone Service Providers (ITSPs) use SIP
to provide basic voice communication structures, enterprises urgently need a technology that can connect
the enterprise IP-PBX network to the ITSP over SIP. This technology is called SIP trunk. A typical network
diagram of SIP trunk is shown in Figure 40.
The SIP trunk function can be embedded into the voice gateway or the firewall deployed at the edge of
an enterprise private network. The device providing the SIP trunk function is called the SIP trunk device,
or the SIP trunk gateway.
96
Features
SIP trunk has the following features:
The SIP trunk device and the ITSP only establish one secure and QoS guaranteed SIP trunk link. The
SIP trunk link can carry multiple concurrent calls, and the ITSP only authenticates the link instead of
each SIP call carried on this link.
The enterprise IP-PBX forwards internal calls. The SIP trunk device forwards outbound calls to the
ITSP, and then the devices in the ITSP forward the calls to the PSTN. Enterprises do not need to
maintain the PSTN trunk, which reduces the costs of hardware and maintenance.
With the SIP trunk device deployed, the entire network can use SIP to better support IP
communication services, such as voice, video conferencing, and instant messaging.
The SIP trunk device initiates a new call request to the ITSP on behalf of the user after receiving a call
request from the user, and both the user and the ITSP communicate only with the SIP trunk device.
The SIP trunk device forwards both SIP signaling messages and RTP media messages.
Typical applications
The SIP trunk device is deployed between the enterprise IP-PBX and the ITSP. All internal calls are placed
by the enterprise IP-PBX. All outbound calls are forwarded by the SIP trunk device to the ITSP through the
SIP trunk link. Figure 41 shows a typical SIP trunk technology.
97
RFC 3261
RFC 3515
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
Enable SIP-to-SIP
calling.
98
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
sip
N/A
4.
Specify a registrar.
registrar registrar-index { ip
ip-address | dns domain-name }
[ port port-number ] [ expires
seconds ] [ refresh-ratio
ratio-percentage ]
5.
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
dial-program
N/A
4.
N/A
5.
codec transparent
By default, codec
transparent transmission is
disabled.
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
dial-program
N/A
4.
N/A
media flow-around
5.
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
3.
dial-program
N/A
4.
N/A
By default, DO-EO
conversion is disabled.
5.
Command
100
Configuration procedure
1.
Configure Router A:
# Specify local number 2000 on POTS entity 2000.
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 2000 pots
[RouterA-voice-dial-entity2000] line 1/0
[RouterA-voice-dial-entity2000] match-template 2000
[RouterA-voice-dial-entity2000] quit
# Configure the called number as 1000, and the destination IP address as 1.1.1.2 (the address of
the interface on the SIP trunk device) on VoIP entity 1000.
[RouterA-voice-dial] entity 1000 voip
[RouterA-voice-dial-entity1000] address sip ip 1.1.1.2
[RouterA-voice-dial-entity1000] match-template 1000
2.
# Create a SIP trunk account that contains number 2000, username 2000, password 2000, and
realm abc.
[TG-voice-sip] credentials number 2000 username 2000 password simple 2000 realm abc
# Configure the destination address as 2.1.1.2 for outbound calls from private phone 2000 to
public phone 1000.
[TG-voice] dial-program
[TG-voice-dial] entity 1 voip
[TG-voice-dial-entity1] address sip ip 2.1.1.2
[TG-voice-dial-entity1] match-template 1000
[TG-voice-dial-entity1] quit
# Configure the destination address as 1.1.1.1 for inbound calls from public phone 1000 to
private phone 2000.
[TG-voice-dial] entity 2 voip
[TG-voice-dial-entity2] address sip ip 1.1.1.1
[TG-voice-dial-entity2] match-template 2000
3.
Configure Router B:
# Specify local number 1000 on POTS entity 1000.
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 1000 pots
[RouterB-voice-dial-entity1000] line 1/0
[RouterB-voice-dial-entity1000] match-template 1000
[RouterB-voice-dial-entity1000] quit
# Configure the called number as 2000, and the destination IP address as 2.1.1.1 (the address of
the interface on the SIP trunk device) on VoIP entity 2000.
[RouterB-voice-dial] entity 2000 voip
[RouterB-voice-dial-entity2000] address sip ip 2.1.1.1
[RouterB-voice-dial-entity2000] match-template 2000
4.
Entity
Registrar Server
Expires Status
------------------------------------------------------------------------------2000
10.1.1.2:5060
1802
Online
The output shows that the private phone number 2000 has registered with the server at
10.1.1.2.
b. Execute the display voice statistics call-active command to verify that all calls between the
private network and public network pass through the SIP trunk device.
c. On the SIP server of the ITSP, you can view only the interface address of the SIP trunk device,
which means that the SIP trunk device can hide the information of private users.
102
Call waiting
When subscriber C calls subscriber A who is in a conversation with subscriber B, the call is not rejected.
Like a normal call, subscriber C hears ringback tones, and subscriber A hears call waiting tones. This is
call waiting.
Subscriber A can answer the new call by pressing the flash hook or hanging up to end the call with
subscriber B. In the former case, subscriber B is held. In the latter case, subscriber A is immediately
alerted and can pick up the phone to answer the call originated by subscriber C.
Call hold
When subscriber A in a conversation with subscriber B presses the flash hook, subscriber B is held (in
silent state or listening to waiting tones). The system plays dial tones to subscriber A, waiting for
subscriber A to initiate a new call. If subscriber A fails to dial within a period of time, the system stops
playing dial tones and subscriber A cannot initiate a new call. Subscriber A can resume the call with
subscriber B by pressing the flash hook again. This is call hold
Call forwarding
When the called party cannot answer a call, the called party notifies the calling party of the
forwarded-to number. The calling party re-initiates a call request to the new destination. This is call
forwarding. For example, subscriber A calls subscriber B who is busy, and subscriber A forwards the call
to subscriber C. Then, subscriber A and subscriber C establishes a call. Subscriber A is the initiator of call
forwarding, subscriber B is the recipient, and subscriber C is the final recipient.
The system supports the following types of call forwarding:
Call forwarding busyForwards an incoming call to the predetermined destination when the
called party is busy.
Call forwarding no replyForwards an incoming call to the predetermined destination when the
called party provides no answer within a configurable period of time.
Call transfer
Subscriber A (the originator) and subscriber B (the recipient) are in a conversation. Subscriber A presses
the flash hook to place the call on hold. Subscriber A dials another number to originate a call to
103
subscriber C (the final recipient). After Subscriber A hangs up, the call between subscriber B and
subscriber C is established. This is call transfer.
There are three types of call transfer:
Call transfer without notificationThe originator hangs up before receiving ringback tones from
the final recipient.
Call transfer with early notificationThe originator hangs up after receiving ringback tones from
the final recipient and before speaking with the final recipient.
Call transfer with notificationThe originator hangs up after speaking with the final recipient.
During the call transfer process, if the recipient does not support call transfer or the final recipient is busy
or provides no reply, the initiator can re-establish the original call.
Call backup
The calling party might fail to receive a response after initiating a call to the called party through the IP
or PSTN network. In this case, if there is a backup link (PSTN link or VoIP link) to the called party, the
calling party can re-initiate a call to the called party over the backup link. This is call backup.
Silent mode (inactive)During call hold, the held party hears silence. This mode is configured on
the holding party (the initiator of call hold).
Unidirectional playing mode (sendonly)During call hold, the held party hears tones or music
played by a third-party music server. To play tones, you need to configure the sendonly keyword on
the holding party. To play music on hold by the third-party music server, you must configure the
sendonly moh-number string option on the SIP trunk device and configure the inactive keyword on
the holding party. For information about configuring a SIP trunk device, see "Configuring SIP trunk."
Command
Remarks
1.
system-view
N/A
2.
voice-setup
N/A
104
Step
3.
Command
Remarks
call-hold-format { inactive |
sendonly [ moh-number string ] }
Command
Remarks
1.
system-view
N/A
2.
dial-program
N/A
3.
N/A
By default, call forwarding is
disabled.
4.
call-forwarding { no-reply |
on-busy | unavailable |
unconditional } number number
105
Router B
Eth1/1
10.1.1.1/24
1000
Telephone A
Eth1/2
10.1.1.2/24
Eth1/1
20.1.1.2/24
Router C
Eth1/1
20.1.1.1/24
3000
Telephone C
2000
Telephone B
Configuration procedure
Before performing the following configuration, make sure Router A, Router B and Router C can reach
each other.
1.
Configure Router A:
# Create VoIP entity 2000, configure the destination IP address as 10.1.1.2, and configure the
called number as 2000.
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 2000 voip
[RouterA-voice-dial-entity2000] address sip ip 10.1.1.2
[RouterA-voice-dial-entity2000] match-template 2000
[RouterA-voice-dial-entity2000] quit
# Create VoIP entity 3000, configure the destination IP address as 20.1.1.2, and configure the
called number as 3000.
[RouterA-voice-dial] entity 3000 voip
[RouterA-voice-dial-entity3000] address sip ip 20.1.1.2
[RouterA-voice-dial-entity3000] match-template 3000
[RouterA-voice-dial-entity3000] quit
# Configure the local number as 1000 for POTS entity 1000, and bind FXS interface line 1/0 to
the POTS entity.
[RouterA-voice-dial] entity 1000 pots
[RouterA-voice-dial-entity1000] line 1/0
[RouterA-voice-dial-entity1000] match-template 1000
2.
Configure Router B:
# Create VoIP entity 1000, configure the destination IP address as 10.1.1.1, and configure the
called number as 1000.
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 1000 voip
[RouterB-voice-dial-entity1000] address sip ip 10.1.1.1
[RouterB-voice-dial-entity1000] match-template 1000
[RouterB-voice-dial-entity1000] quit
106
# Configure the local number as 2000 for POTS entity 2000, and bind FXS interface line 1/0 to
the POTS entity.
[RouterB-voice-dial] entity 2000 pots
[RouterB-voice-dial-entity2000] line 1/0
[RouterB-voice-dial-entity2000] match-template 2000
3.
Configure Router C:
# Configure the local number as 3000 for POTS entity 3000, and bind FXS interface line 1/0 to
the POTS entity.
<RouterC> system-view
[RouterC] voice-setup
[RouterC-voice] dial-program
[RouterC-voice-dial] entity 3000 pots
[RouterC-voice-dial-entity3000] line 1/0
[RouterC-voice-dial-entity3000] match-template 3000
[RouterC-voice-dial-entity3000] quit
# Create VoIP entity 1000, configure the destination IP address as 10.1.1.1, and configure the
called number as 1000.
[RouterB-voice-dial] entity 1000 voip
[RouterB-voice-dial-entity1000] address sip ip 10.1.1.1
[RouterB-voice-dial-entity1000] match-template 1000
107
Router B
Eth1/1
10.1.1.1/24
1000
Telephone A
Eth1/2
10.1.1.2/24
Eth1/1
20.1.1.2/24
Router C
Eth1/1
20.1.1.1/24
3000
Telephone C
2000
Telephone B
Configuration procedure
Before performing the following configuration, make sure Router A, Router B and Router C can reach
each other.
1.
Configure Router A:
# Create VoIP entity 2000, configure the destination IP address as 10.1.1.2, and configure the
called number as 2000.
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 2000 voip
[RouterA-voice-dial-entity2000] address sip ip 10.1.1.2
[RouterA-voice-dial-entity2000] match-template 2000
[RouterA-voice-dial-entity2000] quit
# Configure the local number as 1000 for POTS entity 1000, and bind FXS interface line 1/0 to
the POTS entity.
[RouterA-voice-dial] entity 1000 pots
[RouterA-voice-dial-entity1000] line 1/0
[RouterA-voice-dial-entity1000] match-template 1000
2.
Configure Router B:
# Create VoIP entity 3000, configure the destination IP address as 20.1.1.2, and configure the
number template as 3000.
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 3000 voip
[RouterB-voice-dial-entity3000] address sip ip 20.1.1.2
[RouterB-voice-dial-entity3000] match-template 3000
[RouterB-voice-dial-entity3000] quit
# Configure the local number as 2000 for POTS entity 2000, and bind FXS interface line 1/0 to
the POTS entity.
[RouterB-voice-dial] entity 2000 pots
[RouterB-voice-dial-entity2000] line 1/0
[RouterB-voice-dial-entity2000] match-template 2000
108
3.
On Router C, configure the local number as 3000 for POTS entity 3000, and bind FXS interface
line 1/0 to the POTS entity.
<RouterC> system-view
[RouterC] voice-setup
[RouterC-voice] dial-program
[RouterC-voice-dial] entity 3000 pots
[RouterC-voice-dial-entity3000] line 1/0
[RouterC-voice-dial-entity3000] match-template 3000
Router B
Eth1/1
10.1.1.1/24
1000
Telephone A
Eth1/2
10.1.1.2/24
Eth1/1
20.1.1.2/24
Router C
Eth1/1
20.1.1.1/24
3000
Telephone C
2000
Telephone B
Configuration procedure
Before performing the following configuration, make sure Router A, Router B and Router C can reach
each other.
1.
Configure Router A:
# Create VoIP entity 2000, configure the destination IP address as 10.1.1.2, and configure the
called number as 2000.
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 2000 voip
[RouterA-voice-dial-entity2000] address sip ip 10.1.1.2
[RouterA-voice-dial-entity2000] match-template 2000
[RouterA-voice-dial-entity2000] quit
# Create VoIP entity 3000, configure the destination IP address as 20.1.1.2, and configure the
called number as 3000.
[RouterA-voice-dial] entity 3000 voip
109
# Configure the local number as 1000 for POTS entity 1000, and bind FXS interface line 1/0 to
the POTS entity.
[RouterA-voice-dial] entity 1000 pots
[RouterA-voice-dial-entity1000] line 1/0
[RouterA-voice-dial-entity1000] match-template 1000
2.
On Router B, configure the local number as 2000 for POTS entity 2000, and bind FXS interface
line 1/0 to the POTS entity.
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 2000 pots
[RouterB-voice-dial-entity2000] line 1/0
[RouterB-voice-dial-entity2000] match-template 2000
3.
On Router C, configure the local number as 3000 for POTS entity 3000, and bind FXS interface
line 1/0 to the POTS entity.
<RouterC> system-view
[RouterC] voice-setup
[RouterC-voice] dial-program
[RouterC-voice-dial] entity 3000 pots
[RouterC-voice-dial-entity3000] line 1/0
[RouterC-voice-dial-entity3000] match-template 3000
Configuration procedure
1.
Configure Router A:
# Create VoIP entity 2000 with a priority of 1, configure the destination IP address as 10.1.1.2,
and configure the called number as 2000.
<RouterA> system-view
[RouterA] voice-setup
[RouterA-voice] dial-program
[RouterA-voice-dial] entity 2000 voip
110
# Create VoIP entity 3000 with a priority of 2, configure the destination IP address as 20.1.1.2,
and configure the called number as 2000.
[RouterA-voice-dial] entity 3000 voip
[RouterA-voice-dial-entity3000] address sip ip 20.1.1.2
[RouterA-voice-dial-entity3000] match-template 2000
[RouterA-voice-dial-entity2000] priority 2
[RouterA-voice-dial-entity3000] quit
# Configure the local number as 1000 for POTS entity 1000, and bind FXS interface line 1/0 to
the POTS entity.
[RouterA-voice-dial] entity 1000 pots
[RouterA-voice-dial-entity1000] line 1/0
[RouterA-voice-dial-entity1000] match-template 1000
2.
On Router B, configure the local number as 2000 for POTS entity 2000, and bind FXS interface
line 1/0 to the POTS entity.
<RouterB> system-view
[RouterB] voice-setup
[RouterB-voice] dial-program
[RouterB-voice-dial] entity 2000 pots
[RouterB-voice-dial-entity2000] line 1/0
[RouterB-voice-dial-entity2000] match-template 2000
111
Index
ABCDEFOPST
Configuring SIP UA registration,81
Background,96
Binding a digital voice interface to a POTS entity,45
BSV interfaces,29
E&M interface,1
E1 and T1 interfaces,29
Features,97
FXO interface,1
FXS interface,1
Overview,79
Overview,52
Configuring DTMF,16
Configuring extended SIP functions,86
Configuring R2 signaling,34
112
Typical applications,97
113