IP Voice configuration

About the Voice

The General Description of the Voice Application

The Introduction of the Voice

About QoS

The signaling tone of DSP Detective Switch

Configuring Voice over IP

How VoIP Deals with Telephone Call

Premise for Configuration

The List of Task for Configuring Voice over IP

Configuring Dial-peer

Configuring Voice Port

Example of Voice over IP Configuration

Configuring the Fax Function based on Voice over IP

Configuring Gate Keeper of Voice over IP

Configuring IVR

Configuring connection service number

Configuring the dial flow and related parameters

Configuring ivr Card Telephone  

Configuring ivr direct authentication mode 

Configuring ivr once dial mode

Configuring ivr Recording Mode 

Enabling RADIUS Authentication

Enabling RADIUS Accounting

 

About the Voice

1700, 2600, 3600 Series Router of our company supports the voice transmission. The router of our company supports the voice by using the voice packet technology in which the voice signal is packetized and transmitted according to ITU-T Code H.323. H.323 belongs to the protocol family that is used for transmitting the multi-media (voice, video and data) on the data network. The brief introduction includes three parts:

l         The General Description of the Voice Application

l         The Introduction of the Voice 

l         About QoS 

The General Description of the Voice Application” introduces the invoice technology provided by IP voice equipments. “The Introduction of the Voice” presents the user who has little knowledge of the voice technology the basic introduction of the voice technology. “About QoS” briefs the conception of QoS.

The General Description of the Voice Application

IP telephone equipments of our company use IP (Internet Protocol) to transmit the voice. As the voice flow is transmitted through IP, the user shall configure the parameters related to the voice interface as well as the special-purposed functional elements, such as the dial-peer.

 

dial peer

The key to understand the voice technology of our company is the understanding of the dial-peer. The dial-peer is the binding of the telephone number, target equipment and related character. The role of the dial-peer is to define the feature related to the call leg, and the call leg is the discrete leg of call connection located between two points, just as shown in Chart 1 and 2. Four call legs make a terminal-to-terminal call. Viewing from the source IP telephone equipments, there are two call legs, as shown in Chart 1, and viewing from target router, there are two call legs, as shown in Chart 2. The user employs the dial-peer to apply the designated feature to the call leg and identify the initial point and target point of call.

Chart 1 Call Leg of Dial-peer Viewed from Source IP Telephone Connector

Chart 2 Call Leg of Dial-peer Viewed from Target Router

There are two different kinds of dial-peers:

POTS — It is the dial-peer describing characteristics of the traditional telecommunication connection. POTS points to the specific voice port on IP voice equipments. In configuring the dial-peer of POTS, the key command the user shall configure is “port” and “destination-pattern”. The command “destination-Pattern” defines the telephone number related to POTS dial-peer. The command “port” relates POTS dial-peer to the designated voice port. The usual case is that IP telephone equipments are connected with the telephone or the telephone port of local PBX.

  VoIP —It is the dial-peer describing the characteristics of IP network connection. VoIP dial-peer points to the specific voice network equipments. In configuring VoIP dial-peer, the key command the user shall configure is “destination-pattern” and “session”. The command “destination-pattern” defines the telephone umber related to the dial-peer. The command “session” designates the target IP address of the dial-peer.  

 

Voice Port

The voice port command used for IP telephone equipments defines the parameters related to the specific voice port. IP telephone equipments have three kinds of voice ports:

l         FXS —  Interface of Foreign Exchange Station. FXS port employs RJ-11 interface and allows the connection of the basic telecommunication equipments, such as the telephone.

l         FXO —The interface of Foreign Exchange Office. FXO port employs RJ-11 interface. It allows the gateway to be connected directly to the central office of the Public Switch Telephone Network (PSTN). If the local telecommunication bureau permits, the interface can also be connected to standard PBX interface. The interface is very valuable to the remote extended application.

l         E&M—The interface of Ear and Mouth (or Receive and Transmit). E&M port employs RJ-48 interface and is allowed to connect with PBX junction line.

IP telephone connector currently only provides analog voice port. The port type related to these analog voice ports depends on the interface module installed on the equipment. For example, 1700 Series Router can choose FXS、FXO、E&M voice card with two ports and 3600 Series Router can choose FXS、FXO、E&M voice module with two or four ports.

The Introduction of the Voice

It is very helpful to have some knowledge on the transmission and signaling of analog and digit for the implementation of the voice technology. This chapter presents some basic voice telecommunication information as the basis for configuring Voice over IP. The chapter includes the following subjects:

l         Numbering Mechanism

l         Analog and Digit

l         Coding Decryption

l         Mean Opinion Score (MOS)

l         Delay

l         Jitter

l         Echo

l         Signaling

It is very helpful to have some knowledge on the transmission and signaling of analog and digit for the implementation of the voice technology. This chapter presents some basic voice telecommunication information as the basis for configuring Voice over IP. The chapter includes the following subjects:

It is very helpful to have some knowledge on the transmission and signaling of analog and digit for the implementation of the voice technology. This chapter presents some basic voice telecommunication information as the basis for configuring Voice over IP. The chapter includes the following subjects:

Numbering Mechanism

It is very helpful to have some knowledge on the transmission and signaling of analog and digit for the implementation of the voice technology. This chapter presents some basic voice telecommunication information as the basis for configuring Voice over IP. The chapter includes the following subjects:

 

Analog and Digit

By so far, the telecommunication network is based on the analog basic structure. The analog transmission is not very reliable and efficient in terms of the line noise and voice restoration because the analog signal will weaken after a given distance. In order to ensure the signal transmission, the signals shall be amplified periodically. The amplification of the signal strengthens the voice signal and it also increases the background noise of the line, which leads to the decline of transmitted voice quality.

In order to resolve the above problem, the telecommunication employs the digit transmission of  Pulse Code Modulation (PCM) or Adaptive Differential Pulse Code Modulation(ADPCM). Through the application of these two methods, the analog voice is sampled at 8000 times a second and each sample is converted into digital signal, thus the analog signal is converted in to the digital signal.

Coding Decryption

PCM and ADPCM are the examples of Waveform CODEC technology, which is the compression technique that uses the redundancy feature of the waveform itself. In addition to the waveform CODEC, there is source CODEC. The waveform CODEC compresses the voice by sending the simplified parameter information related to the voice transmission and it demands the relative small bandwidth. The source CODEC includes Linear Predictive Coding (LPC), CELP and multi-pulse-multi-layer quantization (MP-MLQ).    

The most common coding standard used for the telecommunication and voice packet is G Series suggestions of ITU-T.

G.711 —    64kbps PCM voice encoding technology

G.723.1— It is a kind of compression technology that compresses the voice or audio signal at a relative low bit rate. It is a part of H324 series standards. Coding decoding is associated with two kinds of bit rates, namely 5.3 and 6.3kbps. The relative bit rate is based on ML-MLQ technology and offers a relative good sound quality to a certain extent. The relative low bit rate is based on CELP and provides the system designer with more flexibility.

G.726 —     It describes the ADPCM coding carried out at the bit rate of 40kbps, 32kbps, 24kbps and 16kbps. If PBX network is configured into the one supportive of ADPCM, the voice of ADPCM coding can be exchanged between data voice network, PSTN and PBX network.

G.728 —     It describes the delay distortion of 16kbps low rate of CELP voice compression.

G.729 —       It describes CELP compression technology that codes the voice into 8kbps stream. The standard has two distortions, namely, G.729 and G.729 Annex A. The main difference between the two is the complexity of the computing. They provide the voice quality similar to 32kbps ADPCM.

Mean Opinion Score (MOS)

Each coding decoding provides the specific voice quality. The transmitted voice quality is the average valuation result drawn from the subjective response of the hearer. A large number of hearers judges the quality of voice sample (corresponding to the specific coding decoding) on the five-score grading system (1 means the worst and 5 means the best), the means score is the mean opinion score of the voice sample. The form 1 illustrates the relationship between CODEC and MOS.

 

Compression Method

Bit Rate

Size of Frame

Result of MOS

G.711 PCM

64

0.125

4.1

G.726 ADPCM

32

0.125

3.85

G.728 LD-CELP

16

0.625

3.61

G.729 CS-ACELP

8

10

3.92

G.729 x 2 Encodings

8

10

3.27

G.729 x 3 Encodings

8

10

2.68

G.729a CS-ACELP

8

10

3.7

G.723.1 MP-MLQ

6.3

30

3.9

G.723.1 ACELP

5.3

30

3.65

Form 1-Compression Method and MOS Score

 

From the economic point of view, the application of coding decoding algorithm with low bit rate can lower the basic cost. It seems reasonable. However, the more factors should be taken into account in designing low bit rate compressed voice network. The compressed voice has some defects. The major one is the voice distortion caused by multi-coding (called series connection coding). For instance, When G729 voice signal experiences series connection coding for three times, the result of MOS declines from 3.92 (good) to 2.68 (unacceptable). Another defect of low bit rate is the delay resulted from compression and decompression.

 

Delay

The most important design consideration in the implementation of the voice is the minimum one-way end-to-end delay. The voice flow is real-time flow. If the long delay occurs at the time of sending voice packet, the voice will become unidentifiable. The delay is unavoidable on the voice network. It is caused by many different factors. The acceptable delay shall be less than 200 milliseconds.

There are two kinds of delay in the current telecommunication network, namely propagation delay and processing delay. The propagation delay is caused by the propagation speed of light on the copper medium of optical fiber or electric wave. The processing delay (sometimes called serialized delay) is caused by the equipments processing the voice information. Processing delay has remarkable influence on voice quality on the data network. Encoding/decoding delay shall be regarded as processing delay. Table 2 describes delay derived by different encoding/decoding.

 

CODEC

Bit Rate

Compression Delay

G.711 PCM

64

0.75

G.726 ADPCM

32

1

G.728 LD-CELP

16

3 to 5

G.729 CS-ACELP

8

10

G.729a CS-ACELP

8

10

G.723.1 MP-MLQ

6.3

30

G.723.1 ACELP

5.3

30

Form 2- The Delay caused by the different coding decoding

 

Another processing delay is the time needed for generating voice packet. When G.729 coding method is employed, DSP generates a frame every 10 millisecond. When two frames are put into a voice packet, the delay of the voice packet is 20 milliseconds.

The source of another processing delay is the time needed for moving the voice packet to the output queue.

Jitter

Jitter is another factor that affects the delay. When the difference exists between the expected time of receiving the voice packet and the real time of receiving the voice packet, the jitter will take place, leading to the discontinuity of real-time voice stream. IP telephone connector repeat the voice in a smooth way to compensates the jitter by setting up a playback buffer.

 

End-to-End Delay

If the user knows the end-to-end signal path/data path, coding decoding technology and effective load of the packet, it will not be very difficult to understand the end-to-end delay. The coder delay (G711 and G726 are 5 millisecond, G729 is 10 millisecond), packaging delay, the fixed part of the network delay plus the delay from the end point to coding decoding on the two ends contribute to the delay of end-to-end connection. 

 

Echo

The echo is the voice of the user heard in the telephone receiver when the user is having a telephone session. So long as the timing is right, the echo will be cleared. If the echo exceeds 25 milliseconds, it will worsen the voice and halt the session. On the traditional telecommunication network, the echo is usually caused by impedance unlatch at the time of switching from four- wire network to two-wire local ring and is controlled by the echo eliminator. In the voice packet network, the echo eliminator is embedded into the low bit rate codec and runs on each DSP. The echo eliminator shall be constrained by the total of the time for waiting for receiving the echo. The total of the time is commonly called “echo track”. It is usually 32 milliseconds.

 

About QoS

The Conception of QoS

QoS (Quality of Service) means that a network is able to provide better service to the chosen network communication by using the various basic technologies. QoS is able to control the network transmission and offer the different levels of services based on a variety of policies for the business like the voice, video frequency and others.

 

End-to-End QoS Model

The service model describes a group of End-to-End QoS ability, namely the service ability needed for transmitting the special network communication from one end to the other end. QoS supports three types of service models: Best Effort Service, Integrated Service and Differentiated Service.

 

Best-Effort Service

It is a single service model. Under this model, application can send the unlimited quantity of data when necessary without applying for permit or advising the network in advance. For the Best-Effort Service, if the condition permits, the network is able to transmit the data without the ensuring reliability, delay scope or throughput. The implementation of QoS function of Best-Effort Service is the first in, first out queue.

 

Integrated Service

It is an integrated service model and is able to meet various QoS demands. Under this service model, application requests for a special type of service to the network through QoS signaling before sending data. The request is made for the purpose of enabling the network to advise the survey of the communication of the application and applying for some special type of service that can meets the requirement on the bandwidth and delay. Only when the network has obtained the confirmed information, the application is able to send the data. What’s more, the data sent by the application shall agree with the survey of the communication described before.

 

Based on the information from the application and the network resource available, the network can achieve the right control and implement the QoS demand of the application so long as the communication load is kept within the scope speicified in the request by maintaining the status of each stream and conducting the operation of intelligent queue based on the different streams.

 

QoS uses the Resource Reservation Protocol (RSVP) to provide the function of the Controlled Load Service and Guaranteed Rate Service. The Controlled Load Service permits the application to maintain short delay and high throughput even at the time of network congestion. For this purpose, QoS offers Weighted Fair Queuing.

 

Differentiated Service

The difference between the Differentiated Service and the Integrated Service is that the application that uses the Differentiated Service does not need to send the signal to the router definitely before sending the data.

For the Differentiated Service, when the network needs to send a special service, it shall designate the corresponding QoS identifier in each data packet. The designation can be embodied in different forms, such as IP priority setting used in IP packet. The router makes the classification by using this QoS stipulation and accomplishes the task of intelligent queue. The Weighted Random Early Detection provided by QoS, Custom Queue and Priority Queue can be used for sending the Differentiated Service.

 

Signaling of QoS

 The signaling function of router QoS offers a method for the end station and network node, enabling router QoS to send the signal so as to apply for the special treatment of some communication. The signaling function of QoS helps QoS dispatch to be carried out in a better way and configures the successful and complete end-to-end QoS service for the whole network. The signaling function of QoS employs IP protocol. Either In-band signaling function (IP priority) or Out-of-Band signaling function (RSVPprotocol) indicates that any special communication class is expected to get some special QoS service.

 

IP telephone equipment provides two functions of IP priority and RSVP protocol. Each voice packet will be labeled with the corresponding identifier.

The detailed information can be referred to the related document of QoS.

 

The signaling tone of DSP Detective Switch

Detect

The command “sense cptone port slot_num/port_num dial dial_string tone_type freq_type”.

 

 “slot_num/port_num” means the port number of the signaling tone to be detected, “dial_string” means the number to be dialed for detecting some signaling tone, “tone_type” means the type of the signaling to be detected, totaling 8 in type, they are DIALTONE_PBXDIALTONE_EXTALERTTONE_PBXALERTTONE_EXTBUSYTONE_PBXBUSYTONE_EXTEMPTYTONE_PBXEMPTYTONE_EXT. Each two types form a group, once means “dialing tone”, “ringing back tone”, “busy tone” and “empty tone”. PBX means the signaling tone of PBX directly connected with DSP port. EXT means the signaling tone of EXT not directly connected with DSP port. The four kinds of PBX signaling tones are compulsory configuration (the national standard is set as the default value), the remaining four kinds of PBX signaling are optional and can be configured with 12 signaling tones at the most, which are mainly used for configuring the “busy tones”. freq_type is the type of frequency and can be classified into single frequency and dual frequency. The detail can be referred to the Directions for Use of PBX.

 

Each group of signaling tone has four group of parameters, namely high_freq (high frequency, 2001 is ineffective value and is used at the time of single frequency), low_freq (low frequency), time_on (duration of wave crest), time_off (duration of wave trough).

 

The detailed description below is based on the example of four kinds of PBX signaling tones.

1. Dialing Tone

When the dialing tone is detected (tone_type is DIALTONE_PBX or DIALTONE_EXT), “dial_string” can be filled with a randm number string but will not work. “time_on” of dialing tone is 300, “time_off” is 1023 (ineffective value).

 

2. Ringing Back Tone

When Ringing Back Tone (tone_type is ALERTTONE_PBX or ALERTTONE_EXT), “dial_string” shall be the telephone number of another port on PBX connected with the DSP port and is not occupied, namely the number corresponding to current “dial_string” is dialed through.

3. The Busy Tone

When the busy tone is detected (tone_type is BUSYTONE_PBX or BUSYTONE_EXT), another telephone (the telephone number is dial_string) on PBX connected with DSP port is picked up, then the command is used for detecting.

 

4. Empty Tone 

When Empty Tone is detected (tone_type is EMPTYTONE_PBX or EMPTYTONE_EXT), “dial_string” shall be set as the telephone number that does not exist on PBX connected with DSP port. The command can be used for detecting. Some switches do not have Empty Tone and uses the busy tone as the alternative. The detection result is consistent with that of busy tone. At this moment, the setting of Empty Tone for PBX is optional. When the signaling tone of EXT is detected, the similar method can be applied,

 

Configuration

When the signaling tone is configured (the command “cptone slot_num” is used), it should be under cptone configuration status. The four kinds of PBX signaling tones shall be configured; otherwise the default value shall be used. The dialing tone needs to be configured with only two parameters, i.e., the high frequency and low frequency. time_ontime_off are the designated values of the system (they are 300 and 1023 respectively). The remaining signaling tones shall be configured with four parameters. If the switch is the single frequency switch, the high frequency uses the ineffective value of 2001 at the time of configuration.

 

The configuration will be effective when the configuration is completed and the configuration status exits. All DSP and ports of the same slot uses the same configuration. It is recommended that the connection with different switches should use the ports that are not on the same slot unless otherwise the parameters of these two switches are consistent.

After the configuration is accomplished, the value of the configuration will be used all the time. The command “default cptone slot_num” is used for restoring the default setting of the signaling sound of the slot.

It is recommended that the command “default cptone slot_num” should be used for restoring cptone that corresponds to some slot to its default value at the time of configuring cptone before joining in the configuration status of cptone. Otherwise the consequent configured ext signaling tone will be added to the back of original ext signal and pbx signaling tone overrides the previous configuration. If cptone of the slot has no ext type signaling tone, the command “default” will not be used for resetting before configuration.

 

Configuring Voice over IP

The chapter introduces how to configure VoIP (Voice over IP) on IP telephone equipment. VoIP means the voice transmission on IP network. VoIP is the software function in essence. In order to implement the function on 1700 Series, 2600 Series and 3600 Series router of our company, voice interface module (VIM) or Voice Interface Card (VIC) shall be installed and each interface card corresponds to the special signal type related to the voice interface.

 

VoIP has the advantages below:

l         Saving the cost

l         Offering remote switch (private branch (telephone) exchange, PBX) through telnet (wide area network, WAN)

l         Integrating the transmission of voice /data

l         IP telephone gateway provides remote extension.

 

How VoIP Deals with Telephone Call

Before configuring VoIP, it is very helpful to know the event that takes place on the level of application program at the time of setting up telephone call. The usual steps of using VoIP to make the voice call at the two ends are as follows:

l         When the user picks up the telephone transmitter, the off-hook signal is sent to the signaling processing program of VoIP telephone.

l         The signaling processing program of VoIP makes the dialing tone, waiting for the user to dial number.

l         When the user has dialed the number, the dialed number is stored in the signaling processing program.

l         When the accumulated telephone number is matched with the pre-configured telephone number, the telephone number will map to IP host. IP host will connect with target telephone or PBX (private branch (telephone) exchange)

l         The session application program sets up all-directional transmission and receive channel on IP network in accordance with H.323 protocol. If the call is processed by PBX, PBX will send the call to the target telephone. If RSVP is configured (Resource Saved Program), RSVP will start to work so as to achieve expected service quality of IP network.

l         At this moment, the connected two ends employs the same coding decoding technology and uses RTP/UDP/IP as protocol stack for transmitting the voice.

l         When the call is on hook on either end, RSVP (if RSVP is used) will cancel the original reserved resource and session application program will terminate the all-directional transmission and receive channel and the session is terminated. Each end is under idle state, waiting for the next off hook to trigger the next call.

 

Premise for Configuration

Before configuring IP telephone equipments for using Voice over IP, the user shall do the work below first:

l         Setting up the working IP network

l         Installing the voice network module and voice interface card to the router.

l         Accomplishing the planning of telephone number

l         Establishing the working telecommunication network based on the telephone number planning of corporation.

l         Integrating the telephone number planning and telecommunication network into the existing IP network topology and combining IP and telecommunication network together according to the specific IP and telecommunication network topology. Usually our company recommends the user to use the standardized number as much as possible. The important is to avoid the clear difference of numbering system on the different IP telephone connectors to make the route and dial-peer user more transparent, for example, to make the second switch to avoid the second dialing tone.

l         Contacting the PBX supplier to learn how to re-configure PBX interface and to have E&M interface.

l         After the user analyzes the complete telephone number planning and decides how to integrate it into the existing IP network, the user has made the preparation for configuring network equipments.

 

The List of Task for Configuring Voice over IP

The user shall complete the configuration of the dial-peer for configuring IP voice on IP telephone connector.

First, under global configuration modem there is the command “dial-peer terminator”. The default mode has no setting of “terminator”. The dial mode is that the called number is matched once when the user presses key each time. The command can be used for setting ’#’ or ’*’ as dial terminator. Thus only when the user enters the dial terminator, the called number will be matched.

The command “dial-peer” is used for defining the dial-peer and joining into the dial-peer configuration mode. Each dial-peer defines the feature related to a call leg. The call leg is the discrete leg of call connection between the two connected points. The end-to-end call contains four call legs, two of them come from the source access server and the other two come from destination access server. The dial-peer is used for adding the attribute to the call leg and identifying call source and destination. There are two kinds of dial-peers:

 

POTS—It describes the dial-peer connected with the traditional telecommunication network. POTS peer points to the specific voice port on the voice network equipment. In order to have the minimum configuration of POTS dial-peer, the two features shall be configured: the related destination telephone and logic interface. The command “destination-pattern” is used for associating telephone number with POTS.

 

VoIP—It describes the dial-peer of connection feature of packet network. For Voice over IP, it is IP network. VoIP peer points to the specific VoIP equipment. In order to have the minimum configuration of VoIP peer, the two features shall be configured: the related destination telephone number and destination IP address. The command “destination-pattern” is used for defining the destination telephone number related to VoIP peer. The command “session” is used for designating the destination IP address of VoIP peer.

 

Notes:

l         The binding telephone number of two dial-peers shall not be the same because it will lead to the situation that a number maps to multiple ports (POTS) or maps to multiple IP address (VoIP), being at loose ends.

   

If “dial-peer terminator” is not set under the global configuration mode, the dial match will be overlapped mode, namely, the number is matched a time when the user presses the key each time. Thus any dial-peer binding telephone number A is the prefix of telephone number B, which leads to the dial of A rather than the intended the dial of B.

 

A port can bind multiple telephone numbers (POTS) and an IP address can also bind multiple telephone number (VoIP). As a matter of fact, the illegal configuration of 1 and 2 has no chance to be implemented by the command.

The additional information of configuring dial-peer and dial feature can be referred to the part of “Configuring Dial-peer”.

 

Configuring Dial-peer

The understanding of the dial-peer is the key to know how Voice over IP works. Each dial-peer defines the feature related to the call leg, just as shown in Chart 1 and Chart 2. The call leg is the discrete leg of the call connection of two connected points. All the specific connected call legs have the same connection ID.

 

Four call legs form a end-to-end call, two of which come from source router, as shown in Chart 1, and the other two come from the destination router, as shown in Chart 2. The dial-peer relates to the each of these call legs. The dial-peer is used for applying the attribute to the call leg and used for identifying call source and destination.

The dial-peer is not only used for inbound call leg but also used for outbound call leg. The inbound call leg originates from outside the router and outbound call leg originates from inside call leg. When the inbound call leg is set up, it relates to POTS dial-peer. POTS dial-peer associates telephone number with the specific voice port, which enables the incoming call of the telephone number to be received. When the outbound call leg is set up, it relates to VoIP dial-peer. VoIP dial-peer associates telephone number with the destination IP address, which enables the outgoing call of the telephone number to be implemented.

Chart 3 –the Connection of IP Voice Equipments

 

The command below shall be entered on 265010.1.2.2for configuring the connection between the source and destination as show in Chart 3.

 

2650_config#dial-peer voice 1 pots

2650_config_dialpeer#destination pattern 2601

2650_config_dialpeer#port 1/0

2650_config_dialpeer#exit

2650_config#dial-peer voice 2 pots

2650_config_dialpeer#destination pattern 2602

2650_config_dialpeer#port 1/1

2650_config_dialpeer#exit

2650_config#dial-peer voice 3 voip

2650_config_dialpeer#destination-pattern 170.

2650_config_dialpeer#session target ipv4: 10.1.1.2

2650_config_dialpeer#exit

 

The command below shall be entered on 175010.1.1.2

 

1750_config#dial-peer voice 1 pots

1750_config_dialpeer#destination-pattern 1701

1750_config_dialpeer#port 1/0

1750_config_dialpeer#exit

1750_config#dial-peer voice 2 pots

1750_config_dialpeer#destination-pattern 1702

1750_config_dialpeer#port 1/1

1750_config_dialpeer#exit

1750_config#dial-peer voice 3 pots

1750_config_dialpeer#destination-pattern 1703

1750_config_dialpeer#port 2/0

1750_config_dialpeer#exit

1750_config#dial-peer voice 4 pots

1750_config_dialpeer#destination-pattern 1704

1750_config_dialpeer#port 2/1

1750_config_dialpeer#exit

1750_config#dial-peer voice 5 voip

1750_config_dialpeer#destination-pattern 260.

1750_config_dialpeer#session target ipv4: 10.1.2.2

1750_config_dialpeer#exit

In the above example of configuration, the last number of the telephone number of dial-peer 3 of 2650 is replaced by wildcard character “. ”, which means that, on 265010.1.2.2, any call number followed by one-digit number beginning with the number “170” will lead to the connection of 1750 router (10.1.1.2 ) and also means that 1750router10.1.1.2serves the numbers followed by one digit beginning with these numbers.

 

Under the dial-peer configuration mode, the commands “shutdown” and “codec” etc are not used in this example and they can be referred to the part “The Voice Related Commands”.

 

Setting up Dial-peer Configuration Table

Before configuring the dial-peer, the user shall identify the specific data corresponding to each dial-peer. The method for accomplishing the task is to create dial-peer configuration table.

 

Here is example of Chart 4. 1750 router whose IP address is 10.1.1.2 is put in the sales office (two pieces of two-port FXS voice cards are inserted.), 3660 router with IP address of 10.1.2.2 is put in the head office (a piece of two-port FXO voice card). Through right planning and configuration, IP voice connection can be implemented between the sales office and head office. The four telephones in the sales office need to be created into the dial-peer. Router 3660 is the main network node that access to the head office and it needs to be connected with PBX of corporation. The four units in the head office needs to be created into the dial-peer, these four units are the basic telephones connected to PBX.

Chart 4- Example of VoIP Voice Network

 

The following is peer to peer point configuration table for the example in Chart 4

Dial-peer

Telephone Number

Type

Voice Port

Destination IP Address

1750

 

 

 

 

1

1701

POTS

1/0

 

2

1702

POTS

1/1

 

3

1703

POTS

2/0

 

4

1704

POTS

2/1

 

10

900.

VoIP

 

10.1.2.2

3660

 

 

 

 

1

170

VoIP

 

10.1.1.2

2

900.

POTS

1/0 and 1/1

 

Chart 3- The peer-to-peer point configuration table for the example in Chart 4

 

Configuring POTS Dial-peer

In order to configure POTS dial-peer, the user needs to define its telephone number and relates it to the related voice port. The following commands should be used under global configuration mode for joining in dial-peer configuration mode and choosing POTS type.

 

Command

Subcommand and Parameter

Function

dial-peer

voice num pots

Joining in dial-peer configuration mode for configuring POTS peer. The “num” value in the command exclusively identifies the dial-peer.

The command below shall be used under the dial-peer configuration mode for configuring the identified POTS dial-peer.

Step 

Command

Subcommand and Parameter

Function

1

destination-pattern

STR[T]

Defining the telephone number related to POTS dial-peer.

2

port

slot/port

Associating POTS dial-peer with the specific voice port.

3

trim_prefix

number

The command has different meanings in the different process.

In POTS type dial-peer, if the local called port is FXO port, the program will automatically peel off the front numbers and dial the backward numbers back to PBX automatically.

The meaning of VOIP type dial-peer can be referred to the description of VOIP type dial-peer.

 

Configuring VoIP Dial-peer

In order to configure VoIP dial-peer, the user needs its destination telephone number and destination IP address. The command below shall be used under global configuration mode for joining in the dial-peer configuration mode (and choosing VoIP type).

Command

Subcommand sand Parameter

Function

dial-peer

voice num voip

Joining in dial-peer configuration mode for configuring VoIP peer. The “num” value in the command identifies exclusively the dial-peer.

 

The following command can be used under dial-peer configuration mode for configuring the identified VoIP

Steps

Command

Subcommand and Parameter

Function

1

destination-pattern

STR[T]

Defining the destination telephone number related to VoIP dial, string is the telephone number less than 15 in length, of which the wildcard character .

2

Session

target {ipv4: ip_addr | terminal | ras}

destination-address is the point-divided format of IP address designated by the dial-peer.

Terminal means this dial-peer is used for calling H.323 terminal equipment (such as Microsoft Netmeeting). At this moment, the command “destination-pattern” is used for identifying IP address of H.323 terminal equipment. The format is the point-divided format of of IP A.B.C.D of destnation H.323 terminal equipment. The area that is less than 3 digit after the point is removed shall be filled up with the preset 0 to form a number. If the command “trim-prefix” is configured, the designated prefix shall be removed, then IP address shall be sorted according the above rules.

Ras means that the destination address information that binds to the dial-peer is obtained through RAS dynamic process resolution.

3

codec

codec_type

Configuring the coding format used for session

4

trim_prefix

number

The command has different meaning in different process.

In VOIP type dial-peer, when “session terminal” is configured, the command bears the function of peeling off the front numbers keyed by the user and then sorting the backward numbers into destination IP address.

The function of POTS type dial-peer can be referred to the description of POTS type dial-peer.

5

require-qos

 

Meaning that the session process related to the dial-peer needs ensuring voice quality.

codec_type:     {g711ar64 | g711ur64 | g729r8 | g729-compatible | g723r53 | g723r63 | g726r32 | g726r40 | g727r32 | g727r40 }

 

Configuring the Alternative of VoIP Dial-peer

In configuring dial-peer, the alternative dial-peer can be configured so that the ID designated alternative dial-peer can be used for dialing when the dial-peer does not work. For this purpose, the command below can be used:

Command

Subcommand and Parameter

Function

alternative

num preference num

Configuring the alternative dial-peer for use and its preference

After the function is configured, when the dial fails, the alternative dial-peer will be used for dialing on the sequence of preference till the dial succeeds or the entire alternative dials fail. The alternative configured in the alternative dial-peer will not be used in this dial.

 

Verification Technique 

The dial-peer configuration of the user can be checked by accomplishing the following tasks:

l         If the user only configures a relative small number of dial-peers, the command “show run” can be used for verifying whether the configured data is right, displaying the designated dial-peers or showing all the configured dial-peer.

 

Debug Techniques

When the user meets the trouble at the time of connection call and think that the trouble may be related to the dial-peer configuration, the trouble can be shot by accomplishing the following tasks:

l         For FXS port, when the connected off-hook telephone makes no dialing tone, it should be checked whether FXS port is configured with POTS type dial-peer.

l         Checking whether the binding telephone number of VoIP type dial-peer and IP address is right or not.

l         Ping the relevant IP address to confirm the connectivity of network.

l         The command “show run” is used on the connectors of local and remote IP telephone for verifying whether these IP telephone connectors have right configuration.

l         The debug command “debug vpm, debug h323, debug H225 and debug H245” is used.

 

Configuring Voice Port

At present, the voice ports of serial products of our company includes three types: FXSFXO and E&M. They have different configuration commands. Generally, the default configuration of the port is used. The detail of the configuration command can be referred to the index of IP voice commands.

The command below is used under global configuration mode for changing the voice port configuration:

Command

Subcommand and Parameter

Function

voice-port

slot/port

Joining in voice port configuration mode to configure the corresponding voice port.

Slot is the number of the slot where the port is located; port is the number of port.

 

General Configuration Command

Here listed are some general configuration commands

Command

Subcommand and Parameter

Function

comfort-noise

 

Setting whether the background noise is exported at the time of mute of session.

connection-plar

STR

After the voice port receives the off-hook of the opposite part, the port starts a VOIP call according to the hotline dial configured on the port.

description

STR

Adding the description to the designated voice port so as to avert the confusion.

output-gain

NUM

Configuring the volume of the voice port played to the user.

Shutdown

 

Shutting down the current voice port.

 

Special Configuration of E&M Port

Considering the different venders and different settings, the general-used configuration commands below are used for configuring E&M port:

Command

Subcommand and Parameter

Function

operation

{2-wire | 4-wire }

Configuring the wiring system linked to the voice port.

emsignal-in

{immediate | wink-star t |delay-dial}

Configuring the signaling employed on the port when the switch calls the local port.

emsignal-out

{immediate | wink-start | delay-dial}

Configuring the signaling employed on the port when the local port calls the switch.

type

{1 | 2 | 3 | 5}

Configuring the mode of connection of the voice port.

 

The Special Configuration of FXO Port

The frequency of switch shall be detected and adjusted at the time of using FXO port (the detail can be referred to the part “The Signaling Tone of DSP Detective Switch”).

Command

Subcommand and Parameter

Function

sense

cptone port slot/port dial [STR] tone_type freq_type

The command is used for detecting the frequency and wave pattern of various signaling tones of the switch connected with router FXO port directly or indirectly. By setting the different dial_string and making the corresponding operation (similar to the operation of ordinary telephone) at the time of detection, the various signaling tones on the different switches can be detected.

cptone

slot

Setting the parameter configuration of the signaling tone of some slot, at this moment, all the ports on the slot shall be under idle state.

 (Note: The command “sense” is used under global configuration mode)

 

 

Example of Voice over IP Configuration

The real configuration work of Voice over IP completed by the user depends on the actual topology structure. The examples below provide a starting point for the user. These examples of configuration may need to be customized to the clients so as to agree with the topology structure of the user’s network.

 

We will provide the configuration steps for the solution below:

l      Connection between FXS and FXS

l     Using Gateway Connection of PSTN Connected with FXO

l        Using IP circuit to connect two FXO 

l        Using the configuration of E&M Connection 

l       Configuring the Dial Alternative

 

The following parts introduce these examples:

Note: Each computer shall be configured with IP address used by the voice gateway.

In example 1, the configuration of 1750_1 is:

1750_1_config#gateway-cfg

1750_1_config_gw#gateway ipaddr 10.1.1.1

 

Connection between FXS and FXS

In this example, a small corporation with two separate offices in two places needs to integrate the Voice over IP into the existing IP network. As two telephones are connected with 1750_1, 1750_1 only needs to configure two POTS peer and one VoIP peer. As two telephones are connected with 1750_2, 1750_2 only needs to configure two POTS peers and a VoIP peer. The Chart 5 exhibits the example of Topology with FXS-to-FXS connection.

Chart 5—The Example of FXS-to-FXS Connection

The configuration of 1750_1 

 

1750_1_config#interface e1/0

1750_1_config_e1/0#ip address 10.1.1.1 255.255.255.0

1750_1_config_e1/0#exit

1750_1_config#ip route default 10.1.1.2

1750_1_config#dial-peer voice 1 pots

1750_1_config_dialpeer#destination-pattern 4117

1750_1_config_dialpeer#port 0/0

1750_1_config_dialpeer#exit

1750_1_config#dial-peer voice 2 pots

1750_1_config_dialpeer#destination-pattern 4118

1750_1_config_dialpeer#port 0/1

1750_1_config_dialpeer#exit

1750_1_config#dial-peer voice 3 voip

1750_1_config_dialpeer#session target 10.1.20.1

1750_1_config_dialpeer#destination-pattern 412.

1750_1_config_dialpeer#exit

1750_1_config#wr

The configuration of 1750_1 

1750_2_config#interface e1/0

1750_2_config_e1/0#ip address 10.1.20.1 255.255.255.0

1750_2_config_e1/0#exit

1750_2_config#ip route default 10.1.20.10

1750_2_config#dial-peer voice 1 pots

1750_2_config_dialpeer#destination-pattern 4121

1750_2_config_dialpeer#port 0/0

1750_2_config_dialpeer#exit

1750_2_config#dial-peer voice 2 pots

1750_2_config_dialpeer#destination-pattern 4122

1750_2_config_dialpeer#port 0/1

1750_2_config_dialpeer#exit

1750_2_config#dial-peer voice 3 voip

1750_2_config_dialpeer#session target 10.1.1.1

1750_2_config_dialpeer#destination-pattern 411.

1750_2_config_dialpeer#exit

1750_2_config#wr

Using Gateway Connection of PSTN Connected with FXO

The example below demonstrates how to configure Voice over IP for using FXO port to connect PSTN. In this example, the subscriber of 2650 in Shanghai can access to the subscriber of PSTN in Beijing through 1750 (a two-port FXO voice card). The 1750 router in Beijing connects with PSTN directly through FXO interface. Here is an assumption. The telephone number of the possible subscriber in Beijing is 8 digits (area digit of Beijing is 010), 2650 in Shanghai uses FXS ports to connect with the telephone, 1750 router in Beijing uses FXO port to connect with the telephone port on PSTN. It is assumed that the number of the port on PSTN is A, the subscriber in Shanghai can use the telephone numbered 80118012in the chart to directly dial the random telephone in Beijing. When the subscriber of PSTN in Beijing needs to call the telephone numbered 80118012, the number A shall be dialed first, then 80118012is dialed at the time of hearing the second dialing tone. Chart 6 demonstrates the topology structure of the example.

 

Chart 6- The Remote Connection between FXS and FXO

Notes: The example assumes that the Corporation has set up IP connection between its two remote offices in separate places.

 

Configuration of 2650 

 

2650_config#interface e1/0

2650_config_e1/0#ip address 192.168.1.3 255.255.255.0

2650_config_e1/0#exit

2650_config#ip route default 192.168.1.1

2650_config#dial-peer voice 1 pots

2650_config_dialpeer#destination-pattern 8011

2650_config_dialpeer#port 1/0

2650_config_dialpeer#exit

2650_config#dial-peer voice 2 pots

2650_config_dialpeer#destination-pattern 8012

2650_config_dialpeer#port 0/1

2650_config_dialpeer#exit

2650_config #dial-peer voice 10 voip

2650_config_dialpeer#session target 192.168.20.11

2650_config_dialpeer#destination-pattern 010........

2650_config_dialpeer#exit

2650_config#wr

Configuration of 1750 Dial-peer

1750_config#dial-peer voice 1 pots

1750_config_dialpeer#port 1/0

1750_config_dialpeer#destination-pattern 010........

1750_config_dialpeer#exit

1750_config#dial-peer voice 2 voip

1750_config_dialpeer#session target 192.168.1.3

1750_config_dialpeer#destination-pattern 8011

1750_config_dialpeer#exit

1750_config#dial-peer voice 3 voip

1750_config_dialpeer#session target 192.168.1.3

1750_config_dialpeer#destination-pattern 8012

1750_config_dialpeer#exit

1750_config#wr

Using IP circuit to connect two FXO

Under some circumstances, it is very useful to connect two PBX by using IP network. The example below demonstrates how to configure Voice over IP for using IP circuit and FXO port to connect the different PSTN. The general PBX uses model 5 and 4-wiring system, exports the signaling “immediate”, and imports the signaling “delay-dial”. The example here is configured on this model and uses the method of creating call through GK. 3660_3 is set as GK.

 

Chart 7-IP Connection between FXOs

 

The configuration of 3660_1

 

3660_1_config#inter e1/0

3660_1_config_e1/0#ip address 10.1.1.1 255.255.255.0

3660_1_config_e1/0#exit

3660_1_config#gatekeeper

3660_1_config_gatekeeper#zone local gkD-Link D-Link.com interface Ethernet0/0

3660_1_config_gatekeeper#gw-type-prefix 20.... gw ipaddr 10.1.1.2

3660_1_config_gatekeeper#gw-type-prefix 10.... gw ipaddr 10.1.1.3

3660_1_config_gatekeeper#no shutdown

3660_1_config_gatekeeper#exit

3660_1_config#

Configuration of 3660_2

3660_2_config#inter e1/0

3660_2_config_e1/0#ip address 10.1.1.2 255.255.255.0

3660_2_config_e1/0#exit

3660_2_config#dial-peer voice 1 pots

3660_2_config_dialpeer#destination-partten 200000

3660_2_config_dialpeer#port 1/0

3660_2_config_dialpeer#exit

3660_2_config#dial-peer voice 2 pots

3660_2_config_dialpeer#destination-partten 20....

3660_2_config_dialpeer#port 1/0

3660_2_config_dialpeer#trim-prefix 2

3660_2_config_dialpeer#exit

3660_2_config#dial-peer voice 3 voip

3660_2_config_dialpeer#destination-partten 10....

3660_2_config_dialpeer#session target ras

3660_2_config_dialpeer#exit

3660_2_config#gateway

3660_2_config_gateway#gateway ipaddr 10.1.1.2

3660_2_config_gateway#gateway gkid gkD-Link D-Link.com ipaddr 10.1.1.1

3660_2_config_gateway#no shutdown

3660_2_config#

Configuration of 3660_3 

 

3660_3_config#inter e1/0

3660_3_config_e1/0#ip address 10.1.1.3 255.255.255.0

3660_3_config_e1/0#exit

3660_3_config#dial-peer voice 1 pots

3660_3_config_dialpeer#destination-partten 100000

3660_3_config_dialpeer#port 1/0

3660_3_config_dialpeer#exit

3660_3_config#dial-peer voice 2 pots

3660_3_config_dialpeer#destination-partten 10....

3660_3_config_dialpeer#port 1/0

3660_3_config_dialpeer#trim-prefix 2

3660_3_config_dialpeer#exit

3660_3_config#dial-peer voice 3 voip

3660_3_config_dialpeer#destination-partten 20....

3660_3_config_dialpeer#session target ras

3660_3_config_dialpeer#exit

3660_3_config#gateway

3660_3_config_gateway#gateway ipaddr 10.1.1.3

3660_3_config_gateway#gateway gkid gkD-Link D-Link.com ipaddr 10.1.1.1

3660_3_config_gateway#no shutdown

3660_3_config#

 

Using the configuration of E&M Connection

The example below demonstrates how to configure Voice over IP for using E&M port to connect the different PSTN.

General PBX uses model 5, 4-wiring system, exports the signaling “immediate”, imports the signaling “delay-dial”. The example is configured on this model.

 

Chart 8-The remote interlink between E&M

 

Configuration of 3660_1

 

3660_1_config#inter e1/0

3660_1_config_e1/0#inter e1/0

3660_1_config_e1/0#ip address 10.1.1.2 255.255.255.0

3660_1_config_e1/0#exit

3660_1_config#voice-port 1/0

3660_1_config_voiceport#type 5

3660_1_config_voiceport#operation 4-wire

3660_1_config_voiceport#emsignal-in immediate

3660_1_config_voiceport#emsignal-out delay-dial

3660_1_config_voiceport#exit

3660_1_config#dial-peer voice 1 pots

3660_1_config_dialpeer#destination-partten 20

3660_1_config_dialpeer#port 1/0

3660_1_config_dialpeer#exit

3660_1_config#dial-peer voice 2 voip

3660_1_config_dialpeer#destination-partten 10....

3660_1_config_dialpeer#session target ipv4: 10.1.2.2

3660_1_config_dialpeer#exit

3660_1_config#

Configuration of 3660_2

 

3660_2_config#inter e1/0

3660_2_config_e1/0#inter e1/0

3660_2_config_e1/0#ip address 10.1.1.2 255.255.255.0

3660_2_config_e1/0#exit

3660_2_config#voice-port 1/0

3660_2_config_voiceport#type 5

3660_2_config_voiceport#operation 4-wire

3660_2_config_voiceport#emsignal-in immediate

3660_2_config_voiceport#emsignal-out delay-dial

3660_2_config_voiceport#exit

3660_2_config#dial-peer voice 1 pots

3660_2_config_dialpeer#destination-partten 10

3660_2_config_dialpeer#port 1/0

3660_2_config_dialpeer#exit

3660_2_config#dial-peer voice 2 voip

3660_2_config_dialpeer#destination-partten 20....

3660_2_config_dialpeer#session target ipv4: 10.1.1.2

3660_2_config_dialpeer#exit

3660_2_config#

 

Configuring the Dial Alternative

l          Planning: Dialing 1234, if the dial fails, the sequence configuration is replaced: 23453456.

 

Configuration Command:

Router_config#dial-peer voice 10 voip

Router_config_dialpeer#destination 1234

Router_config_dialpeer#session target ras

Router_config_dialpeer#alternative 20 preference 0

Router_config_dialpeer#alternative 21 preference 2

Router_config_dialpeer#ex

Router_config#dial-peer voice 20 voip

Router_config_dialpeer#destination 2345

Router_config_dialpeer#session target ras

Router_config_dialpeer#ex

Router_config#dial-peer voice 21 voip

Router_config_dialpeer#destination 3456

Router_config_dialpeer#session target ras

Router_config_dialpeer#ex

The same numbered dialpeer is allowed to be used for alternative. When the configuration is made, the primary dial-peer shall be placed on the front, the alternative dial-peer shall be placed on the back (the command “show running” can be used for checking) because the dial inquiry is carried out on the marshaling sequence and the principle of “first matched, first used”. It is recommended that the primary dial-peer shall be configured first, then the alternative dial-peer is configured, or ID setting of the alternative dial-peer be larger than that of the primary dial-peer. When all the dial-peers have been configured, the command “aline-dialpeer” is used under global configuration model for sequencing the dial-peer in accordance with ID.

Notes:

l          The configured telephone number in the alternative dial-peer shall be the workable number, namely the number without dot and letter T.

The command “aline-dialpeer” is irreversible. It is recommended that it be used after the perfect planning.

 

Configuring the Fax Function based on Voice over IP

The configuration method is configuring fax-protocol t38/rtp under voip type dialpeer. When these two commands are not configured, bypass mode fax is used.

The language product of our company supports the fax modes of T38 RTP and fax rate only supports the default 14400pbs. The fax configurations of FXS and FXO ports are consistent.

 

Bypass Fax

It is the default fax mode of D-Linkrouter. This fax mode is recommend when the bandwidth of network is enough. Currently, when our equipments designate codec as g711ar64, g711ur64, g726r32, g726r40, g727r32, g727r40, bypass mode fax can be made. As PCM coding is a damage-free coding method, the fax signal is the best under this coding method. This fax mode occupies the biggest bandwidth. It is recommended that codec be configured as g711ar64 or g711ur64 at the time of using this fax mode.

 

T38 Fax

This fax mode occupies the minimum bandwidth. Our equipments support T38 fax whose version is above cisco12.2.x IOS. When Cisco equipments are configured with fax protocol t38, our equipments can communicates with the Cisco equipments normally. It should be noted that the value of high-speed redundancy and low-speed redundancy implemented by our T38 is 0 (default value). When t38 fax communication with Cisco router is going on, its configuration of high-speed redundancy and low speed redundancy is 0 (default value). In addition, the voce coding of g729r8 and g729-compatible do not support t38 fax.

 

For example, t38 fax is going on between D-Link1750 and Cisco2620, just as shown in the chart below:

The configuration on D-Link1750

 

1750_config#interface e1/0

1750_config_e1/0#ip address 10.1.1.1 255.255.255.0

1750_config_e1/0#exit

1750_config#dial-peer voice 1 pots

1750_config_dialpeer#destination-pattern 4117

1750_config_dialpeer#port 1/0

1750_config_dialpeer#exit

1750_config#dial-peer voice 2 voip

1750_config_dialpeer#session target ipv4:10.1.20.1

1750_config_dialpeer#destination-pattern 4122

1750_config_dialpeer#codec g723r53

1750_config_dialpeer#fax t38

1750_config_dialpeer#exit

1750_config#write

Configuration on Cisco2620

 

2620_config#interface e0/0

2620_config_e0/0#ip address 10.1.20.1 255.255.255.0

2620_config_e0/0#exit

2620_config#dial-peer voice 1 pots

2620_config_dialpeer#destination-pattern 4122

2620_config_dialpeer#port 1/0

2620_config_dialpeer#exit

2620_configr#dial-peer voice 2 voip

2620_config_dialpeer#session target ipv4:10.1.1.1

2620_config_dialpeer#destination-pattern 4117

2620_config_dialpeer#codec g723r53

2620_config_dialpeer#fax protocol t38

2620_config_dialpeer#exit

2620_config#write

RTP Fax

configured into fax protocol t38 mode, and fax rate 14400 and fax train-mode ppp need to configured on Huawei router at the time of configuration because t38 mode fax of Huawei router is transmitted in RTP form. In addition, the voice coding of g729r8 and g729-compatible do not support rtp fax.

 

For example, rtp fax is going on between D-Link1750 and Huawei 1760.

 

Configuration on D-Link1750

 

1750_config#interface e1/0

1750_config_e1/0#ip address 10.1.1.1 255.255.255.0

1750_config_e1/0#exit

1750_config#dial-peer voice 1 pots

1750_config_dialpeer#destination-pattern 4117

1750_config_dialpeer#port 1/0

1750_config_dialpeer#exit

1750_config#dial-peer voice 2 voip

1750_config_dialpeer#session target ipv4:10.1.20.1

1750_config_dialpeer#destination-pattern 4122

1750_config_dialpeer#codec g723r53

1750_config_dialpeer#fax rtp

1750_config_dialpeer#exit

1750_config#write

 

Configuration on Huawei 1760 

1760_config#interface e0/0

1760_config_e0/0#ip address 10.1.20.1 255.255.255.0

1760_config_e0/0#exit

1760_config#dial-peer voice 1 pots

1760_config_dialpeer#destination-pattern 4122

1760_config_dialpeer#port 1/0

1760_config_dialpeer#exit

1760_configr#dial-peer voice 2 voip

1760_config_dialpeer#session target ipv4:10.1.1.1

1760_config_dialpeer#destination-pattern 4117

1760_config_dialpeer#codec g723r53

1760_config_dialpeer#fax protocol t38

1760_config_dialpeer#fax rate 14400

1760_config_dialpeer#fax train-mode ppp

1760_config_dialpeer#exit

1760_config#write

Configuring Gate Keeper of Voice over IP

VoIP gateway of our products offers the connection of PSTN and IP network and provides the appropriate translation between the transmission formats (such as from H.225.0 to H.221) and between communication programs (such as from H.245 to H.242). The purpose is to reflect the feature from a network terminal to a SCN terminal or the reverse feature. VoIP Gatekeeper provides the management function of gateway, including the registration management, bandwidth management, connection management, area management and call management, etc, it can be selected to exist in H.323 system.

 

The Brief Introduction of the Chapter:

l          Configuring Voice over IP Gateway 

l          Configuring Voice over IP Gatekeeper 

l         The example of configuration of voice gateway and voice gatekeeper 

 

Configuring Voice over IP Gateway

This part describes how to configure VoIPVoice over IPgateway on IP telephone equipment.

Configuring gateway information

The command below is used under global configuration mode for configuring the voice gateway and joining in the voice gateway configuration mode:

Command

Subcommand and Parameter

Function

gateway-cfg

 

Joining in the voice gateway configuration mode for configuring.

 

The basic steps for configuring the voice gateway are as follows:

Steps 

Command

Subcommand and Parameter

Function

1

gateway

ipaddr ipaddr

Configuring the address used by the gateway, ipaddr shall be already existed local address (the virtual address is supported).

2

gkid gkname ipaddr ipaddr [port]

Configuring the gatekeeper information of gateway registration. If port is not configured, the default port (1719) will be used. It is generally recommended that the default port be used (it shall be the same as the port configuration on GK).

3

h323id string

Configuring H.323 ID used by the gateway and registering the ID with gatekeeper. If the form of domain name is used, the postfix of the domain name shall be the same as that of GK configuration.

4

tech-prefix string

Configuring the technical prefix registered with gatekeeper. Multiple technical prefixes can be configured, 8 at the most.

 

Notes: The command “gateway tech-prefix” is ineffective to the equipment registered with D-Link GK because the definition of the command “gw-type-prefix” supported by D-Link currently is different from the definition of CISCO. The command is effective to the equipment registered with CISCO GK.

 

Verification Techniques

The voice gateway configuration of the user can be checked by completing the tasks below:

l          The command “show gateway” is used for showing the voice gateway configuration state.

l          The command “show running” is used for showing the content of voice gateway configuration.

 

Debug Techniques

When the user meets the trouble at the time of connection call and thinks that the trouble may be related to the voice gateway configuration, the trouble can be shot by accomplishing the following tasks:

l          Examining IP address of gateway, gatekeeper information and others.

l          The command “show getekway” is used on the equipment for verifying the right configuration of the voice gateway on these equipments.

l          The command “debug voip event asndebug voip event rasdebug voip event gw” is used.

 

Configuring Voice over IP Gatekeeper

This part presents how to configure VoIPVoice over IPgatekeeper on IP telephone equipment.

 

Configuring Gatekeeper Information

The command below is used under global configuration mode for configuring the voice gatekeeper and joining in the voice gatekeeper configuration mode:

Command

Subcommand and Parameter

Function

gatekeeper-cfg

 

Joining in the voice gatekeeper configuration mode for configuring.

 

The basic steps for configuring the voice gatekeeper are as follows:

Step 

Command

Subcommand and Parameter

Function

1

zone

local gkname domain ipaddr

Configuring local area information, ipaddr shall be the already existed local address.

2

remote gkname domain ipaddr [port]

Configuring the remote area information. If port is not configured, the default port will be used. It is generally recommended that the default port be used.

3

prefix gkname string

Configuring the prefix information of the area.

 

The Verification Techniques

The voice gatekeeper configuration of the user can be checked by implementing the tasks below:

l          The command “show gatekeeper” is used for displaying the voice gatekeeper configuration state

l          The command “show running” is used for displaying the content of the voice gatekeeper configuration

 

Debug Techniques

When the user meets the trouble at the time of connection call and thinks that the trouble may be related to the voice gatekeeper configuration, the trouble can be shot by implementing the following tasks:

l          Examining IP address of gatekeeper, gatekeeper information and others.

l          The command “show getekeeper” is used on the equipment for verifying the right configuration of the voice gatekeeper on these equipments.

l          The command “debug voip event asndebug voip event rasdebug voip event gw” is used.

 

The example of configuration of voice gateway and voice gatekeeper

As shown in the chart below, the network configuration of two 1750 gateways’ registration with two 2650 gatekeeper:

Chart 10 The interconnection between gateway and gatekeeper

Configuration of 2650_1 

2650_config#interface e1/0

2650_config_e1/0#ip address 10.1.1.20 255.255.255.0

2650_config_e1/0#exit

2650_1_config#gatekeeper-cfg

2650_1_config_gk#zone local gk1 zone1.com 10.1.1.20

2650_1_config_gk#zone remote gk2 zone2.com 20.1.1.20

2650_1_config_gk#zone prefix gk2 20..

2650_1_config_gk#exit

2650_1_config#wr

 

Configuration of 1750_1

 

2650_config#interface e1/0

2650_config_e1/0#ip address 10.1.1.10 255.255.255.0

2650_config_e1/0#exit

1750_1_config#gateway-cfg

1750_1_config_gw#gateway ipaddr 10.1.1.10

1750_1_config_gw#gateway gkid gk1 ipaddr 10.1.1.20

1750_1_config_gw#gateway h323id 10@zone1.com

1750_1_config_gw#exit

1750_1_config#wr

 

Configuration of 2650_2

 

2650_config#interface e1/0

2650_config_e1/0#ip address 20.1.1.20 255.255.255.0

2650_config_e1/0#exit

2650_2_config#gatekeeper-cfg

2650_2_config_gk#zone local gk2 zone2.com 20.1.1.20

2650_2_config_gk#zone remote gk1 zone1.com 10.1.1.20

2650_2_config_gk#zone prefix gk1 10..

2650_2_config_gk#exit

2650_2_config#wr

Configuration of 1750_2

2650_config#interface e1/0

2650_config_e1/0#ip address 20.1.1.10 255.255.255.0

2650_config_e1/0#exit

1750_2_config#gateway-cfg

1750_2_config_gw#gateway ipaddr 20.1.1.10

1750_2_config_gw#gateway gkid gk1 ipaddr 20.1.1.20

1750_2_config_gw#gateway h323id 10@zone2.com

1750_2_config_gw#exit

1750_2_config#wr

 

Configuring IVR

IVR is one of the audio products of Boda, It is a function module responsible for the voice interactive and providing the voice authentication and accounting service. Its accounting function needs the support of RADIUS server. If the authentication function selects RADIUS, it also needs to configure RADIUS server. This chapter introduces the basic configuration of IVR.

 

The AAA operation for IP voice call and recording the detailed information of each call needs to know the information regarding the identity of the calling party. The information (the identifier of the calling party) can be the calling number or a preset card number/password peer. If the information is the later, the IP telephone subscriber needs to enter a group of number before entering the card number/password so as to tell IP telephone system that the card number/password will be entered (otherwise the system will view the card number/password as the called number and make resolution to the number). The number is known as the connection service number.

 

In fact, the subscriber using the calling number for identity authentication also can use the connection service number, which can be considered for the purpose of uniform accounting (a group uses a connection service number) or for the convenience of managing session authority (for example, a telephone can dial the local call without using the connection service number and can dial the domestic and international call by using the connection service number). Thus, there are three kind of dial process (the first one is called once dial process, the later two are called two-time dial process).

 

l          Directly dialing the called number

l          Dialing the connection service number first, then dialing the called number.

l          Dialing the connection service number first, then entering the card number/password peer, and entering the called number finally.

The voice RADIUS is able to offer above three different basic connection flows on the configuration of the subscriber and set the parameter of flow attribute (such as the times of redial, the digit of card number/password, etc).

The master switch of whole IVR service function is provided for the purpose of masking the entire IVR configuration. When the master switch is closed, all the IVR will cease to run. Under default state, it is enable mode. The configuration below shall be made under ivr configuration state:

 

Command

Subcommand and Parameter

Function 

ivr

{enable | disable}

Enabling /Closing down IVR service

        The Default configuration is “enable”.

The contents to be introduced below are:

l         Configuring connection service number

l         Configuring the dial flow and related parameters

l        Configuring ivr Card Telephone  

l          Configuring ivr direct authentication mode 

l          Configuring ivr once dial mode

l         Configuring ivr Recording Mode 

l       Enabling RADIUS Authentication

l          Enabling RADIUS Accounting

 

Configuring connection service number

For the two-time dial subscriber, some specific connection service number shall be dialed for obtaining the service of IP telephone. To this end, the corresponding connection number shall be configured on the router first before two-time dial service is opened.

 

 

 

The configuration shall be made under dial-peer ivr configuration state.

Command

Subcommand and Parameter

Function 

destination-pattern

des-num

Configuring ivr connection service number

 

Configuring the dial flow and related parameters

 

Configuring the dial flow

The connection service number is the number of dial process and a series of flow attribute parameters shall be set for the connection service number. Each parameter has default value. If the connection service number is not configured with the parameters, it can provide the most basic service.

The two-time dial can be classified into two flows: calling number flow (the calling number authentication) and card number flow (card number/password authentication). Each connection service number needs to indicate its dial flow.

 

Configuring ivr Card Telephone

The configuration in dial-peer of IVR type

In order to configure the card telephone, the model name and ivr connection service number shall be configured first under dial-peer configuration status. To this end, the configuration below shall be made under dial-peer configuration status.

Command

Subcommand and Parameter

Function

destination-pattern

des-num

Configuring ivr connection service number

app

ivrl_card

Configuring ivr model as card telephone

Only when the card telephone is configured, the length of card number and password can be configured. The configuration of the times of re-authentication after authentication failure has the direct impact on the authentication times of direct authentication mode.

The situation that the subscriber does not enter the telephone number for a long time should be avoided. The time from the start of pressing the first key to the actual press of the first key is called the first time dialing time. The time from the real press of the first key to the end of whole dialing process is called the whole time of pressing keys. Here the waiting time of the first time dialing and whole dialing process can be configured. When the waiting time exceeds the configured one, the system will think the card number is ineffective.

Configuring authentication Information

It includes configuring authentication account information, configuring length of card number and password and the times of re-authentication (The configured length of card number and password does not include end key), configuring the waiting time of authentication, the first dialing and whole dialing process.

 

The configuration below shall be made under ivr configuration status:

Command

Subcommand and Parameter

Function

authen

card card-len key-len times

Configuring the length of card number of card telephone, the length of password and authentication times allowed.

time-out time1 time2

Configuring the waiting time of the first dialing of authentication and whole dialing process.

 

The first parameter of the first command is the length of default card number, the second parameter is the length of default password, the third parameter is the configured times of re-authentication. The first parameter of the second command is the waiting time of the first dial, the second parameter is the waiting time of whole dialing process. Under default state, the default length of card number is of 10 digits while the length of password is of 10 digits and the times of re-authentication 3 times.

Configuring the Dial Information

It includes configuring the digits of the connected telephone number, the waiting time of the first key-press and whole dialing process. The time from the start of pressing key to the real press-key is called the first time press-key time. The time from the first press-key to the end of press-key is called the waiting time of whole dial.

 

For the configured number length, when the dialed number is too short to end with end key (the default is "#") or the length of the number to be dialed is longer than the set length, the dial process will cease at the set length of the number. For example, if the configured length of the number is 5, the 4-digit number can be dialed through after pressing the end key and the 6-digit number cannot be dialed through as the dialing process ceases at the configured 5-digit length after having dialed five digits.

 

Configuring the waiting time of first press-key and whole dialing process. It should be noted that the configuration of the command would have impact on dial length of all the models and waiting time. The configuration below shall be made under ivr configuration status.

 

Command 

Subcommand and Parameter

Function

dial

dialing numlen dialing-time

Configuring the default length of the dialed number and re-dialed times allowed.

timeout time1 time2

Configuring the waiting time of the first dial and whole dial.

The first parameter of the first command is default number length of dial; the second parameter is the maximum times in dialing period. The first parameter of the second command is the waiting time of the first dial, the second parameter is the waiting time of whole dial. Under default state, the default dial length is of 10 digits, the default redial times are 3 time. 

Configuring the switch of balance warning tone and rate

When the subscriber uses the card telephone, the account balance should needs to be known to the subscriber. So the function of account balance query is necessary. However, some subscribers feels unacceptable to the account balance warning tone after the authentication is passed and want to close the account balance warning tone, the switch of account balance warning tone can be used at this time. The configuration below should be made under ivr configuration status.

 

Command

Subcommand and Parameter

Function

account-audio

 

Configuring the audible account balance query.

Under default state, account-audio is not configured.

Configuring the amount corresponding seconds in the corresponding server at the time of the query by the subscriber. Actually the seconds stored at the server is the seconds that can be used by the subscriber. When the account balance query is made, the seconds will be converted into the corresponding balance. The rate is set for the covert. The configuration below shall be made under ivr configuration status:

Command

Subcommand and Parameter

Function

account-rate

rate

Configuring the amount corresponding to 6 seconds, the unit is cent.

Under default state, the rate is 3 cents/6 seconds

 

Configuring ivr direct authentication mode

The configuration in dial-peer of IVR type

In order to configure direct authentication mode, the mode name and ivr connection service number shall be configured under dial-peer configuration status first. The configuration below shall be made under dial-peer configuration status:

Command

Subcommand and Parameter

Function

destination-pattern

des-num

Configuring ivr connection service number

app

ivrl_direct_authen

Configuring ivr mode into direct authentication mode

Configuring dial information

Configuring the digits of connected telephone number, the waiting time of the first press-key and whole dial process. The configured connected number length does not include the end key.

 

The waiting time from the start of pressing key to the real press-key is called the first time press-key waiting time. The time from the first press-key to the end of press-key is called the waiting time of whole dial. The first time press-key waiting time and the waiting time of whole dial shall be configured. It should be noted that the configuration of the command has impact on the dial length of all the modes and waiting time.

The configuration below shall be made under ivr configuration status:

Commands

Subcommand and Parameter

Function

dial

dialing numlen dialing-time

Configuring the default length of dialed number and the redial times allowed.

timeout time1 time2

Configuring the waiting time of the first dial and whole dial.

Under default state, numlen = 10,  dialing-time = 3  ,dial timeout time1 = 30 seconds, time2 = 60 seconds

 

Configuring ivr once dial mode

The once dial subscriber (the direct dial subscriber without connection service number) shall be enabled. Due to the lack of connection service number, the authentication function of single subscriber catering to some individual cannot be enabled. So the uniform authentication function shall be implemented to the whole once dial subscriber to enable the operation.

 

The command “gw-authen-h323” can be configured under global configuration mode for configuring once dial mode:

Command

Subcommand and Parameter

Function

gw-authen-h323

 

Enabling once dial authentication mode

Under default state, the switch is not configured

 

Configuring ivr Recording Mode

The configuration in dial-peer of IVR type

In order to configure the recording mode, the mode name ivr connection service number shall be configured under dial-peer configuration status. The configuration below shall be made under dial-peer ivr configuration status:

 

Command

Subcommand and Parameter

Function

destination-pattern

des-num

Configuring ivr connection service number

app

ivrl_record

Configuring the recording function of ivr mode.

For the parameters configured for recording, the destination-integrated filename of recording shall be configured first, thus the recorded file can be obtained. The default filename integrated is “user”. The files recorded by the user each time are the sub-files after the router is restarted and these filenames expand from “1”.

For example, the supposed integrated filename is “user”, the first recorded file is “user/1”, and the second will be

Configuring integrated audio filename

The configuration below is made under ivr configuration status

Command

Subcommand and Parameter

Function

file

record-gather-name filename

Configuring integrated filename

Parameter is the character string of integrated audio filename. Under default state, the integrated audio filename is “user”

After the recording filename is configured, the recording time should be considered. We provide the once recording time and the user can choose precision to a second or 0.1 second, which makes better recording effect.

 

Configuring the default time of recording

Two time parameters with different units are provided, but they are the time of once recording. Among these two time parameters, the one that expresses the less time will be effective. Under default state, the default value is 300(s) 100(0.1 s). The configuration below shall be made under ivr configuration status:

 

Command

Subcommand and Parameter

 

record

time time1 time2

Configuring the default time of recording

The first parameter and second parameter are the default time of once recording. The unit of the first parameter is second; the unit of the second parameter is 0.1 second. Under default state, the default once recording time is 15 seconds.

Example

       record time 12 10

           It indicates that the default recording time is configured, namely 12 seconds and 10* 0.1 = 1 second. The effective one is the one with less time, i.e. 1 second.

Configuring the Location for playing the file 

After the recording is finished, the recorded audio files needs to be put into the designated location for playing. The process is illustrated below. When the users get the audio filename “user/1” according to the above recording steps, the audio file can be stored in the customized location to turn it into the audio file applied in ivr voice mode. Under default state, the voice provided by Boda Corporation is applied.

For setting play-start filename, entering the username prompt tone and password prompt tone, authentication failure prompt audio filename, dial-start audio filename, dial failure audio filename, account balance use-up audio filename, the configurations below shall be made under ivr configuration status:

  

Command

Subcommand and Parameter

Function

file

play-start filename

Setting play-start filename of card telephone.

authen-user-start filename

Setting the prompt tone for entering username of card telephone.

authen-key-start filename

Setting the prompt tone for entering password of card telephone.

authen-failed filename

Setting the authentication failure audio prompt filename.

dial-start filename

Setting the audio prompt filename of dial start.

dial-failed filename

Setting the audio prompt filename of dial

interrupt-start filename

Setting the audio prompt filename of insufficient account balance.

        This parameter is the filename that the user has recorded. It name can be “user/1”.

 

Enabling RADIUS Authentication

The once dial subscriber (the direct dial subscriber without connection service number) shall be enabled. Due to the lack of connection service number, the authentication function of single subscriber catering to some individual cannot be enabled. So the uniform authentication function shall be implemented to the whole once dial subscribers to enable the operation.

 

The intercommunication between RADIUS Server and RADIUS Client on the network layer shall be ensured and the user list of all the once-dial subscribers is configured on RADIUS Server. Under default state, the authentication function of the once dial subscriber is not enabled.

The configuration below shall be made under global configuration mode

Command

Subcommand and Parameter

Function

gw-authen-h323

 

Enabling/Closing down once dial authentication mode

The twice-dial subscriber is enabled. The authentication function of the twice dial subscriber is fixed in the mode. Under default state, the authentication function is opened and cannot be changed.

 

Enabling RADIUS Accounting

Although the authentication functions of once-dial subscriber and twice-dial subscribers are separated, their accounting function is enabled simultaneously. When the accounting function is enabled on RADIUS Client, the system will produce the accounting information for the calls of whole once dial subscriber and twice dial subscriber and sent the information to the designated RADIUS Server for accounting. Under default state, the accounting function of RADIUS is not enabled. The configuration below should be made under global configuration mode.

Command

Subcommand and Parameter

Function

gw-accounting-h323

 

Enabling/Closing off the accounting function of all the subscribers.

Configuring the method for sending RADIUS accounting information.

Under our default state, there is no configuration method for responding to RADIUS accounting request made by RADIUS Client. The configuration below shall be made under global configuration mode:

Command

Subcommand and Parameter

Function

aaa

accounting connection h323 {none | wait-start | stop_only | start_stop}

Configuring the method for sending accounting information

Verification Techniques

IVR configuration of the user can be checked by completing the following tasks:

l         The command “show voip ivr configuration” is used for showing the related configuration of ivr.

Debug Techniques

When the user meets the trouble in using ivr voice interactive and in the process of accounting authentication and thinks the trouble comes from ivr module, the trouble can be shot by accomplishing the tasks below:

Examining radius configuration and aaa configuration

l         The command “show voip ivr configuration” is used for verifying ivr configuration on these equipments.

l         The commands “show voip ivr ivri-session” and “show voip ivr call-instance” are used for checking the structural information of the running ivr.

l         The debug commands “debug voip event ivridebug voip event ivrcdebug voip event ivrp” are used.

The Example of IVR Authentication Accounting Configuration

Chart 11 –IVR Authentication Accounting Configuration

Example 1 –Configuration of different modes of ivr and the specific illustration of configuration

The configuration of 2650_1

2650_1_config#dial-peer voice 10 pots

2650_1_config_dialpeer #des 1001

2650_1_config_dialpeer #exit

2650_1_config#dial-peer voice 11 pots

2650_1_config_dialpeer #des 1002

2650_1_config_dialpeer #exit

  /* In the above, the telephone number 1001 and 1002 are registered on the router. */

2650_config#aaa authentication login def radius

  /*Configuring Authentication Method */ 

2650_config#aaa accounting connection h323 wait-start radius

  /*The wait-start method is applied for configuring accounting packet.  */ 

2650_config#interface e1/0

2650_config_e1/0#ip address 192.168.0.1 255.255.255.0

2650_config_e1/0#exit

2650_1_config#gw-accounting-h323

  /*Enabling all the accounting service */ 

2650_1_config#gw-authen-h323

  /*Enabling once dial authentication service */ 

2650_1_config#radius server 192.168.0.2

2650_1_config#radius key 1111

 

  /*Configuring the secret key consistent with that on radius server */ 

2650_1_config#dial-peer voice 01 ivr

  /*It means that dial-peer type of 01 is ivr type */ 

2650_1_config_dialpeer#des 101

  /*It means ivr special service number of 01 is 101*/  

2650_1_config_dialpeer#application ivrl_card

  /*It means that ivr twice dial mode of 01 is card telephone mode */ 

2650_1_config_dialpeer#exit

2650_1_config#dial-peer voice 02 ivr

  /*It means that dial-peer type of 02 is ivr type. */ 

2650_1_config_dialpeer#des 102

  /*It means that ivr special service number of 02 is 102.   */ 

2650_1_config_dialpeer#application ivrl_direct_authen

  /*It means that ivr twice dial mode of 02 is direct authentication mode */ 

2650_1_config_dialpeer#exit

2650_1_config#dial-peer voice 03 ivr

  /*It means that dial-peer type of 03 is ivr type. */ 

2650_1_config_dialpeer#des 103

  /*It means that ivr special service number of 03 is 103. */ 

2650_1_config_dialpeer#application ivrl_record

 

  /*It means that ivr of 03 is to provide recording function service. */ 

2650_1_config_dialpeer#exit

2650_1_config#wr

  /* 

After the configuration of the above total ivr type is completed, ivr is able to run. If the user needs to set its own ivr parameter and apply the specific recording service, the parameters shall be configured and the rate shall be made in turn.

       

*/ 

2650_1_config#ivr-cfg

 /* Joining in ivr configuration status */ 

2650_1_config_ivr#account-audio

 /* Enabling the automatic switch of account balance after the authentication is passed. */ 

2650_1_config_ivr#default account-audio

 /* Restoring the default value of the rate and closing the switch */ 

2650_1_config_ivr#account-rate 4

 /* Configuring rate, setting the rate of 6 seconds as 4 cents, the default is 3 cents. */ 

2650_1_config_ivr#default account-rate

 /* Restoring the default value of rate is 3 cents/6 seconds */ 

2650_1_config_ivr#authen card 6 7 3

 /* Configuring the length of card number of card telephone as 6 digits, the password 7 digits, and the times of re-authentication is 3. */ 

2650_1_config_ivr#authen timeout 10 20

 /* Configuring the waiting time of authenticating the first dial of card telephone as 10 seconds, the time of dialing process as 20 seconds. */ 

2650_1_config_ivr#default default authen

 /* Restoring the default parameter configuration of authentication of card telephone, the length of card number is 10, the times of re-authentication is 3, the time of first dial is 30 seconds, the waiting time of dialing process is 60 seconds. */ 

2650_1_config_ivr#dial dialing 6 4

 /* Configuring the length of the dialed number as 6 digits, the times of re-dial is 4. */ 

2650_1_config_ivr#dial timeout 40 50

 /* Configuring the waiting time of the first dial is 40 seconds; the time of dialing process is 50 seconds. */ 

2650_1_config_ivr#default dial

 /* Restoring the default parameter configuration of dialing-out of card telephone, the default dialing-out number 10 digits, the times of redial is 3 times, the time of the first dial is 30 seconds, the waiting time of the dialing process is 60 seconds. */ 

2650_1_config_ivr#file record-gather-name user2

 /* Setting integrated audio filename obtained through recording as user2, the default is user. */ 

2650_1_config_ivr#file play-start user/1   

/*Getting the recording file of play-start to point to user/1 that has been recorded and existed. The configurations followed are: prompt tone for recording, the prompt tone for restart recording, the prompt tone for recording failure, the prompt tone for entering username for authentication, the prompt tone for entering password for authentication, the prompt tone for authentication failure, the prompt tone for starting dialing, the prompt tone for dial failure and prompt tone for insufficient account balance. */

2650_1_config_ivr#file record-start user/2

2650_1_config_ivr#file record-again user/3

2650_1_config_ivr#file record-failed user/4

2650_1_config_ivr#file authen-user-start user/5

2650_1_config_ivr#file authen-key-start  user/6

2650_1_config_ivr#file authen-failed  user/7

2650_1_config_ivr#file dial-start user/8

2650_1_config_ivr#file dial-failed user/9

2650_1_config_ivr#file interrupt-start user/10

2650_1_config_ivr#def file

 /* Restoring the above-configured files to the default state, namely using the audio file provided by Boda Corporation. */ 

2650_1_config_ivr#record time 5 34

 /* Configuring the default time of once recording 34 seconds, the process of judging 5 seconds >3.4 seconds, the shorter recording time is applied. */ 

2650_1_config_ivr#record key * 1

 /* Configuring recording key, the key of starting recording is ’*’, the key of re-starting recording is ’1’ */.

2650_1_config_ivr#default record

 /* Restoring the default parameter configuration of recording, the default once recording time is 15 seconds. */ 

2650_1_config_ivr#stop-key *

 /* Setting the end key of recording and dial as’*’  */

2650_1_config_ivr#default stop-key

 

 /* Restoring the end key of recording and dial */ 

2650_1_config#wr

 

Example 2—The example of card telephone

2650_1_config#dial-peer voice 10 pots

2650_1_config_dialpeer #des 1001

2650_1_config_dialpeer #exit

2650_1_config#dial-peer voice 11 pots

2650_1_config_dialpeer#des 1002

2650_1_config_dialpeer#exit

2650_config#aaa authentication login def radius

2650_config#aaa accounting connection h323 wait-start radius

2650_config#interface e1/0

2650_config_e1/0#ip address 192.168.0.1 255.255.255.0

2650_config_e1/0#exit

2650_1_config#gw-accounting-h323

2650_1_config#radius server 192.168.0.2

2650_1_config#radius key 1111

2650_1_config#dial-peer voice 01 ivr

2650_1_config_dialpeer#des 101

/*Configuring connection service number */ 

2650_1_config_dialpeer#app ivrl_card

/*Configuring card number mode */ 

2650_1_config_dialpeer#exit

2650_1_config# ivr-cfg

2650_1_config_ivr#account_audio

/*Enabling the automatic prompt of account balance */  

2650_1_config_ivr#exit

2650_1_config#wr

 

Example 3 –Example of Direct Authentication

2650_1_config#dial-peer voice 10 pots

2650_1_config_dialpeer #des 1001

2650_1_config_dialpeer #exit

2650_1_config#dial-peer voice 11 pots

2650_1_config_dialpeer#des 1002

2650_1_config_dialpeer#exit

2650_config#aaa authentication login def radius

2650_config#aaa accounting connection h323 wait-start radius

2650_config#interface e1/0

2650_config_e1/0#ip address 192.168.0.1 255.255.255.0

2650_config_e1/0#exit

2650_1_config#gw-accounting-h323

2650_1_config#radius server 192.168.0.2

2650_1_config#radius key 1111

2650_1_config#dial-peer voice 01 ivr

2650_1_config_dialpeer#des 101

/*Configuring Connection Service Number */ 

2650_1_config_dialpeer#app ivrl_direct_authen

/*Configuring Direct Authentication Mode */ 

2650_1_config_dialpeer#exit

2650_1_config#wr

Example 4 ---Example of once authentication configuration

2650_1_config#dial-peer voice 10 pots

2650_1_config_dialpeer #des 1001

2650_1_config_dialpeer #exit

2650_1_config#dial-peer voice 11 pots

2650_1_config_dialpeer#des 1002

2650_1_config_dialpeer#exit

2650_config#aaa authentication login def radius

2650_config#aaa accounting connection h323 wait-start radius

2650_config#interface e1/0

2650_config_e1/0#ip address 192.168.0.1 255.255.255.0

2650_config_e1/0#exit

2650_1_config#gw-accounting-h323

2650_1_config#gw-authen-h323

/*Configuring once dial switch */ 

2650_1_config#radius server 192.168.0.2

2650_1_config#radius key 1111

2650_1_config#wr

 

Example 5—The two steps are needed for configuring the alternative play-start of recording.

Step 1

2650_1_config#dial-peer voice 10 pots

2650_1_config_dialpeer #des 1001

2650_1_config_dialpeer #exit

2650_1_config#dial-peer voice 11 pots

2650_1_config_dialpeer#des 1002

2650_1_config_dialpeer#exit

2650_config#aaa authentication login def radius

2650_config#aaa accounting connection h323 wait-start radius

2650_config#interface e1/0

2650_config_e1/0#ip address 192.168.0.1 255.255.255.0

2650_config_e1/0#exit

2650_1_config#gw-accounting-h323

2650_1_config#gw-authen-h323

2650_1_config#radius server 192.168.0.2

2650_1_config#radius key 1111

2650_1_config#dial-peer voice 01 ivr

2650_1_config_dialpeer#des 101

2650_1_config_dialpeer#app ivrl_record

2650_1_config_dialpeer#exit

2650_1_config#ivr-cfg

2650_1_config_ivr#file record-gather-name user

2650_1_config_ivr#record time 30 100

/*Configuring the default time of once recording as 10 seconds. */

2650_1_config_ivr#exit

2650_1_config#wr

 

/*Then starting recording and obtaining the recording file user/1 */

Step 2 

2650_1_config_ivr#file play-start user/1

2650_1_config_ivr#exit

2650_1_config#wr