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PBX: Private Branch Exchange

Subscriber Card: Provides interfaces for connecting the internal phones.

Trunk Card: Provides the interface to connect the telephone company or other PBX.

Switched Network: Its to connect an incoming call to an outgoing one.

Signaling Processor: Used to receive and send signaling information.

Control System: Its the heart of the PBX, all control actions needed to set up/tear down a
connection are executed by the system.

When you speak into an analog phone, the sound signals are converted into electrical
signal. The volume and pitch that you use when speaking result in different variations of
electrical current. Electrical voltage, frequency, current, and charge are all used in some

combination to represent the properties of your voice.

Analog phone lines use the properties of electricity to convey changes in voice over
cabling. Of course, the process of sending the audio over phone lines is more complexity
than simply sending audio signals. Additionally, The analog phones you use at home must

carry many different types of signaling, too. Signaling may include messages such as dial
tone, dialed digits, busy signals, and so on. These signaling types are discussed later in this
chapter.

For now, analog cabling connectivity operation is summarized as following:

Each analog circuit is composed of a pair of wires. One wire is the ground, or
positive side of the connection (often called the tip). The other wire is the battery, or
negative side of the connection (often called the ring). Youll commonly hear phone
technicians talk about these wires as the tip and ring. These two wires are what
power the analog phone and allow it to function.
When the phone is lifted off hook, the phone connects the two wires, causing an
electrical signal (48V DC voltage) to flow from the phone company central office
(CO) into the phone. This is known as loop start signaling.

Analog signaling was a massive improvement compared to tin cans and string, but they
have their problems. First, an analog electrical signal experience degradation (signal
fading) over long distances. To increase the distance that the analog signal could travel,

the phone company had to install repeaters to regenerate the signal as it becomes weak.

Unfortunately, as the analog signal was regenerated, the repeater device was unable to
differentiate between the voice traveling over the wire and line noise. Each time the
repeater regenerates the voice, it also amplifies the line noise. Thus, the more times a

phone company regenerated a signal, the more distorted and difficult to understand the
signal became.

The second difficulty encountered with analog connections was the sheer number of wires
the phone company had to run to support a large geographical area or a business with a
large number of phones. Because each phone required two wires, the bundles of wire
became massive and difficult to maintain (imagine the hassle of a single pair of wires in the
bundle breaking). A solution to send multiple calls over a single wire was needed. A digital
connection is that solution.

Line signaling is responsible for off-hook, ringing signal, answer, ground start, on-hook
unidirectional supervision messaging in each direction from calling party to called party
and vice versa.
Ground start signaling originated from its implementation in so called: pay phone systems.
Many years ago, when a person lifted the handset of a pay phone, he did not receive a
dial tone until he dropped in a coin. The coin would brush past the tip and ring wires and
temporarily ground them. The grounding of the wires signaled the phone company to

send a dial tone on the line.

Loop start signaling is the typical signaling type used in home environments. Loop start
signaling is susceptible to a problem known as glare. Glare occurs when the subscriber
picks up the phone to make an outgoing call at the same time as another call comes in on

the phone line before the phone has a chance dial.

Many other types of signaling exist in the analog world. These include supervisory
signaling (on hook, off hook, ringing), informational signaling (dial tone, busy, ring back,
and so on), and address signaling (dual-tone multi-frequency (DTMF) and Pulse).

Channel-associated signaling, also known as PTS(Per-Trunk Signaling ), is a form of digital


communication signaling.

R2 signaling is a CAS system developed in the 1960s that is still in use today in Europe,
Latin America, Australia, and Asia.

Line signaling is used for supervisory signals for call setup and teardown.

Line signaling have 3 types

R2-Digital

R2-Analog

R2-Pulse

Inter-register signaling is used for address, priority levels, congestion notification, or


charges messaging.

Inter-register signaling uses multifrequency tones in the timeslot used for the call so that it
is in-band signaling.

CCS(Common Channel Signaling),in the US also common-channel interoffice signaling


(CCIS)
PRA: Primary Rate Adapter

A signaling point code (SPC) identifies the address information of a signaling point (SP) or
a signaling transfer point (STP) in a Signaling System Number 7 (SS7) network.
An SPC specifies the source and destination of an SS7 signaling message so that node in
the signaling network can send, forward, or receive signaling messages correctly.

There are three types signal nodes as followings:

Signaling Point (SP). A node that sends or receives signaling messages in a signaling
network.
Service Switching Point (SSP). The telephone exchange that initially responds, when
a telephone caller dials a number, by sending a query to a central database called a
service control point (SCP) so that the call can be handled. SSP uses the SS7
protocols which are responsible for the call setup, management, and termination

with other service switching points.

Service Control Point (SCP): The standard component of the Intelligent Network (IN)
telephone system which is used to control the service. SCPs are connected with
either SSPs or STPs.
Signaling Transfer Point (STP): A node that transfers messages received from a
signaling link to another. Element of an SS7-based Intelligent Network that performs
routing of the SS7 signaling.

PRAPrimary Rate AccessSignaling: also called PRI(Primary Rate Interface) signaling or


DSS1(Digital Subscriber Signaling System No.1) signaling, is an interface of ISDN(Integrated
Services Digital Network).
PRI corresponds to two primary rates: 1.544 Mbps (T1) and 2.048 Mbps (E1);
PRI operate in a master-slave mode (or network-user mode). You must decide which
device will be the network side and the other device will be user side.

B-channel: Bearer-channel

D-channel: Data-channel

The Primary Rate Access (PRA) protocol is one of the interface protocols defined by the
Digital Subscriber Signaling No.1 (DSS1) signaling system. The DSS1 signaling system is a
group of interface protocols between ISDN users and networks. It consists of the physical

layer, data link layer, and network layer, corresponding to the three bottom layers of the
OSI system.

Message List

SETUP: The message is sent by the calling party to the network and by the network
to the called party to establish the call.
ALERTING: The message is sent by the called party to the network and by the
network to the caller to indicate that called alerting has been initiated.
CALL PROCEEDING: The message is sent by the called party to the network or by the
network to the calling party to indicate that requested call establishment has been
initiated and no more call establishment information will be accepted.

CONNECT: The message is sent by the called party) to the network and by the
network to the calling party to indicate call acceptance by the called party.
CONNECT ACKNOWLEDGE: The message is sent by the network to the called party
to indicate the user has been awarded the call. It may also be sent by the calling
party to the network to allow symmetrical call control procedures.

DISCONNECT: The message is sent by the user to request the network to clear an
end-to-end connection or is sent by the network to indicate that the end-to-end
connection is cleared.

Document fax machine can be divided into the following categories according to the
different transmission speed and modulation mode:

Double-sideband modulation technology, transmission speed for each page (16) is


about 6 minutes, known as the first machine;

The bandwidth compression technology, transmission speed for per page is about 3
minutes, known as the second machine;
Less source redundancy digital processing technology, transmission speed for per
page is about 1 minute, referred to as the third machine;

The fax machine that can combine with the computer, store information and near
real-time transmission speed, classified as fourth machine.

The most common example is a voice band modem that turns the digital data of a
personal computer into modulated electrical signals in the voice frequency range of a
telephone channel. These signals can be transmitted over telephone lines and

demodulated by another modem at the receiver side to recover the digital data.

1. AB;
2. ACD.

VoIP: Voice over Internet Protocol, A value-added service technology for IP calls. The VoIP
service is a new IP telecom service. It can run on fixed and mobile networks and support
flexible access points. Fees for VoIP subscribers are relatively low. Calls between VoIP

subscribers who belong to the same carrier are free of charge.

Toll-bypass: Long distance voice calls across the WAN instead of the PSTN enables you to
reduce the overall telephony expenditure.

An analog or analogue signal is any continuous signal for which the time varying feature
(variable) of the signal is a representation of some other time varying quantity.
A digital signal is a physical signal that is a representation of a sequence of discrete values
(a quantified discrete-time signal).

Sampling is the conversion of a continuous analogue signal to a discrete signal. A common


example is the conversion of a sound wave (a continuous signal) into a sequence of
samples (a discrete-time signal).
Quantization is the process of converting each analog sample value into a discrete value
that can be assigned a unique digital code word.
Coding is the process by which information from a source is converted into symbols to be
communicated.
Decoding is the reverse process, converting these code symbols back into information
understandable by a receiver.

The requirement of voice compression:

Under a certain code rate, voice quality should be as high as possible.

Decoding time delay is small, the total delay is generally not more than 65 ms.

Code rate fits with the common voice channel transmission, generally the coding
rate is within 16-2 KB/s.

Decoding algorithm complexity cannot be too big to suitable for large scale
integrated circuit implementation.
Better toughness and better anti-error rate performance.

The advantages of speech coding

Improve the quality of calls (digital + channel coding error correction)

Improve the spectrum utilization efficiency (low bit rate coding)

Improve the system capacity (low bit rate, voice activation technology)

Speech coding classification:

Waveform coding

Parametric coding (source code

Transform the time domain analog voice waveform signal into digital voice
signals after sampling, quantization and coding.
Based on the mechanism of human language pronunciation, find and code the
speech characteristic parameters

Hybrid coding

Hybrid coding combines the advantages of waveform coding and parametric


coding.

Common voice codec technology contain:

PCM(Pulse Code Modulation):Pulse Code Modulation; the PSTN classic voice


encoding; is the standard measure of voice quality; 8K sampling bandwidth of 64K;

ADPCM(Adaptive Differential Pulse Code Modulation):adaptive differential coding


methods; usually applied to the limited bandwidth; voice quality is very close to the
PCM, FAX can be supported under certain restrictions, a bandwidth of 32Kbps;

CELP(Code Excited Linear Predictor)

LD-CELP(Low-Delay Code Excited Linear Predictor):low-delay Code Excited Linear


Predictive Coding approach; is often said that G728; voice quality and ADPCM is
similar or even better; bandwidth of 16Kbps; does not support FAX and the modem;
625us one, you need 10-bit information (codebook index value);

CS-ACELP(Conjugate Structure Algebraic Code Excited Linear Predictor):Conjugate


structure generation digital linear predictive coding; is our most popular G729; close
of ADPCM voice quality; bandwidth of 8 Kbps, a 10ms per frame is 10 bytes;

G.729

Commonly used include the following criteria: G729, G729A, G729B;

G729 is 8kbps a low-rate codec

G729A is an appendix of G729, G729 complexity of the algorithm can be


simplified.

G729B is a supplement for the G729 silence detection technology.

G729 the number of frames, each frame length is 10ms, TMG is the default

support 20ms, 30ms, 160ms; TMG VSU modify the code, can support 10ms,
each package contains a.

G729 payload length, each frame is 10 bytes = 8kbps * 10ms / 8

Long when the recommended package, 20ms, bandwidth 34.4kbps (not using silence

detection technology).

Commonly G729A 8kbps at 20ms package.

The RTP payload type value is 18.

Packet voice needs intensive operations (to complete a large amount of computational
processing operations in a short period of time) and special treatment. The general
microprocessor is not competent for these operations.
DSP is used in voice compression, voice activation detection, echo cancellation, delay
treatment and clock synchronization.
Usually a DSP can support multiple voice channel. Number of voice channel depends on
many factors, one of the main factor is the DSP chip can handle how many MIPS (MIPS,

millions of instructions per second). Another key factor is the memory in DSP. The more
memory, the higher operation efficiency, but the increasing of the on-chip memory can
reduce MIPS values. Therefore, the balance between memory and the MIPS must be
considered. In general, the manufacturer will configure large memory to achieve support
from 1 to 8 port (interface).

IP phones use the digitalized transmission technology in packets based on the IP


technology. The basic principles are as follows:

Compress and encode voice data according to the voice compression algorithm.

Package the voice data based on a certain protocol such as the IP protocol.

Send data packets to the recipient through the IP network.

Decode and decompress voice packets after collecting the voice packets to restore

the voice packets to the original voice signals.

The feature of IP phone

Rapid deployment, unified management.

High-quality voice.

Provides various business functions.

Simple and Intuitive Operation.

A Full Range of Mobile office

Meet the individual needs of staffs.

IP PBX integrates voice communication into enterprise data network through the VoIP
technology, thereby be able to set up a unified voice and data network for connecting
offices all over the world.
IP PBX integrates the call control, media gateway, access control, resource allocation,
protocol processing, routing, authentication, billing function etc.
It can be deployed on a hybrid network with IP phones, remote analog phones, and soft
phones, providing high-quality voice and value-added services for enterprises customers,

such as government departments, electric power companies, and financial companies.

Traditional PBX (TDM PBX)

TDM PBX was commonly used for cost saving internal calls before pure IP PBX and
hybrid IP PBX. Capacity is ranging from dozens of lines to thousands of lines to
access analog users by subscriber board. When more users, user accessing can take
up a very large equipment room space.

TDM PBX only provides basic voice business functions, and can not provide video
and all kinds of business services based on IP, which is used in some enterprise that

business requirements are not very high. Traditional PBX uses the special technique,
lacks of openness and standard, expand business difficulty.

IP PBX

The call server, media gateway, signaling gateway, relay gateway, terminals and

others are independent, communication between them through TCP/IP, to provide


users with VoIP business, pure IP PBX call server based on PC server platform (such
as IBM, DELL server), using the Windows or Linux operating system.

Business functions mainly through IP phone, IAD and other access devices to provide

users with voice in LAN, the network is flexible, expandable, but the stability of the
pure IP PBX based on server architecture may not meet specified requirements, tend
to produce puzzling collapse problem, the solution to this problem is commonly to
use two pure IP PBX to work, and improve reliability.

The advantages of IP PBX compared to TDM PBX

IP PBX supports remote/centralized configuration management based on IP


network, convenient maintenance, timely grasp the system status and fault
Doesn't need another phone lines, can remote access to the user without regional

restrictions

Users can use the configured IP telephone or IAD through Ethernet interface in the
office
Can keep their existed phone numbers
Convenient roaming - due to its characteristics of SIP protocol can transfer the
calling to anywhere in the world
Reduces the cost of internal communication because of using the enterprise Intranet
or private network

The SIP standard dispense with expensive dedicated telephone

Can enlarge shrinks scale - based on the expansion of software license

The impact factors from IP Network:

Packet Loss: If you want to get high-quality voice, let packet loss less than 1 percent;

Latency: ITU-T G.114 recommendation specifies that for good voice quality, no more
than 150 ms of one-way, end-to-end delay should occur.

DefineThe time takes for speech to exit the speaker's mouth and reach the
listener's ear.

Type
Propagation Delay: The delay used for electrons travel through
copper or fiber from calling party to called party.

Handling Delay: Devices that forward the frame through the


network cause handling delay.
Jitter: Jitter is defined as a variation in the delay of received packets.

The impact factors from codec:

Analog to digital conversion

Digital to analog conversion

The voice compression algorithm voice quality reduction

Mean opinion score (MOS) is a method of determining voice quality. Listeners hear a
phrase read over a voice network which uses a special codec and rate the quality of the
audio on a scale of 1 to 5, where 1 is the worst and 5 is the best.

Silence Suppression

In traditional telephony networks, a dedicated bandwidth for each voice stream provides
voice with a guaranteed bandwidth across the network. Because bandwidth is guaranteed
in the time division multiplexing (TDM) environment, there is no variable delay (jitter).

Configuring voice in a data network requires network services with low delay, minimal
jitter, and minimal packet loss. Bandwidth requirements must be properly calculated based
on the codec that is used and the number of concurrent connections. QoS must be
configured to minimize jitter and loss of voice packets. The PSTN provides 99.999 percent
availability, and to match this high availability of the PSTN, the IP network must be
designed with redundancy and failover mechanisms.
Issue

Solution

Packet loss

Design the network to minimize congestion


Prioritize voice packets
Use codecs to minimize small amounts of packet loss

Latency

Jitter

Use jitter buffers

Bandwidth

Calculate bandwidth requirements, including voice


payload, overhead, and data

Increase bandwidth
Choose a different codec type
Fragment data packets
Prioritize voice packets

Flow classification is to classify the packets that enters the DiffServ domain, so that can be
appropriately treated in the network. The main purpose of traffic flow classification is to
let other applications system that handles this message or device know the category of the

message, and to implement some agreed processing based on this category .

Traffic policing is the process of monitoring the network traffic for compliance with a
traffic contract and taking steps to enforce that contract. Traffic sources which are aware
of a traffic contract may apply traffic shaping to ensure their output stays within the

contract and is thus not discarded. Traffic exceeding a traffic contract may be discarded
immediately, marked as non-compliant, or left as-is, depending on administrative policy
and the characteristics of the excess traffic.

The token bucket is an algorithm used in packet switched computer networks and

telecommunications networks to check that data transmissions conform to defined limits


on bandwidth and burstiness (a measure of the unevenness or variations in the traffic
flow).

The token bucket algorithm is based on an analogy of a fixed capacity bucket into which
tokens, normally representing a unit of bytes or a single packet of predetermined size, are
added at a fixed rate. When a packet is to be checked for conformance to the defined
limits, the bucket is inspected to see if it contains sufficient tokens at that time. If so, the
appropriate number of tokens, e.g. equivalent to the length of the packet in bytes, are
removed ("cashed in"), and the packet is passed, e.g., for transmission. If there are

insufficient tokens in the bucket the packet does not conform and the contents of the
bucket are not changed.

Traffic shaping (also known as "packet shaping") is a computer network traffic


management technique which delays some or all datagram to bring them into compliance
with a desired traffic profile. Traffic shaping is a form of rate limiting.
Traffic shaping is used to optimize or guarantee performance, improve latency, and/or
increase usable bandwidth for some kinds of packets by delaying other kinds.
A conforming flow can thus contain traffic with an average rate up to the rate at which
tokens are added to the bucket, and have a burstiness determined by the depth of the

bucket. This burstiness may be expressed in terms of either a jitter tolerance, i.e. how
much sooner a packet might conform (e.g. arrive or be transmitted) than would be
expected from the limit on the average rate, or a burst tolerance or maximum burst size,
i.e. how much more than the average level of traffic might conform in some finite period.

The token bucket can be used in either traffic shaping or traffic policing. In traffic policing,
nonconforming packets may be discarded (dropped) or may be reduced in priority (for
downstream traffic management functions to drop if there is congestion). In traffic
shaping, packets are delayed until they conform. Traffic policing and traffic shaping are
commonly used to protect the network against excess or excessively bursty traffic.
Traffic shaping is commonly used in the network interfaces in hosts to prevent
transmissions being discarded by traffic management functions in the network.

If a new packet comes when packets with a lower priority are being sent, the new packet
will be sent immediately while packets with the lower priority stop to be sent. This enables
the packets related to the core services (such as VoIP) to be handled earlier. The packets

related to ordinary services (such as email) are handled when the core services are all
handled and the network is idle. By doing so, key services are handled earlier and network
resources are also fully used.

1. A;
2. D.

3. BD.

MGCP: Media Gateway Control Protocol

MEGACO: Media Gateway Control

H.248 protocol was the result of collaboration of the MEGACO working group of the
Internet Engineering Task Force (IETF) and International Telecommunication Union
Telecommunication Study Group 16. We called it H.248 in ITU-T, and called it MEGACO in
IETF.

H.323 is a suite of protocols defined by the ITU for multimedia conferences over LANs.
TheH.323 protocol was designed by the ITU-T and initially approved in February 1996. It
was developed as a protocol that provides IP networks with traditional telephony

functionality. Today, H.323 is the most widely deployed standards-based voice and
videoconferencing standard for packet-switched networks.

H.323 is a standard for communication protocols from the International


Telecommunications Union Telecommunication Standardization Sector(ITU-T); Version 4 is

the current version. H.323 was created to provide multimedia communication across a
packet network. The protocol can handle video and data, in addition to audio. H.323
interoperates well with the public switched telephone network (PSTN) in translating
between call setup and control signals used in an IP network and those used in a
switched-circuit network. H.323 is widely used in video conferencing deployments.

Notice: Gateway, Gatekeeper and MCU can be in a single device.


The terminals provide the real-time and bi-direction communication in the packet-switched
networks, and all the terminals compulsory support voice communication and optionally
support video and data applications.

The terminals in H.323 System contains: Telephones, Video phones, Voicemail Systems,
Soft phones, (e.g., NetMeeting ).
The gateway is an option component in H.323 system, and it provides the translation
function between different types of voice networks.

Provide interoperability between different telecommunications systems

Gateways are logically two endpoints, one in each telecom domain

Encoding, protocol, and call control mappings occur in the gateways between the
two endpoints
H.323 Proxy: a special null gateway, with two H.323 endpoints

The gatekeeper is an optional component, and it provides pre-call and call-level control
services to H.323 endpoints. Gatekeepers are logically separated from the other network
elements in H.323 environments.
The Gatekeeper, which is optional in an H.323 system, provides call control services to the

H.323 endpoints.

Address translation

Admission and access control of endpoints

Bandwidth management

MCU Typically consists of multi-point controller (MC) and multi-point processor (MP)

MC - handles control and signaling for conference support


MP - receives streams from endpoints, processes them, and returns them to the
endpoints in the conference

The protocols specified by H.323 include the following:

H.225 call signaling: H.225 call signaling is used to establish a connection


between two H.323 endpoints. This connection is achieved by exchanging H.225
protocol messages on the call-signaling channel. The call-signaling channel is
opened between two H.323 endpoints or between an endpoint and the gatekeeper.

H.225 Registration, Admission, and Status: Registration, Admission, and Status


(RAS) is the protocol between endpoints (terminals and gateways) and gatekeepers.
The RAS is used to perform registration, admission control, bandwidth changes, and
status and disengage procedures between endpoints and gatekeepers. A RAS
channel is used to exchange RAS messages. This signaling channel is opened
between an endpoint and a gatekeeper prior to the establishment of any other

channels.

H.245 control signaling: H.245 control signaling is used to exchange end-to-end


control messages governing the operation of the H.323 endpoint. These control
messages carry information related to the following:

Capabilities exchange

Opening and closing of logical channels used to carry media streams

Flow-control messages

General commands and indications

Audio codec: An audio codec encodes the audio signal from the microphone for
transmission on the transmitting H.323 terminal and decodes the received audio
code that is sent to the speaker on the receiving H.323 terminal. Because audio is
the minimum service provided by the H.323 standard, all H.323 terminals must have
at least one audio codec support, as specified in the ITU-T G.711 recommendation

(audio coding at 64 kb/s). Additional audio codec recommendations such as G.722


(64, 56, and 48 kb/s), G.723.1 (5.3 and 6.3 kb/s), G.728 (16 kb/s), and G.729 (8
kb/s) may also be supported.

Video codec: A video codec encodes video from the camera for transmission on the

transmitting H.323 terminal and decodes the received video code that is sent to the
video display on the receiving H.323 terminal. Because H.323 specifies support of
video as optional, the support of video codec is optional as well. However, any
H.323 terminal providing video communications must support video encoding and
decoding as specified in the ITU-T H.261 recommendation.

The flow shows that the call signaling between the two endpoints communicate directly.

Many IP applications require an exchange of data between associated participants.


Implementation of these applications is complicated by the practices of participants,
because end users may move between endpoint, may be addressable in multiple ways,

sometimes concurrently, and may communicate over several different media. Many
protocols have been authored to carry various forms of real-time multimedia session data
such as voice, video, or text messages.

SIP is an application layer control protocol used to establish, modify and terminate

multimedia sessions via pre-defined signaling message exchanges. It is independent of


underlying transport protocols and the type of session that is being established. It enables
IP endpoints to discover one another and to agree on a characterization of a session that
they would like to share.

RFC 3261(that obsoletes RFC 2543) defines SIP version 2 as an application layer signaling
protocol that defines initiation, modification and termination of interactive, multimedia
communication sessions between users.

The basic functions of SIP include location of an end point, signal of a desire to
communicate, negotiation of session parameters to establish the session, and teardown of

the session once established.

The SIP is an application-layer control protocol

Used to create, modify and terminate sessions with one or more participants.

SIP can also be used to invite participants to already existing sessions, such as
multicast conferences.
Media can be added to (and removed from ) an existing sessions.

Session Description Protocol (SDP) is an application layer control protocol for describing

multimedia sessions. SDP is a text-based protocol. During a SIP call, SDP is used to
negotiate the media type, coding scheme, and address for establishing a session.

Real-Time Transport Protocol (RTP)RTP provides E2E services for real-time data such as
audio and video, and provides a means for checking transmission quality for two
communicating parities.

SIP message classification

Request message

Response message

Response messages are responses to request messages and indicate the success or failure
status of calls. Different response states are distinguished using status codes. A status code
consists of three digits. The first digit indicates the response type, and the other two digits
provide detailed description about the response.
Status
Codes

Response
Message

Description

1xx

Information
response

Indicates that a request message has been received and is


being processed.

2xx

Success
response

Indicates that a request is received, processed, and


successfully accepted.

3xx

Redirection
response

Indicates that a further operation is required to complete a


request.

4xx

Customer
error

Indicates a request message contains a syntax error or the


SIP server cannot process the request message.

5xx

Server error

Indicates the SIP server is faulty and cannot process


messages correctly.

6xx

Global fault

Indicates the a request cannot be fulfilled on any SIP servers.

MGCP is a master/slave protocol that allows a call control device such as MGC(Media
Gateway Controller, also known as Call Agent) to take control of a specific port on a
MG(Media Gateway).
MGCP messages are carried over UDP(User Datagram Protocol). Because UDP does not
guarantee message delivery, messages are retransmitted, if needed.

MGCP is short of description capability, and is limited to big MGW. For big MGW, H.248
protocol is suggested.

MGCP message transaction is born on UDP package over broadband IP network, while
H.248 could be born on UDP/TCP/SCTP.

Compared with the MGCP and H.248 has the following advantages:

Support more types of access technology and mobility of the terminal

Support a large-scale network applications and protocols expand, stronger flexibility

In a VoIP network, the actual voice data (conversations) are transported across the
transmission media using Real-Time Transport Protocol (RTP).RTP defines a standardized
packet format for delivering audio and video over the Internet.
RTP defines a standardized packet format for delivering audio and video over the Internet.
It was developed by the Audio/Video Transport Working Group of the IETF and first
published in 1996 as RFC 1889, which was made obsolete in 2003 by RFC 3550.
Applications that use RTP are less sensitive to packet loss but are typically very sensitive to

delays, so UDP is a better choice than TCP for such applications.

RTP typically runs on top of UDP so that it can use the multiplexing and checksum services
of that protocol. RTP does not have a standard TCP or UDP port on which it
communicates. The only standard that it obeys is that UDP communications are done via

an even port, and the next higher odd port is used for RTCP communications. Although
there are no standards assigned, RTP is generally configured to use ports 16384 to 32767.

RTP provides end-to-end network transport functions that are intended for applications
transmitting real-time requirements, such as audio and video. Those functions include
payload type identification, sequence numbering, time stamping, and delivery monitoring.
RTP is a critical component of VoIP because it enables the destination device to reorder
and retime the voice packets before they are played out to the user. An RTP header
contains a time stamp and sequence number, which allows the receiving device to buffer
and to remove jitter and latency by synchronizing the packets to play back a continuous

stream of sound. RTP uses sequence numbers to order the packets only. RTP does not
request retransmission if a packet is lost.

Datagram protocols, such as UDP, send the media stream as a series of small packets. This
is simple and efficient; however, packets can be lost or corrupted in transit. Depending on
the protocol and the extent of the loss, the client may be able to recover the data with

error correction techniques, may interpolate over the missing data, or may suffer a data
dropout. RTP and the RTCP were specifically designed to stream media over networks.
They are both built on top of UDP.

RTCP is a sister protocol of the RTP. It was first defined in RFC 1889, which was replaced

by RFC 3550. RTP provides out-of-band control information for an RTP flow. It works
along side RTP in the delivery and packaging of multimedia data, but it does not transport
any data itself. Although it is used periodically to transmit control packets to participants in
a streaming multimedia session, the primary function of RTCP is to provide feedback on
the quality of service being provided by RTP.

RTCP is used for QoS reporting. It gathers statistics on a media connection and information
such as bytes sent, packets sent, lost packets, jitter, feedback, and round-trip delay.
Applications use this information to increase the quality of service by perhaps limiting flow
or using a low-compression codec instead of a high-compression codec.

DTMF signal in telephones has two purposes: one is used for DTMF to dial, to control the
switch to connect the called user phone; Second is used to remote control the various
actions of telephones through DTMF signal, such as broadcasting messages, voice mail,

etc., and can be done to remote control home appliances equipment by additional circuit
such as open or close functions. The former solves the DTMF signals sending and coding,
the latter solves the DTMF signals receiving and decoding.

The DTMF keypad is laid out in a 44 matrix, with each row representing a low

frequency, and each column representing a high frequency.

The essential difference between in-band and out-of-band is whether the DTMF is
transmitted in the media stream.
Carrying DTMF signal in INFO of SIP signaling.

This method is to use the INFO of the SIP signaling to clearly define DTMF signal.
Main defects is the SIP control signaling and media transport (RTP) is separate, it is
easy to cause DTMF signal and the media package are not synchronized. Simple, for
example, in the Voice Mail applications, the users input a DTMF signal according to

the prompt, then leave messages. Servers began to save the user's message after
receiving the DTMF signal. The DTMF signal is transmitted by SIP signaling, and the
media stream is transmitted by RTP , the users may receive the RTP packets firstly,
however, the INFO messages of the DTMF signal delay, lead to Server without
saving the user's voice mail until you receive the INFO message.

Transparent transmission

Transparency is to transmit directly RTP packets of DTMF audio digital signal without
any processing in the IP network. It should transmit possibly together with the user
voice media stream. Program wants to know which package has DTMF signal and

which kind of the DTMF signal, must check media streaming data in each RTP
packet real-time, analyzing the frequency domain. Because of the influence of the
network packet loss, sometimes resulting in DTMF signal loss, and the mixed DTMF
tones in the voice packets prone to bias, making signal distortion.

RFC 2833

RFC 2833 DTMF is to combine the DTMF signal into a packet according to the rules
and format, are identified with special RTP packets, can be known that the package
is DTMF package in the RTP packet head domain, and know the content of DTMF
signal that RFC2833 has defined. This approach has the advantage of tolerant of
packet loss is high and recognition error rate is low. So the RFC2833 way widely is
used at present.

Transparent fax: Fax signals are transmitted transparently as G.711 packets. G.711 faxes
feature low delay and simple implementation, but they occupy a high bandwidth (fixed at
64 kbit/s) and are easily affected by network conditions. Therefore, G.711 faxes are

recommended on a good network condition and not recommended when network jitter
or packet loss frequently occur. G.711 faxes are applicable to high-speed and low-speed
faxes.

T.38 fax: T.30 fax signals are converted to T.38 packets for transmission on a packet

switching network. T.38 faxes occupy a low bandwidth, provide high reliability with
redundant frames and forward error checking (FEC), and are slightly affected by the
network condition. However, the implementation is complicated. T.38 faxes are applicable
only to low-speed fax services due to delay generated by the packet switching network.

Transparent fax: Fax signals are transmitted transparently as G.711 packets. G.711 faxes
feature low delay and simple implementation, but they occupy a high bandwidth (fixed at
64 kbit/s) and are easily affected by network conditions. Therefore, G.711 faxes are

recommended on a good network condition and not recommended when network jitter
or packet loss frequently occur. G.711 faxes are applicable to high-speed and low-speed
faxes.

T.38 fax: T.30 fax signals are converted to T.38 packets for transmission on a packet
switching network. T.38 faxes occupy a low bandwidth, provide high reliability with
redundant frames and forward error checking (FEC), and are slightly affected by the

network condition. However, the implementation is complicated. T.38 faxes are applicable
only to low-speed fax services due to delay generated by the packet switching network.

1. D;
2. AC.

IPT: IP Telephon;
The IP Telephony (IPT) system provides efficient, reliable voice communications services and also a
variety of advanced, quality, and convenient supplementary voice services for private network users.
The IPT system employs multiple new technologies to continuously reduce enterprises' operation
costs and increase their operation profits.
The IPT system has the following highlights:

Aligned with actual services to ensure a highly available communications system.

Ensures high reliability (carrier-class) and stability.

Employs state-of-the-art communications technologies to maintain its leading position in the


private network communications field.

Adapts to rapid service changes, as it features high scalability, flexible setting modifications,
convenient operation and maintenance, and rapid system deployment.
Fully uses existing system resources and the telephone network, saving long-term operation
costs.
Standard and open. It can directly or indirectly connect to carrier networks, third-party PBXs,
billing systems, recording systems, and directory systems.

Provides a unified GUI-based management tool for convenient, comprehensive system


management.

With more than two decades of accumulated experience in the IP communications field and deep
understanding of customer requirements, Huawei transplants its telecom products and advanced
design concepts to the voice communication architecture and launches a brand new IP-based voice
communications solution, which provides a reliable, efficient, and future-proof communications
platform.

eSpace IPT solution uses a hierarchical system architecture that features high convergence,
security, and openness.
Terminal

Delivers consistent user experience from anywhere, at any time, on multiple devices
including IP phones, analog phones, fax machines, smart phones, and tablets. IP
phones provide diverse easy-to-use functions and high-quality voice call services,
including speed dial, call transfer, automatic callback, multi-party call, corporate

directory, and voicemail.

An analog phone is a traditional desktop phone that converts vibrations into


electrical frequency signals for transmission. It can connect to the system through a
subscriber board on eSpace U1911/U1960/U1981 or through an eSpace IAD.

An eSpace IAD is a voice over IP (VoIP) and fax over IP (FoIP) media access gateway.
It connects analog phones or fax machines to an IP network for providing efficient
and quality voice or fax services.

IP PBX

A unified gateway is deployed at the call management layer to achieve these


functions. The call management layer provides user registration, call routing, call
control, and media control functions.

eSpace U1911/U1960/U1980/U1981 uses a highly-integrated SIP softswitch as its


core and supports both narrowband and broadband services.

Application

The service application layer provides a variety of services, including the Console,
Unified Message Service, Voice Mailbox, Voice Conference, Call Detail Record (CDR),
Network management and Recording services.

The eSight system is a new-generation comprehensive operation and maintenance


solution developed by Huawei for the network infrastructure, unified
communications, telepresence conferencing, video surveillance, and data center in
enterprises. eSight supports unified monitoring and configuration management over
devices of various types and from various vendors, monitors and analyzes network

and service quality, and delivers unified management over and association analysis
among enterprise resources, services, and users. Meanwhile, eSight offers a flexible
and open platform for enterprises to customize software development and build an
intelligent management system tailored to individual needs.

The eSpace U1900 series unified gateways are switches used in Huawei unified
communications solutions. The eSpace U1900 series unified gateways provide professional
voice over IP (VoIP) servies that meet communication requirements of various enterprises.
The unified gateway uses a highly-integrated SIP softswitch as the core and supports both
narrowband and broadband services, helping effectively improve communication efficiency
and reducing operating expense (OPEX).

The unified gateway can connect to analog phones and IP phones at the same time:

Connect to local analog phones directly. (U1980 cannot directly connect to local
analog phones.)

Use IP bearer networks to connect to analog phones through eSpace IADs.

Use IP bearer networks to connect to IP phones.

The unified gateway can connect to PSTN networks or dedicated network voice switches
through digital, analog, and broadband SIP trunks.

Huawei eSpace Integrated Access Device (IAD) is used in Huawei IP telephony and Unified
Communications (UC) solution to connect traditional analog users to IP telephony
networks.
By reusing legacy analog endpoints, eSpace IAD helps reduce initial investments in building
IP telephony networks. eSpace IAD also reduces analog lines' routing and maintenance
costs and enterprises' Total Cost of Ownership (TCO) by fully leveraging the transmission
resources of IP networks.

The full series of eSpace IAD products support 2 to 224 analog user channels to fit the
access scenarios of different user capacities.

Video IP Phone:

eSpace 8950 is a smart, sleek video phone that integrates voice, video, and Unified
Communications (UC) applications. It delivers more engaging communication and
collaboration, with unmatched security, High Definition (HD) audio and video,
simplicity, and smooth operation.

Audio IP Phone:

eSpace 7900 series IP phones are new-generation IP phones with superb


performance. The innovative design delivers an intuitive, easy-to-use experience,
enabling users to benefit from highly efficient communications

eSpace 6805 is a cost-effective starter-edition IP phone that provides rich functions


and is easy to use.

The One Number Link You (ONLY) service refers to a service in which a user's
multiple terminals share the same number and supplementary services.
The shared number is the number of an intra-office terminal, and is called the
ONLY number. The intra-office terminal is the primary terminal and can be specified
by the user. When other users dial the ONLY number of the user, the terminals of
the user ring according to the preset rules, and the user can select one terminal to
answer the phone.
A maximum of eight terminals can be configured (including the primary terminal)
that are bound to the ONLY number.
Advantages:

It is easy to remember one number and use this number to contact a


person.

You can answer important calls anytime and anywhere, which improves
the working efficiency.

You can give only the office number but keep your mobile phone
number and home telephone number private.

Enterprises can protect customer resources from being lost due to


employee transfer.

Flexible modes can be used to convene a conference, such as system convening (the
system automatically convenes participants at a specified time), chairman convening, and
active access.
Web-based conference management is supported. Operations such as conference
reservation, announcement, participant attribute modification, and conference delay
setting can be performed on Web pages.
It is recommended to use the conference call service according to the following

specification:

A maximum of 960 conferences with a maximum of 60 parties in each conference


for the U1980. The total number of all parties in all conferences does not exceed
960.

A maximum of 360 conferences with a maximum of 60 parties in each conference


for the U1960. The total number of all parties in all conferences does not exceed
360.
A maximum of 12 conferences with a maximum of 12 parties in each conference for

the U1930. The total number of all parties in all conferences does not exceed 12.

A maximum of 20 conferences with a maximum of 60 parties in each conference for


the U1911. The total number of all parties in all conferences does not exceed 60.
A maximum of 6 conferences with a maximum of 6 parties in each conference for

the U1910. The total number of all parties in all conferences does not exceed 6.

Manager A has two secretaries: secretary B and secretary C who share two lines. The value
of Shared line mode is set to Single. The two shared lines allow two calls at a time.
When there is an incoming to manager A's number, secretaries B and C's phones ring at

the same time. The indicator of the shared line on manager A's phone blinks. One of the
secretaries answers the call and forwards the call to manager A. While manager A is
answering the call, another call is routed to manager A's number. Secretaries B and C's
phones ring at the same time. One secretary answers the new incoming call.

Shared line

When a user makes an incoming call, the shared line indicators on both the manager
and secretary's phones blink. The manager can answer the call directly. Alternatively,
the secretary can answer the call first, and call the manager through the private line

to ask the manager whether to answer the call. The incoming call to the shared line
is held, and the user hears the waiting tone.

The manager and secretary's phones display the same incoming call information and
shared line status such as ringing, talking, and idle. If the manager is using the
shared line for making a call, the shared line is displayed as occupied on the
secretary's phone. On the contrary, if the secretary is using the shared line for
making a call, the shared line is displayed as occupied on the manager's phone.

Private line

The incoming call can be switched between the manager and secretary for any
times. During the call with the user, the manager can press the private line key to
call the secretary and transfer the call to the secretary. The incoming call to the
shared line is held, and the user hears the waiting tone. When the secretary presses
the shared line key, the call is connected to the secretary, the manager hangs up,
and the user talks with the secretary.

The manager private line and secretary private line can be configured as a speed dial
number of each other. In this way, the manager and secretary can call each other
without dialing any numbers.

Each shared line supports only one call. A manager can have a maximum of six
shared lines, while a secretary can also have a maximum of six shared lines.
Each shared line requires two device IDs. One is the manager device ID, which is
configured for a line key on the manager's phone. The other is the secretary device
ID, which is configured for a line key on the secretary's phone. Six shared lines
require twelve device IDs in total. All shared line device IDs of the manager
correspond to the same SIP number.
Multiple shared lines can be implemented on multiple terminals. The number of
terminals subjects to the phone model (for example, the eSpace 7910 IP phone
supports a maximum of three registered accounts, and the eSpace 7950 IP phone
supports a maximum of six registered accounts). When the multiple shared lines of
the manager are configured on multiple IP phones, the simultaneous ringing service
can be used to implement private line calls. When a secretary calls the private line of

the manager, multiple IP phones of the manager ring simultaneously. The manager
can pick up any IP phone to answer the call, and other IP phones stop ringing.

The U1911, U1960, and U1981 unified gateways provide built-in Voice Mailbox Services
(VMS) to implement recording, storage, and retrieval of voice messages for voice users.
When users have unread messages, the system notifies the users of the unread message

for them to read the messages in time.

Voice messages are stored in the SD card of the MTU board. They can be backed up to the
NFS server (prepared by the customers themselves), and the backup voice messages can be
restored from the NFS server.

By default, voice messages are not encrypted, and they can be obtained by message
backup or directly reading the SD card. The voice messages can be encrypted as required.
Others cannot play the encrypted voice messages even if they have obtained them.
The voice mailbox feature requires the corresponding license, and the number of

concurrent messages is limited by the license.

The specifications of the built-in voice mailbox for each eSpace U1900 model:
Model

Maximum Number of Voice Mailbox Users

Maximum Number of Concurrent Messages

U1911

100

12

U1960

1000

30

U1981

1200

30

A contact center agent can log in to the SoftConsole using an account assigned by the
ConsoleServer. After the account is authenticated by the ConsoleServer, the agent can use
agent functions.
When a user dials the access code of a SoftConsole group, the unified gateway routes the
call to an idle agent in the group, who will then answer the call. Incoming calls are queued
if all agents in the group are busy. When one or more agents become available, the system
routes queued calls to agents following the "first-come, first-served" policy.

The recording system uses the latest digital electronics technologies and enables users to
record and query multiple channels of telephony conversation. The recording system is
widely used in governments, financial organizations, energy companies, hospitals,
educational institutions, and other sectors, which helps supervise service quality and
resolve disputes.
The recording system can monitor the recording status and record conversational
information, including the phone number, call time, and conversational content, in real
time.
The recording system consists of a recording module, a recording query and playback
module, a status monitoring module, a storage management module, and a user rights
management template.
Two modes of site recording are supported: automatic recording and on-demand
recording.

automatic recordingThe start and end of automatic recording are signalingcontrolled. That is, the recording system automatically starts recording when a
conversation is started between the specified users, and stops recording when the
conversation ends.
On-demand recording is controlled by users using a special recording button on
phones or a softkey. Recording can be manually stopped by pressing the recording
button or automatically stops when the conversation ends. On-demand recording
takes effect on a per call basis. Users need to press the recording button each time
they want to record a call.

In an enterprise, employees in the same field can be added to a hunt group, with the
group access code configured. When a user dials the group access code, the phones of
the sign-in group members ring simultaneously, cyclically, or sequentially according to the

configured routing policy. The hunt group service enables employees in the same field to
work together for more efficient request processing.

A user can be added to multiple hunt groups to handle different service requirements.

When a call is connected to a hunt group through the hunt group access code, personal

services (such as call forwarding and DND) for the hunt group members, if enabled, will be
unavailable.

Application Scenario

In an enterprise, employees in a department (for example, IT support department)


have been added to a hunt group with a unified access code, and simultaneous
ringing has been configured for the hunt group. During the office hours, all group

members sign in to the hunt group. When a user dials the access code of this hunt
group, the phones of all employees in this hunt group ring at the same time. After
an employee answers the call, the phones of the other employees stop ringing.

If sequential ringing is configured, the phone of the desired employee (whose phone
is expected to ring first according to the configured routing policy) rings first. If the
desired employee cannot answer the call, the call is diverted to the next expected
number.
When no employee in the hunt group answers the call, the call is forwarded to the
forward-to number configured for the hunt group access code. For example, the
hunt group has configured call transfer to the voice mailbox upon ringing timeout.
During the break, if a user dials the hunt group access code, no one answers the
call. When ringing times out, the system asks the user to leave a message. After the
user leaves a message, the MWIs on the phones of all members in the hunt group
are lit up. When the group members go back to work and find that the MWI is on, a
member presses the corresponding button on the phone, enters the password, and
listens to the message. Then, the MWIs on the other phones are turned off.

Sign-in, sign-out, and presence.When a user dials the access code of a hunt group, the
system allocates the call to group members who have signed in to the hunt group.

Dynamic sign-in and sign-out.Hunt group members need to dial Dynamic signin/sign-out prefix + Hunt group access code + # to sign in to or sign out of a hunt
group.
Static sign-in. Members of a hunt group sign in automatically and cannot sign out.
One-button sign-in, sign-out, and presence. Hunt group members can configure the
sign-in and sign-out programmable button on IP phones. Users can press the button
to quickly sign in and out. The indicator of the programmable button is on after
sign-in and off after sign-out.

Requiring a verification password for sign-in and sign-out. Users can configure a
verification password for sign-in and sign-out. A verification password contains 6
digits.

A paging group supports only one channel broadcast at the same time. If an initiator is
broadcasting, the other initiator cannot initiate broadcast.
The audio paging service has a higher priority than all services excluding DND. Users can
also configure the priority between audio paging and DND.

If the emergency call service is configured, users can associate an emergency call prefix
with a paging group. When an intra-office user (the user may not be in the paging
group) dials the emergency call prefix, the communication between the calling party and

the called party will be broadcast on the phones of members in the associated paging
group.

Audio paging service is one to many unidirectional voice broadcast service.

By default, all members in a paging group can be the initiator of voice broadcast; External

users cannot initiate a voice broadcast.

The service process is as follows:

An initiator dials the access code of a paging group, and speaks after hearing a
beep.

The voice broadcast players of all external paging parties in the paging group play
the voice broadcast, and the IP phones of all unicast and multicast parties respond
as follows:

If a recipient is idle or has not started a call after picking up the phone, the
speaker is automatically started and plays the voice broadcast after a beep.
If a recipient is in a call, the voice broadcast is played as a whisper in audio
mixing mode.

If a recipient has enabled the DND service, the system determines whether to
play the voice broadcast to the recipient based on the service priority
configured for the paging group.

The initiator stops speaking and hangs up. The voice broadcast is stopped for all

recipients.

Model

Maximum Maximum Number of Members in a Paging Group


Number
of Paging
Groups
Unicast party + External paging party + Associated
eSpace 2
U1910/
emergency call prefix: 2 (where the maximum number of
U1930
external paging parties is 1)
Multicast party: all SIP users
Unicast party + External paging party + Associated
eSpace 16
U1911
emergency call prefix: 30 (where the maximum number of
external paging parties is 8)
Multicast party: all SIP users
Unicast party + External paging party + Associated
eSpace 32
U1960
emergency call prefix: 30 (where the maximum number of
external paging parties is 8)
Multicast party: all SIP users
eSpace 64
Unicast party + Associated emergency call prefix: 30
U1980
NOTE: The eSpace U1980 does not support multicast parties
or external paging parties.
Unicast party + External paging party + Associated
eSpace 64
U1981
emergency call prefix: 30 (where the maximum number of
external paging parties is 8)
Multicast party: all SIP users

Maximum
Number of
External Paging
Parties Globally
1

A single-node network is suitable for an enterprise with no more than 20,000 users that
does not have branches or has branches that do not require local regeneration. In such a
network, the unified gateway and a variety of service servers are all deployed at the

enterprise's equipment room, and all terminal users register with the unified gateway. If
the enterprise has small-sized branches, eSpace IADs can be deployed at branches and
function as proxies for user registration to the unified gateway, or IP phones can be
deployed at branches and register with the unified gateway.

Networking Description

Analog phones and fax machines connect to the unified gateway through analog
phone cables. IP phones and SoftPhone connect to the unified gateway through IP
network.

The unified gateway connects to the PSTN through the trunk.


The unified gateway provides voice mailbox and voice conference services for
enterprise users.
By default, the unified gateway functions as an IP PBX and can provide basic and

supplementary voice services for enterprise users. The unified gateway can also
function as an access gateway and register with the IMS network through a SIP
trunk, allowing enterprise users to use services provided by the IMS network.

Only the U1911/U1960/U1981 supports the built-in voice mailbox function.

A centralized call management network can be used for an enterprise with no more than
20,000 users that has branches that require local regeneration. In such a network, the
unified gateway functioning as the central node and a variety of service servers are all

deployed at an enterprise's equipment room, and all terminal users register with the
central node in normal conditions. Unified gateways functioning as branch nodes are
deployed at the branches for local outgoing calls and local regeneration.

Networking Description

The unified gateway in the HQ is called the central node, which can be
U1911/U1960/U1980/U1981. The central node can be deployed in active/standby
mode, one being the active node and the other being the standby node.
The unified gateway gateway in a branch is called a local node, which can be

U1911/U1960/U1980/U1981.

The central node and local nodes connect to the PSTN through digital or analog
trunks.
The central and local nodes are connected through SIP trunks and use the heartbeat

mechanism to monitor the running status of the peer devices.

When the central node and local nodes are connected properly:

Users and services are configured and controlled in a unified manner on the
central node.

IP phones on local nodes are directly registered with the central node.

Analog phones on local nodes are registered with the central node through
local node proxies.
Local nodes synchronize SIP user data from the central node in real time.

When the central node is faulty or disconnected from local nodes:

The basic call function remains available for analog users on local nodes.

When detecting that the central node is faulty or disconnected, IP phones on

a local node automatically set the local node as the SIP server. The basic call
function remains available for SIP users on the local node.

When reconstructing or expanding the traditional PBX-based network, you can use the
U1900 to interconnect with the existing PBX to implement the smooth expansion and
provide new services. This scheme is the interconnection networking of the U1900 and

traditional PBX.

The scheme keeps the traditional PBX device and existing PBX subscribers. The networking
mode is that the traditional PBX device connects to the U1900.
Functioning as the IP phone gateway, the U1900 converts and bridges data for voice

services on the traditional PBX, and controls call connections through signaling.

The trunk resources of the PSTN are reserved for the traditional PBX. These resources
function as an alternative option of original PBX subscribers.
For the phone subscribers and terminals on the IP network, the U1900 functions as the

softswitch center, trunk gateway, and signaling gateway. It controls connections, supports
new services, manages user data, and manages charging for VoIP subscribers.

Through interconnections between multiple U1900s, a wide area VoIP network can be
created among business subscribers. In addition, multiple services and functions can be

created, such as, the group service, multi-party conference, call center, and instant
communication in wide area.

The subscribers of the traditional PBX can be transferred to the U1900 network in future.
The engineers can implement the transfer only by adding subscriber boxes or IADs and

transferring the traditional PBX trunk resources to the U1900.

U1900 uses a highly-integrated SIP softswitch as its core and supports both narrowband
and broadband services. Terminals are connected to U1900 in different ways:

Local analog phones: directly connected

Analog phones: connected through the IAD over the IP network

IP phones and console client: connected over the IP network

As a core component of the solution, the unified gateway provides the following

functions:

P2P audio and video calling, conference calling, and diverse supplementary
telephony services.

Active/standby redundancy, remote disaster recovery, and local regeneration

(active/standby redundancy and remote disaster recovery are alternative).

Connection to the PSTN or PBXs in a private network through digital, analog, or SIP
trunks.

U1900 uses a highly-integrated SIP softswitch as the core and supports both narrowband
and broadband services, helping effectively improve communication efficiency and
reducing operating expense (OPEX).

U1900 can connect to analog phones and IP phones at the same time: Connect to
local analog phones directly. (U1980 cannot directly connect to local analog phones)

Use IP bearer networks to connect to analog phones through IAD.

Use IP bearer networks to connect to IP phones.

U1900 can connect to PSTN networks or dedicated network voice switches through
digital, analog, and broadband SIP trunks.

Slots are located on the front panel of the shelf. The U1911 provides one main control
board slot, three interface board slots, one power supply sockets, and one fan tray
assembly slot.

Slots 0 to 2 are service board slots, which are used to install the MTU, ASI, OSU or
BTU boards.
Slot 3 is the main control board slot, which is used to install the SCU board.

The number of U1911 interface boards is determined by the system capacity. Filler panels
must be inserted into blank slots.

To ensure that basic services are functioning properly, at least one SCU board and one
MTU board must be configured for the U1911.

Slots are located on the front panel of the shelf. The U1960 provides one main control
board slot, seven interface board slots, two power supply sockets, and one fan tray
assembly slot.

Slots 0 to 6 are service board slots, which are used to install the MTU, ASI, OSU or
BTU boards.
Slot 7 is the main control board slot, which is used to install the SCU board.

The number of U1960 interface boards is determined by the system capacity. Filler panels
must be inserted into blank slots.

To ensure that basic services are functioning properly, at least one SCU board and one
MTU board must be configured for the U1960.

Slots are located on the front panel of the shelf. The U1981 provides two main control
board slot, six interface board slots, two power supply sockets, and one fan tray assembly
slot.

Slots 0 to 2 and 4 to 6 are service board slots, which are used to install the MTU,
ASI, OSU or BTU boards.
Slot 3 and 7 are the main control board slot, which is used to install the SCU board.

When only one SCU board is installed, the system runs in single-node control mode. When
two SCU boards are installed, the system runs in active/standby control mode and has a
higher reliability.

The number of U1981 interface boards is determined by the system capacity. Filler panels
must be inserted into blank slots.
To ensure that basic services are functioning properly, at least one SCU board and one
MTU board must be configured for the U1981.

SCU: Service Control Unit.


SCU is the control board (a mandatory board) of the U1911/U1960/U1981. The SD card
delivered with an SCUB board contains the default announcements in 11 languages. On
the U1911/U1960/U1981 with an SCUB board, you can directly switch the announcement
language. The system extracts the voice package for the target language from the SD
card, overwrites the original voice package in the version software, and synchronizes the
voice package to all MTU boards.

The SCU board provides the following functions:

Provides the softswitch functions.

Processes media control protocols.

Supports L2 switching and TDM switching.

Panel Components
Panel
Identifier Description
Component
Ports
100/100 Service network port that connects the device to a LAN and
0BASE- functions as an IP service port.
TX
CONSOL Debugging serial port for configuring and commissioning the
E
device.
USB
Reserved for the service to extend interfaces when required
(not used currently).
AUDIO 3.5 mm audio input port for connecting external audio sources.
It is used for playing music on hold (MoH).
Indicators
PWR
Power indicator (green), which indicates the power status.
RUN
ALM
Button

RST

Running indicator (green), which indicates the running status of


the board.
Alarm indicator (red), which indicates the alarm status of the
board.
Press to restart the board.

To make data and transmission more secure and reliable, you are advised to use the dualnetwork-port or triple-network-port mode.
In dual-network-port mode triple-network port mode:

Network port 0 must be used for accessing the web management system, LMT
command tree, and CLI.

Network port 0, 1 or 2 can be used for accessing the web self-service system.

Only one network port can be used to connect to the Bill Server.

The MTU board provides four E1/T1 ports, each of which supports two trunks.

The MTU board is in common used on the U1911/1960/U1981.

Each U1911/U1960/U1981 must be equipped with at least one MTU board.


Panel
Identifier
Description
Component
Ports
E1/T1
Digital trunk port for connecting to the upper-level office (for
example, LE) through digital trunks. An E1/T1 port supports a
DB9 connector.
The MTU board provides two E1/T1 ports, each of which
supports two trunks.
CONSOLE This serial port is disabled.
Indicators
PWR
Power indicator (green), which indicates the power status.
RUN
Running indicator (green), which indicates the running status
of the board.
ALM
Alarm indicator (red), which indicates the alarm status of the
board.
LOS
Signal loss indicator (yellow), which indicates whether signals
are lost.
RFA
Remote alarm indicator (yellow), which indicates whether
peer devices are faulty.
Button
RST
Press to restart the board.
SD card slot None
SD card slot for housing an 16 GB SD card. The SD card is
used to store recording files of the built-in voice mailbox.
Jumper
None
A jumper fitting used during E1/T1 switching. When the T1
trunk is used, remove the jumper.

The BTU board is in common used on the U1911/1960/U1981.

One BRI trunk can provide 2 voice channel.

Panel Components
Panel
Identifier
Component
Ports

Indicators

Button

Description

BRI S/T

BRI trunk port, which supports an RJ-45 jack. It supports


one BRI trunk for two channels of voice services.
Each BTU board provides four BRI ports.

CONSOLE

This serial port is disabled.

PWR

Power indicator (green), which indicates the power status.

RUN

Running indicator (green), which indicates the running


status of the board.

ALM

Alarm indicator (red), which indicates the alarm status of


the board.

BRI trunk port activity indicator (yellow), which is always off.

BRI trunk port connection indicator (green).


Blinking (4 Hz): The line is properly connected.
Off: The line is not properly connected.

RST

Press to restart the board.

The ASI board is used with U1981/U1960/U1911.

Panel Components
Panel
Identifier Description
Component
Ports

Indicators

Button

FXS 1-32

FXS ports for connecting the device to a maximum of 32


analog phones. The port supports a DB68B connector.

CONSOLE

This serial port is disabled.

PWR

Power indicator (green), which indicates the power status.

RUN

Running indicator (green), which indicates the running


status of the board.

ALM

Alarm indicator (red), which indicates the alarm status of the


board.

RST

Press to restart the board.

The OSU board is used with U1981/U1960/U1911.

Panel Components
Panel
Identifier
Component

Ports

Indicators

Buttons

Description

FXS 1-12

FXS ports for connecting the device to a maximum of 12


analog phones. The port supports a DB68B connector.

FXO 1-12

FXO ports for connecting the device to a maximum of 12


analog trunks. The port supports a DB68B connector.

CONSOLE

This serial port is disabled.

PWR

Power indicator (green), which indicates the power status.

RUN

Running indicator (green), which indicates the running


status of the board.

ALM

Alarm indicator (red), which indicates the alarm status of the


board.

RST

Press to restart the board.

U1981/U1960/U1911 share the same module.

The U1911 supports only one power module.

The U1960 and U1981 can be equipped with two power modules. By default, only one
power module is configured and is in slot 2 for the power supply. When two power
modules are configured, the following functions are supported:

Equalizing currents and backing up power supplies When working normally, each
power supply module shares the load with output current. When a power supply
module stops working, the other power supply module continues to work.

Supporting hot swapping You can insert a power supply module directly into a
vacant slot of the power distribution frame without powering off the
U1911/U1960/U1981. When a backup power supply module is installed, you can

remove a power supply module without affecting the running of the


U1911/U1960/U1981.

When two power supply modules are configured, you cannot configure both the AC and
DC power supply modules on an U1960/U1981.

Height6U(1U=44.5mm)

Indicators in the U1980 shelf


Indicator
Power
indicator
(green)
Running
status
indicator
(green)

Alarm
indicator
(red)

Identifier

Description

POWER

Indicates the power status.


On: The power supply is normal.
Off: There is no power supply.

RUN

Indicates the running status of the device.


On: The board is faulty.
Blinking (at 4 Hz): The board is writing data to the Flash
Memory when the system is running.
Blinking (at 0.5 Hz): The system runs normally.
Off: There is no power input, the board is faulty, or the
system is starting.

ALARM

Indicates the alarm status of the device.


Blinking (at 4 Hz): A critical alarm is generated.
Blinking (at 0.5 Hz): A major alarm is generated.
Blinking (at 0.25 Hz): A minor alarm is generated.
Off: No alarm is generated.

Slots 0-1 are SMCU slots used to install the SMCU boards.
Slots 2-9 are interface board slots used to install interface boards, SC1-MRS boards, these
boards can be used together.

SMCU

Provides the SoftSwitch functions.

Processes media control protocols.

Supports L3 switching and Time Division Multiplex (TDM) switching.

Supports hot swap.

Supports 1+1 hot backup. When the active board is faulty, the standby board

automatically replaces the active board.

MRS

The MRS provides the following functions: number allocation, digit collection, voice
playing, recording, voice conference, conversion from the TDM signaling to the VoIP
signaling, T.30 fax, supporting G.711, G.729 and iLBC. The SC1-MRS provides 256
channels for processing the media resources.

The MRS supports load sharing. When all the boards run normally, they share loads
evenly. When a board is faulty, other normal boards share the load to ensure the

normal running of the system.

DTU

The DTU board is a digital trunk interface board. It provides 4/8 E1/T1 ports and a
debugging port.

PDF: Power Distribution Frame.

The power switch is used to power on and power off the U1980.

The power module slots are used to install power modules. The numbers of the
three slots from left to right are 0, 1, and 2. If there is a vacant slot, insert a blank
filter panel into it. If there is no vacant slot, inserting a blank filter panel will
generate a power alarm.
The alarm tone mute button is used to mute the alarm tone when the equipment
gives out alarms.

The power supply module provides the following functions:

Equalizing currents and backing up power supplies By default, an U1980 has one
power supply module installed in slot 0. If an U1980 has two power supply modules
installed, it can equalize currents and back up power supplies. When working
normally, each power supply module shares the load by providing separate output
current. When a power supply module stops working, the other power supply
module continues to work.

Supporting hot swapping You can insert a power supply module directly into a
vacant slot of the power distribution frame without powering off the U1980. When
a backup power supply module is installed, you can remove a power supply module
without affecting the running of the U1980.

When two power supply modules are configured, you cannot configure both the AC
and DC power supply modules on an U1980.

The U1911/U1960/U1981 uses digital trunk cables, high-density user cables, and DC
power cables.

The U1911/U1960/U1981 uses digital trunk cables to connect various inter-office


trunk devices.

75-120-ohm transfer box converts 75-ohm unbalanced E1 signals to 120-ohm


balanced E1 signals for connection using E1 trunks.
Analog phones use high-density user cables to connect to FXS ports of the
U1911/U1960/U1981.

The SCU board (main control module) integrates functional modules such as the main
control module, narrowband switching module, security logic module and two broadband
switching modules.

The MTU board (media resource module + digital trunk module) contains the media
processing module (DSP + CPU), E1/T1 trunk module, and SD card storage module.

The ASI board (analog user module) provides analog user access.

The OSU board (analog user module + analog trunk module) provides analog user access
and analog trunk access.

The BTU board (digital trunk module) provides BRI trunk access.

The system is divided into the following planes based on the function:

System plane: includes the operating system, database, timer, memory management,
task management, and interface commissioning functions.
Driver plane: includes the RS232 serial port communication, network chip, hardware
chip, and FPGA logical drivers.
Protocol plane: processes data for SIP, AT0, POTS, PRI, R2, QSIG, and Q.921.

Transfer plane: includes the board and card management, inter-board communication,
and message forwarding functions.
Service control plane: includes the call control, connection management, resource
management, user management, SoftConsole management, registration management,
and CDR generation functions.
Maintenance and management (M&M) plane: includes the CLI, web management,
LMT, log/alarm processing, tracing, and license management functions.

Messages transmitted among planes are scheduled and encapsulated in a unified manner,

effectively reducing system complexity and further improving reliability.

Signaling
Function
/Protocol
SS7
Enables the communication between the unified gateway and switches
supporting the SS7 signaling and enables the unified gateway to access E1
trunks provided by the switches.
PRA
Enables the communication between the unified gateway and the switches
on the ISDN and enables the unified gateway to access E1/T1 trunks
provided by the switches on the ISDN.
BRI
Enables the communication between the unified gateway and the switches
on the ISDN and enables the unified gateway to access BRI trunks provided
by the switches on the ISDN.
R2
Enables the communication between the unified gateway and traditional
switching devices and enables the unified gateway to access E1 trunks
provided by the traditional switching devices.
QSIG
Enables the communication between the unified gateway and switches
supporting the QSIG signaling and enables the unified gateway to access
E1/T1 trunks provided by the switches.
SIP
Enables the interconnections between unified gateways and connects the
unified gateway to IADs and SIP multimedia packet terminals.
AT0
Connects the PSTN and switches. It is a DC loop analog trunk (also called
narrowband trunk) and enables intra-office users to have voice
communications with outer-office users through the obtained traditional
phone lines, achieving easy narrowband access.
Telnet
Connects LMTs or remote operation and maintenance terminals to unified
gateways.
SSH2
Secures remote logins to an LMT or an operation and maintenance terminal
through an unreliable network to the unified gateway for configuration and
debugging.

The unified gateway in the HQ is called the central node, which can be
U1911/U1960/U1980/U1981. The unified gateway gateway in a branch is called a local
node, which can be U1911/U1960/U1980/U1981.

The central node and local nodes connect to the PSTN through digital or analog
trunks.
The central and local nodes are connected through SIP trunks and use the heartbeat
mechanism to monitor the running status of the peer devices.

When the central node and local nodes are connected properly:

Users and services are configured and controlled in a unified manner on the central
node.

IP phones on local nodes are directly registered with the central node.

Analog phones on local nodes are registered with the central node through local
node proxies.

Local nodes synchronize SIP user data from the central node in real time.

When the central node is faulty or disconnected from local nodes:

The basic call function remains available for analog users on local nodes.
When detecting that the central node is faulty or disconnected, IP phones on a local
node automatically set the local node as the SIP server. The basic call function

remains available for SIP users on the local node.

The disaster recovery mechanism is described as follows:

Two central nodes are deployed in two places, and each central node has one
unified gateway deployed. The two unified gateways work in active/standby mode,
known as an active node and a standby node. The active and standby nodes are
connected using the TCP protocol. They also use a heartbeat mechanism to
constantly check each other's status. The service servers are deployed at the central
node where the active unified gateway resides.

When the active and standby nodes are running correctly, all users register with the
active node. The active node processes all user requests and synchronizes data to
the standby node in real time. At least 1 Mbit/s bandwidth must be reserved for
data synchronization, and the round trip time (RTT) must be shorter than 80 ms.
When the active node fails, the standby node takes over all services from the active
node and processes all user requests.
A trunk gateway can be deployed for connecting to the PSTN. If the trunk gateway
is not deployed, the active and standby nodes both connect to the PSTN. When the
active and standby nodes are running correctly, the active node routes calls to the
PSTN through trunks. When the active node is faulty, the standby node routes calls
to the PSTN through trunks.

1. A;
2. D.

This chapter are based on the


eSpace_U1910&U1911&U1930&U1960&U1981_V200R003C00_Quick_Installation
_Guide_01 and eSpace U1900 Series Unified Gateway Produc Documentation.

For more information, please go to http://e.huawei.com to find more related documents.

The sudden and momentary electric current that flows between two objects at different
electrical potentials caused by direct contact or induced by an electrostatic field.

Antistatic wrist strap:

An antistatic wrist strap, ESD wrist strap, or ground bracelet is an antistatic


device used to safely ground a person working on very sensitive electronic
equipment, to prevent the buildup of static electricity on their body, which
can result in electrostatic discharge (ESD). It consists of an elastic band
of fabric with fine conductive fibers woven into it, attached to a wire with a
clip on the end to connect it to a ground conductor.

ESDElectroStatic Discharge

Electrostatic discharge (ESD) is the sudden flow of electricity between two


electrically charged objects caused by contact, an electrical short, or dielectric

breakdown.

Major installation environment check items:

N
o
.

Check
Item

Requirement

Auxiliary
equipme
nt

The equipment room is equipped with specialized air conditioners, fire control facilities,
and good lighting facilities. Double aluminum alloy windows and doors are installed in
the equipment room. In addition, the auxiliary devices such as cabinets, distribution
frames, and cabling trays have been installed in the equipment room.

Moistureproof

The dehumidity devices such as air conditioners with the dehumidifying function and
special dehumidifiers must be installed in the equipment room where the relative
humidity is higher than 90%. Water seepage, dripping, and dew producing are forbidden
in the equipment room.

Lightning
protectio
n

Lightning-proof equipment such as lightning rods and lightning strips must be installed in
the equipment room. In addition, the lightning-proof grounding (that is, the grounding of
lightning-proof equipment such as lightning rods) must share a grounding body with the
protection grounding of the equipment room.

Dustproof

If the equipment room is close to a dust source, such as a coal mine, a country road or a
farmland, aluminum alloy windows and fireproofing doors must be installed in the
equipment room.

N
o
.

Check
Item

Requirement

Grounding

Joint grounding is used in the equipment room. That is, the working ground, protection
ground, and lightning-proof ground share a grounding body. Various communication
equipment and power supplies should share a protection ground bar. The grounding
resistance is less than 1 ohm.

AC power
supply

Stable AC power supply must be available in the equipment room to meet the power
requirements of the equipment. The AC power distribution switch and AC power cables
are installed properly. The AC voltage is 220 V, the power is 300 W, and the conducting
wire can accept a current of 3.5 A.

DC power
supply

The AC voltage is 48 V, the power is 350 W, and the conducting wire can accept a
current of 9 A.

UPS power
supply

If the UPS power supply is required for the equipment, the UPS power supply must be
installed before deployment, and the output power meets the requirements.

Shipment
The software and hardware purchased by Huawei have been delivered to the customer
information site, with complete packages.

ESD Measures

To guarantee board safety, employ dedicated ESD bags and boxes when transporting

or storing boards.

Wear a well-grounded ESD wrist strap when holding, removing or inserting boards,
and insert the ground end to the ESD jack of the shelf

Unpacking Boards

Electronic circuit is vulnerable to the ESD. When handling a board, wear the ESD wrist
strap that is properly grounded and touch only the edge of the board.
Boards are put into ESD bags in transportation. Before unpacking boards, take ESD
measures. Moreover, note the effect caused by the ambient temperature and humidity.
Generally, an ESD bag contains desiccant to absorb the moisture inside the bag. When
a board is moved from a cold and dry place to a hot and damp place, wait at least 30
minutes before unpacking the board. Otherwise, the moisture condensed on the board
surface may damage the board.

To unpack boards, do as follows:


1. Check the bag of each board for any damages.
2. Wear an ESD wrist strap and ground it properly.
3. Open each bag and check the board inside it for any damages.

T1 trunk notes:

If you want to use the T1 trunk, you must remove the jumper caps on the CVP
board (U1910&U1930) or MTU board (U1911&U1960&U1981).
Switch the trunk mode on U1900.

Built-in VMS service notes:

If you want to use the built-in VMS service (U1911&U1960&U1981) or


customize the system announcements, ensure that you have installed an SD
card in slot 0 of the MTU board.

Connect the main control board of U1900 to the switch according to the networking data
plan. The adaptive network ports support both straight-through and crossover cables.
Notes

By default, the network port mode of the control board is single-network-port mode. So
connect the first network port to the switch, and add more network ports to the switch
if settings are changed.

Connect and fasten one end of the E1/T1 trunk cable to the E1/T1 port on the

CVP or MTU board, and connect the other end to the E1/T trunk cable of the
peer device directly or through the distribution frame. Connection rules:

E1 cable: For the same link number X, connect local TX to peer RX and local
RX to peer TX.

T1 cable: For the same link number X, connect local RX/TX to peer RX/TX

CVP and MTU boards provide LOS and RFA to indicate the E1/T1 connection status. If the
light is off, the connection status is normal. Otherwise, please check the connection is Ok
or not. The two E1 channels in the CVP board share one pair of indicators, only the two

channels both work normally the indicators are off.

For more indicators information in different boards, please refer to the U1900 product
document.

default IP address of the U1910/U1911/U1930/U1960/U1981 is 192.168.1.17, while the


default IP address of the U1980 is 192.168.1.85. Their default gateway addresses are the
same, which is 192.168.1.1.

The default user name and config password are admin and Change_Me.

1. An operation and maintenance terminal (a PC) is available. The PC must:

Have an Ethernet adapter installed and support TCP/IP.

The web browser can only be Internet Explorer 7.0/8.0/9.0 that supports ActiveX and
Javascript.
Support the resolution of 1024 x 768 or higher (1280 x 1024 recommended).

2. Connect the network cables


Select either of the following two modes for networking based on actual network
requirements:

Use a straight-through cable to connect the network port of the unified gateway to
the network adapter port of the PC.

Use a straight-through cable to connect the network port of the unified gateway to
a switch or a hub, which is then connected to the PC.

Notes

In single-network-port mode, you can log in to the web management system


through network port 0 or 1 on the SCU board (U1911/U1960/U1981) or SMCU
board (U1980).
In dual-network-port or triple-network-port mode, you can log in to the web
management system only through network port 0 on the SCU board
(U1911/U1960/U1981) or SMCU board (U1980).

3. Set the IP address of the PC


The IP address of the PC must be on the same network segment as that of the unified
gateway and cannot conflict with IP addresses of other devices.
4. Set the web browser.
To ensure system security, the unified gateway does not support SSL 2.0/3.0. To avoid a

login failure, in the web browser, choose Tools > Internet Options > Advanced, select TLS
1.0/1.1/1.2, and deselect SSL 2.0/3.0.

LMT tools description

go to http://e.huawei.comdownload the LMT tool based on the version of the


U1900.

CLI command description:

Download putty ( support SSH, Telnet and console port )

Here we use the Web management system to view the license information, you can alos
use LMT or CLI to view Tehran license, the detailed command is show license .
License Key parameters
Parameter

Description

Total users

Number of users to be allocated.

CTI users

Number of SoftConsole clients.

Voice Mail nums

Number of concurrent voice mailboxes.

Conference users

Number of concurrent users in an audio conference.


NOTE:This parameter is available only for
the U1911/U1960/U1980/U1981.

Local Survival Right

Local survival switch.

Redundancy Backup Two-node active/standby switch.


Right

Besides supports loading the license by Web management system, it also supports loading
license by LMT or Putty + FTP Server.

The default IP address of the U1910/U1911/U1930/U1960/U1981 is 192.168.1.17, while


the default IP address of the U1980 is 192.168.1.85. Their default gateway addresses are
the same, which is 192.168.1.1.
The default user name and password are admin and Change_Me
Network port 0 must be used for accessing the web management system, LMT command
tree, and CLI.

Network Port Mode of the Control Board

The SCU board (U1911/U1960/1981) and the SMCU board (U1980) support the
following network port modes: single-network-port mode, dual-network-port mode,
and triple-network port mode.

In different network port modes, each network port processes different data streams. The
following table lists the data streams that can be processed by each network port in
different modes. To make data transmission more secure and reliable, you are advised to
use the dual-network-port or triple-network-port mode.

Network port modes


Network
Port Mode

Network Port 0

Network Port 1

Network Port 2

Singlenetworkport mode

Processes all data streams and connects


to the BillServer.

This network port


and network port 0
work in
active/standby
mode.

Not used.

Dualnetworkport mode

Processes the following streams and


connects to the BillServer:Network
management packets: Telnet, Secure
Shell (SSH), Simple Network
Management Protocol (SNMP), Trivial
File Transfer Protocol (TFTP), FTP server
(FTPS), web management server, or
proprietary protocols for
communicating with the business
management unit (BMU), Business
Management Platform (BMP), and LMT
Protocol packets: RADIUS/SNTP

Processes all data


streams and
connects to the
BillServer.

Not used.

Triplenetworkport mode

Processes the following streams and


connects to the BillServer:Network
management packets: Telnet, SSH,
SNMP, TFTP, FTPS, or proprietary
protocols for communicating with the
BMU, BMP, and LMT
Protocol packets: RADIUS/SNTP

Processes all data


streams and
connects to the
BillServer.

Connects to the
BillServer.

In dual-network-port mode and triple-network port mode,

Network port 0 must be used for accessing the web management system, LMT
command tree, and CLI.

Network port 0, 1 or 2 can be used for accessing the web self-service system.

Only one network port can be used to connect to the BillServer.

You can use the Web management system to configure the IP address, and you can also
use the CLI to configure the IP address.

Different countries and regions are in different time zones. Therefore, you must set the
time zone of the unified gateway to that of the country or region where the unified
gateway is deployed. The system default time zone is GMT +8:00.
Some countries or regions use the DST, and you must configure DST rules on the unified
gateway. When DST rules are configured, the unified gateway can display the DST time
correctly.
The unified gateway supports two time setting methods: manual setting and time

synchronization with the NTP server. For more information about the NTP server settings,
please refer to U1900 product document.

The unified gateway supports two working modes: IMS and PBX (default).

PBX mode

PBX is short for private branch exchange, allowing an enterprise to manage


external incoming calls and internal calls in a group manner. When set to PBX
mode, the unified gateway can function as an IP PBX and support basic voice
communication (communication between intra-office users, and
communication based on narrowband and broadband trunks), point-to-point
video communication, fax, and supplementary services (such as voice
conferencing, automatic switchboard, and call transfer).

IMS mode

IMS stands for IP multimedia subsystem. It is a brand-new multimedia service

system that can meet the new and diverse multimedia service requirements of
terminal users. When set to IMS mode, the unified gateway can function as an
access gateway and register with the IMS network through a SIP trunk,
allowing enterprise users to use services provided by the IMS network. Based
on settings, the unified gateway can use its own services in IMS mode

Set the working mode of a device to IMS when the device is connected to the IMS
network. Otherwise, set the working mode of the device to PBX.

Board information

Board Device Model

Description

CVP

U1910/U1930

Provides the softswitch functions.


Processes media control protocols.
Provides Layer 2 switching and TDM switching functions.

ASI

U1910/U1911/U1930/U19
60/U1981

An ASI board connects the unified gateway to plain old telephone


service (POTS) phones. Each ASI board supports a maximum of 32 POTS
phones.

OSU

U1910/U1911/U1930/U19
60/U1981

An OSU board connects the unified gatewayto POTS phones or analog


trunks. Each OSU board supports a maximum of 12 POTS phones or
analog trunks.

SCU

U1911/U1960/U1981

A Service Control Unit (SCU) board functions as a softswitch. It


processes media control protocols and provides Layer 2 (L2) and time
division multiplexing (TDM) switch functions. An SCU board supports
single-network-port, dual-network-port, and triple-network-port
working modes.

MTU

U1911/U1960/U1981

An MTU board provides digital trunk access for connection to the


upper-level office through digital trunks. It also provides digit
collection, announcement playing, recording, conferencing, TDM-toVoIP conversion, and T.30-to-T.38 fax functions.

Board Device Model

Description

BTU

A BTU board provides digital trunk access. Each BTU board provides
four BRI S/T ports and supports eight voice channels.

U1911/U1960/U1981

SMCU U1980

Supports the following functions:


Provides the softswitch functions.
Processes media control protocols.
Supports three working modes: single-network-port mode, dualnetwork-port mode, and triple-network-port mode.
Provides Layer 3 switching and TDM switching functions.

MRS

Is a mandatory board and provides the following functions:


Allocates numbers, collects digits, and plays announcements.
Supports recording, conference, TDM-to-VoIP conversion, and T.30-toT.38 fax.
Supports G.711, G.729, and iLBC voice codecs.
Provides 256 channels for processing media resources.

U1980

DTU4 U1980
/DTU
8

Provides digital trunk access for connecting to the digital trunk of the
upper-layer office.

BTU

A BTU board provides digital trunk access. Each BTU board provides
four BRI S/T ports and supports eight voice channels.

U1911/U1960/U1981

For the U1911/U1960/U1981, to add an MTU board, enter the value for Number of VMS
Channels, select Yes for Conference board, and click OK on the displayed MTU Parameter

Configuration page. The number of VMS channels refers to the license capacity, that is, the
number of calls being concurrently forwarded to mailboxes

In the Parameter configuration area, set Transmission mode and Transcoding codec and
click OK.

Key parameter description:


Parameter

Description

DTU Module

Indicates the switch of the digital trunk module. If you need to configure E1/T1 trunks on
the U1910/U1930, select On; otherwise, select Off.

Transmission
mode

The unified gateway supports the E1 and T1 modes, but does not support mixed use of the
two modes. The PRA and QSIG trunks support the E1 and T1 modes. Other narrowband
trunks support only the E1 mode.
NOTE:If T1 is used as the transmission mode of PRA or QSIG trunks, on the E1/T1 port
controller of the CVP board (U1910/U1930), MTU board (U1911/U1960/U1981), or DTU4
board (U1980), remove or insert the jumper caps (total eight behind the panel) to adjust to
the Balance mode. Otherwise, T1 trunks will be unavailable.

Transcoding
codec

When selecting ALAW or ULAW as the codec, ensure that the configuration is the same as
the configuration of the peer device; otherwise, the call quality between users of the local
and peer devices will be affected.

Transmission mode and Transcoding codec are two independent parameters. Specifically, the
transcoding codec ALAW orULAW can be selected regardless of whether the transmission
mode is E1 or T1. Generally, the transcoding codec is set to ALAWwhen the transmission mode
is E1 and set to ULAW when the transmission mode is T1. The actual configurations are based
on site requirements.

Click Data Save in the upper right corner of the web management system.

Backing up a data file

Log in to the U1900 Web management system, Choose System > File
Management., Click Download.

Loading a data file

Log in to the U1900 Web management system, Choose System > File
Management., Click Browse, select the data file (data.bin) to load, and click
Upload.
As long as the main control board versions and the board type ( including the board
slot) are the same, the sip user and password can be recovered from one U1900 to
another U1900.

Notes

End of Life Announcement

IAD208E(M) is End of Marketing, more information please go to enterprise.Huawei.com.

The IAD132E(T) is 1 U high (1U = 44.45mm), which can be installed in the standard 19inch cabinet meeting the International Electrotechnical Commission (IEC) standard.
The front panel of the IAD132E(T) has four Ethernet ports, one maintenance serial port,
one Reset button and five indicators.

The IAD132E(T) can be inserted with boards of two types, that is, FXO & FXS Unit (OSU)
and Analog Subscriber Interface (ASID) board.
OSU board

Each OSU provides 8 Foreign Exchange Subscriber (FXS) ports and 8 Foreign
Exchange Office (FXO) ports, supporting eight POTS users and 8-port analog trunk.
FXS: foreign exchange subscriber port, used for connecting a POTS telephone.

FXO: foreign exchange office, that is, the interface between a POTS telephone
and a digital telephony switching system, used for connecting to the PSTN and
obtaining a PSTN number. The FXO port can be used for binding an intraoffice number, so that when a PSTN-side user calls the PSTN number of an
FXO port, the called number is switched to an intra-office number.

ASID board

Each ASID board provides 16 FXS ports, supporting 16 POTS users.

Generally, an IAD196 can work with only one main control board and one service board.
You can insert more service boards based on the system capacity. In addition, you must
insert filler panels into vacant slots.

The shelf of the IAD1224 is a standard 2U shelf, which can be installed in the standard 19inch cabinet meeting the International Electrotechnical Commission (IEC) standard. The
dimensions (H x W x D) are 86.1 mm x 442 mm x 310 mm.

Generally, an IAD1224 can work with only one main control board and one service board.
You can insert more service boards based on the system capacity. In addition, you must
insert filler panels into vacant slots.

The CVP boards functions as the main control board. It mainly manages devices, controls calls,
processes media, performs narrow-band switching, and performs switching between the internal
and external Ethernets.

Service network interface FE1/FE2

Used to connect to superior devices.

Used to cascade connection between IAD1224s.

Maintenance network interface LAN

Used to configure and debug the IAD.

Indicator

Name

Color

Status Description

Power
indicator

PWR

Green

When the indicator is steady on, the power supply is normal.


When the indicator is off, no power supply is received.

Running
indicator

RUN

Green

When the indicator blinks at 1 Hz, the board is being started.


When the indicator blinks at 2 Hz, the board is writing data into
the Flash memory when the IAD1224 is started or running.
When the indicator blinks slowly (at 0.5 Hz), the board is
normal.
When the indicator is off, no power supply is received or the
board fails to run.

Alarm
indicator

ALM

Red

When the indicator blinks at 2 Hz, an alarm is generated.


When the indicator blinks at 4 Hz, a critical alarm is generated.
When the indicator is off, no alarm is generated.

An ASI board (POTS interface board) provides 32 POTS user ports. Connect the DB-68
male connector of the user cable to a port on the ASI port, and connect the FXS line to the
user phone port.
ASI indicators

Indicator

Name

Color

Status Description

Power indicator

PWR

Green

When the indicator is steady on, the power supply is normal.


When the indicator is off, no power supply is received.

Running indicator RUN

Green

When the indicator blinks at 4 Hz, the board is loading


software.
When the indicator blinks at 2 Hz, the user is in off-hook
state.
When the indicator blinks at 0.5 Hz, the board is running
properly and in idle state.
When the indicator is off, no power supply is received or the
board fails to run.

Alarm indicator

Red

When the indicator blinks at 2 Hz, an alarm is generated.


When the indicator blinks at 4 Hz, a critical alarm is
generated.
When the indicator is off, no alarm is generated.

ALM

An OSU board (FXO and FXS interface board) provides 12 POTS user ports and 12 FXO
user ports. Connect the DB-68 male connector of the user cable to a port on the ASI port,
and connect the FXS line to the user phone port, and the FXO line to the PSTN port.
OSU indicators

Indicator

Name

Color

Status Description

Power indicator

PWR

Green

When the indicator is steady on, the power supply is


normal.
When the indicator is off, no power supply is received.

Running indicator RUN

Green

When the indicator blinks at 4 Hz, the board is loading


software.
When the indicator blinks at 2 Hz, the user is in off-hook
state.
When the indicator blinks at 0.5 Hz, the board is running
properly and in idle state.
When the indicator is off, no power supply is received or
the board fails to run.

Alarm indicator

Red

When the indicator blinks at 2 Hz, an alarm is generated.


When the indicator blinks at 4 Hz, a critical alarm is
generated.
When the indicator is off, no alarm is generated.

ALM

The IAD1224 uses two power supply modules that can be swapped and work in
active/standby mode. The AC and DC power supplies are supported.

Power switch

AC power socket

Invalid, not
enabled (no
operations
available
through RJ45
ports)

Procedure
The IAD power-off survival function is available only after the cables are connected. The
cables are connected as follows:

Connect the PSTN cable to the FXO port on the IAD. For example, port 8 in slot 1
connects to the yellow and blue FXO line pair.
Connect the phone cable to the FXS port on the IAD. For example, port 0 in slot 1
connects to the red and green FXS line pair.

When powered off, the IAD automatically connects the phone on the FXS port to the
corresponding PSTN cable.

When the IAD power-off survival is enabled, the number of the phone on the FXS port
becomes the number that the PSTN network carrier assigns to the FXO port. The PSTN
network carrier also specifies the dialing rule for the IAD.

Verification

To verify the power-off survival function of the IAD, proceed as follows:

Power off the IAD.

Make a call to an external phone number using a phone that connects to FXS port 0
on slot 1.

If the call can be connected, the power-off survival function is enabled. If the function is

not connected, verify that the FXS port is normally connected to the phone, the FXO port,
and the PSTN cable, and that the FXS port and the FXO port matches.

Mapping between user cable colors, FXO ports, and FXS ports on the OSU board

Prerequisites

If the protocol mode is set to SIP.

When IADs are assigned with static IP addresses, the local-switch function is
available among local IAD or multiple IADs. If the DHCP or PPPoE mode is selected,
the local-switch function is only available for the local IAD.

1.A

The account root for the Web Management System and that for the command line
interface (CLI) are the same. If you change the password of the root, use the new
password to log in to the Web Management System or the CLI.

If you changed the IP address of the IAD, you can do as follows to obtain the IP address.

Dial *127 on the phone connected to the IAD, and then the voice announcement
plays the IP address.
Log in to the IAD through the serial port and run the display ip address command in
the common user mode to view the IP address of the device.
IAD 104 dont provide users with the web management system.

After you log in to the system, you can click Change Password at the upper right corner of
the page to set a new password. After the password is set, record related data.
By default, the protocol is SIP.

There are three methods for obtaining the IP address. Select a method according to the
actual network.

Click Static and enter the IP address, subnet mask, and default gateway.

Click DHCP. The IAD automatically obtains the IP address from the DHCP server.

Click PPPoE and enter the user name and password.

This chapter just introduce the configuration mode, for the detailed configuration step,
please refer to OUC1202 U1900 Intra-Office Call .

1.AC

An Analog Subscriber Interface (ASI) board provides 32 Foreign Exchange Subscriber (FXS)
ports.
An FXO/FXS Interface Unit (OSU) board provides 12 Foreign Exchange Subscriber (FXS)
ports and 12 Foreign Exchange Office (FXO) ports.

UTP: Unshielded Twisted Paired.


IAD:As the media access gateway of Voice over IP (VoIP) and Fax over IP (FoIP). The IAD
converts analog voice data into IP packets and transmits data through the IP network.

eSpace 7903X is a multi-functional IP phone expansion module. It is used with eSpace


7950 to provide various functions, including speed dial, phone lock, phone sleep, and
contact group. With a single eSpace 7903X, eSpace 7950 increases the number of

programmable buttons to 40 and provides optimized viewing through the 5-inch color
LCD screen. eSpace 7903X helps users use eSpace 7950 more efficiently with better user
experience.

IP phone has built-in downward double Ethernet switch port.

Support the bridge mode, sharing network, a network cable can solve all problems.

An internal double port Ethernet switches support the RJ-45 interface directly connected
to the 10/100/1000 base-t Ethernet, and provides a local area network (LAN) connection
that used to connect the phone and PC in the same position.
System administrators can specify different WLAN for PC and phone (802.1 Q), which
provides voice and data traffic with higher safety and reliability.
eSpace 7870, 7850, 7830, 7820 and 7810 support the bridging function. The device
connected to the PC port of an IP phone can access the network connected to the LAN
interface of the IP phone and can communicate with other devices in the network. In this
case, the IP phone acts as a switch with two interfaces but the working mode is different

from the working mode of a normal switch. Special configurations are performed at the
lower layers of an IP phone to separate the broadcast packets between the two interfaces.
Therefore, the IP phone is not affected by a large number of broadcast packets.

Dynamic Host Configuration Protocol (DHCP) is a protocol for dynamically managing and
configuring users in a centralized manner. It uses the Client/Server structure. A DHCP client
sends the DHCP server a request to apply for parameter settings, including the IP address,

subnet mask, and default gateway. Then the DHCP server sends the parameter settings to
the DHCP client.

DHCP features

Plug and play property

The client can obtain the IP addresses and associated parameters without
configuration, simplify client network configuration, reduce maintenance
costs.

Unified management

All the IP address and related information are unified managed and allocated
by the DHCP server.

Use of high efficiency

Enhance the use efficiency of IP address through IP address lease


management.

Achieved across a network segment

Can implement agreement message interaction between the client and DHCP

server among different subnets through the DHCP relay.

DHCP (dynamic host configuration protocol) evolved from the BOOTP protocol is used to
automatically allocate a client computer the IP address of a standard protocol that is
defined in RFC 2131.

PoEPower over Ethernet

support the mode of free-line power supply and mode of signal-line power supply.
When the PoE function is used, the reliable power supply distance is up to 100
meters.

POE system advantages

support the PoE function. When not being connected to a power adapter, a client
can obtain power from a PSE device (a PoE switch such as the S3900) to work
normally.

PoE reduces investment spending and the total electrical equipment deployment
costs in the unified IP infrastructure.
PoE is free for the terminal equipment installation wall power supply, thus reduces
the cost associated with supporting terminal equipment.
Install the network connection device in the place that difficultly deploying local AC
power source, which provides greater flexibility.

POE system

PSE(Power Sourcing Equipment): Power supply equipment for the Ethernet client
device, and also is the power supply manager of the whole process of POE Ethernet,
such as switches.
PD(Power Device): Accept the PSE load, namely the client devices for POE system,
such as IP phone.

In general, retain the default configuration of voice codec priority for deployment. If the
network environment is complex, you can adjust the codec priority according to the actual
network bandwidth.

Access the phone web page as an administrator. Choose Advanced > Media.
Noise Suppression: When noise occurs in the audio collected by the IP phone, the IP phone
automatically suppresses the noise. The voice quality decreases as the noise suppression
capability increases.

Echo Cancellation: The IP phone conceals the acoustic echoes that occur in the
microphone to minimize the effects of the local audio card shift.
Voice Activity Detection: The IP phone conceals the acoustic echoes that occur in the
microphone to minimize the effects of the local audio card shift.

Automatic Gain Control: The IP phone automatically adjusts the volume and level to deliver
optimal audio experience. In the database engine, users can enable and disable the AGC
function and query the running status of the AGC function. AGC classifies into the analog

AGC and digital AGC. In the analog AGC, the IP phone adjusts the microphone input to
provide optimal audio experience. While, in the digital AGC, the IP phone adjusts the PCM
waveform.

eSpace IP phones support the Virtual Local Area Network (VLAN) function. The packets
sent by an IP phone are labeled with tags. This makes packets transmitted in a separate
voice VLAN, and the stability of VoIP packets is ensured.
Each VLAN is a broadcast domain. Hosts in a VLAN communicate with each other as if
they were in a LAN. Hosts in different VLANs cannot directly communicate with each
other.

DiffServDifferentiated Services. Due to development of the market and change of


competition environment, carriers must provide different levels of services for subscriber
groups with different requirements. The solution of differentiated services is promoted to

meet this demand. The services of high qualities are offered to subscribers with high
priorities, while the services of low qualities are offered to subscribers with low priorities.

VLAN auto-configuration mechanisms:


a, the automatic configuration based on the configuration file

b, the automatic configuration based on the LLDP-MED

Phone support SRTP media layer encryption, using AES128 algorithm encrypted
transmission to ensure the transmission safety.
Transport Layer Security (TLS) and its predecessor, Secure Sockets Layer (SSL), are
cryptographic protocols that provide communications security over the Internet.

A device provides ports for clients to access a LAN. The ports support the following access
control modes:

Authorized-force: A port in this mode allows clients to access network resources


without authenticating the clients.

Unauthorized-force: The device does not authenticate clients accessed through a


port in this mode.
Auto: A port in this mode allows clients to send and receive packets but does not
allow clients to access network resources before authentication succeeds. If
authentication succeeds, the port allows clients to access network resources. This
mode is mostly used.

The authentication process is as follows:


1. A client sends an EAPOL-Start packet to a device.

2. The device receives the packet and sends an EAP-Request/Identity packet,


requesting the client to send the user name. The device encapsulates frames from
the client into a packet and sends the packet to the authentication server.
3. The client sends the user name that is contained in the EAP-Response/Identity
packet to the device.
4. The authentication server searches the database for the user name in the packet
and obtains the corresponding password. The authentication server uses a
randomly generated encryption key to encrypt the password and sends the
encryption key to the device through the Access-Challenge packet.
5. The device sends the encryption key to the client.
6. The client receives the EAP-Request/MD5 Challenge packet containing the
encryption key. The client uses the encryption key to encrypt the password,
generates an EAP-Response/MD5 Challenge packet, and sends the packet to the
device. The device sends the packet to the authentication server.
7. The encryption algorithm is irreversible normally.

802.1x authentication:

MD5 authentication: authenticated by user name and password

EAP-TLS authentication: authenticated by the certificated built in an IP phone,


featuring higher security

VLAN configuration mode:

LLDP mode: The data switch delivers audio VLAN information to an IP phone

through the LLDP protocol. To use this mode, the data switch must support the
LLDP function. This mode is enabled for the IP phone by default.

DHCP Option mode: Users can set the Option 132 and 133 fields to the ID and port
number of the audio VLAN respectively. Then, an IP phone can obtain audio LAN
information through the option fields.
Manual configuration: Users can manually configure audio VLAN information on the
network configuration screen on an IP phone.

Answer1.A

If a static IP address is required for eSpace 7910/eSpace 7950, set IPv4 Settings to Static.
Obtain an IP address, subnet mask, gateway address, and DNS address from your network
administrator.

If Point-to-Point Protocol over Ethernet (PPPoE) is required for eSpace 7910/eSpace 7950,
set IPv4 Settings to PPPoEObtain the user name and password for your PPPoE dialup
connection from your network administrator.

Obtain the IP address of the IP phone.

On the phone login screen, choose More > Network > Status > Network and view
thephone's IP address.
On the phone home screen, choose Apps > Status > Network and view the phone's
IPaddress.

Open a web browser and enter the phone's IP address in the address box.

Enter the administrator account and password, and click Log In.

The default administrator account is admin, and the default administrator password is
admin123. To ensuresecurity, change the administrator password at your first login.

If you enter the password incorrectly for three consecutive times, the IP address of

the PC or terminal you are using will be denied to access the IP phone. To continue
to access the IP phone, you can change the IP address of the PC or terminal (a
maximum of ten IP addresses are supported). However, if the IP address has been
changed for ten times and is still denied to access the IP phone, the IP phone will be
locked, denying any IP address to access it. In this case, to access the IP phone, you
need to wait five minutes until the IP phone is unlocked.

Access the phone web page as an administrator. Choose Advanced > Network.

If the authentication is successful, the device can access resources on the LAN.

If the authentication is failed, the device cannot access resources on the LAN.

Access the phone web page as an administrator. Choose Advanced > Network.

LAN Port: VLAN setting parameters for the LAN port on an IP phone. The value Enabled
indicates that the VLAN function is enabled for the LAN port.

ID: ID of the VLAN where the IP phone belongs to.After LAN Port is enabled, the
data packets sent by the IP phone carry VLAN tags.The network administrator
divides the network connected to the switch into N VLAN. Each VLAN has a VLAN
ID.

Priority: Priority of the VLAN where the IP phone belongs to. The value ranges from

0 to 7. A larger value indicates a higher priority.

PC Port: VLAN setting parameters for the PC port on an IP phone. The value Enable
indicates that the VLAN function is enabled for the PC port.

ID: This VLAN ID is carried by data packets transmitted upstream through the PC

port. For data packets transmitted downstream through the PC port, if the VLAN ID
in the data packets is the same as the VLAN ID of the PC port, the PC port removes
the VLAN tag from the data packets before transmitting them. If the VLAN ID in the
data packets is different from the VLAN ID of the PC port, the PC port discards these
data packets.

Priority: Priority of the VLAN where the IP phone belongs to.The value ranges from 0
to 7. A larger value indicates a higher priority.

PC Port VLAN Filtering: If you select Enable, the phone's PC port allows the following data
packets to pass: data packets that do not carry any VLAN ID, data packets carrying the
VLAN ID of the LAN port, data packets carrying the VLAN ID of the PC port, and data

packets carrying the VLAN ID that is set below PC Port VLAN Filtering.

ID: Indicates the VLAN ID, with which packets are allowed to pass through the
phone's PC port when VLAN Filtering on PC Port is enabled.

Fixed IP mode: Users must fill in the IP, subnet mask, network gateway, active and standby
DNSs.
DHCP mode: An IP phone obtains the IP address from the DHCP server automatically.You
can choose to manually enter the DNS server address or enable the IP phone to
automatically obtain an DNS server address.

PPPoE mode: User name and password must be known.

User needs to be registered to the server when the phone is switched on every time.

User needs to be registered again When the addresses of the SIP Client changes.

Registration information must be refreshed periodically, usually the register saves the
registration information to the Location Server.
Holding the AOR address bound to a Contact, and the Proxy look for the called address
easily when in call.

Choose User > SIP User.

Click Create on the SIP User page.

Enter user information on the Add SIP User page that is displayed.

Key parameter description


Parameter

Description

Start device ID

ID of the device from which the configuration starts. The device ID is


the SIP device registration account, for example, SIP user ID
configured during IP phone registration.

Device type

The Device type parameter is used for the connection between the
unified gateway and a SIP device. The parameter value must be the
same as that specified on the SIP device. If the parameter is set to
eSpace, the corresponding number can access the unified gateway
only through an eSpace soft terminal.

Authentication
mode

This parameter specifies the mode for authenticating a SIP device


when it registers with the unified gateway.
To improve account security, you are advised to select the IP address
or password as the authentication mode for SIP number allocation.
NOTE: In password-based authentication, the password by default
must be a string of 831 characters that consists of at least two
types of the following characters: digits, letters, and special
characters. You can run the config system authentication mode
simple command to set the weak password mode. In this mode, the
password is a string of 131 characters and has no complexity
requirements. The password must not be the same as the device ID
or device ID in reverse order.
Except authentication by IP address and password, unified gateway
supports authentication by IP address segment. You can run the
config add addresspool index [0-n] startip x.x.x.x endip x.x.x.x
command to configure trusted IP address segment, and select
authentication by IP address segment when configuring users (These
two configurations can be achieved only using commands).
NOTICE: A weak password may pose security risks. Use a weak
password with caution.

User right level

User right levels are listed in the following ascending order: Default
right < Normal right < Advanced right < Super right. The default
value is Default right. Users with a preset level of rights can use
specified call rights only in specified time segments. Higher-level
users can reserve trunks for themselves. If there is no available trunk,
they can preempt the trunks assigned for lower-level users to make
outgoing calls.

Choose User > SIP User.

Click Create on the SIP User page.

Enter user information on the Add SIP User page that is displayed.

Configure intra-office prefixes for unified gateway users so that unified gateway users
(intra-office users) can make calls with each other.

Log in to the web management system.

Choose Trunk > Prefix Configuration and click Create.

Configure a prefix in the Create Prefix dialog box

Key parameter description


Parameter

Description

Service type Basic service The service type is a basic audio service such as intra-office call
and local call.
Supplementary service The service type is a supplementary service.
Intelligent service The service type is a common intelligent service, and is
currently unavailable.
SoftConsole service The service type is a softconsole service such as the call
waiting and attendant group in the console.
Virtual user The service type is a virtual user service such as the phone
meeting, automatic switchboard, and customized VU.
Card number service The service type is a card number service such as the
call, recharge, password change, balance query, terminal binding, prepaid,
or postpaid service.
Call by Dialing Card Number Users can dial a card number to make intraoffice calls, national toll calls, international toll calls, and emergency calls.

Parameter

Description

Call
property

Intra-office call This option is selected for calls between users under the
same unified gateway.
Local call This option is selected for local calls.
National toll call This option is selected for national toll calls.
International toll call This option is selected for international toll calls.
Emergency call This option is selected for emergency calls, for example, to
911.
Intra-office or local call This option is selected when first digits in a local
number compose a short number. The call with this attribute is routed to
the intra-office user. If the routing fails, the call is then routed to the local
user in another office.
Route by number length This option is selected for outgoing calls with the
same prefix but with different number lengths.
Local regeneration call This option is selected for calls between intra-office
users and local PSTN users when both active and standby devices in the
centralized call management network mode fail and the local generation
function is enabled.

Calling
number
change

Index of a calling number change rule. The calling number change rule is
used to change calling numbers, for example, to a switchboard.
The priorities of the three number change modes are long number and short
number, number mapping, and number change in descending order.

Calling
number
mapping

Indicates whether to display the mapping number of a calling number. If a


user is not configured with the long number, the mapping number is
displayed on the called parties' phones.
The priorities of the three number change modes are long number and short
number, number mapping, and number change in descending order.
Index of a called number change rule. For example, a called number change
rule can be as follows: If the calling party dials 912345678, unified gateway
deletes the outgoing prefix 9 and sends the called number 12345678.
The priorities of the three number change modes are long number and short
number, number mapping, and number change in descending order.

Called
number
change

Called
number
mapping

Indicates whether the mapping number of a called number can be used in


dialing. If a user is not configured with the long number, the mapping
number can be used.
The priorities of the three number change modes are long number and short
number, number mapping, and number change in descending order.
Display long The long number is a number used for short-number users to make
calling
outgoing calls or for outer-office users to call short-number users.
number
The default value is No. When the parameter is set to Yes and the user is
configured with a long number, the calling number is changed to the long
number for outgoing calls.
The priorities of the three number change modes are long number and short
number, number mapping, and number change in descending order.

Network Environment: An IP phone selects an appropriate network environment according


to the onsite conditions:

UC2.X: UC2.X network

UC2.0: UC2.0 network

Asterisk: Asterisk network

Broadsoft: Broadsoft network

CC: CC network

IMS+UC: IMS+UC network

Others: other third-party networks

Note: The IP phone needs to be restarted after its network environment is modified.After
the network environment is changed, the IP phone changes the default configuration file
and the default service rights in service rights management accordingly. This facilitates the
setting and use of the IP phone.

Sequential: The IP phone sends registration messages to a SIP server at intervals of half of
the registration interval. If the active SIP server is fails, the IP phone registers with the
standby SIP server. If the standby SIP server fails, the IP phone registers with the local SIP

server. Services are interrupted during the SIP server switch. After the IP phone registers
with a SIP server successfully, its services become available again.

Simultaneous: The IP phone registers with the active, standby, and local SIP servers at the
same time. If the active SIP server fails, the standby SIP server takes over the services. If the

standby SIP server also fails, the local SIP server takes over the services. The switching
takes about 3s, which cannot be perceived by users. After the switching, services are
running properly.

Log in to U1900 administrator web page.

Choose User > SIP User.

Tick the encoding to be used according the needs, and you can adjust the priority
according moving them up and down. The smaller the priority value, the higher the
priority.

Log in to IP phone Web page.

Choose Advanced > Media to enter the configuration page.

Access the phone web page as an administrator.

Choose Advanced > SIP Signaling.

SIP Registration Cycle: The IP phone sends registration messages to the server at an
interval of half of the registration cycle.
Re-registration Interval: Interval for the IP phone to re-register with the server upon its
registration failure.

Local SIP Port: SIP port used for listening the IP phone and transmitting SIP signaling.

Time Update Mode: The displayed time on the phone can be automatically and manually
updated.

If a user set the displayed time to be automatically updated, he or she must fill in the
server IP address and the update interval.

If a user chooses to manually update the displayed time on the phone, he or she
must enter the system time by himself or herself.

Time Zone: Time zone of the area where a user locates.

Time Format: 12-hour or 24-hour format.

Date Format: Users can customize the date format based on their favorites.

On the web page, choose Button > Expansion Module.


Click the gray button next to the programmable button on the expansion module. The
Select Programmable Button Functions page is displayed.

If the number a user dialed matches a preset dialing rule, the phone automatically converts
the number according to that rule and calls the converted number after the preset delay
time expires.
Parameter description

x or X: Represents a digit of 0-9.For example, 135xxxxxxxx indicates an 11-digit


number starting with 135.
[ ]: Matches any digit in [ ].For example, [2-8] indicates any digit of 2-8; [279]
indicates any digit of 2, 7, and 9.

.: Indicates that the digit before dot (.) can appear for any times (for example, not
appear or appear once).The last digit of a dialing rule cannot be dot (.). For example,
92.3 can indicate 93, 923, 9223, and so on.
<:>: If the dialing rule is A<B:C>D, When the number a user dials matches ABD, the
phone automatically replaces B with C and calls ACD. A, B, C, and D in the dialing
rule indicate character strings consist of digits and common signs like asterisks (*),
number signs (#), and plus signs (+). A, B, and D can also contain dialing rule

parameters.For example, When the dialing rule is 1[278]<111:000>XX, the phone


automatically replaces 111 with 000 if the number a user dials matches
1[278]111XX.For example, if the number a user dials is 1211123, the phone
automatically calls 1200023.

Scenario ConfigurationConfigure dialing rules based on the actual usage scenarios. The
functions a dialing rule can implement and the associated scenarios are described as
follows:
Immediate call.

Delay call For example,.

a user wants the phone to automatically call a mobile phone number starting with
135, 136, 137, 138, or, 139 in 5s after the user dials such a number. In this
scenario, set Dialing Rules to 13[5-9]XXXXXXXX and Delay to 5.

Number replacement

For example, a user wants the phone to automatically call a 5digit number after the
user dials such a number. In this scenario, set Dialing Rules to XXXXX and Delay to
0.

For example, a user wants the phone to automatically replace 0 with 0086 and call
the number after the user dials a mobile phone number starting with 0. In this
scenario, set Dialing Rules to <0:0086> XXXXXXXXXXX.

Number insertion

For example, a user wants the phone to automatically suffix # to a 3digit number
starting with * and call the number after the user dials such a number. In this
scenario, set Dialing Rules to *XX<:#>.

Corporate Directory Server: Server that provides multiple information services including the
corporate directory and corporate presence. You can select Corporate or LDAP from the
Corporate Directory Server drop-down list box.In the UC2.X or UC2.0 network, you can

also choose Corporate(UC2.X) and Corporate(UC1.1).

Software download path:


Log in to http://support.huawei.com/enterpris, choose Downloads > Unified
Communications and Collaboration > Unified Communications > Unified Communications
Endpoints to download the specific version.

Software package describe:


File

Description

eSpace7910V200R003C00.bin

Upgrade files used for online upgrade or


upgrade through eSpace EMS.

UPF-eSpace7910.xml

Policy templates used for configuration file


upgrade.

Config-eSpace7910.xml

Configuration file used for phone deployment.

eSightUI.xml

Configuration templates used for auto


deployment. through eSight UC&C

tools.zip

Tools to make Language Package

1. A

During a call, the party who dials the phone number and initial a call is caller (calling party)
and the other party who answers the call is callee (called party).

PSTN: Public Switched Telephone Network. A telecommunications network established to


perform telephone services for the public subscribers. Sometimes it is called POTS.
POTS: Plain Old Telephone Service. The basic telephone service provided through the
traditional cabling such as twisted pair cables.

E.164 Number Mapping (ENUM) is a set of protocols for unifying telephony systems using
Internet. Dynamic Delegation Discovery System (DDDS), DNS, and E.164 are used for
addressing.

ENUM is a standard of IETF. It uses special DNS record types to translate a telephone number
into a Uniform Resource Identifier (URI) or IP address that can be used in Internet
communications. The objective of ENUM is using a single number to replace multiple numbers
and translating the home phone, office phone, fax, cell phone, and email of each individual to
URIs or IP addresses.

MSISDN: Mobile Subscriber International ISDN/PSTN number

ENUM: E.164 NUmber Mapping

E.164 Number Mapping (ENUM) is a set of protocols for unifying telephony systems

using Internet. DDDS, DNS, and E.164 are used for addressing. ENUM is a standard of
IETF. It uses special DNS record types to translate a telephone number into a URI or IP
address that can be used in Internet communications. The objective of ENUM is using a
single number to replace multiple numbers and translating the home phone, office
phone, fax, cell phone, and email of each individual to URIs or IP addresses.

E.164 is an ITU-T recommendation that defines a numbering plan for the world-wide
PSTN and some other data networks and also defines a general format for international
telephone numbers. Plan-conforming numbers are limited to a maximum of 15 digits,
and a complete number contains an international call prefix. An E.164 number is an
MSISDN number, which is the number that a calling party needs to dial for reaching a
user in the mobile communications network.
The format is CC+NDC+SN or CC+N1N2N3+H0H1H2H3+ABCD.

A call prefix is the string consisting of first few digits or all digits in called numbers. If the
called number is 1234, the call prefix can be:

First digit: 1

First two digits: 12

First three digits: 123

Entire called number: 1234

The prefix groups configured on the U1900 constitute a called number analysis table of
the system. If the called number analysis table contains multiple prefixes for a called
number, the system uses the maximum matching rule to match the called number with a
prefix during number analysis.
Assume that the called number is 1234 and the prefixes 1, 12, and 1234 are configured in
the called number analysis table. According to the maximum matching rule, the system
will select the prefix 1234, which is the closest to the called number. The prefixes 1 and 12
do not comply with the rule.

The basic service prefixes are classified into two types:

Intra-office prefix: used for intra-office and outer-office users to call intra-office
users. For example, if the intra-office number ranges from 8000 to 8099, you can
set the intra-office prefix to 8. When calling an intra-office user, you can simply dial
the user number such as 8008.

Outgoing prefix: used for intra-office users to make outgoing calls, such as national
and international toll calls. For example, an outer-office user number is 12345678. If

an intra-office user wants to dial 912345678 to call the outer-office user, the
outgoing prefix 9 needs to be set. The number conversion rule is that the first digit
of the called number is deleted.

Why do we convert the calling number?

Generally, the numbers assigned by PBX are short number or extension number
which are only valid inside the office but can not be recognized by PSTN. In this
case, when initializing an PSTN outgoing call, system need to convert the private
extension number to public PSTN number. Otherwise, the call will be rejected.

Why do we convert the called number?

Employees are usually required to dial a 0 or 9 followed by the actual called


number.Thus the called number with 0and 9 can not be recognized by the
PSTN. So the system need to delete the 0 and 9 before sending it out.

In this scenario, two number conversion modes are adopted, the number change and the
long number and short number.

Number Change : This topic describes how to define the number change rules for
modifying, inserting, and deleting calling or called numbers.
Long Number and Short Number : A user has two numbers: a short number and a long
number allocated by the PSTN. The short number is used for the calls between intra-office
users. The long number is displayed when you make an outgoing call or is used for outeroffice users' direct dialing in.

Number Mapping :The number mapping is used to implement irregular number changes.

The priorities of the three number change modes are long number and short number,
number mapping, and number change in descending order. If a user meets the conditions
of the three number change modes, only the mode with the highest priority is used. If the
user has a long number, the long number is displayed on the phone of the called party.

The user who does not set the long number can use the mapping number if it is set. The
user who does not set the long number and number mapping uses the changed number.

Number Change

modify

Insert

Delete

Number mapping

Long number and short number

The priorities of the three number change modes are long number and short number,
number mapping, and number change in descending order. If a user meets the conditions
of the three number change modes, only the mode with the highest priority is used. If the
user has a long number, the long number is displayed on the phone of the called party.

The user who does not set the long number can use the mapping number if it is set. The
user who does not set the long number and number mapping uses the changed number.

Office route

There are direct channels between office A and office B and between office A and
office C; therefore, both office B and office C are office routes of office A. There is
no direct channel between office A and office D; therefore, office D is not an office
route of office A.

The U1900 uses the office route number to uniquely identify an office route. For
example, the office route number from office A to office B is 1 and from office A to

office C is 2.

Sub-route

If a direct channel or an alternative channel is available between two exchange


offices, there is a sub-route between the two exchange offices. For example, there

are two sub-routes between office A and office B. Sub-route 1 is a direct sub-route
and can connect office A and office B directly without any convergence connections.

Route

A route is the group of sub-routes between two offices. A route contains one or
more sub-routes and different routes may contain the same sub-route. For example,
the route between office A and office B contains sub-routes 1 and 2 while there is
only sub-route 2 between office A and office D.

Route policy: time-based routing, routing based on charging rate, default route.

You can configure the time segment and charge rate for each office direction (route).

The system can dynamically and intelligently sort and prioritize office directions (routes)
based on specified routing policies. The system also attempts to call peer ends one by one.
Pay attention to the following points:

You are advised to bind each office route ID to a unique office route selection code.

If multiple office route IDs are bound to the same office route selection code, all
office route records under the office route selection code are added to the routing
table on the U1981 unified gateway when you configure the route analysis function,
which may cause that the routing table becomes full.

SS7 is a common channel signaling system that complies with international standards. It
has the following advantages: High speed transmission, Large signaling capacity, Powerful
functionality and High flexibility and reliability.SS7 meets the requirements of the Public

Switched Telephone Network (PSTN) and intelligent network (IN).

PRA refers to the control signal between terminal devices and networks. The signaling
structure is 30B+D (used in Europe and China) or 23B+D (used in North America and
Japan).

R2 is a type of channel-associated signaling. R2 was previously applicable to the


international telecommunication network and telecommunication networks in many
countries. The China No.1 (CNo.1) signaling is a type of R2. R2 consists of line signaling
and register signaling. The definitions of the line signaling and register signaling, however,
vary according to the country.
The U1900 supports Session Initiation Protocol (SIP). Defined by the Internet Engineering
Task Force (IETF), SIP is a protocol at the application layer, which is used to set up, modify,
and end multimedia sessions on the IP network.

The following transmission protocols can be used:

User Datagram Protocol (UDP) is a connectionless protocol. It provides transactionoriented simple and unreliable information transmission. In UDP-based data
transmission, each datagram is independent and contains the source address and
destination address. This protocol, however, is unreliable, as there is no guarantee of
data delivery (including the delivery time) or integrity.

Transmission Control Protocol (TCP) is a connection-oriented protocol. In TCP-based

data transmission, a virtual channel is required between the sender and receiver.

Transport Layer Security (TLS) is a cryptographic protocol that provides


communication security between two communication application programs,
ensuring data confidentiality and integrity. When TLS is used for interconnection

between two devices, ensure that the two devices have the same user certificates,
user private keys, and root certificates.

The difference between QSIG and PRA is the control signaling.


Clock source: The clock source is configured on one device so that the peer device can
synchronize its clock with this device. The configuration of a clock source prevents the
frame slip (voice packet loss).

If the U1900 unified gateway connects to multiple devices through E1 or T1 trunks,


configure the clock source on any of these devices. The U1981 unified gateway will
synchronize time with this device.
If the U1900 unified gateway provides the clock source, you do not need to
configure a clock source.

First layer: The physical layer

Physical layer provides means for establishing, maintaining and releasing the physical
connection, guarantees the information transmitted on a physical circuit. Physical
layer specification refers to electrical characteristics, physical characteristics of the
interface, including the connector specification for mechanical properties.

Second layer: The data link layer

Link layer is on top of the physical layer provides means of establishing, maintaining
and releasing of the data link. Data link layer provides features like multiplexing flow
control, error detection and recovery, and information delivery. PRI Protocol's link
layer standard protocol is Q921.

Third layer: the network layer

Network layer provides services to complete the call control features, including
circuit-switched calls and packet-switched call control, PRI protocol standard
protocols for the network layer is Q931.

Voice Data and Signaling Data Sent Through Different Channels: Signaling data sent by the
U1980 unified gateway must pass through a signaling transfer point (STP). The STP
transfers signaling data to the PSTN switch. Voice data sent by the U1980 unified gateway

is directly transferred to the PSTN switch through the E1 trunk. That is, voice data and
signaling data are sent through different channels.

Voice Data and Signaling Data Sent Through the Same Channel: The U1980 unified
gateway needs to negotiate with the peer device to determine whether to use the ISUP or

TUP trunk to connect to the peer device.

An AT0 trunk is an analog circuit trunk. It uses a analog subscriber line to connect an FXO
port in the local office to an FXS port in the upper-level office to implement the dedicated
line function.
To support concurrent incoming and outgoing calls, you can apply to the local carrier for
multiple lines, and allocate a number to each line. Based on the policy for using line
resources, intra-office users are divided into DID users (that can occupy dedicated lines
exclusively) and non-DID users.

1. AC;
2. BC.

Company A headquartering in Beijing, would like to deploy a small sized IPT network so as
to realize the interconnection between inner users.
There are three types of telephone users:

IP Phone users:6000-6009;

Analog Phone users under IAD: 6100-6109;

Analog Phone users under U1900: 6200-6209.

The first number in Status Code can define the type of response. In SIP/2.0, the first
number has six values, that is, it can define six types of responses in the following.

1xx: Information: The request has been received and it will be processed
continuously.

2xx: Success: The action has been received, understood and accepted successfully.

3xx: Redirection: Further operations are required to fulfill the call request.

4xx: Client-Error: The request has syntax errors or cannot be executed by the server.
The client should modify the request and then resend it.

5xx: Server-Error: The server cannot execute the valid request due to errors.

6xx: Global-Failure: Any server cannot execute the request.

Among them, 1xx response is Provision response, while other responses are Final
responses.

Analog phones are connected to specific port with analog telephone lines.

The analog phone users under IAD register to U1900 by the proxy of IAD.

Analog phones can not be connected to U1980 directly.

Now we are configuring the SIPor IP PhoneUsers under U1900.

Now we are configuring the SIPor IP PhoneUsers under U1900.

IP Phone users: 6000-6009

Service rights for different user right levels:

Key parameter description :

Now we are configuring analog phone (or POTS) users under IAD. The analog phone users
under IAD register to U1900 by the proxy of IAD, so for U1900, all these users are
considered as SIP users.
Analog Phone users under IAD: 6100-6109

Now we are configuring analog phone (or POTS) users under U1900. The analog phones
are directly connected to the OSU or ASI board on U1900 using analog lines.

Analog Phone users under U1900: 6200-6209

Key Parameter Description

A call prefix is the string consisting of first few digits or all digits in called numbers. If the
called number is 1234, the call prefix can be:

First digit: 1

First two digits: 12

First three digits: 123

Entire called number: 1234

The prefix groups configured on the U196xx unified gateway constitute a called number
analysis table of the system. If the called number analysis table contains multiple prefixes
for a called number, the system uses the maximum matching rule to match the called
number with a prefix during number analysis.
Assume that the called number is 1234 and the prefixes 1, 12, and 1234 are configured in
the called number analysis table. According to the maximum matching rule, the system
will select the prefix 1234, which is the closest to the called number. The prefixes 1 and 12
do not comply with the rule.
The basic service prefixes are classified into two types:

Intra-office prefix: used for intra-office and outer-office users to call intra-office
users. For example, if the intra-office number ranges from 7000 to 7099, you can
set the intra-office prefix to 7. When calling an intra-office user, you can simply dial
the user number such as 7001.

Outgoing prefix: used when intra-office users make outgoing calls, such as local
calls, national toll calls, or international toll calls.

Prefix 6 is configured as short number prefix

Config add prefix dn 6 callcategory Basic callattribute Inter cldpredeal No minlen 0


maxlen 4

Prefix 6890 is configured as long number prefix. Parameter PSTN access should be enabled
so that the internal users are accessible for outside users in PSTN.

Config add prefix


minlen 0 maxlen 8

dn 6890 callcategory Basic callattribute Inter cldpredeal No

Sequential: The IP phone sends registration messages to a SIP server at intervals of half of
the registration interval. If the active SIP server is fails, the IP phone registers with the
standby SIP server. If the standby SIP server fails, the IP phone registers with the local SIP

server. Services are interrupted during the SIP server switch. After the IP phone registers
with a SIP server successfully, its services become available again.

Simultaneous: The IP phone registers with the active, standby, and local SIP servers at the
same time. If the active SIP server fails, the standby SIP server takes over the services. If the

standby SIP server also fails, the local SIP server takes over the services. The switching
takes about 3s, which cannot be perceived by users. After the switching, services are
running properly.

If the SIP users on the U1900 unified gateway use POTS phones connected to an IAD, you
must configure the connection data and allocate user numbers to ports on the IAD after
adding SIP users on the U1900 unified gateway.
If SIP users are registered with the U19xx unified gateway through the IAD, you need to
configure the IAD data to ensure the normal connection between the IAD and the U19xx
unified gateway after configuring SIP users on the U19xx unified gateway. Then allocate
the user numbers for the ports on the IAD.

Port numbers of the IAD boards map lines with different colors. For example, the blue and
white twisted pair cable from the OSU in slot 1 of the IAD maps port 7. To view the
mapping between ports and line colors, see "Pins of the User Cable" in the IAD Product

Documentation.

Verification

After the configuration is completed, if configured user numbers can be found, users
are successfully configured; otherwise, users fail to be configured.

Troubleshooting

When a message indicating that a number exists is displayed, do not configure this
number.
When a message indicating license limitation is displayed, the number of configured
users reaches the maximum number of users supported by the license.

1. C;
2. AB.

As business expands, company A sets a branch in Shenzhen and the extension numbers
range from 8000 to 8009, which are also used to interconnect with users in HQ.
Meanwhile, HQ is connected with two carriers via PRA and AT0 to realize the PSTN
outgoing calls.

Requirement:

For outgoing calls, a prefix 9 should be added before each called number.

9 is local prefix;

90 is DDD prefix;

900 is IDD prefix.

The long number of calling party is displayed.

Assume that the called number is 28970139 and the prefixes 2, 289, and 28970 are
configured in the called number analysis table. According to the maximum matching rule,
the system will select the prefix 28970, which is the closest to the called number. The

prefixes 2 and 289 do not comply with the rule.

config add prefix

config delete prefix

config modify prefix

show prefix

Configure Office Route Selection Code:

[%eSpace U1900(config)]#config add office selectcode no 1 repeatroute no

If a direct channel is available between two exchange offices, either of the exchange
offices is the office route of the other exchange office.
Sub-route:

If a direct channel or an alternative channel is available between two exchange

offices, there is a sub-route between the two exchange offices. The direct channel is
the direct sub-route while the alternative channel is the alternative sub-route.

Route:

A route is the group of sub-routes between two exchange offices. A route contains

one or more sub-routes and different routes may contain the same sub-route.

Configure Office Route

[%eSpace U1900(config)]#config add office no 1 officeselectcode 1

NOTE:

You are advised to bind each office route ID to a unique office route selection code.
If multiple office route IDs are bound to the same office route selection code, all
office route records under the office route selection code are added to the routing
table on the U1980 unified gateway when you configure the route analysis function,
which may cause that the routing table becomes full.

The Media stream encryption parameter indicates whether to encrypt RTP media
streams to ensure data security.

Configure intra-office prefixes and outgoing prefixes for U19xx unified gateway users so
that U1900 unified gateway users (intra-office users) can make calls with each other or to
external users.
A call prefix is the string consisting of first few digits or all digits in called numbers. If the
called number is 1234, the call prefix can be:

First digit: 1

First two digits: 12

First three digits: 123

Entire called number: 1234

The prefix groups configured on the U1980 unified gateway constitute a called number

analysis table of the system. If the called number analysis table contains multiple prefixes
for a called number, the system uses the maximum matching rule to match the called
number with a prefix during number analysis.

Assume that the called number is 1234 and the prefixes 1, 12, and 1234 are configured in

the called number analysis table. According to the maximum matching rule, the system
will select the prefix 1234, which is the closest to the called number. The prefixes 1 and 12
do not comply with the rule.

The basic service prefixes are classified into two types:

Intra-office prefix: used for intra-office and outer-office users to call intra-office
users. For example, if the intra-office number ranges from 7000 to 7099, you can
set the intra-office prefix to 7. When calling an intra-office user, you can simply dial
the user number such as 7001.

Outgoing prefix: used when intra-office users make outgoing calls, such as local
calls, national toll calls, or international toll calls.

Configure inter-office short number prefix:

config add prefix dn 8 callcategory basic callattribute local cldpredeal no


officeselectcode 1 minlen 4 maxlen 4 if4pstnprefix no

HQ is connected to two carriers via PRA and AT0 to realize the PSTN outgoing calls.

Requirement:

For outgoing calls, a prefix 9 should be added before each called number.

9 is local prefix;

90 is DDD prefix;

900 is IDD prefix.

The long number of calling party is displayed.

Key Parameter Description:

config add office selectcode no 2 repeatroute No


Parameter

Description

Office route selection


policy

Policy indicating which trunk is used when multiple trunks


are available between two offices.

Failure Processing Index

Index that allows the standby office route selection code


policy to be used when the active office route selection
code policy fails.

Whether to send the


two-stage dialing tone

Indicates whether to play the two-stage dialing tone to a


user when the user dials an outgoing prefix. To play the
two-stage dialing tone, set this parameter to Yes.

Configure office route :

config add office no 2 officeselectcode 2

NOTE:

You are advised to bind each office route ID to a unique office route selection code.
If multiple office route IDs are bound to the same office route selection code, all
office route records under the office route selection code are added to the routing
table on the U1980 unified gateway when you configure the route analysis function,
which may cause that the routing table becomes full.

The Media stream encryption parameter indicates whether to encrypt RTP media
streams to ensure data security.

Configure the number change index 0, deleting the first digit 9 of called number.

config add predeal index 0 changetype delete changepos 0 changelen 1

The priorities of the three number change modes are long number and short number,
number mapping, and number change in descending order. If a user meets the conditions
of the three number change modes, only the mode with the highest priority is used. If the
user has a long number, the long number is displayed on the phone of the called party.
The user who does not set the long number can use the mapping number if it is set. The
user who does not set the long number and number mapping uses the changed number.

Configure local outgoing call:

[%eSpace U1900(config)]#config add prefix dn 9 callcategory basic callattribute


local cldpredeal yes cldindex 0 officeselectcode 2

Configure DDD prefix:

[%eSpace U1900(config)]#config add prefix dn 90 callcategory basic callattribute


ddd cldpredeal yes cldindex 0 officeselectcode 2

Configure IDD prefix:

[%eSpace U1900(config)]#config add prefix dn 900 callcategory basic callattribute


idd cldpredeal yes cldindex 0 officeselectcode 2

SS7 is a common channel signaling system that complies with international standards. It
has the following advantages: High speed transmission, Large signaling capacity, Powerful
functionality and High flexibility and reliability.

SS7 meets the requirements of the Public Switched Telephone Network (PSTN) and
intelligent network (IN).

PRA refers to the control signal between terminal devices and networks. The signaling
structure is 30B+D (used in Europe and China) or 23B+D (used in North America and

Japan).

R2 is a type of channel-associated signaling. R2 was previously applicable to the


international telecommunication network and telecommunication networks in many
countries. The China No.1 (CNo.1) signaling is a type of R2. R2 consists of line signaling

and register signaling. The definitions of the line signaling and register signaling, however,
vary according to the country.

The U1900 supports Session Initiation Protocol (SIP).

Defined by the Internet Engineering Task Force (IETF), SIP is a protocol at the

application layer, which is used to set up, modify, and end multimedia sessions on
the IP network.

The following transmission protocols can be used for U1900 unified gateway.
Protocol

Description

Local Port

UDP

A connectionless transport layer protocol


that provides a simple but unreliable
transaction-oriented information
transmission service. UDP features less
resource consumption and higher
processing speed, but does not guarantee
that data packets can be sent to the
destination safely and completely.

5060 (mandatory)

TCP

A connection-oriented, reliable IP-based


5061 (recommended)
transport layer protocol. The processing
speed of TCP is lower than that of UDP, but
TCP supports retransmission of lost packets
and data verification to ensure that data
packets can be sent to the destination safely
and completely.

TLS

A cryptographic protocol that provides


communication security between two
communication application programs,
ensuring data confidentiality and integrity.

5062 or 5063
(recommended)

Set peer device domain name to U1911, IP address to 10.77.194.45, and number of calls
that a trunk can bear to 20.

Set the office route number to 1, peer device domain name to U1911,the peer port of sip
protocol to 5060, and number of calls that a trunk can bear to 20.

config add peercomponent domainname U1911 ip 10.77.194.45 tkcnum 20

config protocol sip officeno 1 domainname U1911 peerport 5060 maxcallnum 20

Key parameters
Parameter
Domain
name of the
peer device

Description
Domain name that uniquely identifies a peer device. This parameter is
mandatory, and the domain name must be unique.

Transmission There are four types: UDP, TCP. TLSServer and TLSClient.
protocol type NOTE: If you want to encrypt signaling using TLS, configure the
communication parties as TLSServer and TLSClient respectively.
Select office ID of the office route that a SIP trunk belongs to.
route number
Maximum
number of
restricted
calls

Maximum number of calls on a SIP trunk. If the number of incoming


and outgoing calls routed through a SIP trunk exceeds the maximum
number, the system automatically rejects new calls.

ISDN User / network interface in a large number of interface solutions. Uses two types of
interfaces.

BRI: Basic Rate Interface, ie 2B + D and B are the 64kbps rate of the digital channel,
D is 16kbps digital channel.

PRI: Primary Rate Interface, 30B + D or 23B + D, B and D are 64kbps digital channel.

Insert an MTU to slot 2, set the office route number to 2, and set the signaling type to
PRA:

Add a PRA link. Set the link number to 0, slot number to 2, E1 port number to 1. Set the
link position to user. This parameter must be set through negotiations with the peer
device.

config add board slot 2 type mtu officeno 2

config add pralink linkno 0 slot 2 trunkport 1 position user

Configure the clock source:

config system clocksource primaryslot 2 primarye1 1

Parameter

Description

Position

Position in a PRA link. The options are User Side and Network Side.

Link
Timeslot

Circuit ID of a link, which specifies the timeslot in the E1 or T1 link of a


PRA trunk.
When E1 is used, this parameter can be set only to 16.
When T1 is used, the value of this parameter is an integer ranging from 1
to 24.

CRC

CRC is a data verification mechanism. The values of the local office and
peer office must be the same. If the peer office enables the CRC, set this
parameter to Yes.

Whether to
send the
abnormal
tone

Indicates whether to play an announcement based on the returned cause


code when you hang up.
If this parameter is set to No, the system plays a tone when you hang up.
The default value is Yes.

The clock source is configured on one device so that the peer device can synchronize its
clock with this device. The configuration of a clock source prevents the frame slip (voice
packet loss).

If the U1900 unified gateway connects to multiple devices through E1 or T1 trunks,


configure the clock source on any of these devices. The U1900 unified gateway will
synchronize time with this device.
If the U1900 unified gateway provides the clock source, you do not need to

configure a clock source.


Parameter

Description

Select clock
source

Select Local device if the local device is used as the clock source. Select
Peer device and configure the clock source if the clock of the peer device
is used as the clock source.

Active clock
slot No

Slot of the board that uses the QSIG trunk for interconnection.

An AT0 trunk is an analog circuit trunk. It uses a analog subscriber line to connect an FXO
port in the local office to an FXS port in the upper-level office to implement the dedicated
line function.
To support concurrent incoming and outgoing calls, you can apply to the local carrier for
multiple lines, and allocate a number to each line. Based on the policy for using line
resources, intra-office users are divided into DID users (that can occupy dedicated lines
exclusively) and non-DID users.

The requirement:

Configure DDI(Direct Dial-In), to bind the DDI user with a specific AT0 line.

Configure the incoming call to transfer to Switch Board by default. Non-DDI calls will
be transferred to a preset number which is usually Switch Board number.

The calling party can dial the PSTN number corresponding to a dedicated line to directly
call a DID user, or dial PSTN numbers corresponding to other lines to call any intra-office
users through the automatic switchboard.
When a DID user makes an outgoing call, the call must be routed through the dedicated
line of the user. When a non-DID user makes an outgoing call, the system selects a line
from other lines to route the call based on the line selection rule.

1. A;
2. B.

Outgoing call;

Display the long number of caller;

The caller without long number shows switchboard number 68908888;

Delete the first digit 9 of the called number.

If a user meets the conditions of the three number change modes, only the mode with the
highest priority is used.

If the user has a long number, the long number is displayed on the phone of the
called party.

The user who does not set the long number can use the mapping number if it is set.
The user who does not set the long number and number mapping uses the changed
number.

Configure long number & short number:

Config add subscriber eid 6000 dn 6000 longdn 68906000 Dnstep 1 number 10

A user has two numbers: a short number and a long number allocated by the PSTN. The
short number is used for the calls between intra-office users. The long number is displayed
when you make an outgoing call or is used for outer-office users' direct dialing in.

Configure prefix 6890 for long number so that outer-office users can directly dial in.

Config add prefix dn 6890 callcategory Basic callattribute Inter cldpredeal No


uselongcli Yes If4pstnprefix Yes

Configure prefix 9. If the user who dials prefix 9 has a long number, the long number is
displayed on the phone of the called user.
Similarly, when configuring DDD prefix 90 and IDD prefix 900, display long calling
numberneed to be set as yes.

Configure number mapping:

Config add numbermap innernum 6010 externnum 68906543

The number mapping is used to implement irregular number conversion.

Procedure

Login on to the U1960 web configuration page.

Choose Trunk > Office Route Configuration.

Click Number mapping.

Click Create.

Configure the number mapping for user 6010.

Key parameters:

Parameter

Description

Internal number

Short number used in an intra-office call.

External number

Long number allocated by the carrier.

Index 0: The first digit of the number is deleted.

Index 1: The entire number is changed to 68908888.

Index 2: The prefix 5 is added to the number.

Number change method that specifies the way that the system uses to change the
calling or called numbers. The options are as follows: Change Number Change a
specified string in the original number.

Delete Number Delete a specified string from the original number.

Insert Number Add a string to a specified position in the original number.

Insert by Length Add a string to a specified position in the original number based on
the number length.
Change by Length Change a specified string in the original number based on the
number length.
Delete by Length Delete a specified string from the original number based on the
number length.
Insert by Call Source If a number is within a specified call source, prefix the area
code of the call source to the number.

Key parameter description:

Parameter

Description

Number change
index

Unique index of a number change rule.

Change type

Number change method that specifies the way that the system uses to
change the calling or called numbers. The options are as follows: Change
Number Change a specified string in the original number.
Delete Number Delete a specified string from the original number.
Insert Number Add a string to a specified position in the original number.
Insert by Length Add a string to a specified position in the original number
based on the number length.
Change by Length Change a specified string in the original number based
on the number length.
Delete by Length Delete a specified string from the original number based
on the number length.

Number length

Original number length. For example, if the original number is 9999, the
number length is 4.

Starting position

Position from which the number change begins. This parameter specifies the
position to change some digits in, add digits to, or delete digits from an
original called or calling number. Digits are changed from the left in the
original number, and the first position code is 0.

New number

New number after the number change or inserted number.

1. BC

The intelligent routing service automatically selects office routes when an IP trunk or a
TDM trunk is faulty, and provides routing polices to increase communication reliability and
minimize communication costs.
The intelligent routing service includes:

Routing by time segment: Different time indexes are set for different office routes.
Each time index corresponds to a specified time segment (accurate to hour). Based
on the current time, the IP PBX searches for the time index for an outgoing call and

selects the related office route.

The IP PBX allows multiple office route selection codes to share one office route.
When an office route is added to multiple office route selection codes, multiple
routing policies apply to the office route. The IP PBX automatically selects the

routing policy based on the outgoing prefix that a user dials.

Routing by charge rate Different charge rate reference values are set for different
office routes. The IP PBX preferentially selects the office route with the lowest
charge rate reference value for outgoing calls. If all trunk circuits are busy for this
office route, the IP PBX selects the office route with the second lowest charge rate
reference value.

Rerouting upon a routing failure When a call fails to be routed based on the office
route selection code, the IP PBX selects a new route based on the standby office
route selection code corresponding to the failure processing index.
Route load balancing The IP PBX balances traffic among multiple preset routes.

The IP PBX elects office routes based on office route IDs in ascending order till an
office route that contains idle circuits is found. The later calls will elect from the next
office route after an office route is selected by the preceding call.

Routing by percentage Different percentages are set for different office routes.
The IP PBX selects office routes in turn based on the preset percentages. Except the
office routes whose percentages are set to 100%, office routes are polled based on
preset percentages in descending order.
Routing by user right level Different office routes are set for different user right
levels. When a higher-right-level user makes an outgoing call, the IP PBX
preferentially selects the office route that is set for the user right level. If selecting
the office route fails, the IP PBX selects the office route that is set for a lower user
right level.
When routing by user right level is used, the IP PBX preferentially uses an office
route at the user right level to route calls. If no circuit is idle, the office route that is
set for a lower user right level is used. Office routes that are set for the same level
are elected based on office route IDs in ascending order. To ensure good voice
quality of calls for higher-right-level users, these users are allocated with highperformance office routes. Lower-right-level users, however, are not allowed to use
these office routes.

Trunk link balancing The IP PBX preferentially selects a trunk with more idle circuits
to balance loads among available trunk links.

When the call fails to be routed based on the office route selection code, select a new
route based on the standby office route selection code corresponding to the failure
processing index.

Parameter description:

Parameter

Description

Failure Processing Index

Unique index of a failure processing policy.

Rerouting policy upon failure

Failure selection policy used by a failure processing index.

Selection node of Standby


office route

Office route selection policy used for rerouting.

Called number change

Indicates whether to change the called number after rerouting.

Calling number change

Indicates whether to change the calling number after rerouting.

Policy for selecting among office routes with the same office route selection code. The
options are as follows:

None : Office routes are polled based on office route IDs in ascending order until an
office route that contains idle circuits is found.

By Time : Office routes are polled based on office route IDs in ascending order until
an office route that is valid at the current system time and contains idle circuits is
found.
By Charging Rate: Office routes are polled based on charging rates in ascending
order until an office route that contains idle circuits is found. Office routes with the
same charging rate are polled based on office route IDs in ascending order. If no idle
circuit is found, office routes configured with no charge rate are polled based on

office route IDs in ascending order until an office route that contains idle circuits is
found.

By Load Sharing : Office routes are polled based on office route IDs in ascending
order until an office route that contains idle circuits is found. The later calls will poll
office routes from the next office route after an office route is selected by the
preceding call.

By Load Sharing Percentage: Except the office routes whose load percentages are
set to 100%, office routes are polled based on the load percentages in descending
order.
User Level : Office routes are polled based on user priorities in ascending order until

an office route that contains idle circuits and has an equal or lower user priority is
found. Office routes with the same charging rate are polled based on office route
IDs in ascending order.

Load Balancing: Office routes are polled based on office route IDs in ascending
order until an office route that contains the maximum number of idle circuits is
found.

The U1900 polls office routes based on the office route ID in ascending order until an
office route that contains idle circuits is found. Subsequent calls will poll from the next
office route after the office route that is selected by the preceding call.

Configure office route selection policy

config add office selectcode no 1 repeatroute no selecttype loadshare

Configure the office route selection code.

Set the office route selection code to 1. Office route selection code 1 selects office
routes based on load balancing.

Configure the office route

Office route 0 is used for the PRA trunk, with office route selection code being 1

Office route 1 is used for the ISUP trunk, with office route selection code being 1

Configure office route

config add office no 0 officeselectcode 1

config add office no 1 officeselectcode 1

1. C.

This feature applies to UC terminals of all types.


IVR processes and announcements can be customized. A maximum of 10 levels of IVR
menus are supported. The keys for repeatedly listening to an IVR menu and for forwarding
a call to a voice mailbox are configurable. Different announcements can be played within
different time segments.

The U1900 series unified gateway supports a maximum of 256 automatic switchboards
and 245 customized scripts. A maximum of 246 notifications can be recorded. Among

these notifications, one is for the default automatic switchboard, and the other 245 are for
the customized automatic switchboard. Each customized switchboard can have one
notification. The maximum duration for a single IVR file is:

U1981/U1960/U1911: 300 seconds

U1980/U1930/U1910: 32 seconds

NOTE:

For a U1980/U1930/U1910 unified gateway, you can load custom announcements


only by uploading the voice.zip file, instead of using the web system to record
custom announcements or using an IP phone to record custom announcements or
notifications.

Typical Scenario of the VU Service

After the automatic switchboard is configured, outer-office users hear "Thanks for
calling XX. Please dial the extension number. For the agent service, dial 0."when
dialing the automatic switchboard number.The customer dials 0 for the agent
service. The customer dials an extension number to reach the desired enterprise
user.

Generally, the unified gateway at the central node plays the welcome announcement and
IVR menus for the calling party who makes calls to the automatic switchboard. In the
centralized network scenario, if a branch deploys a local node, the administrator can

configure the local node to play announcements and collect digits for the switchboard.
This helps save the bandwidth from the central node to the local node. When a PSTN user
connects to the automatic switchboard from the local node, the local node plays the
switchboard announcements and collects digits for the switchboard

Prerequisites

Numbers have been allocated to intra-office users on the unified gateway.

The trunk that connects to other devices has been configured.

When an external user dials the automatic switchboard number, the user hears the
announcement "Please dial the extension number." The user enters the extension number.
Then, the call is routed to the user inside the enterprise.

VU scripts of the automatic switchboard have been pre-configured into the system by
default. Users can configure the automatic switchboard access code to enable automatic
switchboard functions.
Use an administrator account to log in to the web management system of a U1900 series
unified gateway.
Configure an automatic switchboard access code.

Choose Trunk Management > Prefix Configuration.

Click Create.

Configure prefix 28887888.

NOTE:

The automatic switchboard access code (28887888 in this example) must be


different from user numbers. If the automatic switchboard access code is the same
as a user number, the call to the number is routed to the user, not the switchboard.

Key parameters:

VU loop

Whether the switchboard prompt tone is

returned again when the switchboard cannot be called

VU outgoing right
Whether the switchboard can forward calls to
external extensions. To prevent intra-office users from dialing the switchboard to

make toll calls, it is recommended that you do not set it to DDD or IDD.

Recording and loading a single voice file: using an IP phone to record

For U1911/U1960/U1981, if you do not have a high requirement on the audio


quality and need to modify custom announcements frequently, you can use a phone
to record custom announcements in real time. Recorded custom announcements
will be automatically loaded to the unified gateway.

A user can dial the automatic switchboard number and the recording prefix to
record a notification. After a notification is recorded, it is played prior to the

automatic switchboard announcement when a user dials the automatic switchboard


number.

Recording and loading a single voice file: using dedicated recording software to record

When U1911/U1960/U1981 is used, it is recommended that dedicated recording

software be used to record voice files if you have a high requirement on the audio
quality and the voice files seldom change.

Customizing and loading multiple voice files

U1911/U1960/U1981/U1980/U1930/U1910 allows you to upload a package to


replace customized voice files. For U1980/U1930/U1910, you must this method to
load voice files.

The audio files must meet the following requirements. If the format is incorrect, audio files
cannot be played.

The codec is G.711a.

The announcement recorded is in mono.pcm or .wav format.

The sampling rate is 8 kHz, and the sample precision is 8 bits.

The duration does not exceed 5 minutes, and the size does not exceed 2.5 MB.

Audio files must be named after voice channel ID, for example, 252.pcm.

The length of the audio file name plus suffix name cannot exceed 128 bytes.
Otherwise, an error will occur when loading audio files through the web
management system.

After being loaded, custom announcements are stored in the SD card of the SCUB. Every
time the MTUB restarts or a new MTUB is added, the MTUB reads all custom
announcements from the SD card of the SCUB, The synchronization process takes about 1
minute and the announcements take effect immediately.

Custom announcements can be loaded only if SCUB boards exist on the unified gateway
and SD cards can be inserted into the SCUB boards.

NOTE:

Custom announcements can also be packaged into voice.zip, but the format and
size of audio files for custom announcements must meet the format and size
requirements on default announcements.

Loading Audio Files

Use an administrator account to log in to the web management system of a U1900


series unified gateway.

Choose System > Voice File Management.

Click Create to add a voice file and click OK.

Verification

Dial #79*N# as a user who has the agent rights to verify that the announcement for
voice channel N is the recorded custom announcement. An agent can hear the
announcements for all voice channels.

(Optional) Configuring the Time Segment Index

The system supports period-specific automatic switchboard for playing different


announcements at different time segments. Skip this section if period-specific
automatic switchboard is not required.

Use an administrator account to log in to the web management system of a U1900


series unified gateway.
Choose System > VU Management > Time index.

NOTE:

Do not create two time segment policies that have the same time segment. If such
policies exist, announcement may encounter exceptions if users dial the
corresponding automatic switchboard number within that time segment.
If two time segments overlap, the automatic switchboard corresponding to holidays
is preferentially used among all the overlapping time segments. If the overlapping
time segments all belong to holidays or all do not belong to holidays, the automatic
switchboard corresponding to the time segment that has a smallest index is
preferentially used.

Use an administrator account to log in to the web management system of a U1900 series
unified gateway.

Choose System > VU Management > Customized automatic switchboard Configuration.

Click Create.

If the automatic switchboard is not required to play announcements by time segment, you

only need to set the first time segment. Set the values of the other Time segment fields
to none.

If multi-level voice menus are not required, you do not need to click Create menu to set
related parameters.

By default, the system allows users to press * to return to the previous menu and press #
to return to the root menu. No additional configuration is required.
Several automatic switchboard can share one console.

Parameter

Description

Script name

Name of the script that is generated after the automatic switchboard is configured. The
script name is user-defined. The script can be downloaded so that it can be load to
another automatic switchboard to implement the same functions as this automatic
switchboard.

Extension number Maximum length of an extension number when a call is forwarded from the automatic
length
switchboard to an extension.

Inter-digit timeout Interval between two dialing attempts. If the interval between a user's two conseutive
time
dialing attempts reaches this duration or if the user performs no operation within this
duration after the system plays an announcement, the system determines that the
dialing ends or the operation times out.
Play voice interval Duration after which the system determines a dialing timeout and plays an
announcement again because no operation is performed within this duration. If no digit
is dialed when the number of times the system plays the announcement reaches the
value of Play count, the system releases the call.
Prefixed number

Number that the system automatically prefixes to an extension number when a user
dials the automatic switchboard number and an extension number.
Generally, you do not need to specify this number. If required in some scenarios, set
this parameter based on the data plan.

Time segment

Time segment used to specify the announcements and IVR menus to be played. The
time segments are set on the Time index tab page.
The default value is none, indicating all time segments.

Welcome voice

Number of the voice channel for the welcome announcement that is played when a call
is made to the automatic switchboard in the specified time segment.

Menu voice

Voice menus to be played after the welcome announcement is played. For example, to
query numbers, please dial 0. To query services, please dial 1. To switch to the manual
service, please dial 9.
Voice menus can be recorded separately from or together with the welcome
announcement. If you have recorded voice menus together with the welcome
announcement, set this parameter to none.

Extension support. A user can directly dial the extension number of an intra-office user
Extension not
support
to reach this user after hearing the corresponding announcement.
Extension support Extension not support. A user can only dial a menu code to access a lower-level menu
after hearing the corresponding announcement.
Play count

Times an announcement is replayed upon an incorrect user input.

Create menu

Click to create a lower-level menu.

Parameter

Description

Description of menu items

09, *, #, T/O, Failure transferred. When configuring submenu processes for


an automatic switchboard, define the system response process when a user
presses a single key (digits from 0 to 9, *, or #) and define the system response
process upon situations such as a user operation timeout or a call transfer
failure.
NOTE:By default, the system allows users to press * to return to the previous
menu and press # to return to the root menu. If you have redefined the
functions for the two keys in the submenus, the new definitions take effect.
Switch to assigned number. After a user presses this key, the system forwards
a call to a preset number, for example, a SoftConsole access number or a
manual service number.
Play voice. When a user presses this key, the system plays the announcement
corresponding to a specified channel number, for example, the introduction to
an enterprise.
Switch to voice mailbox. When a user presses this key, the system forwards a
call to a voice mailbox.

If this parameter is empty, the system plays an announcement


notifying that the user is being redirected to a process for
leaving a voice message and prompts the user to enter the user
number to which a voice message is left. Then the user enters
the process for leaving a voice message.

If a user number for a public voice mailbox is entered, the


system plays an announcement notifying that the user is being
redirected to a process for leaving a voice message and then
directly enters the process.
Before setting this parameter, configure the voice mailbox service and ensure
that the voice mailbox permission has been granted to related users.
Get voice message. When a user presses this key, a call is forwarded to a
preset voice mailbox access prefix. Then the user enters the process for
retrieving voice messages. The user can retrieve voice messages as prompted.
Before setting this parameter, configure the voice mailbox service and ensure
that the voice mailbox permission has been granted to related users.
Switch to sub menu. Select this option if you need to create sub menus. You
can also specify the channel numbers for playing sub menu announcements to
notify users of the sub menus.
Before setting this parameter, configure the voice mailbox.
Go back to previous menu. When a user presses this key, the system plays the
announcement for returning to the previous menu.
Replay the menu voice. When a user presses this key, the system repeatedly
plays the announcement for a menu.
Transfer calls to the extension. When a user presses this key, the system
allows the user to dial an extension number.

click an icon in the Operation column next to a script name to modify or download a script
file. The downloaded script file can be uploaded to this device or another device.
NOTE:

The following takes a .cfg script file that is customized on the web management
system for an automatic switchboard as an example. The script files that you upload
or download on the Script Management page are all service VU and IVR script files
supported by the system, for example, the script file for the VMS. The script files are

in .txt format.

Use an administrator account to log in to the web management system of a U1900 series
unified gateway.

Choose Trunk Management > Prefix Configuration.

Click Create.

NOTE:

The access code (28887888 in this example) must be different from user numbers. If

the access code is the same as a user number, the call to the number is routed to
the user, not the switchboard

Verification

Verify the period-specific automatic switchboard service configuration by dialing the

automatic switchboard number.

The system will play the automatic switchboard prompt tone when a user fails to call
someone by dialing an extension number through the switchboard.

Use an administrator account to log in to the web management system of a U1900


series unified gateway.

Choose Trunk > Prefix Configuration.

Create the automatic switchboard prefix and set VU cyclic parameters.

Verification

Make a call to the automatic switchboard.

Pick up the phone and dial automatic

switchboard number 28887888. The call is connected and the announcement


"Please dial the extension number" is played.

Make a call to an extension number.


Dial an extension number 81000 as
prompted.
The call fails and the announcement "Please dial the extension
number" is played.

Verification

Make a call to an extension number. Pick up the phone and dial 2888788881000.
The call is connected.

1. B.

Upon completion of this course, you will be able to:

Understand the voice mailbox function.

Understand the Huawei voice mailbox features.

Master the basic service configuration of VMS.

U1980 can not support the Built-in Voice Mailbox Services function.
The voice mailbox feature requires the corresponding license, and the number of
concurrent messages is limited by the license.
Built-in voice mailbox specifications
Model

Maximum Number of
Voice Mailbox Users

Maximum Number of
Concurrent Messages

U1911

100

12

U1960

1000

30

U1981

1200

30

Users can leave, retrieve, delete, and play voice messages and customize greetings. If a
phone has a message indicator, the indicator turns on when a voice message is received.
Voice mailbox services include call transfer to voice mailbox unconditional (CTVMU), call
transfer to voice mailbox on no reply (CTVMNR), call transfer to voice mailbox on busy
(CTVMB), call transfer to voice mailbox offline (CTVMO).

Call transfer to voice mailbox unconditional (CTVMU) service: A service that allows all the
calls to a registered user to be forwarded to the UMS unconditionally. Then, the caller can

leave messages according to prompts.

Call transfer to voice mailbox on no reply (CTVMNR) service: A service that allows a call to
a registered user to be forwarded automatically to the UMS if this user does not answer
the call within 20 seconds. Then, the caller can leave messages according to prompts.

Call transfer to voice message on busy (CTVMB) service: A service that allows a call to a
registered user to be forwarded automatically to the UMS if this user is busy. Then, the
caller can leave messages according to prompts.
Call transfer to voice mailbox offline (CTVMO: If a user activates the call transfer to voice

mailbox on offline (CTVMO) service, incoming calls are transferred to the user's voice
mailbox if the user is offline. Calling parties can leave voice messages as prompted.

NOTE:

The value of Number of VMS Channels refers to the maximum number of calls that
can be processed at a time for the voice mailbox services, and must be set based on
the license information obtained in last step.

On each MTU board, the value of Number of VMS Channels must be the same as
the value of Voice Mail nums in the license.
The concurrent VMS channel is limited by the U1900 license, the SD card in MTU
board is just used to store the voice data.

Verify the number of SD cards.

Messages in the voice mailbox service are stored in the SD card of the MTU. Only
one SD card can be installed on each MTU board, and the SD card must be installed
in slot SD0. The storage space required for voice messages can be calculated with
the following formula: Storage space required for voice messages = [Size of PCMencoded voice data per second (8 KB) x Maximum duration of a voice message
(120s) + File header and index (0.1 KB)] x Maximum number of voice messages for a

user (20) x Number of users. According to this formula, an 8 GB space can support
the voice mailbox service for about 430 users.

Note:

Information, such as the total space of the SD card, available space of the SD

card, and number of voice messages for intra-office users, can be viewed on
the User > VMS Statistics page of the web management system.

NOTE:

If the scripts do not exist, load the corresponding script. Generally, it is unnecessary
to modify the preloaded VMS script. In case of any special situations, for example,
when the script is deleted, reload the script. When reloading the script, select the
VU type.

The scripts are already loaded by default.

Voice message: Used to record voice messages.

VMS access code: Used to retrieve voice messages and customize personal greetings.

Access the web management system, and choose Trunk > Prefix Configuration.

Click Create, and set the prefix for leaving voice messages and VMS access code.

NOTE:

You can also run the config add prefix dn dn callcategory vu callattribute

<vuvmsleavemsg | vuvmsaccesscode> cldpredeal no command to create a voice


message prefix and a VMS unified access prefix, but do not
set ifshareprefix to yes (default value: no).

Voice message: Used to record voice messages.

VMS access code: Used to retrieve voice messages and customize personal greetings.

Access the web management system, and choose Trunk > Prefix Configuration.

Click Create, and set the prefix for leaving voice messages and VMS access code.

NOTE:

You can also run the config add prefix dn dn callcategory vu callattribute

<vuvmsleavemsg | vuvmsaccesscode> cldpredeal no command to create a voice


message prefix and a VMS unified access prefix, but do not
set ifshareprefix to yes (default value: no).

Access the web management system, and choose User > SIP User.(For a POTS user, choose
User > POTS User.)
Find and select the user number for which you want to enable voice mailbox services and
click Service Configuration.

Under Complex service, click Configure next to Forwarding service.


In the window that is displayed, find the call transfer type to be enabled, and set the
parameters
NOTE:

After a type of call transfer to voice mailbox service is enabled for a user, the user
must activate the service for it to take effect.

Parameter

Order of priorities for call forwarding to phone number and


call forwarding to voice mailbox. If these two services are both
enabled for a user, the selected call forwarding mode prevails.
NOTE:
Users can log in to the self-service portal to adjust the
priorities of their call forwarding services.

Priority

Right
Voice
mailbox
number
Voice
mailbox

Description

Service

Indicates whether to enable the call forwarding to voice mailbox


service permission of the corresponding type.
Number of the voice mailbox created for storing voice
messages.
Indicates whether to register the voice mailbox service of the
corresponding type for the user. You may not register the
service for the user here. Users can register the services
themselves.

Waiting time threshold, crossing which the call forwarding on


Wait time no reply service is triggered. This parameter is available only for
call forwarding to phone number on no reply.

CTVMU : Call transfer to voice mailbox unconditional

CTVMNR : Call transfer to voice mailbox on no reply

CTVMB : Call transfer to voice mailbox on busy

CTVMO : Call transfer to voice mailbox offline

The preset prefix for directly leaving voice messages is *23*. If you need to customize the
prefix base on the user habit, you can create a prefix for directly leaving voice messages as
follows

Access the web management system, and choose Trunk > Prefix Configuration.

Click Create, and create prefix 91003 for directly leaving voice messages.

NOTE:

You can also run the config add prefix dn dn callcategory new callattribute
leavemsgdirect cldpredeal no command to create a direct message prefix, but do
not set ifshareprefix to yes (default value: no).

Activating or Deactivating the Service


If the service is not activated when the voice mailbox service permission is assigned to a
user, the user must activate the service for the service permission to take effect. The
service can be activated or deactivated in either of the following ways:

Method 1: Activate or deactivate the service in the web self-service system.

Log in to the web self-service system.

Choose Self-serviceServiceForwarding service.

Set the service priority, activate or deactivate the service, and click OK.

Method 2: Dial a prefix for activating or deactivating the service on an IP phone.

Service Type

Prefix for Activating the


Service

Prefix for Deactivating the


Service

CTVMU

*91#

#91#

CTVMNR

*93#

#93#

CTVMB

*92#

#92#

CTVMO

*98#

#98#

When entering identity verification information,

Intra-office users need to enter only the password. The intra-office users can also
press the star key (*) to retrieve voice messages for others.
Outer-office users need to enter the voice mailbox user number and password.

When the system is playing a voice message, users can press any button to stop playing
the voice message.

Inter: intra-office call.

Local: local call.

DDD: national toll call.

IDD: international toll call.

Deploy an U1900 series gateway and connect to an PSTN network.


An internal user 6000, can call an internal number 6001, also can call a local operators
number 28562933. This user cannot make a toll call.

A call prefix is the string consisting of first few digits or all digits in called numbers.
Intra-office prefix: used for intra-office and outer-office users to call intra-office users. For
example, if the intra-office number ranges from 7000 to 7099, you can set the intra-office
prefix to 7. When calling an intra-office user, you can simply dial the user number such as
7001.

Outgoing prefix: used when intra-office users make outgoing calls, such as local calls,
national toll calls, or international toll calls.
Configure the prefix:

Log in to the web management system.

Choose Trunk > Prefix Configuration and click Create.

Configure a prefix in the Create Prefix dialog box

Users at different right levels have different rights to make calls. Call rights can be control
by time segment so that users at a level can make calls only in a specified time segment.

Different levels have different new service rights. The service rights of lower-level users are
the sub rights of the service rights of higher-level users.
If trunks are insufficient, reserved trunks can be used based on the preset user level.
If there is no available reserved trunk, higher-right-level users who make outgoing calls can
preempt the trunks assigned for lower-right-level users. For example, a user at the super
level can preempt trunks occupied by users at the advanced, normal, and default levels.
When the trunk of a lower-right-level user is preempted, the user hears the busy tone

(configurable), and the call is released.

Log in to the LMT and access the command tree.


Choose System Management > System Information from the command tree. Double-click
Config Set Workingtime.
Enable and set the specific period.

Log in to the LMT and access the command tree

Set the PriorLevel.

The 32-level right is a customized right. When used with call prefixes, the 32-level right
can prevent specified users from making calls.
The blacklist and whitelist call restriction allows you to accept or reject specified calls.
VoIP domain call restriction is a call restriction policy for SIP users and SIP trunks. Using
this policy, the number of calls for each VoIP domain is restricted for bandwidth control.

9 is the local prefix.

User 6000 and 6001 are the default level, but they have different call rights.

Configure an outgoing prefix.

config modify prefix dn 9 callcategory basic callattribute local cusattribute cus1

Login the web interface for the configuration.


Choose SIP User > User, choose a number you want to set, then click Service
Configuration.

Choose Outgoing call right, click Configure.

Chosse Custom 1.

System blacklist and whitelist

The U1900 series unified gateway adds users to different call barring groups to
control call rights.Call restriction groups include: blacklist, whitelist, and common
call barring groups. Users are added to the common call barring group by default.
The administrator can add users to other groups based on actual needs

Personal blacklist

Users can configure their own blacklists to filter out unwanted calls. A blacklisted
user will hear a call barring prompt tone.

The priority of personal blacklists is lower than that of system blacklists and
whitelists.

The blacklist and whitelist call restriction allows you to accept or reject specified calls.

If the blacklist and whitelist are not configured, the default value is Ordinary.

Add number 9168 to the called party blacklist.

config add blackandwhite dn 9168 calltype callee grouptype blackgroup

Add number 6002 to the calling party whitelist.

config add blackandwhite dn 6002 calltype caller grouptype whitegroup

U1900 Blacklist and whitelist specifications

Item

U1910/U1911

U1930/U1960

Maximum number of users in


a system blacklist or whitelist

Maximum number of users/2

Maximum number of a
personal blacklist

1000

Maximum of blacklists
configured by a user

10

2500

U1980/U1981

10000

A PIN code is a call restriction service password assigned to some users. Common users are
allowed to dial the numbers with certain prefixes depending on the call rights assigned to
the phone numbers they use, while users with PIN codes can dial more numbers within the

PIN-Code service rights. CDRs generated for calls with PIN code authentication record the
user accounts, calling numbers and called numbers corresponding to a PIN code.

After a user picks up an IP phone with the PIN-Code service right enabled and dials a
number:

If the called number prefix is required to have call right authentication, the user
needs to enter the PIN code according to the announcements and make a call based
on the PIN-Code service right.
If the called number prefix is not required to have call right authentication, the user

can directly make a call based on the call rights assigned to the phone number.

NOTE:

The PIN-Code service cannot coexist with the password-based call barring service or
the simple card number service. You can run the show pwdcalllimit command to

query whether the password-based call barring service and the simple card number
service are enabled.

Keep PIN codes secure to avoid unnecessary charge caused by unauthorized use of
the PIN codes.

A PIN code cannot be changed by users. If you want to modify the PIN code, run
the command config modify pwdcalllimit.

The administrator in an enterprise enables the PIN-Code-based call barring service for an
international toll call prefix (for example, 900) on all phones.
To enable the boss to make international toll calls, the administrator configures call barring
accounts and PIN codes for the boss.

Other users can make intra-office, local, and domestic calls, but cannot make international
toll calls starting with 900.
The boss can enter the PIN codes on any phone to make international toll calls through the
IVR dialing mode. User numbers and call barring accounts used to dial phone numbers are
written into call records, which can be checked by the administrator.

PrerequisiteLog in to the device using the LMT. Choose Command Delivery > Script
Load to run the PIN-code-based call barring service script.
PIN-Code Call Restriction Configuration Process
1. Enable the PIN-Code service;
2. Create a PIN code;
3. Enable the service for IP phone user;

4. Configure prefix;
5. Save.

Configuration:
1. Choose User > Global Service > Call Restriction.

2. On the Call restriction type, select Pin-Code and click OK to enable the Pin-Code call
barring service.
3. Click Create and create a PIN-code account and password.

A user can use this account and password to make calls based on the call permission. For
example, you can configure a PIN-code account for making international toll calls.
Configuration:
1. Configure an account and password.
2. Set the Outgoing call rights as International toll call.
3. Click OK.

You can add the service permission for users in a batch. Users with PIN-code accounts and
passwords can use the PIN-code-based call barring service on any phones with the service
permission enabled.
Configuration:
1. Choose User > SIP User or User > POTS User.
2. Choose the number for which you want to configure call barring, click Modify.

3. set Password call limit to Yes, and click OK.

Add the password-based call barring attribute to a prefix.

Configuration:

1. Choose Trunk > Prefix Configuration.


2. Choose the prefix for which you want to configure call barring, click Modify.

3. set Password call limit to Yes, and click OK. For example, to restrict the
international toll call permission, set Password call limit to Yes for prefix 900.

In the upper right corner of the web page, click Save Data to save the configuration.

1. F;
2. F;

3. F;
4. BD

Note:

CDRServer doesnt need the license to support.

CDRServer doesnt support Linux system.

CDRServer doesnt need to install database, it just store the CDR file on the local
disk.

Functions of devices in a CDR system:

Unified gateway: Generates CDRs for all users connected to it and saves the CDRs to its
CDR pool. The unified gateway periodically pushes CDRs to the CDRServer in an
unsolicited manner or upon receiving the CDRServer's request.
CDRServer: Parses the obtained CDRs and stores them as binary files ( .bill) to a specific
location on the CDRServer. The CDRServer can proactively send a request to the unified
gateway for obtaining CDRs after CDR transfer is enabled under CDR Console > Call
record transfer switch of the CDRServer.
BMU: Obtains call records from the CDRServer through an FTPS interface, and writes the
CDRs into the database.
The administrator can manage CDRs on the BMU, such as querying and deleting CDRs.
Enterprise users can query their own CDRs on the BMU self-service platform.
Third-party billing system: Obtains CDR files from the CDRServer through an FTP or FTPS
interface.

CDRServer functions

Function

Description

Querying CDRs

Users can use query CDR files from the CDRServer using any combinations of the
following search criteria: calling number, called number, IP address of the device
where CDRs are generated, and CDR generation time.

Deleting
historical CDRs

Users can delete CDR files from the CDRServer to free up disk space.

Viewing CDR pool


information

Users can view the CDR pool status of eSpace U1911/U1960/U1980/U1981 on the
CDRServer according to the eSpace U1911/U1960/U1980/U1981 IP address.

Controlling CDR tran Users can set the CDR transfer status. When new CDRs are generated in
sfer
the CDR pool of eSpace U1911/U1960/U1980/U1981, the CDRServer determines
whether to move CDRsfrom the CDR pool to the original CDR folder according to
the CDR transfer status.

Functions of devices in a CDR system:

Unified gateway: Generates CDRs for all users connected to it and saves the CDRs to
its CDR pool. The unified gateway periodically pushes CDRs to the CDRServer in an
unsolicited manner or upon receiving the CDRServer's request.

CDRServer: Parses the obtained CDRs and stores them as binary files ( .bill) to a
specific location on the CDRServer. The CDRServer can proactively send a request to
the unified gateway for obtaining CDRs after CDR transfer is enabled under CDR

Console > Call record transfer switch of the CDRServer.

BMU: Obtains call records from the CDRServer through an FTPS interface, and writes
the CDRs into the database.The administrator can manage CDRs on the BMU, such
as querying and deleting CDRs. Enterprise users can query their own CDRs on the

BMU self-service platform.

Third-party billing system(product from Yaxun Hongda is recommended) : Obtains


CDR files from the CDRServer through an FTP or FTPS interface.

Three services are contained in the CDR Server - Running window.

MML: provides interaction between man and machines and parses interactive
commands between the CDRServer and the Unified Gateway, such as the call record
searching command.

CDR: saves CDRs from the Unified Gateway and provides the call record searching
and call record file processing services.

You can start or stop all services at any time.

Start all services:

Choose System > Start All.

Stop all services:

Choose System > Stop All.

The capacity of the CDR pool varies depending on the unified gateway model. For
the U1980 and U1981, a maximum of 480,000 CDRs can be stored; for the U1911 and
U1960, the value is 50,000; for the U1910 and U1930, the value is 30,000.

Network Type

Configuration on the Unified Gateway

Configuration on the
CDRServer
Single-node network Enable the function of generating CDR files and configure the Configure IP address of
CDRServer IP address on the unified gateway at a single node. all unified gateways.
Multi-node
Enable the function of generating CDR files and configure the Configure IP address of
centralized network CDRServer IP address on the unified gateway at each node.
all unified gateways.
In normal network conditions, numbers of all branches are
registered with the central node, the central node routes all
calls, and the central node generates CDRs and send them to
the CDRServer. Unified gateways at branch nodes do not
generate CDRs.
In case of local regeneration, a branch node disconnects from
the central node. The unified gateway at the branch node
generates CDRs and buffers them in their own CDR pool. After
the network recovers, the unified gateway at the branch node
sends the buffered CDRs to the CDRServer.
Multi-node
Enable the function of generating CDR files and configure the
distributed network CDRServer IP address on the unified gateway at each node.
(peer-to-peer)

Configure IP address of
all unified gateways.

Multi-node
Enable the function of generating CDR files and configure the
distributed network CDRServer IP address on the unified gateway at each node
(convergent)
(excluding the convergent gateway).

Configure IP address of
all unified gateways
(excluding the
convergent gateway).

After the CDR switch is turned on in a version earlier than U1900 V200R003C20, all CDR
files will be exported.

config createbill switch on

The CDR switch in U1900 V200R003C20 or a later version will be used to control whether
to export CDR files for internal calls. An internal call refers to a call between two numbers
registerd with the same unified gateway.

config createbill switch on exportinterofficebill yes


Here, the default value of exportinterofficebill is yes, indicating that CDR files of
internal calls will be exported; and value no indicates that CDR files of internal calls
will not be exported.

arameter

Descripti Value
on
Range

Default Value

protocol

Transmis
sion
protocol
type.

Two
tls
options are
available: t
cp andtls.

port

Port
number
of the
CDRServe
r.

The value
range is
102465535.

authtype

Authenti Two
AuthByPwd
cation
options are
type.
available:
NoAuthan
d AuthByP
wd.

protocol = TCP:
2020
protocol = TLS:
2021

How to Set

TLS is recommended.
The CDRServer supports TLS and TCP connections.
When the CDRServer starts, ports 2021 and 2020
are enabled automatically without independent
configuration.
It is an optional parameter.

The parameter must be set to AuthByPwd if


authentication is enabled on the CDRServer. The
parameter can be set toNoAuth or AuthByPwd if
authentication is disabled on the CDRServer.
However, AuthByPwd is recommended to ensure
data communication security.

username User
name for
authentic
ation.

The value The same as the


is a string configuration on
of 1-32
the CDRServer.
characters.

The value is the same as the configuration on the


CDRServer.
The default user name on the CDRServer is admin.

password Password
for
authentic
ation.

The value The same as the


is a string configuration on
of 1-32
the CDRServer.
characters.

The value is the same as the configuration on the


CDRServer.
The default password on the CDRServer
is Change_Me.

1config createbill switch off


2U1960 can support 50,000 CDRs, 50000/500/5=20 daysU1981 can support
480,000 CDRs , 480000/500/5=192 days

Follow-up operation steps:

Click OK.

A message is displayed, prompting you to restart the CDR service for the
configuration to take effect.

Click OK.

Close the CDR Server CDR-Running window.

In the CDR Server window, click Exit to exit the CDRServer.

Start the CDRServer to make configuration take effect.

Notice:

You need to add IP addresses of all unified gateways in a single-node, multi-node


centralized, or multi-node distributed (peer-to-peer) network.

You need to add IP addresses of all unified gateways (excluding the convergent
gateway) in a multi-node distributed (convergent) network.

When the unified gateway has any call services, the system automatically generates
D:\BillL\CurFile and D:\Bak\CurFile directories on the CDRServer.

The BMU periodically obtains CDR files from the D:\BillL\CurFile directory. The
parameter of how often the BMU obtains CDR files from that directory is configured
on the BMU.

The D:\Bak\CurFile directory stores backup CDR files.

If No CDR File in the Directory for Storing CDR Files, the possible causes are as follows:

The unified gateway is incorrectly configured.

The CDRServer is incorrectly configured.

Parameters for importing CDRs are incorrectly configured.

The BMU takes the CDR files.

Parameters for querying call record


Parameter

Description

Caller number

Number of a calling party in a call event.

Callee number

Number of a called party in a call event.

Unified Gateway IP address of a Unified Gateway host. The CDRServer can accept call record files
IP address
generated by multiple Unified Gateway hosts at the same time. To query the call record
files that are uploaded to the CDRServer by a Unified Gateway host, specify the IP
address of the Unified Gateway host.
Received time

Date on which call record are transferred to the CDRServer from the Unified Gateway call
record pool.

Created time

Time when call record are generated in the Unified Gateway call record pool.

Affixal condition

Start Position: indicates that the query starts from the call record with the specified serial
number. Max Records Num: indicates the maximum number of call record that meet
query criteria to be displayed in the query result.

The call record details are the same as those displayed in the query result window. You
can perform this step to view call record details clearly, without the need to view them by
dragging the horizontal scroll bar in the query result window.

1. No. If the CDRServer are connected to third-party billing system or BMU, we can search
the CDRs on the BMU web page or third-party system web page.

UPF, upgrade policy file. A UPF file defines the scope of the IP phones to upgrade and the
upgrade policies. Before an upgrade, an IP phone obtains the UPF file from the
preconfigured upgrade file server(HTTP server).
The global UPF file on the HTTP server is used to configure and upgrade IP phones in
batches.
During DHCP server configuration, a 246 parameter is defined for setting the URL of the
global UPF file. After this parameter is set, the DHCP server sends the URL to the IP phone

that applies for an IP address. The IP phone then downloads the UPF file from this URL.

The UPF file contains the IP addresses of the servers where the firmware version file,
configuration file, ring tone files, and local address book files are stored.
The batch configuration and upgrade of IP phones have the following features:

IP phones of the same model use the same UPF file.

For example, you only need to prepare one UPF file for all eSpace 7850
phones.

IP phones obtain the required firmware version file URL from the UPF file.

After obtaining the global UPF file, IP phones download the firmware version
file based on the URL in the UPF file for batch upgrade.

IP phones obtain URLs from the UPF file to download configuration file, ring tone

files, local address book files, and other files.

How to configure the DHCP and HTTP server, please refer to


eSpace_7910&7950_IP_Phone_V200R003_Administrator_Guide(pdf) document.

After being connected to the DHCP server and powered on, eSpace 7910/eSpace 7950
sends a request to the DHCP server for an IP address. In response, the DHCP server returns
an IP address, together with the URL of the UPF file (containing the addresses of the

configuration file and the file server) carried in the Option 246 parameter. eSpace
7910/eSpace 7950 then obtains the configuration file from the file server to update its
configuration. If the UPF file also specifies other files (for example, firmware version file) to
update, eSpace eSpace 7910/eSpace 7950 also obtains these files for configuration
update.

1. A

The default TR-069 parameter in IP Phone is set to ucems.huawei.com. If a DNS server


has been deployed, add domain name ucems.huawei.com to the DNS server and map it to

the eSights IP address.

IP Phone is able to resolve the value of acs.address carried in the DHCP Option 246 field
to obtain the eSight IP address.

If the DHCP server installed on the Windows Server 2008 is used, set String

value for the Option 246 field to acs.address=http://eSights IP:Port


number/tr069/services/acs, for example,
acs.address=http://192.169.1.37:8089/tr069/services/acs. For details about the
setting, see "Setting the Option246 Parameter" in the IP phone administrator guide.

If the DHCP server installed on the Windows Server 2008 is used, and set String
value for the Option 246 field to acs.address=http://Customized domain name:Port
number/tr069/services/acs, for example,
acs.address=http://ucems.xxx.com:8089/tr069/services/acs. Then we need
DNS server to resolve the domain name to eSight servers ip address.

We can also setting TR-069 Parameters on the Web Page of an IP Phone.

Setting IP Phone Protocol Parameters

Added by Creating groups:

Before connecting IP phones to the eSight and using the configuration and
management functions provided by the eSight, create groups for IP phones. If TR069 parameters are correctly set on an IP phone, the eSight can automatically add
the IP phone to the corresponding IP phone group.

Added in batches:

Set TR-069 parameters on IP phones so that the IP phones can properly connect to
the eSight. If the eSpace 7900 series IP phones (eSpace 7910, eSpace 7950) with
the version V100R001C02 or later or eSpace 8950 IP phones are used onsite, you
can use the access scan function to connect them to the eSight without settings
TR-069 parameters.

The eSight allows you to import IP phone devices in batches. Enter device
information in the Excel template provided by the eSight and import information in
the Excel template to the eSight. Then the eSight will automatically add the devices
specified in the Excel template to the corresponding subnets.

Added by access Scan:

The access scan function applies to eSpace 7910 IP phones and eSpace 7950 IP
phones with the version V100R001C02 or later, and eSpace 8950 IP phones with all
versions.

Prerequisites:

TR-069 parameters have been set on IP Phone devices.

The eSight allows you to import IP phone devices in batches. Enter device information in
the Excel template provided by the eSight and import information in the Excel template to
the eSight. Then the eSight will automatically add the devices specified in the Excel
template to the corresponding subnets.
Procedure:
1. Choose Business > Collaboration > IPPhone Management from the main
menu.
2. In the navigation tree, choose Device Resource Management > IPPhone
Resource Management.
3. Click Import. In the Import area, download the template to your local computer.
4. Enter the IP Phone information in the template and save the template. For details
about how to fill in the template, see the Instructions sheet in the template.
5. Click

next to Upload file and select the Excel template.

6. Click

to upload the template

You can create lower-level groups in an IP phone group. To ensure management


performance, do not create groups of more than 10 levels and control the number of

devices in a group within 500.

he following describes the three types of IP phone groups:

Default group: An IP phone that is added to the eSight is added to the default group
by default if the eSight does not have any IP address group or the IP address of the

IP phone is not included in the management scope of the created IP address group.
After an IP address group is created, the IP phone is automatically grouped to the
corresponding IP address group.

Custom group: Users can create custom groups to manage IP phones that they want

to pay special attention to.

IP address group: Users can create IP address groups to manage IP phones by their
IP addresses.

If the IP address of an IP phone belongs to the IP address segments of multiple groups,

the eSight adds the IP phone to the group with the smallest IP address segment that
covers the IP phone's IP address.

Once the IP address of an IP phone changes, the eSight automatically moves the IP phone
to the matching group.
Procedure
1. Choose Business > Collaboration > IPPhone Management from the main menu.
2. In the navigation tree, choose Device Resource Management > Group
Management.
3. Click Create.

4. Configure the basic information about the group and its manageable IP address
segments.
5. Click OK.

When a great number of eSpace 7910 IP phones and eSpace 7950 IP phones with the
version V100R001C02 or later, and eSpace 8950 IP phones with all versions connect to

the eSight, the eSight uses the access scan function to send auto-configuration server
(ACS) addresses and certificate paths to the IP phones. After IP phones automatically
update their configurations based on the information received from the eSight, the IP
phones automatically connect to corresponding IP phone groups.

When a great number of eSpace 7910 IP phones and eSpace 7950 IP phones with the
version V100R001C02 or later, and eSpace 8950 IP phones with all versions connect to the

eSight, the eSight uses the access scan function to send auto-configuration server (ACS)
addresses and certificate paths to the IP phones. After IP phones automatically update their
configurations based on the information received from the eSight, the IP phones
automatically connect to corresponding IP phone subnets.

Choose Business > Collaboration > IPPhone Management from the main menu.
In the navigation tree on the left, choose Device Resource Management > IPPhone
Resource Management and click Access Scan.
Configure CPE user name and password.

Click next to the Operation.The Edit information of scan config page is displayed.
Modify CPE User Name and CPE Password based on IP Phone device,
click Confirm.
The default value of CPE User Name and CPE Password are huawei and huawei123.

To view scan result details about a record, click in the record. The scan result details
include the start time, end time, time spent, total number of devices scanned, total

number of IP phones detected, and number of devices failed to be scanned.

To search for required scan records, select a search type from the Search Type dropdown list box and enter appropriate search criteria.

PrerequisitesThe configuration file template for the IP phones whose configurations you
want to modify is available. An IP phone model corresponds to a configuration file
template.

The configuration file template for eSpace 7800 series IP phones is in .cfg format.

The configuration file template for 8850 IP phones is in .xml format.

For eSpace 6800 series IP phones, eSpace 7900 series IP phones and eSpace 8950 IP
Phone, compress the configuration file (Config-eSpace68XX.xml, ConfigeSpace79XX.xml, or Config-eSpace8950.xml) and the eSightUI.xml file
provided by the eSight together to a .zip package. This .zip package is the
configuration file template for eSpace 6800 series IP phones, eSpace 7900 series IP
phones and eSpace 8950 IP Phone.

If you need to add a configuration file template for the IP phones to upgrade, perform the
following steps:

Choose Business > Collaboration > IPPhone Management from the main menu.
In
the
navigation
tree
on
the
left,
choose
Deployment
Management > Configuration File Management and click Configuration File
Template Manager.
Click Upload.
On the Upload Configuration File Template page that is displayed, set the
configuration file template parameters.
Click OK. The newly added configuration file template is displayed on the interface.

In the navigation tree on the left, choose Deployment Management > Configuration
File Management.
Click Create and set related parameters in the Basic Information area. For 7900 series
and 8900 series IP phones:

Enter the configuration file name in Configuration file name.

Select the models of IP phones to be configured from Device model. Set


parameters of selected IP phone models in the corresponding panes.For the
description about IP phone parameters, see the corresponding Configuration File
Parameter Description.

For 7900 series and 8900 series IP phones:

When Configuring Policy is set to Full, all parameters are valid in the
configuration file that is generated, and all the parameter values are delivered
to the IP phones of the matching models.

When Configuring Policy is set to Specified item, only selected parameters


are valid in the configuration file that is generated, and the selected parameter
values are delivered to the IP phones of the matching models. The parameters
that are not selected remain unchanged.

NOTICE:

For 7900 series and 8900 series IP phones, set Configuring Policy to Full if the
automatic number allocation function needs to be enabled. Otherwise, automatic
number allocation is unavailable during automatic deployment.

After confirming the configurations, click OK.If you can find the new configuration file on
the Configuration File Management page, the operation is successful.

Set the value of ACS address to the eSight IP address or the DNS domain according to the
site requirements.

Version file management allows you to upgrade IP phone version files in batches.

Choose Business > Collaboration > IPPhone Management from the main menu.In the
navigation tree on the left, choose Deployment Management > Version File
Management.

Click Upload and set version file parameters in the Version Attribute dialog box.

Select the required version file.

eSpace 6800 series: Select a version file in .zip format. The version file is the main
program.

eSpace 7800 series: Select a version file in .rom format. The version file is the main
program.

Version file management allows you to upgrade IP phone version files in batches.

Choose Business > Collaboration > IPPhone Management from the main menu.In the
navigation tree on the left, choose Deployment Management > Version File
Management.

Click Upload and set version file parameters in the Version Attribute dialog box.

Select the required version file.

eSpace 6800 series: Select a version file in .zip format. The version file is the main
program.

eSpace 7800 series: Select a version file in .rom format. The version file is the main
program.

eSpace 7900 series: Select a version file in .zip format. The version file contains the
main program, language package, signal tone, certificate file, configuration file,
wallpaper, and ringtone.

Extract files from the software package. The .binfile that is obtained is the

main program.

Make a language package file, compress it into a.tar file, and rename
it language.tar.
Obtain a signal tone file in .wav format, compress it into a .tar file, and

rename itsignal.tar.

Obtain certificate files in .pem format.

Obtain wallpaper files in .jpg, .bmp, or .pngformat.

Obtain ringtone files in .wav format.

You can select the required packages to compress them into a .zip package
and upload the .zip package to the eSight.

eSpace 8850: Select a version file in .bin format. The version file is the main
program.
eSpace 8900 series: Select a version file in .zip format. The version file is the main
program.

Click OK.The eSight uploads the version file to the file server. You can view the new
version file on the Version File Management page.

The automatic number allocation function is available only for eSpace 7900 and 8900
series IP phones of V200R003C00 or later.

You can bind a number to the SN or MAC address of an IP phone. Then eSight can
allocate the number to the IP phone based on the bound SN or MAC address during
automatic number allocation. The numbers and identifiers are in one-to-one
mappings.

If the numbers and identifiers are not bound, eSight allocates a random number
within the number segment to an IP phone.

Choose Business > Collaboration > IPPhone Management from the main menu.

In the navigation tree on the left, click IPPhone Management Configuration.

Under Support Number Allocation, set IP phone number allocation parameters.

If Allocate numbers by eSight is set to ON, eSight actively allocates numbers to IP


phones based on the defined IP phone number allocation policy.If Allocate
numbers by eSight is set to OFF, the IP phone number allocation function is
disabled.

Set the IP phone identifier to Serial Number or MAC Address based on the site
requirements.

You do not need to manually configure an SIP password for each phone number.

Choose Resource > Resource Management > Collaboration Resource from the main
menu. In the navigation tree, choose Unified Communications.

In the management object list, choose the device type, and click the device name.

In the navigation tree on the left, choose Manage Service > BIN configuration.

Set BIN channel parameters.

Set BIN channel connection switch to ON.


The default value of Port is 8000, and the value must be the same as the BIN
channel of an IP PBX.

Set Password to the password of the IP PBX in Config mode.

Set SSL secure connection switch to the same value on the IP PBX.

Choose Business > Collaboration > IPPhone Management from the main menu.

In the navigation tree on the left, choose Business Resource Management > Number
Resource Management.

Click
. Open the downloaded template file, enter parameters following the
template instructions, and save the settings.

Click Import, select the configured template, and click OK.

We can use the eSight to manage the number resource.

You can bind a number to the SN or MAC address of an IP phone. Then eSight can
allocate the number to the IP phone based on the bound SN or MAC address during
automatic number allocation. The numbers and identifiers are in one-to-one
mappings.
If the numbers and identifiers are not bound, eSight allocates a random number
within the number segment to an IP phone.

Choose Business > Collaboration > IPPhone Management > from the main menu.In
the navigation tree on the left, choose Deployment Management > Automatic

Deployment.

Click in the Associate Configuration File column of the subnet where the IP phones to
deploy are located, and select the configuration files for the IP phone models to deploy.

For 7900 series and 8900 series IP phones, select the configuration files

whose Configuring Policy is set to Full if the automatic number allocation function
needs to be enabled. Otherwise, automatic number allocation is unavailable during
automatic deployment.

Optional: Click in the Associate Version File column and select the version files of the

IP phone models to deploy.

Click

in the Allocated Number Segment column and specify the number segment.
The eSight does not automatically allocate numbers if no number segment is
specified.

Choose Business > Collaboration > IPPhone Management from the main menu. . In
the navigation tree on the left, choose Deployment Management > Manual

Deployment and view the execution result of the automatic deployment task.

The eSight supports the upgrade of the main programs of eSpace 6805, eSpace 6810,
eSpace 7800 series, eSpace 8850, and eSpace 8900 series IP phones, and the upgrade of

the main programs, language packages, certificates, signal tones, wallpapers, and
ringtones of eSpace 7900 series IP phones. eSpace 7800 series IP phones consist of eSpace
7810, eSpace 7820, eSpace 7830, eSpace 7850, and eSpace 7870. eSpace 7900 series IP
phones consist of eSpace 7910 and eSpace 7950. eSpace 8900 series IP phones consist of
eSpace 8950.

The upgrade will result in about a 5-minute service interruption. The communication
between the eSight and the IP phone will also be interrupted for about 5 minutes.

Choose Business > Collaboration > IPPhone Management from the main menu.

In the naClick Next, click Add on the Select Device page, and select the IP phone to
be upgraded.vigation tree on the left, choose Deployment Management > Manual
Deployment.

Click Add Task and set related parameters on the Create Task page.

Enter a task name in Task Name.

Set Select Upgrade Type to IPPhone.

Determine whether to select Scheduled Task based on the site requirements.

Click Next, click Add on the Select Device page, and select the IP phone to be upgraded.

Notice:

If the IP phone that you want to upgrade is unavailable, check whether the version
file for this model of IP phone is uploaded to the file server.

If the IP phone does not belong to any subnet, the IP phone cannot be upgraded.

In Select Upgrade Type, select the program type (for example, Main Firmware) that
you want to upgrade.
In Select Target Version, select the required version file for the model of IP phones to be
upgraded. Only version files that have been uploaded to the file server are available here.
In the Update Configuration File area, select a configuration file that you have uploaded
to the file server.

Failed to upgrade the software:

Choose Business > Collaboration > IPPhone Management from the main
menu. In the navigation tree, choose Device Resource Management > IPPhone
Resource Management. Check the connection status in the management object
list.If an IP phone is not online, select the IP phone, click Refresh, and check
whether the Connection value of the IP phone can be changed to Online. If the IP

phone is still offline, modify the configurations and ensure that the IP phone is
online. Then re-upgrade the IP phone.

If the device is still offline, maybe the network is abnormal. Run the ping command
on the eSight server which associates with the media gateway when the device
connects to the eSight to check the connection between the device and the eSight.
If the connection is abnormal, connect the device to the eSight correctly.

Check whether the CPE user name and password configured on the IP phone are
the same as the user name and password that are configured on the eSight when
adding the IP phone to the eSight. If they are different, seeSetting IP Phone Protocol
Parameters to correct the settings on the IP phone.

1. ABC

Routine Maintenance: maintenance operations carried out by the maintenance staff on a


daily or regular basis.
Version Maintenance: version upgrade and rollback

Routine maintenance serves the following purposes:

Discovering alarms generated by the equipment or existing defects of the equipment


in time, and accordingly taking proper measures. This ensures the stability of the
equipment and lowers the rate of failure.

Recognizing abnormal cases that occur during the running of the charging system
and handling it in time. This reduces economic loss due to loss of bills.
Grasping the running statuses of both equipment and network in real time and
identifying the running tendency of both equipment and network. This improves the
efficiency of maintenance staff in handling emergencies.

Carrying out periodic checks, backup, tests and cleaning for identifying the defects
in the equipment such as, natural aging, function failure, and deterioration of

performance; and taking proper measures to eliminate these defects.

Checking the Electromagnetic Interference in the Equipment Room. The electromagnetic


criteria is: l Electric field intensity 130 dB (V/m) l Magnetic field intensity 800 A/m.
We need special instrument to check the electromagnetic interference. This is an optional
item.

Item

Action

Expected Result

Fire
prevention
condition

Check the power cable and


connector of the power
module.

No hidden danger of fire exists.


Firefighting devices are in good
condition.

Dust-proof
condition

Check the external surfaces of


the device, the computer, the
floor and the desktop.

Each position is dust-free.

Theft proof
condition

Check the doors, windows,


and theft-proof iron net.

All the facilities are in good


condition.

For Running Status of the Boards, choose Show Board.

For the status of Fast Ethernet Port, the FE port is marked with 100BASE-T,

For CPU usage, choose Trace > Resource Trace > Resource Trace Start Board in the main
window of the OMU client. Then select Resource Trace > Slot No. > Board to view the CPU
usage of the SC-MCU board in the slot.

Alarms on LMT can be categorized into:

Critical

Major

Minor

Warning

Uncertain

Cleared

Different indicators: power indicator, running indicator, alarm indicator, signal loss
indicator, remote alarm indicator.

Indicators in different status represent the running status of the device.

Power indicator, On: The power supply is normal, Off: There is no power supply.

Running indicator, Steady on: The board is faulty. Blinking (4 Hz): The board is writing data
into the Flash memory while the system is running. Blinking (2 Hz): The board is writing
data into the Flash memory while the system is started. Blinking (0.5 Hz): The board is
running normally. Off: No power is supplied or the board is faulty.

Alarm indicator, Steady on: An alarm exists. Blinking (0.5 Hz): A major alarm exists.
Blinking (0.25 Hz): A minor alarm exists. Off: No alarm exists.

Signal loss indicator, Steady on: Signals are lost. Off: Signals are normal.

Remote alarm indicator, Steady on: The peer device fails to receive data. Off: The peer
device is running properly.

If trunk circuit is in ISOLATE state, Check the following items:

Run the config cancel isolate board/card command to cancel the isolation of the board.

Verify that the board where the trunk circuit is located is normal.

If PRA links is in fault state, Check the following items:

Verify that the connection of the E1 trunk line is correct. That is, verify that the local-end TX
is connected to the peer-end RX, and the local-end RX is connected to the peer-end TX.

The board where the PRA links resides is normal.


The E1 trunk line is connected correctly (the local-end TX is connected to the peer-end RX,
the local-end RX to the peer-end TX).
The PRA links is configured correctly (if the local-end link is user, the peer-end link is network;
if the local-end link is network, the peer-end link is user).

If subscriber is in fault state, Check the following items:

If a subscriber is in BLOCKED state, run the unblock subscriber command to unblock the
subscriber.

If a subscriber is in LOCKED state, check whether the subscriber does not dial any number
after hook-off or the subscriber does not hang up when a call ends. If the subscriber does
not dial any number after hook-off or the subscriber does not hang up when a call ends,
hang up, and then check whether the fault is rectified. If the fault is caused because of other
reasons, check the circuit to determine whether a short circuit occurs.

Monthly maintenance refers to the tasks carried out by the maintenance staff every month.
It covers the tasks on shelf, terminal system, and spare parts.

The temperature in the warehouse is 0C45C.

The relative humidity in the warehouse is 5%95%, without condensation.

Yearly maintenance refers to the tasks carried out by the maintenance staff every year.

[%U1900(config)]$debug print cli history info place backup


CLI count = 9
Time
Tick
Session Result Command
2009-12-26 114612 3364
0
Success Back board data syn have
finished!
2009-12-26 114652 7302
6
Success enable
2009-12-26 114657 7841
6
Success show s i
2009-12-26 114737 11831
6
Success config s gateway ip 10.77.220.1
2009-12-26 114810 15178
6
Success config sys hostip 10.77.221.17
2009-12-26 114817 15801
6
Fail
save
2009-12-26 114830 17170
0
Success TotalBackup is completed for the
first time!
2009-12-26 114931 23111
6
Success save
2009-12-26 114944 24411
6
Success reboot
====

Command executed success !

====

To back up the data,


Step 1: Use the save command to save the current configuration and service data to the
FLASH memory.
Step 2: When the system prompts "Saving flash in the background finished!" (it means
the saving is finished), use the config upload file data command to back up the data
in the FLASH memory to the load directory on the TFTP server.
Step 3: After successful backup, check that the data.bin file is contained in the load
directory on the TFTP server. The data.bin file is the backup file of the database.
Step 4: Rename the data.bin file, back it up for restoring the data when necessary.

To restore the data of the specified time, load the backed-up data file to the host. To
restore the data,
Step 1: Store the data file to be imported in the load directory on the TFTP server. The
data file need to be named data.bin. If not, you need to rename it. Otherwise, you
will fail to import the data.

Step 2: Use the config download file data command to load the data to the FLASH
memory.
Step 3: After running the config download file data command successfully, run the
reboot command to reboot the system so that the restored data takes effect.

You can also execute the Back up&Restore operations on U1900 webpage.

Choose System > File Management.

Backing up a data file

Click Download.

Loading a data file

Click Browse, select the data file (data.bin) to load, and click Upload.

Ensure that the hostip has been configured as the IP address of TFTP server and the
directory is rightly set.

The Local Maintenance Terminal (LMT) is a management system for the U1911 unified
gateway. It provides various functions such as alarm management, configuration
management, signaling tracing, upgrade, log collection, and offline operations.
The LMT connects to the U1911 unified gateway using SSH/Telnet. A maximum of four
U1911 unified gateways can be connected to the LMT
Alarm management

Displays and queries real-time alarms.

Displays and queries historical alarms.

Queries alarms by parameters such as the alarm severity, alarm type, alarm ID, time,
and device IP address.

Collects statistics on alarm by level.

Acknowledges and clears alarms.

Synchronizes and filters alarms.

Exports alarms.

Preparation before upgrade

Obtain software packages and reference documents.

Check the required items.

Back up data.

Obtain the license file for the target version. This file must be obtained before site
deployment.

Stop services before the upgrade. This operation is required only when a conference
is being held using the gateway.

The basic steps:

Log in to the U1900 using LMT.

Run the save command to save the current data.

Choose Version > Upgrade .

Click Browse... to select the target version.

Select the device to upgrade, and click Upgrade, click OK.

Verify the services on the U1900 gateway.

If the service verification succeeds, the upgrade continues.

Load voice.zip file.

The basic steps:

Log in to the device using the LMT.

Choose Version > Rollback .

Select the device for the rollback, select a source version from the Select the version
(rollback) drop-down list box, and click Rollback.
The rollback takes about 15 minutes. After the rollback is completed, Rollback

success is displayed in the ExecutionStatus column.

The operator user can execute all commands except for those for modifying password of
the superior account, modifying MAC address and modifying the registered domain name
of the MG device.

If you click Save as carrier setting, you can restore data to the carrier settings as required.
If you click Save as ordinary setting, you cannot restore data to the carrier settings.

Version information collection is the basis of fault location. If the IAD software being used
is not the latest version and the faults in this version are already solved in the upgraded
versions, you can upgrade the software version to solve the fault.
You can collect following information on the same page.

Main Board Version;

Other Board Version;

Network Port;

User Register Status;

Group Register Status;

Vlan Information;

Current Config;

Alarm History.

You can collect following information on the same page.

Main Board Version;

Other Board Version;

Network Port;

User Register Status;

Group Register Status;

Vlan Information;

Current Config;

Alarm History.

The network connection issue may be caused by incorrect device IP address configuration.
For example, the IP address of the IAD is not set in the same network segment as the
upper-level switch, or the MAC address of the IAD conflicts with other devices.

NOTE:

When the CPU usage is higher than 85%, the CPU is overloaded and a CPU
overload alarm is reported. You can end several calls to decrease the CPU usage,
and then continue conversation after the CPU usage becomes normal.

When the CPU usage is lower than 75%, the alarm is cleared.

A single IAD upgrade:

Web manually (HTTP/FTP)

Users can log in the Web management system with the IP address of the IAD,
and select the FTP or HTTP upgrade mode on the Web page. Choose System
Tool > Upgrade Version in the navigation bar to enter the version upgrade
page. Upgrade the software as prompted. After the upgrade is complete,
choose System Tool > Restart Device to restart the IAD.

CLI manually (TFTP/FTP)

Users can log in to the IAD for upgrade in Telnet mode or through a serial port
on the maintenance terminal. After the upgrade is complete, restart the IAD.

Batches of IADs upgrade

Web automatically (FTP) or CLI automatically (FTP)

Users can log in to the IAD for upgrade in Telnet mode or through a serial port
on the maintenance terminal to enable the automatic upgrade function. After
the automatic upgrade function is started, IADs can periodically detect the
latest IAD software update and then automatically install it. After the upgrade

is complete, the IAD automatically restarts.

Through the eSight

Users can upgrade IADs managed by eSight in batches. For details on how to
upgrade IADs in eSight, see Upgrade the IAD Manually and Automatically

Upgrading the IAD in IAD Management in the eSight Help.

Automatic upgrade: Set the upgrade interval for an IP phone. When the upgrade interval
expires, the IP phone automatically checks for new version files on the server and if there is
any new version file, automatically starts its version upgrade.
A UPF file defines the scope of the IP phones to upgrade and the upgrade policies. Before
an upgrade, an IP phone obtains the UPF file from the preconfigured upgrade file server.
Obtain the UPF template, and modify the parameters in the UPF template depending on
your needs.

Prerequisites

Hardware version number of the IP phone to upgrade.

Address of the HTTP server where the files for the IP phone to upgrade are stored.

Target version number.

UPF template (which is delivered with the version release).

It is recommended that you use the Apache server (a type of HTTP server) as the upgrade
server.
Choose Start > All Programs > Apache HTTP Server 2.2 > Monitor Apache Servers.
In the following ways you can start Apache server: Choose Start and click it; Or choose
Open Apache Monitor, In the Apache Service Monitor window that is displayed, click
Start.

Copy files that IP phones will access to the root folder of the Apache server.

Change the Option246 parameter value of the DHCP server to the configuration file
URL.8.5 Setting the Option246 Parameter document describes how to set the Option246
parameter.

The configuration file URL specified by the Option246 parameter has the highest
priority than other specified URLs.

It is optional to specify the configuration file name. The IP phone will automatically
search for and download the configuration file mapping its model.

The Option246 parameter value of the DHCP server should be in the form of
upfprofile.address=http://172.168.30.10/UPF-eSpace7950.xml.

Basic steps:

Enable automatic upgrade. Specifically, on the phone web page, choose Advanced
> Upgrade > Automatic Upgrade. On the page that is displayed, select
Enable for Automatic Upgrade.

Save the upgrade files to the root directory of the upgrade server.

When the message "Upgrade?" is displayed on the phone screen, select Yes.

Enter the UPF file URL and enable automatic upgrade on the IP phone. Specifically,
access the phone web page as an administrator, choose Advanced > Server >
Version Download Server > Upgrade Server, and enter the UPF file URL on the
page that is displayed.

After the upgrade is complete, the message "Upgrade Success" is displayed on the
phone screen.

Working Principle

After being powered on, a phone obtains the IP address from the DHCP server.
Then the DHCP server delivers the configuration file URL to the phone using the
Option246 parameter.
After receiving the URL, the phone obtains the global configuration file from the
file server to update the phone configurations, and downloads files such as the
firmware version file from the URLs specified in the configuration file.

If some phones failed to be configured or upgraded, the possible cause is that too many
phones send configure and upgrade requests to the server at the same time, and the
server cannot handle all those requests. You are advised to restart these phones. The

phone downloads the configuration file and firmware version from the file server during
the restart.

Basic steps:

Access the phone web page as an administrator and choose Advanced > Upgrade.

In the Manual Upgrade area, click Browse next to Import Upgrade Package.

Locate the version file to be upgraded on the local PC. Click Upgrade.

1. A;
2. AC;

Step 1: Collect fault information.


When technical problems occur in the system or device, you need to collect as much
information as possible in the following ways: fault complaint, cyclic check of the
device, alarm from the OMU, alarm from the device, and dialing test.
Step 2: Check whether the device is running normally.
The fault of a hardware device causes service congestion, for example, breakdown of
the SMCU board causes global service congestion. Thus, you need to make sure that
the hardware device runs normally before handling any problem. The methods used
for checking hardware devices are as follows:
Make sure that the power supply of the U1900 device runs normally. That is,
check the shelf and boards inside the shelf are powered on.
Through viewing the indicator light or the alarm information on the OMU client,
check whether all the boards are in normal state.
Step 3: Take action to solve the device fault.

If the system fails to work, and you have confirmed that U1900 device is down, you
need to take action to solve the fault immediately according to the predefined
scheme.
Step 4: Take action to solve service fault.

If the system fails to work, and you have confirmed the service fault, you need to take
action to solve the fault immediately according to the pre-prepared scheme.

Step 5: Make sure that the fault is removed.


After handling the emergent fault, you need to check that the fault has been removed
in the ways such as NMS, dial test and service verification. That is, you need to
check that the device works properly and the service recovers.

Step 6: Collect fault information.


It is necessary to collect the fault information in time. This helps the technical engineers
to analyze and locate the fault, and prevents the fault from recurring. If the fault is a
device fault, the collected information helps the vendor improve on its devices.

For this reason, it is necessary to collect the fault information regardless of whether the
fault has been successfully resolved or not. You can work out a summary report on
the fault. The report contains the following items: the time, nature, symptom and
solution of the fault.
Step 7: Report for help.
If you fail to clear the fault according to the related emergent fault handling process,
ask for technical support from the Customer service center or local office of Huawei
by telephone or fax.

To minimize the time of handling the fault, you must: record your emergency handling
in detail, notify Huawei of the type of the boards to be replaced, and apply for the
necessary spare boards based on the warranty clauses.

Checking Power Supply of the Shelf:

Power failure of the shelf causes the failure of the host device.

Any of the following cases indicates that the shelf is powered off:

The POWER or RUN indicator light on the front panel of the shelf is off.

The RUN indicator light on the panel of all the boards is off.

The fans in the fan box stop running.

To remove the fault, do as follows:


Step 1: Make sure that the connector of the power module is connected properly
and the power switch at the back of the shelf is turned on.

Step 2: If the connector of the power module is connected properly and the
power switch at the back of the shelf is turned on, turn off the power switch
and remove the connector of the power module.
Step 3: Replace the power module and the power cable.

Step 4: Connect the connector of the power module properly, and then turn on
the power switch to power on the shelf.

Checking Whether the Main Control Board(s) Is Normal:

For U1980, if all the installed SMCUs (one or two in total) are faulty, the host device
breaks down. If the ALM indicator light on the panel of the SMCU flashes, it
indicates that the SMCU is faulty.

To remove the fault of the SMCU, do as follows:


Step 1: Reboot the system.
Step 2: After the SMCU starts up, observe the ALM indicator light on the front
panel. If the ALM indicator light is off, it means that the board is in the normal

state. The ALM indicator light flashes, it means that the board is in the
abnormal state. In this case, proceed to Step 3.
Step 3: Wear the ESD-preventive wrist strap properly.
Step 4: Check whether the fault is resulted by incomplete insertion of the SMCU.
Open the ejector lever to power off the SMCU. Remove the SMCU, and reinsert it. Observe whether the fault is cleared. If the fault is cleared, proceed to
Step 6.If the fault is not cleared, in the case of two SMCUs, re-insert the other
SMCU by following the same procedure. If the fault is not cleared, proceed to

Step 5.
Step 5: Remove the faulty SMCU, and insert a backup SMCU.

Taking Bill Immediately:

When handling the fault, you can perform critical operations, for example, resetting

boards. To ensure the bill security, transfer the bills from the host to the directory
specified on the bill server before performing the operations. The method is to
choose BillConsole> Control bill transfer on the bill server, and then check the bill
transfer status is Started.

Check Whether Host Device Breakdown Is Caused by Modifying the Configuration Data:

Any of the following cases indicates that the board faults:

The alarm center on the OMU client shows some alarm information on the fault of
the board.
The RUN indicator light on the board panel is always on or off.

To remove the fault of the board, do as follows:


Step 1: Reset the faulty board.

Use the reset board slot command or press the Reset button.

Step 2: After the board starts up, observe the RUN indicator light on the front panel.

If the RUN indicator light flashes, it indicates that the board is normal. Proceed
to

Step 3: Wear the ESD-preventive wrist strap properly.


Step 4: Open the ejector lever of the board to power off the board.
Step 5: Remove the faulty board, and re-insert it. Check whether the fault is cleared.

If the fault is not cleared, proceed to Step 6.

Step 6: Remove the faulty board, and then insert a backup board.
Step 7. Otherwise, proceed to Step 3.

Log in to the U1900 using the LMT.

On the LMT page, choose Host Log > Call Analyse.

Select Online Analyse.

Set up a call. The cal log information is displayed.

Click Analyze Call Log.

Click Analyze Call Log.


On the Call Analyse page, click a call record. The call flow chart of the selected call is
displayed.

The priority for rectifying faults depends on the alarm severity.

Critical: global alarms that severely affect device operations. For example, a power
supply fault, critical alarms must be handled immediately; otherwise, the system may
crash.
Major: board or physical circuit alarms that occur in a certain scope. For example,
the fault of physical circuit, major alarms must be handled immediately; otherwise,
the services cannot run normally.

Minor: general faults or event alarms that describe the working status of each board
or physical circuit. You must locate the cause to remove potential problems.
Warning: prompt message. For example, a message indicating that a device has
been restored, warning alarms do not affect system performance and services. No
action is required.

The procedure for handling an alarm is as follows:

Find the critical and major alarms among all alarms.

View the cause of each alarm and handle the alarm.

Verify that the alarm is cleared.

If the alarm is cleared, the fault is rectified.

If the fault is not rectified, find out other possible causes.

Use the LMT to view alarm information.

Log in to the LMT and choose Alarm Management > Alarm. All alarms of the
current device are displayed.
Double-click a required alarm. The detailed alarm information is displayed.

Click Alarm Suggestions tab. Clear the alarm as suggested.

When the alarm is cleared, it turns grey.

Logs record key events and information that are generated while programs are running. By
viewing and analyzing logs, you can locate and rectify faults.
Online Analysis

Log in to the U1900 using the LMT.


On the LMT, choose Log Management > Run Log. The run log information is
displayed.

Signaling records the process of establishing a call. By analyzing the signaling, you can
find out possible causes of call failure and rectify the fault.
Online Analysis

Log in to the U1900 using the LMT.

On the LMT page, click Trace.

Double-click the signaling to monitor (for example, SIP).

Enter the IP address of the U1900 and click Ok. The signaling information page is
displayed.

Perform signaling tracing.

Click a signaling in the tracing result. The signaling analysis result is displayed.

Use the signaling analysis result to determine the fault cause.

Using the ping command to view the status of network connection, delay time and loss
package.
IP bearer network quality

Network Quality

One-Way Delay (ms)

Packet Loss Rate

Jitter (ms)

Good (recommended)

40

0.1%

10

Poor

100

1%

20

Terrible

400

5%

60

Data service usage

Service Type
Good
Fax
Transparent Available
transmission
(G.711a/u)
T.38
Available
Modem
Transparent Available
transmission
(G.711a/u)

Poor
Unavailable

Terrible
Unavailable

Available
Unavailable

Unavailable
Unavailable

Run the following commands to check the MTU status:

show board slot <2-9>

show device status

If the MTU board is faulty, restart it.

If the fault persists after the board restarts, replace the board.

The fault is rectified.

Run the ping command to verify connectivity between an IP phone and the eSpace
U1900.
Decoding

The fault may be caused by conflict between the codec type of different IP terminals
and that of the eSpace U1900. To rectify the fault, set the highest codec type for all
IP terminals and the eSpace U1900. If all terminals support G.711A, you are advised
to set the highest codec type on all terminals to support G.711A.

No corresponding prefix is configured on the eSpace U1900. Therefore, you must run a
command to add a prefix for the called number.
[%U1900(config)]#config add prefix

Verify that the called number is correct.


[%U1900(config)]#show subscriber

Possible cause

The IP phone is faulty.

The network is disconnected.

A service restriction is configured on the IP PBX.

Dial *125# on the IP phone. If no announcement is played, the IP phone is not registered
with the IP PBX. Verify the network connection, account settings, and that the IP phone
runs properly. If an announcement is played, the IP phone is registered with the IP PBX.
Check whether outgoing call rights are configured for the user.

Possible cause

The echo cancellation function is disabled on the IAD.

Possible cause

The bill pool is full and no bill is deleted.

Suggestion: If no Huawei or third-party bill server is deployed to retrieve bills onsite,


disable the bill generating function when deploying the unified gateway to avoid service
interruption due to the full bill pool.

Possible cause

The long number and short number of the called party are the same. As a result, the
system processes the number dialed by the calling party as the long number.

Check PRA link status.


[%U1980(config)]# show pralink

Check status of PRA trunk circuits.


[%U1980(config)]#show tkc office no 1
[%U1980(config)]#config cancel isolate card

Confirm with engineers and find that an engineer deleted the DTU by running
a command.
The fault is rectified.

Check whether the eSpace U1900 is effectively grounded.

Check whether the E1 cable is longer than 100 m.

Run the command for displaying the system clock source to check whether the clock
source is obtained from the peer office.

If the resistance of the peer office is 120 ohms, check whether the connector is
correct and perform the test again after replacing the connector.

Connect the two E1 ports on the eSpace U1900 to form a self-loop and check
whether the board is faulty.
Check whether frame slip occurs on the two E1 ports connected in a self-loop.

Run the following command to configure the local crystal oscillator as the clock source and

then check whether frame slip occurs:


[%U1980(config)]#config system clocksource oscillator

Possible cause

Physical trunk links are faulty.

Trunk connection configurations are incorrect or links are disconnected.

Configurations of routing upon failures are incorrect.

Step 4: (Replace xxx with the domain name of the SIP trunk). Check whether the
HeartBeat parameter is set to YES. If no, run the config modify peercomponent
domainname xxx heartbeat yes heartbeatperiod 5 command to enable the heartbeat
function.
If the fault persists, contact Huawei technical support.

Possible cause

The international toll call right is not enabled for the account.

The called number does not exist.

By default, the international toll call right is disabled for users on the IP PBX to prevent
users from making international toll calls.

Possible cause

The DND service is enabled.

The CFU service is enabled.

DND: do not disturb

CFU: call forwarding unconditional

Possible cause

The conference is scheduled as an encrypted conference, but the SRTP flag is set to
OFF on the IP phone so that it does not support encryption. As a result, the user
fails to join the conference.

Possible cause

By default, an instant conference supports a maximum of three participants. When


the moderator adds four or more participants, an announcement is played,
indicating that the number of participants reaches the maximum.

Possible cause

The network is disconnected.

Registration information is incorrect.

The IP PBX is faulty.

Most registration failures are caused by network disconnection.

For details about how to rectify network disconnection faults, see Other Fault

Analysis (2).

Some users fail to register with the IP PBX because the user name, password, or SIP server
IP address is incorrectly set.

You can log in the SIP server to query registration information such as the user name
and password.

In addition, the SIP server IP address may be incorrectly entered during the batch
configuration of IP phones.

Log in to an IP phone as the administrator and enter the correct SIP server IP
address on the Advanced > Server page.

Possible cause

The network connection is abnormal.

The network cable is connected to the PC port instead of the LAN port.

Network configurations are incorrect.

Possible cause

By default, the interval for refreshing the registered SIP server is set to 3600s on an
IP phone, which is too long.

1. AB;
2. D.

Appendix I. Glossary

Number

3WC: Three-Way Calling

AC: Alternative Current

AAC-LD: Advanced Audio Coding with Low Delay

ACD: Automatic Call Distributor

AG: Access Gateway

AN: Access Network

Apps Applications

AR: Access Router

ARP: Address Resolution Protocol

A-SBC: Access Session Border Controller

B2BUA: Back-to-Back User Agent

BEST: Branch Exchange for Survivable Telephony

BHCA: Busy Hour Call Attempts

BHCC: Busy Hour Completed Call

BLF: Busy Lamp Field

BMU: Business Management Unit

BRA: Basic Rate Access

BS: Billing System

BTU BRI Trunk Unit BRI

BRIBasic Rate Interface

BYOD Bring Your Own Device

CACCall Admission Control

CAS: Channel Associated Signaling

CC: Conference Call

CDMA: Code Division Multiple Access

CDR: Call Details Records

Appendix I. Glossary

CFB: Call Forwarding Busy

CFNR: Call Forwarding No Reply

CFO: Call Forwarding Offline

CFU: Call Forwarding Unconditional

CTVMNR: Call Transfer to Voice Mailbox on No Reply

CTVMU: Call Transfer to Voice Mailbox Unconditional

CTVMB: Call Transfer to Voice Message on Busy

CH: Call Hold

CLI: Command Line Interface

CLIP: Calling Line Identification Presentation

CLIR: Calling Line Identification Restriction

CLIRO: Calling Line Identification Restriction Override

CNIP: Calling Name Identification Presentation

CNIR: Calling/Connected Name Identification Restriction

CODEC: Code And Decode

COLP: Connected Line Identification Presentation

CONP: Connected Name Identification Presentation

CRC: Cyclic Redundancy Check

CT: Call Transfer

DC: Direct Current

DID: Direct-Inward-Dialing

DDI: Direct-Dialing-In

DMZ: Demilitarized Zone

DND: Do-Not-Disturb

DSP: Digital Signal Processor

DSS1: Digital Subscriber Signaling No.1

DTMF: Dual Tone Multiple Frequency

ECS: Enterprise Communication Suite

Appendix I. Glossary

EMC: Electromagnetic Compatibility

EMS: Element Management System

ESD: ElectroStatic Discharge

FXS: Foreign eXchange Station

FXO: Foreign eXchange Office

GUI: Graphical User Interface

HME: Huawei Media Engine

IAD: Integrated Access Device

IPCC: IP Contact Center

IPT: IP Telephony

ISDN: Integrated Services Digital Network

IVR: Interactive Voice Response

LAN: Local Area Network

LDAP: Lightweight Directory Access Protocol

LE: Local Exchange

LMT: Local Maintenance Terminal

MAA: Mobile Access Agent

MAS: Mobile Agent Server

Meeting MS: Meeting Media Server

MGC: Media Gateway Controller

MGW/MG: Media Gateway

MoH: Music on Hold

MRS: Media Resource Serve

Appendix I. Glossary

NGN: Next Generation Network

OMU: Operation Maintenance Unit

ONLY: One Number Link You P

PBX: Private Branch Exchange

PCM: Pulse Code Modulation

PGND: Protection Ground

POTS: Plain Old Telephone Service

PRA: Primary Rate Adaptation

PRI: Primary Rate Interface

PSTN: Public Switched Telephone Network

QoS: Quality of Service

RBT: Ring Back Tone

RTCP: Real-time Transport Control Protocol

RTP: Real-time Transport Protocol

RX: Receive

SCE: Service Creation Environment

SDP: Session Description Protocol

SG: Signaling Gateway

SNMP: Simple Network Management Protocol

SNTP: Simple Network Time Protocol

SIP: Session Initiation Protocol

SMTP: Simple Mail Transfer Protocol

SRU: Service Router Unit

SS: Supplementary Service

Appendix I. Glossary

TCOTotal Cost Of Ownership

TDM: Time Division Multiplex

TFTP: Trivial File Transfer Protocol

TMG: Trunk Media Gateway

TX: Transmit

UA: User Agent

UACUser Agent Client

UASUser Agent Server

UC: Unified Communications UCE: eSpace UC Express

UM: Unified Media

UMS: Unified Message System

USM: Unified Session Manager

USB: Universal Serial Bus

VLAN: Virtual Local Area Network

VMS: Voice Mail Service

VoIP: Voice over IP

VRP: Versatile Routing Platform

VU: Virtual User

WAN: Wide Area Network

Key parameter

Run Date

Deadline of the license. After this date, license will return to default status.

Default license setup supports X number of users. If you need a support for more
users number (lets say N), then you should apply for a proper new license to
support such number. At the time the new license expires, only default license users
are supported after new license expiration. For example, assuming that you have
created X+N users before running U1900 system and you have a valid license to
support X+N users, after the license expires, the system only support X users. Note
that all the services are blocked until you delete the excess users or renew the
license

License Current Config Information: show the capability that the license supports .

Local Survival Right: This control item is for a local node. The value 0 indicates that this
item is controlled by a license, and the value 1 indicates that this item is not controlled by
a license. When this item is not controlled by a license, the following controls are

implemented:

"User port license" item is set to the maximum numbers supported in the system,
the others items keep the number supported by the license.
User data can be synchronized only through the central node. User data cannot be

configured using the CLI or network management system. Other data, such as trunk
configuration, can be configured using the CLI or network management system.

Information about the central node must be configured on the local node, and the
local node also needs to have heartbeat communication with the central node. If the

accumulative days of losing heartbeat communication with the central node exceed
30 days, the license becomes invalid. The license takes effect again after the local
node reconnects to the central node.

Redundancy Backup Right : This control item is for a standby node. The value 0 indicates
that this item is controlled by a license, and the value 1 indicates that this item is not
controlled by a license. When this item is not controlled by a license, the following controls
are implemented:

"Local regeneration license" item keeps the number supported by the license, the
others items(except "Open interface platform license" item) are set to the maximum
numbers supported in the system.
User data can be synchronized only through the active node. User data cannot be
configured using the CLI or network management system. Other data, such as trunk
configuration, can be configured using the CLI or network management system.
Information about the active node must be configured on the standby node, and
the standby node also needs to have heartbeat communication with the active
node. If the accumulative days of losing heartbeat communication with the active
node exceed 30 days, the license becomes invalid. The license takes effect again
after the standby node reconnects to the active node.

License Use Information: show the license that has been used.

User ports are classified into two types: analog user ports and SIP user ports.

Default number of user ports supported by U1900 without a license:

eSpace U1981/U1980/U1960/U1911: 4

eSpace U1910: 50

eSpace U1930: 100

Number of users supported after a license is loaded:

eSpace U1981/U1980/U1960/U1911: Number supported by the license

eSpace U1910 50 + number supported by the license

eSpace U1930 100 + number supported by the license

The number of voice conference concurrence user in eSpace U1910 and eSpace U1930 is
open by default , and the maximum number is 12.

eSpace U1980 provides voice conference resources for 60 parties by default. After a
license is loaded, the total number equals 60 plus the number supported by the license.

eSpace U1960 provides voice conference resources for 30 parties by default. After a
license is loaded, the total number equals 30 plus the number supported by the license.

eSpace U1911 provides voice conference resources for 16 parties by default. After a
license is loaded, the total number equals 16 plus the number supported by the license.

eSpace U1981 provides voice conference resources for 60 parties by default. After a

license is loaded, the total number equals 60 plus the number supported by the license.