Voice over IP (VoIP) has many applications and one of them is commonly known as the IP phone. VoIP makes it possible for an IP network to carry voice services, such as the Plain Old Telephone Service (POTS). VoIP is based on the packet switching technology. To transmit voice data over an IP network, the voice data need to be processed by the Digital Signal Processor (DSP), encapsulated into frames, and then stored in packets for transmission. VoIP provides voice services using software but it requires hardware support, including voice interfaces provided by voice gateways or other voice terminal devices.

In the earlier 1995, the software providing the long-distance telephone service over the Internet was introduced for the first time. Such service is known as the Internet phone, the primary form of IP phone. Through the development for these years, the IP phone has gained worldwide development as a brand-new telephone service and increasingly threatens the plain phone service.

The development of IP phone is achieved due to the gigantic technology strides and market force.

l The technology achievements in these years make the voice-to-IP-packets transformation technology more mature and practical than before. The rapid development of Integrated Circuit (IC) technology enormously decreases the cost of DSP, the core component of IP phone, and hence makes the large-scale promotion of the IP phone possible in terms of technology.

l Market force is another element that cannot be ignored in the rapid development of IP phone. The VoIP network constructed using the devices like IP voice Gateways can bypass the long-distance calls onto the data network, and thus save a large amount of call charges for the subscribers.

From the earlier 1990s until now, IP phone has been evolved into the IP voice gateway from the IP phone software. So far, the applications of the VoIP technology have been developed from the simple PC products of voice services into the telecommunication services that provide the voice/fax/data transmission featured by multi-service, high reliability and good quality.

At present, the IP voice gateway can interconnect the PSTN and the Internet. In addition, PC-to-phone, phone-to-PC, and phone-to-phone technologies are sophisticated enough to significantly improve the voice quality. For these reasons, VoIP can fully meet subscribers’ requirements.

1.1.2 VoIP Basic Structure and the Related Protocols

I. Basic structure

In the plain voice service approach, all the functions from the calling party to the receiving party are implemented through PSTN. The undertaking of VoIP, however, is different.

Figure 1-1 Basic structure of VoIP system

As shown in Figure 1-1, the IP voice gateways provide telephone ports. When a calling subscriber establishes a connection with the IP voice gateway at the calling side, the voice gateway transforms analog signals to digital signals and compresses them into packets that can be transmitted over the IP network. These packets are then transmitted to the IP voice gateway at the called side, where the voice packets are restored to the recognizable analog voice signals and finally transmitted to the called party. This is a complete phone-to-phone communication process. In practice, gatekeepers might be required to perform routing and access control on an IP network.

VoIP uses User Datagram Protocol (UDP) to transmit voice data and uses Transport Control Protocol (TCP) to process and control calls on the transport layer. As UDP provides connectionless and unreliable datagram service, it cannot ensure orderly and jitter-free transmission. To solve this problem, Real-time Transport Protocol (RTP) is run on UDP, because the time stamp in the RTP header can guarantee real-time transmission and synchronization for voice and fax information.

Figure 1-2 Format of the VoIP packet

II. H.323 protocol stack and its related protocols

For the VoIP implementation, almost all venders use the standard H.323 protocol stack established by ITU-T at present. Implemented on the application layer, the H.323 protocol stack describes the terminals, devices, and services used for multimedia communication on a LAN without committed QoS, including H.225.0, H.245, G.729, G.723.1, G.711, H.261, H.263, and T.120 series protocols.

G.723.1, G.729, and G.711 are protocols for audio coding/decoding, H.263 and H.261 for video coding/decoding, H.225.0 and H.245 for system control, and T.120 series for multimedia data transmission.

RTP and RTCP (RTP Control Protocol) together guarantee the real-time transmission of voice information. The functionality of RTP is enhanced by RTCP, which mainly works to provide feedback information on data transmission quality. The information can help the application system adapt to varied network environments and troubleshoot the network.

Figure 1-3 The H.323 protocol stack

1.1.3 Basic Phone Calling Flow of VoIP

The following is the process of VoIP application program processing.

1) A subscriber picks up the phone, and the VG detects the subscriber’s off-hook action in real-time.

2) The voice interface module sends the off-hook signal to the VoIP signal processing module of the VG.

3) After hearing the dialing tone played by the session application of VoIP, the subscriber should dial before the dialing tone times out.

4) The session application of VoIP collects the number dialed by the subscriber.

5) During the process of collecting numbers, the session application matches it with configured called number template in real-time.

6) After a configured called number template is successfully matched, the phone number will be mapped into a voice gateway. This voice gateway, as the called gateway, is directly connected to the destination telephone or PBX.

7) The caller voice gateway initiates a new call to the called gateway based on H.323 protocols through IP network, and sets up a channel for receiving and transmitting the voice data.

8) The called voice gateway receives the H.323 call transmitted from IP network side, and searches for the destination phone number according to the matched called number template configured on itself. If the call needs to be processed by the PBX, the called voice gateway will pass the call between PBXs using PSTN signaling till arriving at the ultimate destination.

9) During the calling connection setup, both sides across the connection will negotiate the encoding/decoding methods to be used for the call, using the RTP protocol to transmit voice data.

10) RTP voice channel is used to transmit the prompt signals during the calling connection, and other signals that can be transmitted in-band across the IP network.

11) If any party of the call hangs up the phone, the session application of VoIP will terminate and wait for a new calling connection.

1.1.4 IP Voice Features Supported by the Voice Gateway

I. Voice subscriber line

The VG provides:

l An FXS (Foreign eXchange Station) analog voice subscriber line, usually called plain telephone service port. It is directly connected to the terminals with FXO subscriber lines (such as common analog phone sets), providing ringing current, voltage and dialing tone.

l An FXO (Foreign eXchange Office) analog voice subscriber line, known as 2-wire loop trunk port. It is connected to the analog telephone port of a PSTN central office (PBX) or the FXS interface.

l An E1VI interface, that is, the digital E1 voice subscriber line. The bandwidth of the interface is 2.048 Mbps, which is shared by 32 timeslots (TS 0 to TS 31), each having 64 kbps. It supports R2, DSS1, QSIG and digital E&M signaling.

The device with FXS ports must be connected to the device with FXO ports, such as an ordinary telephone set. The ring current that FXS and FXO receive is signals of 25 Hz and 60 VAC.

Table 1-1 Interface connections between a voice gateway and a traditional SPC switch

Voice gateway	PBX	Capacity (channel)
FXS (Phone)	Analog subscriber line (ASL)	1
FXO (Line)	Analog trunk (AT)	1
Digital E1 trunk	Digital E1 trunk	30

II. IP voice features

l Silence compression

VoIP implementation in the VG can automatically detect the time ranges of silence in a session and hence stop generating voice traffic in these time ranges. Thus, it reduces the volume of transmitted voice traffic.

l Comfortable noise

By generating some background noise, the presence of unnatural voice due to the silence compression can be eliminated.

l Jitter-free and disorder prevention

Jitter is caused by a variation in speed of inbound packets due to various delays on the network, and the difference on the sequences of inbound and outbound packets. To compensate the distorted voice caused by jitter and disorder, jitter buffer is added on the voice device at the receiving side to buffer the packets sufficiently until the slowest packet reaches. Thus, these packets can be processed in sequence. In addition to that, the system can fragment packets to transmit the packets at uniform speed to the voice interface module.

l QoS

As the voice service is highly time-sensitive, the priority transmission of voice packets must be guaranteed. The VG supports setting the ToS field in the IP packets, thereby providing QoS.

l IP fax

The IP fax system is based on VoIP to establish fax channels to transmit/receive fax data. The implementation of IP fax includes modulation and demodulation, fax protocol processing, and IP channel maintenance.

l One-stage dial and two-stage dial

The VoIP implementation of the VG supports both one-stage dial and two-stage dial access functions, which well tolerates the differences for various PBXs to transmit called numbers to the voice gateway. When a call is made from a PBX to the voice gateway, if the PBX sends the called number to the voice gateway, the voice gateway will connect the subscriber by one-stage dial access; if the PBX does not send the called number to the voice gateway, the voice gateway will adopt the two-stage dial access mode and play prompts to guide the subscriber to enter some other information, like the called number.

l Automatic busy-tone detection

The busy tones played differently on different PBXs, with different spectrum features. Therefore, it is difficult to recognize the busy-tone features with a fixed threshold. The smart busy-tone identification technology of the voice gateway can sample, calculate, and analyze the incoming busy-tone, so as to obtain a set of parameters closest to the busy-tone features. By configuring these parameters on the port, busy-tone detection can be well implemented.

III. Voice protocol standard

The IP voice standards on a voice gateway include:

l G.711 (A law and m law)

l G.729 (G.729 and G.729A)

l G.723.1 (G.723r53 and G.723r63)

l H.225.0

l H.245

l RFC1889, 1890, and so on.

IV. Special-service numbers support as provided by an SPC switch

l Do-not-disturb

With the “do-not-disturb” service activated, the called subscriber will refuse any incoming call, regardless of whether line is idle, and the calling party will hear the busy tone.

l Call transfer on busy

With the call transfer on busy function activated, a new incoming call will be transferred to a specified number if the called subscriber’s line is busy.

l Call transfer unconditional

With the service of call transfer unconditionally activated, the incoming calls will be transferred to a specified number regardless of whether the called subscriber line is busy.

l Alarm call service

With the alarm call service activated, the telephone will ring for 60 seconds at the time specified by the subscriber and automatically disconnect the line after that. This function is only valid for 24 hours.

l Lines group access

Relevant users (such as employees of a department or an enterprise) are grouped into a user group. Each user in a group has two numbers, one of which is the PSTN number of the user or a number uniformly allocated within the whole enterprise, called long number; the other is the short number used within the group. Users can dial short numbers for intra-group calls, while the long number must be dialed for inter-group calls.

1.2 Basic VoIP Settings

Basic VoIP settings mean that with the default settings in a voice gateway and in a simple networking environment, phone calls can be made. The settings include:

l Enter the voice entity view

l Configure the POTS voice entity

l Configure the VoIP voice entity

To correctly configure VoIP, you must understand voice entities first. In a complete phone-to-phone connection, the call can be divided into four call segments, according to different positions of users (the calling side and the called side), with each segment corresponding to a voice entity.

Figure 1-4 Call segments viewed from voice gateways at both ends

As shown above, there are two types of voice entities used in VoIP communications:

l POTS voice entities. POTS means plain old telephone service, corresponding to the local call (or PSTN) side. The configuration of POTS voice entities is to associate the physical voice subscriber lines with the local telephone devices.

l VoIP voice entities, the configuration of which is to map telephone numbers with IP addresses (the IP address of the gateway where the number is located, or the IP address of the GK that can resolve this number). Compared with POTS voice entities, VoIP voice entities correspond to the IP side.

1.2.1 Entering the View for Configuring a Voice Entity

I. Entering voice view

To configure the related parameters for VoIP, use the voice-setup command to enter voice view.

& Note:

VoIP in this manual refers to both VoIP and FoIP, if not specified.

Perform the following configuration in system view:

Table 1-2 Enter voice view

Operation	Command
Enter the voice view	voice-setup

II. Entering voice dial program view

To configure the POTS voice entity and VoIP voice entity, use the dial-program command to enter voice dial program view.

Perform the following configuration in voice view.

Table 1-3 Enter voice dial program view

Operation	Command
Enter voice dial program view	dial-program

1.2.2 Configuring the POTS voice entity

The basic configuration of POTS voice entity includes:

l Create a POTS voice entity

l Configure the called-number template for the voice entity

l Configure the local subscriber-line number

I. Creating a POTS voice entity

Perform the following configuration in voice dial program view:

Table 1-4 Create a POTS voice entity

Operation	Command
Create a POTS voice entity and enter the POTS voice entity view	entity number pots
Delete the POTS voice entity	undo entity { number \| all \| pots }

II. Configuring the called-number template for the POTS voice entity

The called-number template of the voice entity is used to define the telephone number associated with the POTS voice entity, namely the telephone number of the phone set connected with the local voice gateway.

Perform the following configuration in POTS voice entity view:

Table 1-5 Configure the called-number template for the voice entity

Operation	Command
Configure the called-number template for a voice entity	match-template string
Delete the called-number template for a voice entity	undo match-template

No called-number template for the voice entity is set by default.

III. Configuring the local subscriber-line number

The local subscriber-line number configuration of the voice entity is to associate the POTS voice entity with a physical voice subscriber line, which usually connects the voice gateway to the local central office (the FXS and FXO voice subscriber lines and digital voice subscriber lines generated by the E1VI interface).

Perform the following configuration in POTS voice entity view.

Table 1-6 Configure the local subscriber-line number

Operation	Command
Configure the local subscriber-line number	line line-number
Remove the local subscriber-line number	undo line

By default, no local subscriber-line number is configured.

1.2.3 Configuring the VoIP Voice Entity

The required configuration tasks for the VoIP voice entity include:

l Create a VoIP voice entity

l Configure the called-number template for the VoIP voice entity

l Configure the routing program for reaching the called voice gateway

I. Creating a VoIP voice entity

Perform the following configuration in voice dial program view.

Table 1-7 Create a VoIP voice entity

Operation	Command
Create a VoIP voice entity and enter VoIP voice entity view	entity number voip
Delete the VoIP voice entity	undo entity { number \| all \| voip }

II. Configuring the called-number template for the VoIP voice entity

The called-number template of the VoIP voice entity is used to define the called number that is associated with the VoIP voice entity, namely the telephone number of the user at the IP side.

Perform the following configuration in VoIP voice entity view:

Table 1-8 Configure the called-number template for the voice entity

Operation	Command
Configure the called-number template for a voice entity	match-template string
Delete the called-number template for a voice entity	undo match-template

By default, no called-number template for the VoIP voice entity is configured.

III. Configuring the routing program for reaching the called voice gateway

When the IP address is directly configured, the point-to-point routing program is adopted. The calling gateway initiates the call connection directly to the called gateway. If the voice gateway needs the voice service provided by the gatekeeper and manages voice calls through the gatekeeper, the RAS signaling must be used.

Perform the following configuration in VoIP voice entity view.

Table 1-9 Configure the routing program for reaching the called voice gateway

Operation	Command
Configure the routing program for reaching the called voice gateway	address { ip ipaddress \| ras }
Delete the configured routing program	undo address

No default routing program is configured for reaching the called voice gateway.

1.3 Advanced VoIP Settings

With advanced VoIP settings you can adjust or configure the VoIP-related advanced parameters, enable or disable some auxiliary functions, and optimize the system performance. These settings are not needed unless in special applications.

The advanced settings include (the order is insignificant):

l Configure the voice gateway H.323 descriptor

l Configure the duration of the timers in H323 protocol stack

l Configure the voice subscriber line

l Perform the advanced settings for the POTS voice entity

l Perform the advanced settings for the VoIP voice entity

l Configure the dial parameters and the number match policy

l Configure the default values for global voice parameters

l Configure the numbers of special services

l Configure busy-tone detection for the FXO port

l Configure off-hook/on-hook detection sensitivity

l Reset voice card

l Configure fast connection

l Configure the transmission mode of DTMF codes

l Configure the tunneling function

l Configure voice Caller ID Display (CID)

l Configure voice performance

l Configure T.38 capacity description compatibility

1.3.1 Configuring the voice gateway H.323 descriptor

Perform the following configuration in voice view.

Table 1-10 Configure the voice gateway H.323 descriptor

Operation	Command
Configure the voice gateway H.323 descriptor	voip h323-descriptor descriptor
Delete the voice gateway H.323 descriptor	undo voip h323-descriptor

The default H.323 descriptor is “Voice-Gateway”.

& Note:

The default value of the H.323 descriptor is used in configuration. You are suggested not to change the descriptor unless necessary. If you do need to change it, perform the configuration under the guide of networking engineers.

1.3.2 Configuring the Duration of the Timers in H.323 Protocol Stack

Perform the following configuration in voice view.

Table 1-11 Configure the duration of the timers in H.323 protocol stack

Operation	Command
Configure the duration of the timers in H.323 protocol stack	voip h323-timer { socket-create \| twaitalerting \| twaitconnect \| twaitsetup } seconds
Restore the default duration of the timers in H.323 protocol stack	undo voip h323-timer { socket-create \| twaitalerting \| twaitconnect \| twaitsetup }

By default, the duration of the socket-created timer is 30 seconds, the timer that waits for the Altering message 20 seconds, the timer that waits for the Connect message 200 seconds, and the timer that waits for the Setup message 20 seconds.

1.3.3 Configuring the voice subscriber line

The voice gateway provides analog voice subscriber lines for VoIP implementation, including two basic types: FXS and FXO. FXS and FXO ports on a voice gateway are numbered together. The following figure shows how the voice subscriber lines on the H3C VG 10-41 voice gateway are numbered, and the FXO ports are numbered right next to the FXS ports.

Figure 1-5 Number the subscriber-line of VG 10-41

The configurations of a voice subscriber-line mainly include the configuration of some physical features of its port. Normally, you can use the default settings of the voice subscriber-line parameters instead of reconfiguring them.

& Note:

The VG 10-41 supports the emergency function, that is, the FXS 3 port and the FXO port (Line ports) stay connected when the voice gateway is powered off, so that subscribers can continue to place PSTN calls.

I. Entering voice subscriber-line view

You must enter voice subscriber-line view before configuring voice parameters for the FXS port.

You can use the following command in voice view.

Table 1-12 Enter the voice subscriber-line view

Operation	Command
Enter the voice subscriber-line view	subscriber-line line-number

II. Configuring voice parameters on an FXS subscriber-line

The FXS port is directly connected to ordinary telephones, fax machines, or PBXs through the standard RJ-11 telephone cable and they exchange signaling through level changes of Tip and Ring lines to provide ringing, voltage and dialing tone.

Voice parameter configuration on the FXS port includes:

l Configure voice subscriber-line description

l Enable/Disable voice subscriber-line

l Configure the comfortable noise

l Enable private-line auto-ringing

l Configure private-line auto-ringing mode

l Enable echo cancellation

l Configure echo duration

l Configure echo cancellation parameters

l Configure voice receive-gain and transmit-gain

l Configure sensitivity level of DTMF code detection

l Configure dial time parameters of the voice subscriber-line

l Allow the FXS to send called numbers

l Configure the polarity reverse function of the FXS interface

l Configure the electric impedance of the voice subscriber-line

l Configure packet loss compensation mode

Perform the following configuration in FXS voice subscriber-line view.

Table 1-13 Configure FXS voice subscriber-line

Operation	Command
Configure voice subscriber-line description	description string
Delete the voice subscriber-line description	undo description
Enable a voice subscriber-line	undo shutdown
Disable a voice subscriber-line	shutdown
Enable the comfortable noise	cng-on
Disable the comfortable noise	undo cng-on
Specify an E.164 telephone number for private-line auto-ring	private-line string
Delete the E.164 telephone number for private-line auto-ring	undo private-line
Configure the ring mode of private line	private-type { delay \| quick }
Restore the default ring mode of private line	undo private-type
Enable echo cancellation	echo-canceller enable
Configure echo cancellation parameters	echo-canceller { tail-length milliseconds \| parameter { convergence-rate value \| max-amplitude value \| mix-proportion-ratio value \| talk-threshold value } }
Restore the default values of echo cancellation parameters	undo echo-canceller { enable \| tail-length \| parameter { convergence-rate \| max-amplitude \| mix-proportion-ratio \| talk-threshold } }
Configure voice receive-gain	receive gain value
Restore the default voice receive-gain	undo receive gain
Configure voice transmit-gain	transmit gain value
Restore the default voice transmit-gain	undo transmit gain
Configure a sensitivity level for DTMF code detection of analog voice subscriber line (FXS and FXO)	dtmf threshold index value
Restore the default sensitivity level of DTMF code detection of analog voice subscriber line (FXS and FXO)	undo dtmf threshold index
Configure the sensitivity level of DTMF code detection	dtmf sensitivity-level { high \| low }
Restore the default value of the sensitivity level of DTMF code detection	undo dtmf sensitivity-level
Configure the timeout time of waiting for dialing the first digit	timer first-dial seconds
Configure the timeout time of waiting for dialing the next digit	timer dial-interval seconds
Restore the default settings of the intervals concerning voice subscriber-line dial	undo timer { first-dial \| dial-interval }
Enable the FXS to send called numbers.	send-number
Disable the FXS to send called numbers.	undo send-number
Configure the electric impedance of the FXS voice subscriber-line	impedance impedance-mode
Restore the default electric impedance of the FXS voice subscriber-line	undo impedance
Enable the polarity reverse function of the analog voice subscriber-line	polarity-reverse
Disable the polarity reverse function of the analog voice subscriber-line	undo polarity-reverse
Configure the packet loss compensation mode	plc-mode { general \| specific }
Restore the default packet loss compensation mode	undo plc-mode

Perform the following configuration in voice view.

Operation	Command
Enable the polarity reverse function of the analog voice subscriber-line globally	vi-card polarity-reverse { all \| line number }
Disable the polarity reverse function of the analog voice subscriber-line globally	undo vi-card polarity-reverse { all \| line number }

For details about the above commands, refer to H3C VG Series Voice Gateways Command Manual.

III. Configuring voice parameters on an FXO subscriber-line

FXO is a 2-wire loop trunk. With an RJ-11 telephone cable, the FXO port can connect local calls to the PSTN central office or the PBX. It also implements signaling interaction through the level changes of the Tip and Ring lines. The utility of FXO subscriber-line can only be connected to the utility of FXS subscriber-line.

Voice parameters configuration on FXO subscriber-line includes:

l Configure voice subscriber-line description

l Enable/Disable a voice subscriber-line

l Configure comfortable noise

l Configure private-line auto-ring

l Enable echo cancellation

l Configure echo duration

l Configure echo cancellation parameters

l Configure voice receive-gain and transmit-gain

l Configure sensitivity level of DTMF code detection

l Configure dial time parameters for the voice subscriber-line at the called side

l Configure the electric impedance for the voice subscriber-line

l Configure the called side to generate ring-back tone to the peer during fast connection

l Configure the off-hook mode of FXO voice subscriber line

& Note:

Most of the configuration tasks listed above are the same as the previous section “Configure voice parameters on an FXS subscriber-line”. So only the unique configurations for the FXO subscriber-line will be described in the following.

Perform the following configuration in FXO voice subscriber-line view.

Table 1-14 Configure the FXO voice subscriber-line

Operation	Command
Configure the duration for the calling party to send DTMF codes	delay dtmf milliseconds
Configure the interval at which the calling party sends DTMF codes	delay dtmf-interval milliseconds
Configure the start-dial delay	delay start-dial seconds
Restore the default duration and interval for the calling party to send DTMF codes as well as the default start-dial delay	undo delay { dtmf \| dtmf-interval \| start-dial }
Configure the electric impedance for an FXO voice subscriber-line	impedance impedance-mode
Restore the default electric impedance for an FXO subscriber-line	undo impedance
Configure the FXO off-hook mode	hookoff-mode { immediate \| delay }
Restore the default FXO off-hook mode	undo hookoff-mode

1.3.4 Advanced Settings for the POTS Voice Entity

POTS voice entity settings include (optional):

l Configure the voice entity description

l Enable/disable a voice entity

l Configure the voice compression mode

l Configure to send called number

l Enable the voice activity detection

l Configure the length of time for a voice packet

l Configure the permitted caller

I. Configuring the voice entity description

Perform the following configuration in voice entity view:

Table 1-15 Configure voice entity description

Operation	Command
Configure the voice entity description	description string
Delete the voice entity description	undo description

No default voice entity description is configured.

II. Enabling/Disabling a voice entity

Perform the following configuration in POTS voice entity view.

Table 1-16 Enable/disable a voice entity

Operation	Command
Enable a voice entity	undo shutdown
Disable a voice entity	shutdown

A voice entity is enabled by default.

III. Configuring the voice compression mode

Perform the following configuration in POTS voice entity view.

Table 1-17 Configure the voice compression mode

Operation	Command
Configure the voice compression mode	compression { 1st-level \| 2nd-level \| 3rd-level \| 4th-level } { g711alaw \| g711ulaw \| g723r53 \| g723r63 \| g729a \| g729r8 }
Restore the default voice compression mode	undo compression { 1st-level \| 2nd-level \| 3rd-level \| 4th-level }

The default voice compression mode is g729r8. Currently, the VG 20-16, VG 20-32 and VG 21-08 do not support the G.723 voice compensation mode.

IV. Enabling to send the called number

When the voice gateway receives a call that is to be sent to the called party through the POTS voice subscriber-line, the system determines if it should truncate the fixed number before the wildcard “.” in the match-template command according to the configuration in the send-number command. If the called number is truncated, only the number corresponding to the wildcard is sent through the POTS subscriber-line, that is, the digits before the wildcard are removed. Otherwise, the whole number is sent.

Perform the following configuration in POTS voice entity view.

Table 1-18 Configure number sending modes

Operation	Command
Configure number sending modes	send-number { digit-number \| all \| truncate }
Truncate the called number	undo send-number

By default, the system sends the truncated number.

V. Enabling the voice activity detection

Perform the following configuration in POTS voice entity view.

Table 1-19 Enable the voice activity detection

Operation	Command
Enable the voice activity detection	vad-on
Disable the voice activity detection	undo vad-on

The voice activity detection is disabled by default.

& Note:

The G.711 voice compensation mode does not support voice activity detection. After the voice activity detection function is enabled, if the negotiated voice compensation mode is G.711, the voice activity detection function will not work.

VI. Configuring the length of time for a voice packet

Perform the following configuration in POTS voice entity view.

Table 1-20 Configure the length of time for a voice packet

Operation	Command
Configure the length of time for a voice packet	payload-size
Restore the default length of time for a voice packet	undo payload-size

By default, g711 is 20, g723 is 30, and g729 is 30 in milliseconds. Currently, VG 20-16, VG 20-32 and VG 21-08 do not support G.723.

VII. Configuring the permitted caller

For the sake of voice communication security, you can use the caller-permit command to permit an incoming call. Only the reliable callers are allowed to call in, that is, voice entity can only be used by the specified callers.

If you have not defined caller-permit, any caller can call this voice entity. If you have, only the callers specified in caller-permit are allowed to call the voice entity.

Perform the following configuration in voice entity view:

Table 1-21 Configure the permitted caller

Operation	Command
Configure the calling number to permit an incoming call	caller-permit permit-num
Delete the calling number of an permitted incoming call	undo call-permit { permit-num \| all }

By default, any incoming call is allowed.

VIII. Configuring the priority of a voice entity

In the voice entity match policy, if the same number match template, voice entities with higher priority will be fist matched. Together with the command select-rule rule-order, this configuration can enable special call control and routing policies.

Table 1-22 Configure the priority of a voice entity

Operation	Command
Configure the priority of the voice entity	priority priority-order
Restore the default priority of the voice entity	undo priority

By default, the priority is 0. The smaller the value, the higher the priority. Namely, 0 represents the highest priority and 10 represents the lowest priority.

1.3.5 Advanced Configurations for the VoIP Voice Entity

Advanced settings for the VoIP voice entity include:

l Configure the area ID

l Configure Port

l Configure the precedence for IP data packets

l Configure voice compression mode

& Note:

Most of the configuration tasks listed above are the same as those in the previous section “Advanced Settings for the POTS Voice Entity”, except that these configurations are performed in VoIP voice entity view. So only the unique and important configurations for the VoIP voice entity will be introduced in the following.

I. Configuring the area ID

Perform the following configuration in VoIP voice entity view.

Table 1-23 Configure the area ID

Operation	Command
Configure the area ID	area-id string
Delete the area ID	undo area-id

No default area ID is configured.

II. Configuring the precedence for IP data packets

To ensure voice quality and reduce the time delay when data packets are transmitted together with voice and fax packets, you need to assign a higher precedence to voice data with the ip-precedence command. This command configures the precedence of voice and fax packets (the ToS field of IP packets) related to some voice entities.

Perform the following configuration in VoIP voice entity view.

Table 1-24 Configure the precedence of voice and fax packets

Operation	Command
Configure the precedence for voice and fax IP data packets	ip-precedence tos-value
Restore the default precedence of voice and fax IP data packets	undo ip-precedence

The default precedence of voice and fax packets is 0. The lager the value, the higher the precedence.

III. Configuring the port number

The port number of a VoIP voice entity is the one used by the calling voice gateway to set up a call (TCP) connection with the called voice gateway. Normally the well-known number 1720. But for some devices, this port number cannot be used and you need to configure one as required by the called gateway.

Perform the following configuration in VoIP voice entity view.

Table 1-25 Configure the port number in the VoIP voice entity

Operation	Command
Configure the port number of the VoIP voice entity.	peer-signal-port port-number
Restore the default port number of the VoIP voice entity.	undo peer-signal-port

The port number of the VoIP voice entity defaults to 1720.

IV. Configuring the voice compression mode

Perform the following configuration in VoIP voice entity view

Table 1-26 Configure the voice compression mode

Operation	Command
Configure the voice compression mode	compression { 1st-level \| 2nd-level \| 3rd-level \| 4th-level } { g711alaw \| g711ulaw \| g723r53 \| g723r63 \| g729a \| g729r8 }
Restore the default voice compression mode	undo compression { 1st-level \| 2nd-level \| 3rd-level \| 4th-level }

The default voice compression mode is g729r8. Currently, the VG 20-16, VG 20-32 and VG 21-08 do not support the G.723 voice compensation mode.

1.3.6 Configuring the Dial Parameters and Number-Match Program

I. Introduction to the dial program

With increasingly wide application of VoIP, more and more dialing schemes are created to enhance service flexibility and satisfy users’ requirements. Thus, reasonable and operable number management becomes increasingly important. To manage voice gateway numbers and create a management policy for all the numbers, we have launched a flexible dial program solution. It helps manage the out-of-order numbers in the system and optimizes the management, ensuring more convenient and reasonable number management.

In terms of dial program application, the calling and called sides have different processes. They are discussed in the following sections.

1) Dial program process of calling side

The following figure shows the dial program process of calling side.

Figure 1-6 Dial program process of the calling side

The calling/called number of a voice call coming from PSTN or initiated by the local voice subscriber line must first undergo number substitute related to the voice subscriber line. Then the system determines if the called number is of special services. If yes, it implements the process appropriate to special services. Otherwise, it implements global number substitute. After that, it selects the suitable voice entity according to certain rules (voice entity select rule), and then substitutes the calling/called number under voice entity. Finally, it initiates a call to the called party and sends the calling/called number.

2) Dial program process of called side

The following figure shows the dial program process of called side.

Figure 1-7 Dial program process of the called side

When the called side receives a call (called number), it converts the calling/called number in global scope. After that, it selects the suitable voice entity according to certain rules. If the called party is connected to a local voice subscriber line, it connects the subscriber line directly. If the called party is on PSTN network, it initiates a call to PSTN, and sends the calling/called number, and indicates PBX of PSTN to connect the call.

3) Configure outgoing call routing program

By default, the system randomly selects the voice entity to match a called number, thereby determining the outgoing route. In a routing program, the match-template commands in both VoIP voice entity and POTS voice entity views are configured with the same called number, so as to achieve the following:

l PSTN backup IP routing: Find a match among VoIP voice entities first, and then among POTS voice entities. For a called number, the system checks VoIP voice entities first, and only when the peer voice gateway defined by the VoIP voice entity is unreachable (they cannot communicate on the network layer), will the system find a match among the POTS voice entities according to the same called number.

l Backup GK routing: Find a match among VoIP voice entities by GK routing first, then among the VoIP entities by defining the fixed IP addresses, and finally among POTS voice entities. For a certain called number, when the GK server defined by VoIP voice entity via the ras parameter is unreachable, the system will find a match among POTS voice entities according to the same called number.

When finding a match among VoIP voice entities, the system supports three VoIP voice entity routing programs.

l Static routing program: Use the address ip command, and find the address of the destination voice gateway in static mode.

l Dynamic routing program: Use the address ras command. The voice gateway and GK Server interact with the RAS (Registration, Admission, and Status) information and GK Server dynamically returns the peer voice GW address that matches the called number.

l Static and dynamic integrated routing program: If two voice entities have the same match-template configuration (they have the same called number) and are configured with the address ras command and the address ip command respectively, the system first select a route according to the configured routing rule; if the selected entity fails, the system will select another voice entity to complete the call, thus to support the dynamic and static integrated routing program.

4) Regular Expressions

Regular expressions are a powerful and flexible tool for pattern matching and substitution. They are not restricted to a language or system and have been widely accepted.

When using a regular expression, you need to construct a matching pattern according to a certain rule, and then compare the matching pattern with the target object. The simplest regular expressions exclude all metacharacters. For example, you can specify a regular expression “hello”, which only matches the character string “hello”.

For flexible matching mode construction, regular expressions are allowed to contain some special characters, called metacharacters, to define how other characters appear in the target object. The following table describes the metacharacters.

Table 1-27 Metacharacters

Symbol	Description
0-9	Each digit, among 0 and 9, represents a digit.
# and *	Each represents a valid digit.
.	Wildcard, which can match any digit of valid number. For example, “555. . . .” can match any number string that starts with “555” and has four additional characters.
-	Connector, used to connect two values (it is preceded by the smaller one and followed by the larger one) to express a range. For example, “1-9” represents the range of 1 to 9 (including 1 and 9).
[ ]	Represents a character selection range. It can be used together with “!”, “%” or “+”. For example, “[235-9]” represents only one single character of “2” or “3”, or between “5” and “9” can be matched.
( )	Represents a group of characters. For example, “(086)” represents the character string “086”. This pair is usually used together with “!” “%” or “+”. For example, “(086)!010” can match the two character strings “086010” and “010”.
!	Specifies that the preceding sub-expression can be absent or present once. For example, “(010)!12345678” can match “12345678” and “01012345678”.
+	Specifies that the preceding sub-expression can be present one or more times. However, when it appears at the beginning of a whole number, it means that this number is an E.164 number. In this case, the “+” character neither represents any specific number nor means number repetition. For example, “(1)9876(54)+” can match “987654”, “98765454”, “9876545454” and so on. “(2)+110022” means that 110022 is an E.164 number.
%	Specifies that the preceding sub-expression can be absent or present multiple times. For example, “9876(54)%” can match “9876”, “987654”, “98765454”, “9876545454” and so on.

& Note:

The control characters “!” “+” and “%” mean that the preceding sub-expression (a character or a group of characters) can be presented for specific times. For example, “(100)+” can match 100, 100100, 100100 and so on. During the number matching, a number is considered completely matched if it matches any one of the numbers. In the case of longest matching, the system does not wait for succeeding dialed numbers when the number is completely matched. Refer to the T-mode related sections for circumstances where the system needs to wait for succeeding dialed numbers.

Characters “\” and ”|” are mainly used in regular expressions but not in user-configuration.

Character “\” is used for meaning transfer. For example, “\+” represents the normal character “+”, but not a control character, which is considered when “+” is separately used in a regular expression. Similarly, all control characters need to be preceded with a meaning transfer character when they are to be used as normal characters.

Character “|” represents the logical OR relationship between character(s) on its left and the character(s) on it right. For example, “0860108888|T” means “0860108888” or “T”.

Dial program configuration tasks include:

l Configure global number-match method

l Configure dialing terminator

l Configure voice entity select rule

l Configure and bind the max-call set

l Configure number-substitute

l Configure number information attribute

l Configure number sending modes

II. Configuring global number-match method

To satisfy the diversified subscriber dial-up schemes, the longest number matching and the shortest number matching are included in the number-match method.

During the process of dialing and call connection, direction of the call is determined by the match template, i.e., the preset number in match-template. If the longest number matching is adopted, the system tries to match a longer number. During dialing process, if there is a possibility that the input number will be matched a longer match template, the system waits for a longer input number. If no result is received within the time limit, the shortest number matching is selected. If the shortest number matching is used, it will initiate a call as long as the input number can be matched a match template. The number that is dialed by the subscriber after that will be ignored.

Perform the following configuration in voice dial program view.

Table 1-28 Configure global number-match program

Operation	Command
Configure the global number-match program	number-match { longest \| shortest }

By default, the shortest number matching method is adopted.

& Note:

When R2 CAS is adopted in E1 voice, the longest number matching method is currently not supported.

III. Configuring dialing terminator

Dialing terminator is mainly used to notify the voice gateway that dialing is completed. And it should establish a call connection with the existing number without delay, even when number-match is set to longest.

Perform the following configuration in voice dial program view.

Table 1-29 Configure dial terminator

Operation	Command
Configure dialing terminator	terminator character
Delete dialing terminator	undo terminator

By default, no dialing terminator is configured.

IV. Configuring voice entity select rule

If multiple voice entities can match a call number, the system follows some rules to select a voice entity. That means the preference of the voice entities is determined according to the configured rules. Voice entities can be selected according to the type-first rule or priority rules.

Type-first rule means different types of voice entities (VoIP, and POTS) are configured to different priorities. By default, their type priorities are ignored.

Other rules include precise match, priority, random select, and longest unused. The system takes out one to three rules from the above four rules and place them in order. It determines the priorities of voice entities according to the first rule, and follows the second rule to determine the sequence of those of the same priority, and so on.

& Note:

Usually, type-first rule is used first, and then other rules are used according to other indexes.

During the search for voice entities, you can configure the maximum number of voice entities found, so as to avoid infinite search. The system stops searching when the maximum number is reached. You can also configure it to stop searching when it has searched the specified voice entity.

1) Configure the type-first rule for voice entity selection

Perform the following configuration in voice dial program view.

Table 1-30 Configure the type-first rule for voice entity selection

Operation	Command
Configure the type-first rule for voice entity selection	select-rule type-first 1st-type 2nd- type
Delete the type-first rule	undo select-rule type-first

By default, voice entity is not selected according to their types (VoIP and POTS).

2) Configure the priority rules for voice entity selection

Perform the following configuration in voice dial program view.

Table 1-31 Configure the priority rules for voice entity selection

Operation	Command
Configure the priority rules for voice entity selection	select-rule rule-order 1st-rule [ 2nd-rule [ 3rd-rule ] [ 4th-rule ] ]
Restore the default value	undo select-rule rule-order

By default, the priority rules for voice entity selection accords with the sequence of “precise match -> priority -> random selection”.

3) Configure the maximum number of voice entities found

Perform the following configuration in voice dial program view.

Table 1-32 Configure the maximum number of voice entities found

Operation	Command
Configure the maximum number of voice entities found	select-rule search-stop max-number
Restore the default value	undo select-rule search-stop

By default, the maximum number of voice entity that the system searches for is 128.

4) Configure to stop the search for voice entities

Perform the following configuration in voice dial program view.

Table 1-33 Control the search for voice entities

Operation	Command
Disable the search for voice entities	select-stop
Re-enable the search for voice entities	undo select-stop

By default, the search for voice entity is enabled.

V. Configuring and bind the max-call group

You can set one or more voice entities as a call connection group and limit the total number of call connections for one or several voice entities according to the network scale, so as to control communication quantity. Using the commands described in Table 1-34, you can configure a max-call set, with parameters of a group label and the maximum number of call connections. Then, by using the commands described in Table 1-35, you can bind it to a group of voice entities. If the total number of call connections of the current call group has not reached the preset maximum number of call connections, the voice entities in the group can initiate new calls; otherwise they are not allowed to initiate new calls. If it is necessary to limit the total number of IP calls that can be initiated by the voice gateway, you can perform the binding operation on the VoIP entity; if it is necessary to limit the total number of IP calls that can be received by the voice gateway, you can perform the binding operation on the POTS entity.

1) Configure a max-call group

Perform the following configuration in voice dial program view.

Table 1-34 Configure a max-call group

Operation	Command
Configure a max-call group	max-call group-number max-number
Delete the max-call group	undo max-call { group \| all }

By default, no max-call group is configured.

2) Bind the max-call group

Perform the following configuration in voice entity view.

Table 1-35 Bind the max-call group

Operation	Command
Bind the max-call group	max-call group
Delete the binding	undo max-call

By default, the max-call group is not bound. Namely, the voice entities do not belong to any max-call group, and there is no limitation on their call connections.

VI. Configuring number-substitute

According to network requirements, you can configure a number-substitute list, and define the number-substitute rules, dot match rules, and the first-used number-substitute rule. Then, you can use these rules in global scope, voice entities, and voice subscriber lines, and thereby substitute the calling/called numbers in a flexible way.

If there are many number-substitute rules in a list, the system firstly matches the first-used number-substitute rule. If it works, the system substitutes the number according to that rule and returns. If it does not work, the system matches other rules in order. It substitutes the number and returns as soon as it successfully matches a rule. In short, a number-substitute list only substitutes a number once at most.

A number-substitute list prescribes the substitute method. It can be used wherever number-substitute is required. And there is no limitation on where and how many times it is used. Therefore, one number-substitute list can be bound to the calling/called number substitute of the global scope, on voice entities, and on subscriber lines.

l Global number-substitute: According to the number-substitute rules configured in a dial program, the system substitutes the calling/called number of all the incoming/outgoing calls passing the voice gateway. Different from the number-substitute on voice entities and subscriber lines, multiple number-substitute lists can be bound to any of the four global number-substitute cases (incoming/outgoing X calling/called number). Thus, if there is no matchable rule in the first number-substitute list, it can turn to other lists.

l Number-substitute of voice entities: It binds the number-substitute list to voice entities, and substitutes the calling/called number that match the rule.

l Number-substitute of the specified subscriber line: It substitutes the calling/called numbers of the incoming calls reaching the subscriber line according to the number-substitute rule configured on the subscriber lines.

1) Set a number-substitute list

Perform the following configuration in voice dial program view.

Table 1-36 Set a number-substitute list

Operation	Command
Set a number-substitute list and access voice number-substitute view	number-substitute rulelist-number
Delete the number-substitute list	undo number-substitute { rulelist-number \| all }

2) Configure a dot match rule

Perform the following configuration in voice number-substitute view.

Table 1-37 Configure dot match rules

Operation	Command
Configure a dot match rule	dot-match { end-only \| left-right \| right-left }
Restore the default dot match rule	undo dot-match

By default, the dot match rule is end-only.

3) Configure a number-substitute rule

Perform the following configuration in voice number-substitute view.

Table 1-38 Configure a number-substitute rule

Operation	Command
Configure a number-substitute rule	rule rule-number input-number output-number
Delete a number-substitute rule	undo rule rule-number

No default number-substitute rule is configured.

4) Configure the first-used number-substitute rule

Perform the following configuration in voice number-substitute view.

Table 1-39 Configure the first-used number-substitute rule

Operation	Command
Configure the first-used number-substitute rule	first-rule rule-number
Restore the default first-used number-substitute rule	undo first-rule

By default, the rule with the smallest serial number is used first.

5) Bind the number-substitute list

Use the substitute command in voice entity view and voice subscriber-line view; use the substitute incoming-call and substitute outgoing-call commands in voice dial program view.

Table 1-40 Bind the number-substitute list

Operation	Command
Bind the number-substitute list to the calling/called numbers of voice subscriber lines or voice entities	substitute { called \| calling } list-number
Delete the number-substitute list	undo substitute { called \| calling }
Bind the number-substitute list to the calling/called numbers of the incoming/outgoing calls from the network side	substitute { incoming-call \| outgoing-call } { called \| calling } list-number
Delete the number-substitute list	undo substitute { incoming-call \| outgoing-call } { called \| calling } { all \| list-number }

By default, no number-substitute list is bound, that is, the system does not substitute numbers.

VII. Configuring number information attribute

Calling/called number information includes number types and numbering plans. Attributes of the calls from the voice gateway contain the number types and numbering plans that are attached to voice entities. Number types and numbering plans comply with the Q.931 standard of ITU-T.

1) Configure the calling/called number type for a voice entity

Perform the following configuration in voice entity (POTS and VoIP) view.

Table 1-41 Configure the calling/called number type for a voice entity

Operation	Command
Configure the calling/called number type for a voice entity	type-number { called \| calling } { abbreviated \| international \| national \| network \| reserved \| subscriber \| unknown }
Restore the calling/called number type	undo type-number { called \| calling }

The default calling/called number type of a voice entity is unknown.

2) Configure the calling/called numbering plan for a voice entity

Perform the following configuration in voice entity (POTS and VoIP) view.

Table 1-42 Configure the calling/called numbering plan for a voice entity

Operation	Command
Configure the calling/called numbering plan for a voice entity	plan-numbering { called \| calling } { data \| isdn \| national \| private \| reserved \| telex \| unknown }
Restore the default calling/called numbering plan	undo plan-numbering { called \| calling }

The default calling/called numbering plan of a voice entity is unknown.

VIII. Configuring number sending modes

1) Configure the dial prefix

You can configure the dial prefix when PBX sends a number to PSTN. When POTS voice entities initiate a call, you can use the dial-prefix command to configure the prefix and add it to the called number.

Perform the following configuration in POTS voice entity view.

Table 1-43 Configure the dial prefix

Operation	Command
Configure the dial prefix	dial-prefix string
Delete the dial prefix	undo dial-prefix

No default dial prefix is configured.

2) Configure number sending modes

You can control how to send the called number when the PBX sends a number to PSTN. There are three control modes at present, as shown in the following;

l Send some digits of the called number. That means it sends the digits according to the value of digit-number you configure in the following command.

l Send all digits of the called number

l Send the truncated called number. If the match-template command of the corresponding voice entity includes end wildcard, it only sends the digits that match the wildcard.

Perform the following configuration in POTS voice entity view.

Table 1-44 Configure number sending modes

Operation	Command
Configure number sending modes	send-number { digit-number \| all \| truncate }
Restore the default number sending mode	undo send-number

By default, it sends the truncated number.

1.3.7 Configuring Global Default Voice Parameter

For each voice entity of the voice gateway, if a certain command (such as the compression command) is not performed specially to configure a parameter, the voice entity will be automatically assigned value of the preset default value of the system (which the subscriber cannot revise).

When there are too many voice entities on one voice gateway and the manually configured values of most voice entity parameters are the same as the default values, the default values can be adopted to improve the efficiency. On the contrary, in some cases when the manually configured values of most voice entity parameters are different from the default values and the manually configured values of most voice entity parameters are almost the same, to specify suitable voice parameters for these voice entities one by one will waste much time. Rather, the default command can be performed to update the attribute defaults of all existed global voice entities into new values. In this way, the attribute of each voice entity will directly inherit the default values and the newly created voice entity will be assigned values according to the new defaults, a more flexible, simple and convenient configuration is thus realized.

I. Difference between the default command and the undo command

The default command (such as the compression command) is used to configured the parameters of each voice entity globally and take them as the default values later. The undo default command is used to restore the global parameters to the fixed system defaults. The command command is used to configure voice parameters of a voice entity as the default values (that is, values specified by the default command), without changing the global values. The values of each parameter in different stages by using the default and undo default commands are described below in detail.

Step 1: Take the system fixed parameter value A by the command command as the default value of default, and configure voice entity 1 with parameter B, voice entity 2 with the default A and voice entity 3 with parameter C.

Figure 1-8 Take the system fixed value as the default value of default

Step 2: Using the default command to revise the parameter default of the command command into C, with the parameter of voice entity 1 still being B and parameters of voice entity 2 and voice entity 3 changing into C automatically.

Figure 1-9 Revise the parameter default of the command command

Step 3: Using the undo default command to restore the parameter default of the command command to the system fixed value A, with the parameter of voice entity 1 still being B and parameters of voice entity 2 and voice entity 3 changing into A automatically.

Figure 1-10 Restore the default parameter of the command command to the system fixed value

II. Configuring the default command

Perform the following configuration in voice dial program view.

Table 1-45 Configure the global voice parameter defaults

Operation	Command
Configure the default compression mode globally	default entity compression { 1st-level \| 2nd-level \| 3rd-level \| 4th-level } { g711alaw \| g711ulaw \| g723r53 \| g723r63 \| g729a \| g729r8 }
Configure the system to default to enable the voice activity detection globally	default entity vad-on
Configure the default duration of DSP packets globally	default entity payload-size { g711 \| g723 \| g729 } time-length
Configure the default fax parameters globally	default entity fax { baudrate { 2400 \| 4800 \| 9600 \| 14400 \| disable \| voice } \| ecm \| level level \| local-train { threshold threshold } \| nsf-on \| protocol { h323-t38 \|pcm { g711alaw \| g711ulaw } \| t38 } \| support-mode { rtp \| vt } \| train-mode { local \| ppp } \| redundancy { hb-redundancy \| lb-redundancy } number }
Configure the system to default to enable the fast connection mode globally	default entity fast-connect

The default values of voice parameters are as follows:

l The system truncates the called number

l The voice compression mode is g729r8 (currently, the VG 20-16, VG20-32 and VG 21-08 do not support G.723)

l DTMF codes are transmitted in the in-band mode

l Voice activity detection is disabled

l The fast connection mode is disabled

l The carrier transmitting level is -15

l The fax local training threshold is 10

l The T.38 protocol is used for fax

l The number of low-rate and high-rate redundant packets is 0

l The fax rate is determined in the voice mode

l Fax negotiation is implemented based on the standard capability

l The RTP mode is used for interoperation with devices of other vendors

l The point-to-point training mode is adopted

l The ECM mode is not used for fax

Perform the following configuration in voice view.

Table 1-46 Configure the global voice parameter defaults

Operation	Command
Configure the high-frequency and low-frequency DTMF gain values of all voice subscriber lines	default subscriber-line dtmf gain { hf \| lf } value
Configure the receiving gain value of all voice subscriber lines	default subscriber-line receive gain value
Configure the transmitting gain value of all voice subscriber lines	default subscriber-line transmit gain value

1.3.8 Configuring Cptone

I. Introduction to Cptone

Cptone (call progress tone), also called signal tone, is a combination of separate single-frequency tones and is repeatedly played continually or at a fixed ON/OFF duration ratio. The cptone can be applied as the dial tone, ringing tone and busy tone to indicate different status of the call process. Not regulated by a unified international standard, cptones differ between countries. Therefore you can customize the cptone to satisfy the actual demands.

You need to set the following parameters to configure a cptone:

l Cptone frequency: Most signal tones occupy one or two frequencies.

l ON/OFF duration: Duration time of alternant tone-playing and silent periods. Always referred to as the signal ON/OFF ratio.

l Combination mode: The way that frequencies are combined, which can be overlapped, multiplied or alternant.

Cptone-related protocols/standards include:

l ITU-T E.180/Q.35: TECHNICAL CHARACTERISTICS OF TONES FOR THE TELEPHONE SERVICE

l ITU-T E.180S2:VARIOUS TONES USED IN NATIONAL NETWORKS

The cptone can be configured following locale mode or customized mode. You can modify the cptone amplitude through configurations.

II. Configuring cptone in locale mode

Cptone standards differ from countries and areas. Therefore you need to make cptone configurations according to the location where you use the VG. The system stores cptone parameter settings for 62 countries/areas, and you can select the ones that suit your location by using the cptone locale command in voice subscriber-line view.

Table 1-47 Configure cptone in locale mode

Operation	Command
Enter system view	system-view
Enter voice view	voice-setup
Enter voice subscriber-line view	subscriber-line line-number
Configure the cptone	cptone { locale \| cs } [ { type \| all } amplitude value ]

Caution:

The cptone locale command takes effect on all voice ports on the board where the current voice subscriber line is attached.

III. Configuring cptone in customized mode

You can customize the cptone as needed by operations in Table 1-48.

Table 1-48 Configure cptone in customized mode

Operation	Command	Description
Enter system view	system-view	—
Enter voice view	voice-setup	Required
Set cptone parameters	vi-card cptone-custom type arg0 arg1 arg2 arg3 arg4 arg5 arg6	Required
Enter voice subscriber-line view	subscriber-line line-number	Required
Execute the cptone cs command	cptone { locale \| cs } [ { type \| all } amplitude value ]	Required

& Note:

The cptone parameters customized through the cptone-custom command will not take effect until you run the cptone cs command in voice subscriber-line view. The cptone locale command takes effect on all voice ports on the board where the current voice subscriber line is attached.

IV. Customizing cptone amplitude

You may need to modify the cptone amplitude when the VG connects with different phone sets or devices. For example, exceeding amplitude can cause failure in detecting the first dialed digit; inefficient busy tone amplitude or over attenuation will cause loss of busy tone detection. To avoid these problems, use the amplitude customization function as shown in Table 1-49.

Table 1-49 Customize cptone amplitude

Operation	Command	Description
Enter system view	system-view	—
Enter voice view	voice-setup	Required
Enter voice subscriber-line view	subscriber-line line-number	Required
Execute the cptone cs command	cptone { locale \| cs } [ { type \| all } amplitude value ]	Required

1.3.9 Configuring Special-Services Numbers

I. Introduction to special services

The voice gateway supports the special-services numbers of program-control switch. The special-services numbers supported currently include the following:

1) Do-not-disturb

After setting the “do-not-disturb” service, the subscriber telephone will refuse the incoming call requests, regardless of whether it is idle. In this case, the calling party will hear the busy tone. To set this service on a dual-tone telephone connected to the voice gateway, you can press “*56#” after picking up the phone. To cancel the setting, press “#56#”.

& Note:

l * is the beginning of a string that sets a special service or turns to the next stage of the special service.

l # is the beginning of a string that cancels the special service or terminates the special service number string.

2) Call transfer on busy

After setting the service of call transfer on busy, a new incoming call will be transferred to a specified number if the subscriber’s line is busy. To set this service on a dual-tone telephone connected to the voice gateway, you can press “*58*ABCD#” after picking up the phone. To cancel the setting, press “#58#”.

& Note:

“ABCD” represents the telephone number to which the calls will be transferred. Note that this function is only available for the telephones connected to the FXS subscriber-line on the voice gateway. In addition, you can only specify a telephone connected to the same voice gateway as the call transfer destination. Otherwise, the setting will be invalid.

After special service configuration, if you want to set a new special service number, you must cancel the previous special service settings first.

3) Call transfer unconditionally

If you set the service of call transfer unconditionally, the incoming calls will be transferred to a specified telephone regardless of whether the subscriber is busy. To set this service on a dual-tone telephone connected to the voice gateway, you can press “*57*ABCD#” after picking up the phone. To cancel the setting, press “#57#”.

& Note:

4) Alarm call service

After setting the alarm call service, the telephone will ring for 60 seconds at the time specified by the subscriber and automatically disconnect after that. This function is only valid for 24 hours. To set this service on a dual-tone telephone connected to the voice gateway, you can press “*55*HHMM#” after picking up the phone. To cancel the setting, press “#55#”.

& Note:

l “HH” represents hour, which can be any integer from 0 to 23. “MM” represents minute, which can be any integer from 0 to 59.

l The alarm time set by a subscriber is based on the system time of the VG.

5) Lines group access

For the detailed configuration of lines group access, refer to 1.5.9 Lines Group Access.

II. Enabling special services numbers for local or remote users

Before using the special-services numbers provided by the voice gateway, you first need to use the command to enable the special services.

Perform the following configuration in voice dial program view.

Table 1-50 Enable special-services numbers for local or remote users

Operation	Command
Enable special-services numbers for local or remote users	special-service { local \| remote }

By default, the special-services numbers are disabled.

III. Enabling/disabling special service dial tone

You can choose to change the dial tone to a special tone after you have configured the special services.

Table 1-51 Enable/disable special service dial tone

Operation	Command
Enable special service dial tone	special-service switch-dialtone
Disable special service dial tone	undo special-service switch-dialtone

1.3.10 Configuring Busy Tone Detection on the FXO Subscriber-line

I. Introduction to Busy Tone Detection

There are various switches at the actual spot of user application and different devices use different signaling standards. So, the prompt voice played on switches has different indices and different spectrum features, and it is impossible to identify the busy tone feature with a fixed threshold.

The VG series adopts smart busy-tone identification technology to sample, calculate, and analyze the incoming busy tone, so as to obtain a set of parameters closest to the busy-tone features. You can complete the busy-tone detection by configuring these parameters on the FXO subscriber-line. The voice gateway provides automatic busy-tone detection to resolve the problem that different switches have different busy tone indices. Pay attention to the following three points:

l Busy tone frequency: most busy tones contain one or two frequency at present.

l Duty time: duration specification of high/low level composing busy tone signal, i.e., the common called signal duty ratio. Different regions have different specification about duty time of busy tone.

l Detection threshold: detection used to judge the signal energy. The signal energy higher than the detection threshold is regarded as a high level; otherwise it is regarded as a low level.

The typical network diagram of automatic busy tone detection is shown in the following. The telephone Tel1 is connected to a PBX, which is connected to the FXO port on the voice gateway through a common telephone cable. The configuration of the peer is similar to this.

Figure 1-11 Network diagram of busy tone detection

1) Dial the number 1002 at telephone Tel1 (010-1001). After the call is connected, the FXO subscriber-line on the voice gateway plays dialing tone to the switch and transmits the tone to telephone Tel1 through the switch. Dial the number 07552001 now. Pick up the telephone Tel2 when it rings. Thus, the telephone Tel2 is connected.

2) Hangs up Tel1. The switch will play busy tone to the voice gateway through the telephone cable between them to provide input for busy tone check.

3) On the console of Voice gateway 1, use the vi-card busy-tone-detect command to start the check. To ensure the capture of busy tone signal sent by the switch, it is recommended to run this command two seconds after Tel1 hangs up.

4) The console terminal displays “Detecting, please wait ...” to prompt that busy tone is being performed. After check, the terminal displays “Detect finish!” Use the save command to save the detected busy tone parameter.

5) Authenticate whether automatic busy tone check succeeds. Repeat the operation of step (2). If it succeeds, the Tel1 will automatically disconnect after the switch plays busy tone to the voice gateway for about four seconds and the “Clatter” sound can be heard by the Tel2.

Configuration tasks of busy tone detection on the FXO subscriber-line include:

l Configure the standard of busy tone detection

l Configure custom busy tone detection parameters

l Set the threshold of busy tone detection

l Set the auto on-hook of silent detection

l Enable busy tone detection

II. Configuring the standard of the busy tone detection

Perform the following configuration in FXO voice subscriber-line view.

Table 1-52 Configure the standard of busy tone detection

Operation	Command
Configure the standard of busy tone detection	area { asia \| custom \| europe \| north-america }
Restore the default standard of busy tone detection	undo area

By default, European busy tone standard (europe) is adopted.

III. Configuring the custom busy tone detection parameters

After successfully checking the busy tone of FXO subscriber-line with vi-card busy-tone-detect auto command, the system will automatically calculate relevant parameters of busy tone detection. With the display current-configuration command, the busy tone parameters value of the vi-card custom-toneparam command can be displayed. These parameters contribute to configuring and adjusting busy tone detection parameters manually.

Perform the following configuration in voice view.

Table 1-53 Configure custom busy tone detection parameters

Operation	Command
Configure custom busy tone detection parameters for the FXO subscriber-line	vi-card custom-toneparam area-number index arg0-arg9
Delete custom busy tone detection parameters for the FXO subscriber-line	undo vi-card custom-toneparam index

Caution:

After the settings are complete, use the area custom command to make them effective.

IV. Setting the threshold of the busy tone detection

Different PBXs sends different number of busy tone signals. When the FXO interface detects busy tone signals, deviations of busy tone detection parameters may lead to erroneous detection of busy tone signals. By carrying out this command, you can increase the number of busy tone signals to be detected. The system deems that the PBX is sending busy tone signals only when the number of busy tone signals consecutively detected by the FXO interface reaches the set number of busy tone signals. This reduces the possibility of erroneous detection of busy tone signals caused by inaccurate busy tone parameters.

Perform the following configuration in FXO subscriber-line view.

Table 1-54 Set the threshold of the busy tone detection

Operation	Command
Set the threshold of the busy tone detection	busytone-t-th time-threshold
Restore the threshold to the default settings	undo busytone-t-th

By default, the threshold of the busy tone detection is set to 2.

V. Setting the auto on-hook of the silent detection

Auto on-hook is used to prevent the FXO port from being halted when the busy tone detection fails. You can set the silent threshold. When the voice signal of two adjacent sampling points sent from the exchange is smaller than the value, system considers it in the state of silence. You can se the value to limit the silence detection time. When the link is in the state of silence for a time longer than the configured value, system ends the call automatically.

It is recommended that you set the silent threshold to 20 seconds and the auto on-hook time to 10 seconds.

Perform the following configuration in FXO subscriber-line view.

Table 1-55 Set the auto on-hook of the silence detection

Operation	Command
Set the auto-hook-up of the silence detection	silence-th-span threshold time-length
Restore the default settings	undo silence-th-span

By default, the silence threshold is 20 and the auto on-hook of the silence detection is 7200 seconds.

VI. Enabling the busy tone detection

Perform the following operation in voice view.

Table 1-56 Enable busy tone detection

Operation	Command
Enable busy tone detection on FXO port	vi-card busy-tone-detect auto index line-number [ free \| time ]

The system can record four types of busy tone characteristics, which are identified by the index argument. A device supports busy tone detection on only one channel at a time.

1.3.11 Configuring Online Detection on the FXO Subscriber-line

FXO port on the VG is responsible for the trunk connection between IP phone sets and the PSTN, and the trunk status (online/offline) decides whether calls can be connected. You can configure online detection on the FXO port, so that the VG can get realtime information of the trunk line status, and terminate the call when it finds line abnormality. Online detection works only on the FXO port but not FXS ports.

For VG 21-08 (version C), online detection monitors FXO online status only when calls are proceeding. For VG 10-41 (version C), online detection monitors FXO online status in a realtime manner, and the following messages will be given in case of plugging in or plugging out.

l The message for plugging out is: Subscriber-line channelnumber is down.

l The message for plugging in is: Subscriber-line channelnumber is up.

Perform the following operations in voice view.

Table 1-57 Enable/disable online detection on the FXO port

Operation	Command
Enable online detection on the FXO port	fxo-monitoring enable
Disable online detection on the FXO port	undo fxo-monitoring enable

1.3.12 Configuring On-hook/Off-hook Detection Sensitivity

Perform the following configuration in voice view.

Table 1-58 Configure on-hook/off-hook detection sensitivity

Operation	Command
Configure on-hook/off-hook detection sensitivity	vi-card hook-sensitivity { low \| middle \| high }
Restore the default value of on-hook/off-hook detection sensitivity	undo vi-card hook-sensitivity

By default, the on-hook/off-hook detection sensitivity is set to the middle level. This command is applicable to VG 20-16 and VG 20-32 only.

1.3.13 Resetting Voice Interface Card

Perform the following configuration in voice view.

Table 1-59 Reset voice interface card

Operation	Command
Reset voice interface card	vi-card reboot

1.3.14 Configuring Fast Connection and Tunneling Function

The VG series provides interoperability with mainstream voice gateways in the industry using fast connection.

According to the specification of H.323 protocol, the fast connection means: during establishing call connection according to H.225.0 protocol, establish voice channel in advance through exchanging H.245 call capability parameter (e.g., voice code) in Q.931-like control message, thereby omitting the establishment process of H.245 TCP connection. So the fast connection accelerates the connection speed.

Fast-connect enables the system to provide such special services as DTMF transparent transmission, call transfer, and so on.

I. Configuring fast connection

According to the specification of H.225.0 protocol, fast connection carries H.245 message (e.g., message of opening logical channel) in Setup, CallProceeding, Alerting or Connect message, which makes RTP/RTCP voice channel be established before GW receives Connect message to avoid subsequent TCP connection establishment and H.245 message exchange process, thereby shortening connection time.

There is no capability negotiation process in fast connection mode, so the capability of both parties is determined by called GW. When fast connection mode is adopted, the Setup message send from calling GW carries coding/decoding parameters supported by local end. After receiving this message, called GW will select one coding/decoding supported by it to notify calling GW through one message of CallProceeding, Alerting and Connect, so this coding/decoding is adopted by both parties to communicate.

The calling GW can set whether to use fast connection mode for originated every line call on the voice gateway. The called GW determines whether to use fast connection mode to initialize call according to the configuration of voip call-start command if the calling GW adopts fast connection. Only after fast connection mode is started, can tunnel function be configured.

Perform fast-connect command configuration in VoIP voice entity view and perform voip call-start command configuration in voice view.

Table 1-60 Configure fast connection

Operation	Command
Enable fast connection at the calling side	fast-connect
Disable fast connection at the calling side	undo fast-connect
Configure initial call mode at the called side	voip called-start { fast \| normal }

By default, the calling side does not enable fast connection. If fast connection is started, the called GW will return fast connection parameter fast in Alerting message by default.

II. Configuring to send the ringback tone

In fast connection mode, the ringback tone is sent remotely. If you want to broadcast the tone locally, enable this function.

Perform the following configuration in VoIP voice entity view.

Table 1-61 Configure to enable the local to send the ringback tone

Operation	Command
Enable the local to send ringback tone	send-ring
Disable the local to send the ringback tone	undo send-ring

By default, the local does not send the ringback tone. This command is only available in VoIP entity and fast connection.

1.3.15 Configuring Transmission Mode of DTMF Code

Dual Tone Multi-Frequency (DTMF) uses the combination of two given single-frequency signals to represent digits and functions. Usually, eight kinds of frequency – 697 Hz, 770 Hz, 852 Hz, 941 Hz, 1209 Hz, 1336 Hz, 1477 Hz and 1633 Hz – are adopted internationally. They are classified into two frequency groups: high frequency group (1209, 1336, 1477, 1633) and low frequency group (697, 770, 852, 941). Any frequency from the high frequency group is randomly combined with that from the low frequency group. Thus, 16 combinations can be made to represent 16 characters of telephone digital keys, including 0 to 9, A, B, C, D, * and #. However, for common telephone, A to D are excluded.

During the communication between calling and called subscriber, DTMF code can be transmitted transparently through two kinds of modes: in-band transmission and out-of-band transmission. In-band mode means that DTMF code is encapsulated in RTP voice packet to transmit. Out-of-band mode means that DTMF code is encapsulated in H.245 or H.225 message to transmit.

In either fast or non-fast connection mode, DTMF code can be directly transmitted through out-of-band of H.245 and H.225 protocols.

When fast connection mode is adopted, the party for which DTMF H.245 out-of-band transmission mode is configured decides the transmission method according to whether the tunnel function is enabled during the call. If the tunnel function is enabled, DTMF code will be encapsulated in the Facility message whose h323-message-body is set as empty for transmission. If disabled, DTMF code will be encapsulated in H.245 UserInput message for transmission.

In actual configuration, to implement the transparent transmission of DTMF code, it is necessary to perform some configuration in VoIP voice entity of calling GW and in POTS voice entity of called GW at the same time.

Perform the following configuration in VoIP and POTS voice entity view.

Table 1-62 Configure transmission mode of DTMF code

Operation	Command
Configure DTMF code to be transmitted in out-of-band mode	outband { h245 \| h225 }
Restore transmission mode of DTMF code to in-band mode	undo outband

By default, DTMF code is transmitted in in-band mode.

1.3.16 Configuring Tunneling Function

In fast connection mode, the tunnel function encapsulates non-standard H.245 message (e.g., transparent transmission capability of DTMF code) in Facility message of H.225.0 protocol to complete capability negotiation and call transfer, which makes it unnecessary to establish one independent H.245 TCP connection for the transmission of H.245 message.

The tunnel function can be used on a VG only when fast connection is enabled on it. When the VG functions as a calling GW, you can enable/disable tunneling for each channel. When the VG functions as a called GW, whether the tunnel function is enabled is determined by whether this function is enabled on the calling GW and the configuration of the voip calledtunnel command on the called GW. Namely, if tunneling is enabled on the calling GW, it is enabled on the called GW as well; otherwise, tunneling will not be used.

Perform the following configuration in VoIP voice entity view and configure the voip calledtunnel command in voice view.

Table 1-63 Configure the tunneling function

Operation	Command
Enable the tunneling function at the calling side	tunnel-on
Disable the tunneling function at the calling side	undo tunnel-on
Configure the tunneling switch of the called side	voip calledtunnel { enable \| disable }

By default, the calling side does not enable the tunnel function. When fast connect is enabled, the called gateway defaults to enable the tunnel function.

& Note:

The VG supports being used as the called voice gateway in tunnel mode.

1.3.17 Configuring Voice CID

CID service means that the terminal device of called subscriber displays such calling identity information as calling number, call data and time, including: calling side sends calling number and called side receives calling number. With the rapid development of VoIP communication, CID function shows more and more marketing demands, especially in band, public security and frontier defense department etc, which conduces to telephone communication security. On one hand, CID technology ensures that called subscriber can learn the number information of calling subscriber in time and effectively locates the calling subscriber. On the other hand, it also prompts the calling subscriber not to dial maliciously.

The CID function of the H3C VG implements transmitting and receiving calling number and calling time of single-data-message format while the device is in the on-hook state, conforming to the requirement of YDN069-1997 of China’s Ministry of Information Industry. The CID function can combine with call forwarding unconditional and call forwarding busy, i.e., calling identity information can be transmitted according to these new services. When single-data-message format is adopted, the message includes the following fields:

l Date and time of voice call (minute, hour, day and month).

l If the voice gateway is configured as display-permitted, it includes the calling number.

l If the voice gateway is configured not to display calling number, it will transmit a “P”.

l When the called PBX cannot obtain the calling number (e.g., the originating point does not send the calling number), it will transmit an “O”.

The CID function of the VG needs the help of FXS and FXO module plug-in card.

The FXS interface module sends the calling number to a subscriber’s telephone. The calling number is sent to the called party's telephone through the FXS subscriber line using FSK modulation between the first and second ringing. To send and receive the calling number correctly, the subscriber must pick up the phone after the second ringing; otherwise the calling number may not be displayed.

The FXO interface module receives the calling number sent from PBX. The FXO subscriber line receives the calling number modulation information sent from PBX between the first and second ringing. After FSK demodulation and parity check, it will find whether the CID function is enabled if the parity check is correct. If the CID function is enabled, it sends the calling number to the IP side. Otherwise, it sends a “P” or “O” to the IP side.

I. Enabling calling number display

Perform the following configuration in FXS voice subscriber-line view.

Table 1-64 Enable/disable displaying calling number

Operation	Command
Enable displaying calling number	cid display
Disable displaying calling number	undo cid display

By default, FXS subscriber lines display the calling number.

& Note:

To implement CID function, besides that the VG supports receiving calling number in off-hook status, the hardware and software of PBX must support this service and the subscriber’s telephone must support CID function, i.e., the telephone must be able to correctly receive and display caller identity information (such telephones are CID I or CID II class) in off-hook or conversation status.

II. Enabling the FXO subscriber line to receive calling numbers

To receive the calling number data in some format on an FXO subscriber line, you should use the command in the following table to enable it to receive the calling number in FXO voice subscriber-line view.

Perform the following configuration in FXO voice subscriber-line view.

Table 1-65 Enable an FXO interface to receive calling numbers

Operation	Command
Enable an FXO subscriber line to receive calling numbers	cid enable
Disable the FXO subscriber line to receive calling numbers	undo cid enable

By default, an FXO subscriber line has been enabled to receive calling numbers.

& Note:

After being enabled to receive calling numbers, the off-hook speed of the FXO subscriber line will become lower. In practice, you should determine whether to enable this function depending on the actual situation.

III. Configuring the voice gateway to send calling numbers to the IP side

Perform the following configuration in FXS or FXO voice subscriber-line view.

Table 1-66 Configure the voice gateway to send calling numbers to the IP side

Operation	Command
Enable the voice gateway to send calling numbers to the IP side	cid send
Disable the voice gateway to send calling numbers to the IP side	undo cid send

By default, calling numbers are sent to the IP side.

IV. Configuring message format of sending calling number

Perform the following configuration in FXS or FXO voice subscriber-line view.

Table 1-67 Configure message format of sending calling number

Operation	Command
Configure message format of sending calling number	cid type { complex \| simple }

By default, both FXS and FXO subscriber lines send calling number using complex data message format.

& Note:

The call date and time transmitted in single-data-message format is the system time of the VG. To find the exact time when a call occurs, use the clock command to keep the clock in the VG synchronous with the local standard time.

1.3.18 Configuring Voice Performance

I. Enabling fast receiving/sending of voice data

The VG realizes fast forwarding of voice data, which not only improves the forwarding performance of voice data but also improves voice quality effectively.

The forwarding of voice data includes common and fast forwarding and fast forwarding includes fast receiving and fast sending. The subscriber can configure the voice performance switch flexibly according to the demand. Comparing with common receiving, the fast receiving properly reduces memory application and data copy process, i.e., the data are sent to service module for processing directly, which accelerates the receiving speed of voice data. The fast sending process recurs to the interruption mechanism. It performs packet processing to voice data and then sends data packets directly to network layer for forwarding process according to route and link information.

Perform the following configuration in voice view.

Table 1-68 Enable/disable fast receiving/sending of voice data

Operation	Command
Enable/disable fast receiving/sending process of voice data	vqa performance { receive \| send } { fast \| normal }

By default, the fast receiving/sending process of voice data is enabled.

II. Enabling voice data statistics

To locate the faults in VoIP call correctly and fast and to debug them, you can enable the voice data statistics function. The statistics information includes succeeding times of searching voice table, total received data packets, times of searching voice table in fast and common mode and the voice and fax information of receiving/sending channel.

The voice data statistics function is mainly used for debug purpose. So to obtain the higher performance of voice data processing, it is recommended to disable the statistics function when the service works normally.

Perform the following configuration in voice view.

Table 1-69 Enable/disable voice data statistics

Operation	Command
Enable voice data statistics	vqa data-statistic enable
Disable voice data statistics	vqa data-statistic disable

By default, the voice data statistics is disabled.

III. Configuring jitter buffer depth

The bad network condition will arouse the abnormal transmission speed change of data packets, which results in the inconsistency of receiving packet sequence with sending packet sequence, thereby resulting in jitter. To alleviate the bad effect of network condition on voice packets, Jitter Buffer is used to handle voice data packet. These measures includes compensating lost voice packets, adjusting out-of-sequence voice packet, deleting voice packets arousing jitter and discarding voice packets with repeat serial number.

Perform the following configuration in voice view.

Table 1-70 Configure Jitter Buffer depth

Operation	Command
Configure Jitter Buffer depth	vqa jitter-buffer depth

By default, the Jitter Buffer depth is 3.

IV. Configuring precedence for all voice signaling packets

To ensure the fast and accurate setup of voice calls in the case that voice signaling packets and common data packets are being transmitted simultaneously, you can upgrade the precedence of voice packets by means of the command in the following table. This command configures the precedence (namely the ToS field of the IP packets) for all voice signaling IP packets.

Perform the following configuration in voice view.

Table 1-71 Configure precedence for all voice signaling packets

Operation	Command
Configure precedence for all voice signaling packets	vqa ip-precedence tos-value

By default, the precedence of voice signaling packets is 0, that is, voice signaling packets and data packets have the same precedence.

& Note:

The ip-precedence command in the voice entity view of an entity is only used to configure the precedence of the entity-related voice or fax data, while the vqa ip-precedence command is used to configure the precedence of all voice signaling IP packets.

V. Monitoring DSP buffer-time

After you set a buffer-time, DSP automatically drops the expired voice data that it buffers to keep voice delay within an acceptable range.

Perform the following configuration in voice view.

Table 1-72 Monitor DSP buffer-time

Operation	Command
Monitor time-length of the data buffered by DSP.	vqa dsp-monitor buffer-time time
Delete the monitoring time-length of the DSP to buffer the data	undo vqa dsp-monitor buffer-time

By default, data buffered by DSP is monitored with the duration of 270 milliseconds.

1.3.19 Configuring T.38 Capacity Description Compatibility

As NetMeeting does not support T.38 capacity description parsing, you must disable the voice gateway in H.323 slow-start mode to carry the T.38 capacity description in its capacity set in order to work with NetMeeting.

Perform the following configuration in voice view.

Table 1-73 Configure T.38 capacity description compatibility

Operation	Command
Enable the voice gateway to carry the T.38 capacity description in its capacity set when it is in H.323 slow-start mode.	voip h323-config tcs-t38
Disable the voice gateway to carry the T.38 capacity description in its capacity set when it is in H.323 slow-start mode.	undo voip h323-config tcs-t38

By default, the T.38 description is carried.

1.4 Displaying and Debugging VoIP

After the above configuration, execute the display command in any view to display the running of the VoIP configuration, and to verify the effect of the configuration.

Execute the command in any view.

Table 1-74 Display and debug VoIP

Operation	Command
Reset voice data statistics information	reset voice voip data-statistic
Display call history record information	display voice call-history-record { callednumber called-number \| callernumber calling-number \| cardnumber card-number \| remote-ip-addr ip-address \| last [ last-number ] } [ brief ]
Display conversation information of voice subscriber-line	display voice call-history-record line line-number
Display the content in the call information table	display voice call-info { brief \| detail \| mark TAG }
Display current default value and system fixed default value	display voice default all
Display configuration information of different types of voice entities	display voice entity { all \| pots \| voip \| mark entity-tag }
Display the information of the custom number-substitute list	display voice number-substitute [ list-tag ]
Display call control module information of RCV software module	display voice rcv ccb
Display call statistics information between RCV software module and other software modules	display voice rcv statistic { all \| call \| error \| proc \| timer \| vas \| vcc \| vpp }
Display the statistics information of the IPP module	display voice ipp { ccb \| statistic { all \| h225 \| h245 \| ras \| socket \| timer \| vcc \| vpp } }
Display voice subscriber-line information	display voice subscriber-line line-number
Display voice data statistics information	display voice voip data-statistic [ channel channel-number \| detail ]
Display various statistics information in VPP software module	display voice vpp [ channel channel-number ]
Enable H.225.0 negotiation packet or event debugging	debugging voice h225 { asn1 \| event }
Enable H.245 negotiation packet or event debugging	debugging voice h245 { event }
Enable H.323 protocol stack module debugging	debugging voice ipp { all \| error \| rtp-rtcp \| socket \| timer \| vcc \| vpp }
Enable RCV software module debugging	debugging voice rcv { all \| cc \| error \| vcc \| timer \| vas \| vpp }
Enable voice data flow debugging	debugging voice data-flow { all \| detail \| error \| fax [ error ] \| jitter [ error ] \| receive \| send \| vpp }
Configure the interval of debugging information display	trace interval [ packets ]
Enable VAS software module debugging	debugging voice vas { all \| buffer \| channel channel-number \| cid \| command \| dsp \| error \| fax \| rcv \| receive \| send }
Enable VPP software module debugging	debugging voice vpp { all \| codecm \| error \| vcc \| rcv \| timer \| vas }
Enable voice dial program debugging	debugging voice dpl { all \| error \| general }
Enable MIB debugging	debugging voice vmib { aaaclient \| all \| analogif \| callactive \| callhistory \| dialcontrol \| digitalif \| error \| general \| gkclient \| h323statistic \| voiceif }

1.5 VoIP Typical Configuration Examples

1.5.1 Configuring FXS Interface to Implement Interconnection

I. Network requirements

Two voice gateways are connected to each other through the WAN.

The network diagram is shown in the following figure.

For example, if the subscriber with the phone number of 0101001 at the VG A side wants to talk to the subscriber with the number of 07552001 at the VG B side, he/she needs to dial 07552001 and waits for the called party to pick up the phone to start conversation.

II. Network diagram

Figure 1-12 Voice gateways are connected to common dual tone telephone sets directly

III. Configuration procedure

& Note:

l The configuration of the scenario is made assuming that the route between VG A and VG B is reachable. For the configurations of the Ethernet interface and the default route, refer to section 1.5.1 “Configuring FXS Interface to Implement Interconnection”.

l Configure the IP address and phone number correctly according to the actual network environment.

1) VG A configuration:

# Configure the VoIP voice entity.

[VGA] voice-setup

[VGA-voice] dial-program

[VGA-voice-dial] entity 0755 voip

# Configure called number and “.” is wildcard.

[VGA-voice-dial-entity755] match-template 0755....

# Configure called IP address.

[VGA-voice-dial-entity755] address ip 2.2.2.2

# Configure the local POTS voice entity connected to Tel.1.

[VGA-voice-dial-entity755] entity 1001 pots

# Configure phone number of Tel.1.

[VGA-voice-dial-entity1001] match-template 0101001

# Associate POTS voice entity 1001 with subscriber-line 0.

[VGA-voice-dial-entity1001] line 0

# Configure the local POTS voice entity connected to Tel.2.

[VGA-voice-dial-entity1001] entity 1002 pots

# Configure phone number of Tel.2.

[VGA-voice-dial-entity1002] match-template 0101002

# Associate POTS voice entity 1002 with subscriber-line 1.

[VGA-voice-dial-entity1002] line 1

[VGA-voice-dial-entity1002] return

# Configure the default route and IP address of the Ethernet.

[VGA] ip route-static 0.0.0.0 0 1.1.1.2

[VGA] interface ethernet 0

[VGA-Ethernet0] ip address 1.1.1.1 255.255.255.0

2) VG B configuration

# Configure VoIP voice entity.

[VGB] voice-setup

[VGB-voice] dial-program

[VGB-voice-dial] entity 010 voip

[VGB-voice-dial-entity10] match-template 010....

[VGB-voice-dial-entity10] address ip 1.1.1.1

# Configure the local POTS voice entity connected to Tel.3.

[VGB-voice-dial-entity10] entity 2001 pots

[VGB-voice-dial-entity2001] match-template 07552001

[VGB-voice-dial-entity2001] line 0

# Configure the local POTS voice entity connected to Tel.4.

[VGB-voice-dial-entity2001] entity 2002 pots

[VGB-voice-dial-entity2002] match-template 07552002

[VGB-voice-dial-entity2002] line 1

[VGB-voice-dial-entity2002] return

# Configure the default route and IP address of the Ethernet.

[VGB] ip route-static 0.0.0.0 0 2.2.2.1

[VGB] interface ethernet 0

[VGB-Ethernet0] ip address 2.2.2.2 255.255.255.0

1.5.2 Example of Basic Configuration on the FXO Port

I. Network requirements

VG B has a FXO port, through which it is connected to the PBX.

For example, if the subscriber with the phone number of 07552001 at VG B side wants to talk to the subscriber with the phone number of 0101001 at VG A, he/she needs to dial 07552003 first and then dial 0101001 after hearing the second dialing tone, and waits for the called party to pick up the phone to start conversation.

II. Network diagram

Figure 1-13 Basic configuration on the FXO port

III. Configuration procedure

& Note:

l Configure the IP address and phone number correctly according to the actual network environment.

1) VG A configuration:

[VGA] voice-setup

[VGA-voice] dial-program

[VGA-voice-dial] entity 0755 voip

[VGA-voice-dial-entity755] match-template 0755....

[VGA-voice-dial-entity755] address ip 2.2.2.2

[VGA-voice-dial-entity755] entity 1001 pots

[VGA-voice-dial-entity1001] match-template 010....

[VGA-voice-dial-entity1001] line 0

2) VG B configuration

[VGB] voice-setup

[VGB-voice] dial-program

[VGB-voice-dial] entity 010 voip

[VGB-voice-dial-entity10] match-template 010....

[VGB-voice-dial-entity10] address ip 1.1.1.1

[VGB-voice-dial-entity10] entity 2001 pots

[VGB-voice-dial-entity2001] match-template 0755....

[VGB-voice-dial-entity2001] send-number all

[VGB-voice-dial-entity2001] line 4

1.5.3 Configuring the FXO Subscriber-line to Work in Private-line Auto-ring Mode

I. Network requirements

Configure FXO subscriber-line on VG B to work in private-line automatic ring mode and the default remote connection number is 0101001.

When the subscriber attached to the PBX with the number of 07552001 dials 07552003, the subscriber is first connected to VG B. Because the FXO subscriber-line is working in private-line auto-ring mode, it will automatically use the preset connection number to request the remote subscriber attached to VG A to establish a connection with it.

II. Network diagram

Figure 1-14 The FXO subscriber line on VG B works in private-line automatic ring mode

III. Configuration procedure

& Note:

l Configure the IP address and phone number correctly according to the actual network environment.

1) VG A configuration

[VGA] voice-setup

[VGA-voice] dial-program

[VGA-voice-dial] entity 0755 voip

[VGA-voice-dial-entity755] match-template 0755....

[VGA-voice-dial-entity755] address ip 2.2.2.2

[VGA-voice-dial-entity755] entity 1001 pots

[VGA-voice-dial-entity1001] match-template 010....

[VGA-voice-dial-entity1001] line 0

2) VG B configuration

[VGB] voice-setup

[VGB-voice] dial-program

[VGB-voice-dial] entity 010 voip

[VGB-voice-dial-entity10] match-template 010....

[VGB-voice-dial-entity10] address ip 1.1.1.1

[VGB-voice-dial-entity10] entity 2001 pots

[VGB-voice-dial-entity2001] match-template 0755....

[VGB-voice-dial-entity2001] send-number all

[VGB-voice-dial-entity2001] line 4

# Perform the following configuration to FXO Subscriber-line 4.

[VGB-voice-dial-entity2001] quit

[VGB-voice-dial] quit

[VGB-voice] subscriber-line 4

[VGB-voice-dial-line4] private-line 0101001

1.5.4 Configuring Voice Fast Connection

I. Network requirements

VG A and VG B are connected to each other through the WAN. The call from VG A to VG B adopts fast connection mode and supports out-of-band transmission of DTMF code. The call from VG B to VG A does not adopt fast connection mode and supports out-of-band transmission of DTMF code.

II. Network diagram

Figure 1-15 Typical network diagram for voice fast connection

III. Configuration procedure

& Note:

l The configuration of the scenario is made assuming that the route between VG A and VG B is reachable. For the configurations of the Ethernet interface and the default route, see 1.5.1 “Configuring FXS Interface to Implement Interconnection”.

l Configure the IP address and phone number correctly according to the actual network environment.

1) VG A configuration:

# Configure the VoIP voice entity.

[VGA] voice-setup

[VGA-voice] dial-program

[VGA-voice-dial] entity 0755 voip

[VGA-voice-dial-entity755] match-template 0755....

[VGA-voice-dial-entity755] address ip 2.2.2.2

# Enable fast connection and out-of-band transmission of DTMF code for VoIP voice entity.

[VGA-voice-dial-entity755] fast-connect

[VGA-voice-dial-entity755] outband h225

# Configure the local port and telephone number for Tel.1.

[VGA-voice-dial-entity755] entity 1001 pots

[VGA-voice-dial-entity1001] match-template 0101001

[VGA-voice-dial-entity1001] line 0

# Enable out-of-band transmission function of DTMF code for POTS voice entity corresponding to Tel.1.

[VGA-voice-dial-entity1001] outband h225

2) VG B configuration:

# Configure the VoIP voice entity.

[VGB] voice-setup

[VGB-voice] dial-program

[VGB-voice-dial] entity 010 voip

[VGB-voice-dial-entity10] match-template 010....

[VGB-voice-dial-entity10] address ip 1.1.1.1

# Enable out-of-band transmission function of DTMF code for the VoIP voice entity.

[VGB-voice-dial-entity10] outband h225

# Configure the local port and telephone number for Tel.2.

[VGB-voice-dial-entity10] entity 2001 pots

[VGB-voice-dial-entity2001] match-template 07552001

[VGB-voice-dial-entity2001] line 0

# Enable out-of-band transmission function of DTMF code for POTS voice entity corresponding to Tel.2.

[VGB-voice-dial-entity2001] outband h225

1.5.5 Application of Dial Program

I. Network requirements

There are two PBX local telephony networks, one in city A and the other in city B. They are connected by two voice gateways. The subscribers within the PBX networks can dial remote ordinary telephone numbers using VoIP.

There are financial, marketing, and sales departments in city A and city B respectively. If someone working in one of these departments wants to dial a number of another department in the other city, he or she just needs to know the local telephone number of the department and the area code of the other city. For example, you are in city B financial department and dial 3366 to call the local marketing department. If you want to call city A marketing department from city B financial department, you can dial 0103366. The CID in city A marketing department is 0211234, that is, area code of city B + telephone number of the financial department.

This setting is very flexible. You can dial 0106565 to call city A marketing department from city B financial department, as long as there is no number collision.

This function is very convenient for the army, because the military departments have the same organization system.

II. Network diagram

Figure 1-16 Network diagram of voice dial program

III. Clews

City B calling side: Changes the called number to a middle number.

City A called side: Changes the middle number to the local corresponding number, and initiates the call.

The processes for the calling number are similar.

IV. Configuration procedure

& Note:

l Configure the IP address and phone number correctly according to the actual network environment.

1) Configure the City B voice gateway:

# Configure a number-substitute list for the called number:

[VG] voice-setup

[VG-voice] dial-program

[VG-voice-dial] number-substitute 21101

[VG-voice-dial-subst21101] rule 1 0101688 0001

[VG-voice-dial-subst21101] rule 2 0103366 0002

[VG-voice-dial-subst21101] rule 3 0102323 0003

# Configure a number-substitute list for the calling number:

[VG-voice-dial-subst21101] quit

[VG-voice-dial] number-substitute 21102

[VG-voice-dial-subst21102] rule 1 1688 0210001

[VG-voice-dial-subst21102] rule 2 3366 0210002

[vg-voice-dial-subst21102] rule 3 2323 0210003

# Configure the VoIP voice entity destined to city A.

[VG-voice-dial-subst21102] quit

[VG-voice-dial] entity 10 voip

[VG-voice-dial-entity10] match-template 010....

[VG-voice-dial-entity10] address ip 1.1.1.1

[VG-voice-dial-entity10] substitute called 21101

[VG-voice-dial-entity10] substitute calling 21102

2) Configure the City A voice gateway:

# Set the Ethernet interface:

[VG] interface ethernet 0

[VG-Ethernet0] ip address 1.1.1.1 255.255.255.0

# Configure a number-substitute list for the called number:

[VG] voice-setup

[VG-voice] dial-program

[VG-voice-dial] number-substitute 101

[VG-voice-dial-subst101] rule 1 ^0001$ 1234

[VG-voice-dial-subst101] rule 2 ^0002$ 6788

[VG-voice-dial-subst101] rule 3 ^0003$ 6565

# Configure a number-substitute list for the calling number:

[VG-voice-dial-subst101] quit

[VG-voice-dial] number-substitute 102

[VG-voice-dial-subst102] dot-match left-right

[VG-voice-dial-subst102] rule 1 ^...0001$ ...1234

[VG-voice-dial-subst102] rule 2 ^...0002$ ...6788

[VG-voice-dial-subst102] rule 3 ^...0003$ ...6565

# Set to apply the number-substitute rules

[VG-voice-dial-subst102] quit

[VG-voice-dial] subst incoming-call called 101

[VG-voice-dial] subst incoming-call calling 102

# Configure the local port Line 0.

[VG-voice-dial] entity 1010 pots

[VG-voice-dial-entity1010] match-template ....

[VG-voice-dial-entity1010] line 0

[VG-voice-dial-entity1010] send-number all

# Configure the local port Line 1.

[VG-voice-dial-entity1010] quit

[VG-voice-dial] entity 2010 pots

[VG-voice-dial-entity2010] match-template ....

[VG-voice-dial-entity2010] line 1

[VG-voice-dial-entity2010] send-number-all

1.5.6 Example of Busy-Tone Detection

I. Network requirements

VB B has a FXO port, through which it is connected to the PBX using automatic busy-tone detection.

II. Network diagram

Figure 1-17 Diagram of busy-tone detection

III. Configuration procedure

# Enable the busy-tone detection on FXO port 4 on VG B.

[VGB] voice-setup

[VGB-voice] vi-card busy-tone-detect auto 0 4

& Note:

l Other configuration steps are the same as those described in section 1.5.2 “Example of Basic Configuration on the FXO Port”.

l Configure the IP address and phone number correctly according to the actual network environment.

1.5.7 Configuring CID

I. Network requirements

Disable the FXS port on VG A from sending caller identity to the subscriber with the number of 0101001.

VG B has a FXO port, through which it is connected to PBX and can receive CID from the PBX in simple format.

II. Network diagram

Figure 1-18 Network diagram of CID

III. Configuration procedure

1) Disable CID on VG A.

[VGA] voice-setup

[VGA-voice] subscriber-line 0

[VGA-voice-line0] undo cid display

2) Configure CID on VG B in simple format.

[VGB] voice-setup

[VGB-voice] subscriber-line 4

[VGB-voice-line4] cid type simple

& Note:

l Other configuration steps are the same as those described in section 1.5.2 “Example of Basic Configuration on the FXO Port”.

l Configure the IP address and phone number correctly according to the actual network environment.

1.5.8 Adjusting Volume

I. Network requirements

l The VGs support volume control. The administrator can adjust the signal intensity of the input/output VG according to the strength of line signals.

l Shown in the following network diagram, Tel.1, a user on VG A, complains that the volume of the opposite side is very low during conversation, whereas according to Tel.2, another user on VG A, the volume is low only in conversation with Tel.3, a user on VG B.

l Based on the complaints of users, the administrator confirms that the problem is caused by weak output signals of Line 0 on VG A and weak input signals of Line 0 on VG B.

l Accordingly, the administrator adjusts the volume for the said voice subscriber lines on VG A and VG B.

II. Network diagram

Figure 1-19 Network diagram of volume adjustment

III. Configuration procedure

1) Configure VG A

# Configure the default route and Ethernet IP address.

[VGA] ip route-static 0.0.0.0 0 1.1.1.2

[VGA] interface ethernet 0

[VGA-Ethernet0] ip address 1.1.1.1 255.255.255.0

# Configure VoIP voice entity.

[VGA] voice-setup

[VGA-voice] dial-program

[VGA-voice-dial] entity 0755 voip

[VGA-voice-dial-entity755] match-template 0755....

[VGA-voice-dial-entity755] address ip 2.2.2.2

[VGA-voice-dial-entity755] quit

# Configure the local port connected to Tel.1.

[VGA-voice-dial] entity 1001 pots

[VGA-voice-dial-entity1001] match-template 0101001

[VGA-voice-dial-entity1001] line 0

[VGA-voice-dial-entity1001] quit

# Configure the local port connected to Tel.2.

[VGA-voice-dial] entity 1002 pots

[VGA-voice-dial-entity1002] match-template 0101002

[VGA-voice-dial-entity1002] line 1

[VGA-voice-dial-entity1002] return

# Adjust the transmitting gain of voice subscriber line0.

[VGA] voice-setup

[VGA-voice] subscriber-line 0

[VGA-voice-line0] transmit gain 5

2) Configure VG B

# Configure the default route and Ethernet IP address.

[VGB] ip route-static 0.0.0.0 0 2.2.2.1

[VGB] interface ethernet 0

[VGB-Ethernet0] ip address 2.2.2.2 255.255.255.0

# Configure VoIP voice entity.

[VGB] voice-setup

[VGB-voice] dial-program

[VGB-voice-dial] entity 010 voip

[VGB-voice-dial-entity10] match-template 010....

[VGB-voice-dial-entity10] address ip 1.1.1.1

[VGB-voice-dial-entity10] quit

# Configure the local port connected to Tel.3.

[VGB-voice-dial] entity 2001 pots

[VGB-voice-dial-entity2001] match-template 07552001

[VGB-voice-dial-entity2001] line 0

[VGB-voice-dial-entity2001] quit

# Adjust the receiving gain of voice subscriber line0.

[VGB] voice-setup

[VGB-voice] subscriber-line 0

[VGB-voice-line0] receive gain 6

1.5.9 Lines Group Access

I. Network requirements

l Set up a VoIP network by using voice gateways, and allocate the number resource reasonably by means of the lines group access function.

l Users under VG A belong to Group A. In this example, the short numbers of user 66004000 and user 66004001 are 4000 and 4001 respectively.

l Users under VG B belong to Group B. In this example, the short numbers of user 88003000 and user 88003001 are 3000 and 3001 respectively.

II. Network diagram

Figure 1-20 Network diagram for lines group access

III. Configuration procedure

1) Configure VG A:

# Configure the IP address of the Ethernet interface:

[VGA] interface ethernet 0

[VGA-Ethernet0] ip address 10.1.1.1 255.255.255.0

# Configure the VoIP voice entity:

[VGA] voice-setup

[VGA-voice] dial-program

[VGA-voice-dial] entity 8800 voip

[VGA-voice-dial-entity8800] match-template 8800....

[VGA-voice-dial-entity8800] address ip 10.1.1.2

[VGA-voice-dial-entity8800] quit

# Configure the POTS voice entities for users in Group A:

[VGA-voice-dial] entity 4000 pots

[VGA-voice-dial-entity4000] match-template (6600)!4000

[VGA-voice-dial-entity4000] line 0

[VGA-voice-dial-entity4000] quit

[VGA-voice-dial] entity 4001 pots

[VGA-voice-dial-entity4001] match-template (6600)!4001

[VGA-voice-dial-entity4001] line 1

[VGA-voice-dial-entity4001] return

2) Configure VG B

# Configure the IP address of the Ethernet interface:

[VGB] interface ethernet 0

[VGB-Ethernet0] ip address 10.1.1.2 255.255.255.0

# Configure the VoIP voice entity of Group A:

[VGB] voice-setup

[VGB-voice] dial-program

[VGB-voice-dial] entity 6600 voip

[VGB-voice-dial-entity6600] match-template 6600....

[VGB-voice-dial-entity6600] address ip 10.1.1.1

[VGB-voice-dial-entity6600] quit

# Configure the POTS voice entities for users in Group B:

[VGB-voice-dial] entity 3000 pots

[VGB-voice-dial-entity3000] match-template (8800)!3000

[VGB-voice-dial-entity3000] line 0

[VGB-voice-dial-entity3000] quit

[VGB-voice-dial] entity 3001 pots

[VGB-voice-dial-entity3001] match-template (8800)!3001

[VGB-voice-dial-entity3001] line 1

[VGB-voice-dial-entity3001] quit

1.6 Troubleshooting

Fault 1: Hear busy tone immediately after finishing dial

Troubleshooting: perform according to the following steps.

l First to check whether the opposite route exists. You can use ping command to ping opposite IP address.

l Check whether the configuration of voice entity is correct.

l Check whether the configuration of phone number is correct. You can use display voice call-history-record line line-number command to check call history record.

Fault 2: Unable to hang up.

Show as in following figure: telephone Tel1 is connected to PBX1, which provides 1 line common telephone line to connect with the FXO subscriber-line on VG 1. The opposite configuration is similar to this.

Figure 1-21 Network diagram of busy tone detection

To describe conveniently, it is supposed that Tel1 hangs up first after conversation. PBX_1 plays busy tone to local VG_1, which will hang up and send the disconnect message to VG_2 after checked this busy tone signal. After receiving the message, VG_2 starts hang-up indication to PBX_2 and PBX_2 releases busy tone to Tel2 after receiving this disconnect message. The hang-up process is completed in this way.

If FXO subscriber-line cannot be able to check the busy tone signal sent by switch, it will keep conversation continuously, thereby resulting in being unable to hang up. Such faults should be resolved through busy tone check.

Troubleshooting: Perform according to the following steps.

l Simplest condition (switch uses North American Standard). Because the voice gateway uses busy tone identification of Europe Standard by default, it should be changed into North American Standard. On FXO line0 of the voice gateway run the area north-america command to configure busy tone type as North American Standard. If the problem is not resolved after running the area command, continue the next step:

l Automatic busy tone check: the voice gateway can identify busy tone with intelligent software, i.e., the software works out one set of parameters closest to busy tone features through sampling, calculating and automatically analyzing input busy tone. The subscriber perform configuration on corresponding FXO subscriber-line with this set of parameter and the busy tone check can be finished.

& Note:

The busy tone types on all subscriber-lines are consistent. But you must use the leftmost subscriber-line, i.e., subscriber-line 0 during configuration and detection.

Detect busy tone following the steps for the automatic busy tone detection in the earlier section. If failed, it may be that the operation of checking busy tone parameter failed. Repeat above operations until the busy tone parameters are detected.

Chapter 2 Fax Configuration

2.1 Fax Overview

I. Introduction to fax

Traditional fax transmission is over PSTN. The fax service allows the transmission of various types of information in a high speed and is easy to handle. Due to these benefits, the fax service has gained wide application through years of development. By far, G3 fax machines are the fax terminals dominant in the fax communications. A G3-category fax machine is the communications device that adopts the digital signal processing technology. It digitizes and compresses the image signals internally, converts the signals into the analog signals via a modem, and then transmits them into the switch via the connected common subscriber-line.

Fax over IP (FoIP), as the name implies, is to send and receive faxes over the Internet. The VG series provides users, including the PSTN users, with the FoIP function in addition to the VoIP function. This allows users to send international and national faxes at a rather low cost.

The following figure shows the FoIP system structure:

Figure 2-1 Structure of the FoIP system

II. Protocol-compliance of FoIP

The real-time FoIP is compliant with ITU-T T.30 and T.4 at the PSTN side and H.323 and T.38 at the IP side.

l T.30 applies to document faxing over PSTN. It describes and provisions in detail the communication flow of G3 fax machines over common telephone networks, as well as signal format, control signaling, and error correction mode adopted in the communications.

l T.4 is a standard protocol involving the G3 fax terminals for file transmission. It provisions the standards of G3 fax terminals in image encoding, signal modulation and speed, transmission duration, error correction, and file transmission mode.

l T.38 is about the real-time G3 FoIP. It describes and provisions the communication mode, packet format, error correction, and a portion of the communication flow of the real-time FoIP.

III. IP Faxing flow

In the real-time faxing, the calling terminal and the called terminal set up a call, shake hands, make training, transmit packets, and tear down the call totally at real time. From the perspective of a user, it is the same as faxing over PSTN.

The signals that a G3 fax machine receives and sends are the modulated analog signals, therefore a VG processes the fax signals in a way different from telephone signals. First, it should perform fax signals modulation or demodulation, i.e. analog-to-digital demodulation on signals from PSTN or digital-to-analog modulation on signals from IP. Besides, the VG needs not compress the fax signals.

A real-time faxing process is made up of 5 phases:

1) Phase A: Fax call setup stage, which is similar to the process of telephone call setup. The difference is that the fax individual tone identifying sender/receiver is included;

2) Phase B: Pre-message stage, where fax facilities negotiation and training are performed.

3) Phase C: Messaging stage where fax packets are transmitted in compliance with T.4 provisions and packet transmission is controlled (including packet synchronization, error code detection and correction, and line track).

4) Phase D: Post-messaging stage where the control operations such as packet authentication, messaging completion, and multi-page continuous transmission are performed.

5) Phase E: Fax call is released to terminate the call.

2.2 FoIP Configuration

Before configuring FoIP, the user should configure VoIP. For the VoIP configuration procedure, see Chapter 1 “VoIP Configuration” of this manual.

After completing the VoIP configurations, you should be able to get through IP phone calls. Usually, the default IP Fax configurations are adequate for you to send and receive faxes so long as a fax machine is connected in this case. Configuring IP Fax is mainly to set the specific IP Fax parameters or make configurations for some special purposes.

The FoIP configuration tasks include:

l Enable ECM for faxing

l Set fax facilities signal transmission mode

l Set the maximum fax rate

l Set fax training mode

l Set the fax local training threshold percentage

l Set the transmitting level of gateway carrier

l Set the fax interoperating protocol

l Set the fax transmission format

l Set the number of redundant packets to send

l Set the threshold for CNG/CED signal detection

l Set the default fax parameter values in global scope

2.2.1 Configuring the Fax to Work in ECM Mode

As provisioned in ITU-T, Error Correction Mode (ECM) is required for fax message transmission using the ITU-T V.34 half-duplex and half modulation system. In addition, the G3 fax terminals working in full duplex are required to support half-duplex, that is, to support ECM.

The fax machines using ECM can correct errors, provide Automatic Repeat Request (ARQ), and transmit fax packets in the HDLC frame format. On the contrary, the fax machines using non-ECM (which must be supported on fax machines) cannot correct errors and transmit fax information in the form of binary strings.

In practice, non-ECM will be used as long as the GW is configured to work in the non-ECM mode, even if both fax machines at the two ends support ECM. Non-ECM will also be adopted if either or neither of the fax machines supports ECM. ECM can be adopted only if it is supported at both fax machines and the GW as well.

Perform the following configurations in voice entity view.

Table 2-1 Configure ECM for faxing

Operation	Command
Adopt ECM for faxing.	fax ecm
Disable ECM for faxing.	undo fax ecm

By default, ECM is not adopted on GWs.

2.2.2 Configuring Fax Facilities Signal Transmission Mode

In a normal fax application, the participating fax terminals negotiate with the standard facilities (such as V.17 or V.29 rate). In this case, they do not send each other Non-Standard Faculties (NSF) message frames. In some circumstances (such as encrypted fax transmission), NSF is somewhat important for fax communications. Therefore, before starting the negotiation, the participating fax terminals will first exchange the NSF message frames and complete the subsequent fax negotiation in NSF mode for communications. NSF is a type of standard T.30 message, which carries private information.

Perform the following configurations in voice entity view.

Table 2-2 Configure fax facilities signal transmission mode

Operation	Command
Enable NSF mode for fax facilities signal transmission.	fax nsf-on
Restore the default fax facilities signal transmission mode.	undo fax nsf-on

By default, standard facilities mode is adopted for fax negotiation.

2.2.3 Configuring the Maximum Fax Rate

The user can configure the maximum fax rate for the participating fax terminals depending on the applied fax protocol. If a rate is set to a value other than “disable” or “voice”, the fax protocol associated with the speed will be first used for rate negotiation. The specified rate is the permitted maximum rate, which is not necessarily used.

If “voice” is adopted, the maximum fax rate will first be determined by an appropriate voice codec protocol.

l If G.711 voice codec protocol is used, the allowable maximum fax rate is 14400 bps and the fax protocol for it is V.17;

l If G.723.1 Annex A voice codec protocol is used, the allowable maximum fax rate is 4800 bps and the fax protocol for it is V.27.

l If G.729 voice codec protocol is used, the allowable maximum fax rate is 9600 bps and the fax protocol for it is V.29.

If “disable” is set, fax function will be disabled.

Perform the following configurations in voice entity view.

Table 2-3 Configure fax rate

Operation	Command
Configure the maximum fax rate.	fax baudrate { 14400 \| 2400 \| 4800 \| 9600 \| disable \| voice }
Restore the default fax rate.	undo fax baudrate

By default, voice mode is first used for determining the fax rate.

& Note:

The VG 20-16, VG 20-32 and VG 21-08 voice gateways do not support G.723.

2.2.4 Configuring Fax Training Mode

If GWs participate in the rate training between the fax machines at two ends, it is called local rate training. In this approach, the training is first implemented between the fax machine and the attached GW at each end. Then, the receiving GW sends the training result of the called terminal to the GW of the calling terminal, and the sending GW will determine the ultimate packet transmission rate by comparing the training results of both ends.

In the peer-to-peer training approach, however, the GWs do not participate in the rate training between the fax machines at both ends. In this approach, the rate training is conducted between the two fax terminals and is transparent from the perspective of the GWs.

Perform the following configurations in voice entity view.

Table 2-4 Configure fax training mode

Operation	Command
Configure fax training mode.	fax train-mode { local \| ppp }
Restore the default fax training mode.	undo fax train-mode

By default, the peer-to-peer mode (PPP) is adopted.

2.2.5 Configuring the Threshold Percentage of Local Fax Training

When two fax machines carry out the rate training, the sender will first send the TCF data of “0”s to the receiver for 1.5±10% seconds, and the receiver will determine whether the current rate is acceptable accordingly.

If you have set the training mode to local training, use the command in the following table to configure the threshold percentage of local training. If errors occur to the TCP data during its transmission, there will be the presence of “1”s in the received TCF data. If the percentage of the received “1”s to the overall TCP data is lower than the specified threshold, the current rate training, which will be otherwise regarded a failure, is successful.

Perform the following configurations in voice entity view.

Table 2-5 Configure threshold percentage of the local fax training

Operation	Command
Configure threshold percentage of the local fax training.	fax local-train threshold threshold
Restore the default threshold percentage of the local fax training.	undo fax local-train threshold

By default, threshold percentage (threshold) of the local fax training is set to 10.

2.2.6 Configuring GW Carrier Level

Normally, the default GW carrier level is adequate for transmission. If the fax cannot be set up yet on the premise that all other configurations are correct, the subscriber can attempt to adjust the GW carrier level. The bigger the level value is, the greater energy is; the smaller the level value is, the higher the attenuation is.

Perform the following configurations in voice entity view.

Table 2-6 Configure the GW carrier level

Operation	Command
Configure the GW carrier level.	fax level level
Restore the default GW carrier level.	undo fax level

By default, the GW carrier level (defined by the parameter level) is -15, namely the transmit carrier level is -15dBm. .

2.2.7 Configuring Fax Interconnection Protocols

The VG support the standard packet format defined by ITU-T T.38.

The VGs support two fax starting modes: H.323 negotiation mode and VG negotiation mode, corresponding to fax protocols H.328-T.38 and T.38 respectively.

The Fax Passthrough technology was primarily developed for the purpose of transmitting the T.30 fax packets over packet switching networks. With this technology, the devices at two ends can directly communicate on a transparent IP connection, and the VGs do not discriminate fax calls from voice calls. After successfully detecting the fax tone in an established VoIP call, the VG will load the Passthrough parameters to be used in the fax communications while terminating the voice coding/decoding process. Then, it will switch over to the Fax Passthrough mode, with which the fax information will be encapsulated in the GW-to-GW RTP packets in the form of uncompressed G.711 codes. Transmitting fax information with this approach will occupy a fixed bandwidth of 64K. Fax Passthrough is sensitive to the elements like loss rate, jitter, and latency. Therefore, even though the packet loss in a network can be alleviated by introducing the packet redundancy mechanism, it is important to ensure synchronization of the clocks at the two ends participating in the communications. The Fax Passthrough function is called Voice Band Data (VBD) by ITU-T. That is, fax or modem signals are transmitted in voice channels over a packet switched network by using a proper coding method. So far, only two types of coding/decoding methods support Passthrough transmission: G.711 A-law and G.711 µ-law. In addition, it is recommended that you disable Voice Activity Detection (VAD) to avoid fax failure when using the Passthrough function.

The VGs support fax passthrough transmission through the following two types of configurations:

l The PCM mode is enabled as the fax protocol.

l The negotiated voice compensation mode is G.711, and the fax baudrate is set to “disable” (fax forwarding is disabled), and the VAD is disabled to avoid fax failure. This mode is applicable to passthrough transmission with other devices.

& Note:

l T.38 is adopted to interoperate with H3C devices, while H.323-T.38 is used to interoperate with devices of other manufactures including Cisco.

l The fax redundancy command is available only if H.323-T.38 or T.38 is used as the fax protocol.

Perform the following configurations in voice entity view.

Table 2-7 Configure fax interoperation protocols

Operation	Command
Adopt the H323-T.38 fax protocol.	fax protocol h323-t38
Adopt the T.38 fax protocol.	fax protocol t38
Adopt the Passthrough fax mode.	fax protocol pcm { g711alaw \| g711ulaw }
Restore the default fax protocol.	undo fax protocol

By default, T.38 fax protocol is adopted.

2.2.8 Configuring Fax Transmission Format

Normally, if T.38 is adopted, RTP mode (defined by the parameter rtp) will be used. But VT mode (defined by the parameter vt) should be used in order to interoperate with a VocalTec's GW.

Perform the following configurations in voice entity view.

Table 2-8 Configure the fax transmission format

Operation	Command
Configure the fax transmission format	fax support-mode { rtp \| vt }
Restore the default fax transmission format	undo fax support-mode

By default, RTP mode is adopted.

2.2.9 Configuring the Number of Redundant Packets

The VG series supports the control over high-speed and low-speed redundant packets. Increasing the number of redundant packets can improve transmission reliability and reduce the loss resulted from packet drops. However, too many redundant packets will greatly the consumption of bandwidth. On a low-bandwidth network, this will seriously degrade the fax quality. Therefore, you should carefully select an appropriate number of redundant packets depending on the network bandwidth.

Perform the following configuration in voice entity view.

Table 2-9 Configure the number of redundant packets

Operation	Command
Configure the number of redundant packets	fax redundancy { hb-redundancy \| lb-redundancy } number
Restore the default number of redundant packets	undo fax redundancy { hb-redundancy \| lb-redundancy }

By default, the number of fax redundant packets is 0.

2.2.10 Configuring Threshold for Detecting CNG/CED Signals

Calling Tone (CNG) is generated when the fax at the calling end starts. Called Terminal Identification (CED) is generated when the fax at the called end starts.

A VG confirms the fax state by detecting the CNG/CED, however, as it is applied in various environments, sometimes it may fail to detect or have error detection. In these circumstances, you can use this command to set the detection parameter to adjust the sensation and reliability of device.

The greater the threshold and times values are, the more reliability the detection is. But this does not mean the detection never fails. The times parameter specifies the lower limit of CNG/CED duration. For instance, if the default value of times is 10, the CNG/CED signal lasting at least 300ms is regarded valid. 30 ms is added to duration for every 1 increase in the times value.

Perform the following configuration in voice entity view.

Table 2-10 Configure the threshold for detecting CNG/CED signals

Operation	Command
Set the threshold parameter for detecting CNG/CED signal	cngced-detection threshold times
Restore the default value of the threshold parameter	undo cngced-detection

By default, the threshold parameter is 0 and the times parameter is 10.

& Note:

This function is applicable to VG 10-40 and VG 10-41 voice gateways only.

2.2.11 Configuring Global Fax Default Parameters

Perform the following configurations in voice dial program view.

Table 2-11 Configure the default values of the fax parameters in the global scope

Operation	Command
Configure the GW transmitting level in the global scope.	default entity fax level level
Configure the local fax training threshold in the global scope.	default entity fax local-train threshold threshold
Configure the protocol interoperating with other devices in the global scope.	default entity fax protocol { h323-t38 \| pcm { g711alaw \| g711ulaw } \| t38 }
Configure the number of redundant packets of high baudrate and low baudrate in the global range	default entity fax redundancy { hb-redundancy \| lb-redundancy } number
Configure the maximum fax speed in the global scope.	default entity fax baudrate { 12000 \| 14400 \| 2400 \| 4800 \| 7200 \| 9600 \| disable \| voice }
Configure the transmission mode of fax facilities signal in the global scope.	default entity fax nsf-on
Configure the device interoperating mode in the global scope.	default entity fax support-mode { rtp \| vt }
Configure the fax training in the global scope.	default entity fax train-mode { local \| ppp }
Enable ECM in the global scope.	default entity fax ecm

By default, the GW carrier level is set to -15, local training with the threshold of 10 is adopted, the number of slow baudrate and fast baudrate packets is 0, the fax rate is acknowledged in voice mode, standard fax facilities negotiation approach is adopted, the RTP transmission format is adopted for the sake of device interoperation, and ECM is disabled.

2.3 Displaying and Debugging Fax

After completing the configurations described above, you can use the display command in any view to verify the configuration effect after configuring various fax parameters.

Execute the following command in any view.

Table 2-12 Display and debug the fax information

Operation	Command
Display the fax statistics of the Fax module	display voice fax statistics
Enable all fax debugging of the Fax module.	debugging voice fax all
Enable the API function debugging of the Fax module.	debugging voice fax api
Enable the main task debugging of the Fax module.	debugging voice fax cc
Enable the debugging of a specified channel of the Fax module.	debugging voice fax channel channel-number
Enable the controller debugging of the Fax debugging.	debugging voice fax controller
Enable the debugging of the Fax module in errors at all levels.	debugging voice fax error-all
Enable the debugging on the information between the Fax and the IPP modules.	debugging voice fax ipp
Enable the T.38 information debugging of the Fax module.	debugging voice fax t38
Enable debugging of the fax data read and written between the VAS software module and the voice card.	debugging voice vas fax
Enable the debugging of fax data flow	debugging voice data-flow fax
Enable the debugging of fax data flow error information	debugging voice data-flow fax error
Clear fax statistics information	reset voice fax

2.4 Fax Configuration Examples

2.4.1 FoIP Configuration Examples

I. Configuring T.38

1) Network requirements

VG A and VG B send and receive faxes via an IP network, and both of them are directly connected to fax terminals via FXS subscriber-lines. The default fax protocol T.38 is adopted.

The FXS subscriber-lines connected VG A and VG B to the fax machines are respectively numbered 0101001 and 07552001; the IP addresses of the interfaces connected VG A and VG B to the Internet are respectively 1.1.1.1 and 2.2.2.2.

2) Network diagram

Figure 2-2 Network for the typical FoIP configuration

3) Configuration procedure

& Note:

All the configurations and discussion in this example are based on the assumption that the route between VG A and VG B is reachable. For the configurations of Ethernet interface IP addresses and default route, see the examples in Chapter 1 “VoIP Configuration”.

# Configure VG A:

[VGA] voice-setup

[VGA-voice] dial-program

[VGA-voice-dial] entity 0755 voip

[VGA-voice-dial-entity755] match-template 0755....

[VGA-voice-dial-entity755] address ip 2.2.2.2

[VGA-voice-dial-entity755] entity 1001 pots

[VGA-voice-dial-entity1001] match-template 0101001

[VGA-voice-dial-entity1001] line 0

# Configure VG B:

[VGB] voice-setup

[VGB-voice] dial-program

[VGB-voice-dial] entity 010 voip

[VGB-voice-dial-entity10] match-template 010....

[VGB-voice-dial-entity10] address ip 1.1.1.1

[VGB-voice-dial-entity10] entity 2001 pots

[VGB-voice-dial-entity2001] match-template 07552001

[VGB-voice-dial-entity2001] line 0

II. Configuring G.711 Passthrough transmission

1) Network requirements

VG A and VG B send and receive faxes via an IP network, and both of them are directly connected to fax terminals via FXS subscriber-lines. The G.711 Passthrough function is adopted.

2) Network diagram

Figure 2-3 Network for the typical FoIP configuration

3) Configuration procedure

& Note:

# Configure VG A:

[VGA] voice-setup

[VGA-voice] dial-program

[VGA-voice-dial] entity 0755 voip

[VGA-voice-dial-entity755] match-template 0755....

[VGA-voice-dial-entity755] address ip 2.2.2.2

[VGA-voice-dial-entity755] fax protocol pcm g711alaw

[VGA-voice-dial-entity755] entity 1001 pots

[VGA-voice-dial-entity1001] match-template 0101001

[VGA-voice-dial-entity1001] fax protocol pcm g711alaw

[VGA-voice-dial-entity1001] line 0

# Configure VG B:

[VGB] voice-setup

[VGB-voice] dial-program

[VGB-voice-dial] entity 010 voip

[VGB-voice-dial-entity10] match-template 010....

[VGB-voice-dial-entity10] address ip 1.1.1.1

[VGB-voice-dial-entity10] fax protocol pcm g711alaw

[VGB-voice-dial-entity10] entity 2001 pots

[VGB-voice-dial-entity2001] match-template 07552001

[VGB-voice-dial-entity2001] fax protocol pcm g711alaw

[VGB-voice-dial-entity2001] line 0

2.4.2 Configuring Fax ECM Mode

I. Network requirements

VG A and VG B send and receive faxes via an IP network, and both of them are directly connected to fax terminals via FXS subscriber-lines. ECM is adopted for receiving and sending faxes.

II. Network diagram

Figure 2-4 Network for the typical FoIP configuration

III. Configuration procedure

# Configure VG A:

[VGA] voice-setup

[VGA-voice] dial-program

[VGA-voice-dial] entity 0755 voip

[VGA-voice-dial-entity755] match-template 0755....

[VGA-voice-dial-entity755] address ip 2.2.2.2

[VGA-voice-dial-entity755] fax ecm

[VGA-voice-dial-entity755] entity 1001 pots

[VGA-voice-dial-entity1001] match-template 0101001

[VGA-voice-dial-entity1001] fax ecm

[VGA-voice-dial-entity1001] line 0

# Configure VG B:

[VGB] voice-setup

[VGB-voice] dial-program

[VGB-voice-dial] entity 010 voip

[VGB-voice-dial-entity10] match-template 010....

[VGB-voice-dial-entity10] address ip 1.1.1.1

[VGB-voice-dial-entity10] fax ecm

[VGB-voice-dial-entity10] entity 2001 pots

[VGB-voice-dial-entity2001] match-template 07552001

[VGB-voice-dial-entity2001] fax ecm

[VGB-voice-dial-entity2001] line 0

2.4.3 Configuring the Fax Transmission format

I. Network requirements

VG A and VocalTec Voice GW send and receive faxes via an IP network, and both of them are directly connected to fax terminals via FXS subscriber-lines. For configuration of a VocalTec voice gateway, refer to the configuration manual related to the VocalTec voice gateway.

The FXS subscriber-lines connected VG A and VocalTec Voice GW to the fax machines are respectively numbered 0101001 and 07552001; the IP addresses of the interfaces connected VG A and VocalTec Voice GW to the Internet are respectively 1.1.1.1 and 2.2.2.2.

II. Network diagram

Figure 2-5 Network for the typical FoIP configuration

III. Configuration procedure

& Note:

l All the configurations and discussion in this scenario are made assuming that the route between VG A and VocalTec Voice GW is reachable.

l For the configurations of Ethernet interface and default route of VG A, see the examples in Chapter 1 “VoIP Configuration”.

l The configurations of VocalTec Voice GW are beyond the scope of this example.

l Configure the IP address and phone number correctly according to the actual network environment.

# Configure VG A:

[VGA] voice-setup

[VGA-voice] dial-program

[VGA-voice-dial] entity 0755 voip

[VGA-voice-dial-entity755] match-template 0755....

[VGA-voice-dial-entity755] address ip 2.2.2.2

[VGA-voice-dial-entity755] fax support-mode vt

[VGA-voice-dial-entity755] entity 1001 pots

[VGA-voice-dial-entity1001] match-template 0101001

[VGA-voice-dial-entity1001] fax support-mode vt

[VGA-voice-dial-entity1001] line 0

Chapter 3 E1 Voice Configuration

3.1 E1 Voice Overview

3.1.1 Introduction to E1 Voice

I. E1

The E1 system recommended by ITU-T is mainly used in Europe and some countries outside of Europe.

E1, has the sampling frequency of 8 kHz, PCM frame length of 125 ms, 8 bits per code and time-slot bit rate of 64 kbps. E1 adopts 13-segment A-law coding/decoding. Each PCM primary frame of E1 contains 32 time slots. Each PCM primary frame of E1 contains 256 bits. E1 provides 2.048 Mbps bandwidth

II. E1 voice function

E1 voice implements VoIP or VoFR function on E1 line, and provides a voice transmission mode compatible with data transmission. To implement this function, corresponding E1 voice interface must be provided on the router and a series of functions suiting for transmit voice on E1 line.

This function is available only on voice gateways that provide the E1 interface (e.g. H3C VG 80-20 Voice Gateway). The networking adopting E1 circuit to transmit voice is basically the same as general VoIP networking applications, except that the PSTN switch is connected to a voice gateway through an E1 trunk. When E1 line is adopted, the supported signaling include R2 signaling (similar to China signaling No.1), digital E&M signaling, and DSS1 user signaling and QSIG signaling on ISDN PRI interface. The basic networking is shown in the Table 3-1:

Figure 3-1 Basic composition of E1 voice system

If E1 voice mode is adopted, the voice gateway can provide more channels for voice communication, which greatly helps the voice gateway offer more services.

3.1.2 Application of E1 Interface

I. Application of CE1/PRI interface

The physical interface of E1 voice is CE1/PRI interface and this interface is physically divided into 32 time slots. The corresponding numbers range from TS0 to TS31. The methods to use this interface are described as follows:

Dividing time slots logically and TS16 time slot is used as signaling channel. On E1 interface, PRI set or TS set can be created, so TS16 time slot is used to transmit DSS1 user information or R2 signaling.

l When E1 interface is used as ISDN PRI interface, DSS1 user signaling is adopted. TS16 time slot is used as D channel to transmit connection signaling, so only other time slots except TS0 and TS16 (TS16 transmits signaling) can be bundled as one interface whose logical feature equals to ISDN dialing interface.

l When E1 interface is used as CE1 interface having signaling channel, every 32 time slots comprises of one primary frame (e.g., PCM30 frame structure) if R2 signaling is adopted. Here, TS0 is used for frame synchronization, TS16 is used to transmit the control information of digital line signaling, and other 30 time slots are used to transmit voice information. Every 16 primary frames comprises one multiframe. In each multiframe, TS0 of even primary frame is used to transmit FAS (flag of frame synchronization) and TS0 of odd primary frame is used to transmit NFAS (flag of non-frame-synchronization) that transmits status information about link and provides control signaling for basic rate multiplexing. The upper 4 bits of TS16 of first primary frame (Frame0) of each multiframe transmit the multiframe synchronization flag (MFAS) and the lower 4 bits transmit non-multiframe-synchronization flag (NMFAS). TS16 of other 15 primary frames respectively transmit line status of two time slots. For example, TS16 of second basic frame (Frame1) transmits the digital line signaling of TS1 and TS17, and TS16 of third basic frame (Frame2) transmits the digital line signaling status of TS2 and TS18, and so on.

l When E1 interface is used as CE1 interface having signaling channel, E1 interface will be used as digital E&M interface if digital E&M signaling is adopted. The time slot dividing and functions of various time slots are completely same as those when R2 signaling is adopted.

& Note:

E1 voice modular (E1VI) has the basic features of common E1 interface. For example, after creating pri-sets, the system will automatically generate corresponding logical serial interfaces and perform command configuration with synchronous serial interface mode. In addition, if a TS set is created on the E1 voice interface to configure R2 signaling, the system will automatically generate a voice subscriber-line corresponding to this TS set. If a PRI set is created to configure DSS1 signaling, the system will automatically generate the voice subscriber-line corresponding to this PRI set. So, the E1 voice interface has the features of both serial interface and voice subscriber-line.

3.1.3 Features of E1 Voice

E1 voice not only supports basic VoIP function but also has the following characteristics:

I. Signaling mode supported

E1 as the ISDN PRI interface supports DSS1 and QSIG user signaling, E1 voice interface also supports R2 signaling and digital E&M signaling.

l DSS1 user signaling is adopted on the D channel between ISDN user and network interface (UNI) and is composed of data link layer protocol and third layer protocol used for basic call control. It mainly controls the basic call of frame mode and switch virtual connection on UNI interface.

l QSIG is an international signaling standard issued by ISO, IEC and JTC1 for dedicated ISDN telecommunication networks. The signaling mechanism used in QSIG supplementary services is very similar to that used in ISDN DSS1 and ISUP protocols.

l R2 signaling (similar to China signaling No.1) follows ITU-T specification, including digital line signaling and interregister signaling. Digital line signaling is transmitted in the TS16 (ABCD bit) time slot of E1 trunk and it is mainly used to monitor the occupation, release and block status of trunk line. Interregister signaling transmits address information, international call voice bit and authentication bit, echo cancellation suppress information, call number attribute and called number attribute with Multiple Frequency Control (forward and backward) in every time slot.

l The transmission mode of digital E&M signaling is similar to that of R2 signaling. In TS16 time slot of E1 trunk, the call control signal similar to E (recEive) and M (transMit) of analog E&M signaling. TS0 transmits synchronous signal and other time slots transmit voice signal. The digital E&M signaling checks and sends connection signaling through checking the signal change of TS16 time slot of E1 trunk. The digital E&M signaling provides immediate, wink and delay start modes to adapt to the application of devices with different response time and start modes, so it can establish connection more reliably.

II. Fax function

FXS and FXO interfaces support IP fax, and E1 voice interface also supports fax, allowing establishment of fax channel and transmission/reception of fax data.

III. Protocols and standards supported

E1 voice supports protocols in ITU-T H.323 framework, and ITU-T G.711, G.729, and the 5.3K and 6.3K compression algorithm in Annex A of G.723.1

3.2 E1 Voice R2 Signaling Configuration

E1 voice R2 signaling configuration includes:

l Create TS set

l Configure voice subscriber line corresponding to TS set

l Configure POTS voice entity

l Configure VoIP voice entity

l Configure basic parameters of E1 interface

l Configure relevant parameter of R2 signaling

l Maintain MFC channel and circuit

3.2.1 Creating TS Set

TS set means the logical voice subscriber-line abstracted through defining time slot list on actual E1 interface, and it serves for voice transmission. All time slots contained in this TS set are used to transmit voice. Only one TS set can be defined on one E1 interface. It is convenient to configure signaling for E1 line through configuring signaling type for various TS sets and relevant parameters for various signaling.

After successfully configuring TS set, the system will generate voice subscriber-line corresponding to this TS set according to current E1 interface number and TS set number. The voice subscriber-line is “E1 interface number: TS set number”.

In system view, use controller e1 command to configure and use other commands to configure in CE1/PRI interface view.

Table 3-1 Create TS set (R2 signaling type)

Operation	Command
Enter CE1/PRI interface view	controller e1 e1-number
Create TS set of R2 type	timeslot-set ts-set-number timeslots-list signal r2
Delete specified TS set	undo timeslot-set set-number

By default, TS set is not created.

3.2.2 Configuring Voice Subscriber Line Corresponding to TS set

Voice subscriber-line corresponding to TS set configuration includes:

l Enter voice subscriber-line view

l Configure description information of voice subscriber-line

l Enable/disable voice subscriber-line

l Enable comfort noise function

l Configure automatic ringing function of leased line

l Configure dial-back cancellation function

l Configure voice input gain and output gain

l Configure sensitivity level of checking DTMF code

& Note:

Most of above configuration are same as configuring voice subscriber-line in last chapter “VoIP Configuration”. Refer to the steps in last chapter. The following only introduces the particular configuration of voice subscriber-line corresponding to TS set.

3.2.3 Entering Voice Subscriber-Line View

Perform the following configuration in voice view.

Table 3-2 Enter voice subscriber-line view

Operation	Command
Enter voice subscriber-line view	subscriber-line e1-number :ts-set-number

3.2.4 Configuring the Sensitivity Level of DTMF Code Detection

Perform the following configuration in digital voice subscriber-line view.

Table 3-3 Configure the Sensitivity Level of DTMF Code Detection

Operation	Command
Configure the sensitivity level of DTMF code detection	dtmf threshold { 0 \| 1 }
Restore the default sensitivity level of DTMF code detection	undo dtmf threshold

3.2.5 Configuring POTS Voice Entity

The detailed configuration steps are same as configuring POTS voice entity in last chapter “VoIP Configuration”. The difference only lies in the following description.

Perform the following configuration in POTS voice entity view.

Table 3-4 Configure POTS voice entity

Operation	Command
Configure POTS voice entity to correspond to logical voice line of TS set	line e1-number :ts-set-number
Cancel the corresponding of POTS voice entity to logical subscriber-line	undo line

3.2.6 Configuring VoIP Voice Entity

The detailed configuration steps are same as configuring VoIP voice entity in last chapter “VoIP Configuration”. Refer to that.

3.2.7 Configuring Basic Parameter of E1 Interface

To synchronize the communication of devices at both ends of E1 trunk, E1 clock source will be configured to the devices at both ends. Now, there are two methods to select clock: generating clock through self or extracting clock from line.

In system view, use controller e1 and interface serial command to perform configuration and use other commands to perform configuration in CE1/PRI interface view.

Table 3-5 Configure basic parameter of E1 interface

Operation	Command
Enter CE1/PRI interface view	controller e1 e1-number
Configure work mode of CE1/PRI interface	using { e1 \| ce1 }
Configure clock source of CE1/PRI interface	clock { master \| slave }
Configure frame format of CE1/PRI interface	frame-format { crc4 \| no-crc4 }
Configure line code format of CE1/PRI interface	code { ami \| hdb3 }
Start loopback/dial-back	loopback { local \| remote }

By default, the work mode of CE1/PRI interface adopts interface channelized mode (ce1), clock source adopts line clock (slave), frame format is non-CRC4 frame (no-crc4), line code format is HDB3 format (hdb3) and loopback/dial-back is forbidden (undo loopback).

3.2.8 Configuring Parameters of R2 Signaling

I. Introduction to R2 signaling

ITU-T T.400 – T.490 Series protocols define the standards of R2 signaling. However, R2 signaling is implemented according to different standards in different countries or regions. R2 signaling in different countries is the variant of ITU-T, and China Signaling No.1 is one of the subset of R2 signaling.

R2 signaling is divided into digital line signaling and interregister signaling. Digital line signaling mainly functions to monitor seized, released, and blocked statuses of the trunk. Interregister signaling adopts Multiple Frequency Control (MFC) mode to transmit information like addresses. Normally, the calling side works as the originating point, and the called side works as the terminating point. The signals sent from the originating point to the terminating point are called forward signals, and the reverse signals are called backward signals, as shown in the following figure:

Figure 3-2 Relevant elements of R2 signaling

1) ITU-T digital line signaling

Digital line signaling is responsible for changing call statuses and conditions of a line. It mainly functions to identify and detect these four statuses: calling party picks up the phone and seizes the line, called party picks up the phone and answers the call, calling party releases the call, and called party releases the call. Accordingly, it sets the line to be idle or seized. This signaling is transmitted in the 16th multiframe time slot of PCM system. The two transmission directions of each line respectively have four bits (A, B, C and D) as flag bit, while C and D bits are fixed: 01 (C and D bits of China Signaling No.1 are 11). Therefore, the forward line signaling adopts a_f and b_f bits and the backward line signaling adopts a_b and b_b bits. Their meanings are shown in the following table:

Table 3-6 Signal bit meanings of line signaling

	Meaning	Vale=0	Value=1
a_f	Identify working state of device at the originating point and indicate state of the calling party line	Off-hook, seized	On-hook (idle)
b_f	Indicate fault state from the originating point to the terminating point	Normal	Faulty
a_b	Indicate state of the called party line (on-hook or off-hook)	Off-hook by called party	On-hook by called party
b_b	Indicate state of device at the terminating point (idle or seized)	Idle	Seized or blocked

Table 3-7 State code of line signaling

Sate of the Circuit	Signaling Code
	Forward		Backward
	a_f	b_f	a_b	b_b
Idle	1	0	1	0
Seized	0	0	1	0
Seizure-ack	0	0	1	1
Answer	0	0	0	1
Clear-back	0	0	1	1
Clear-forward	1	0	0/1	1
Blocked	1	0	1	1
Unblocked	1	0	1	0

Description of typical R2 digital line signaling exchange

l Call establishment: When the trunk circuit is idle, the originating point sends a forward seizure signal to the terminating point, which sends back a seizure-ack signal after it recognizes the seizure signal. At this time, the circuits of the both sides are seized, and they begin interregister signaling exchange. When the called party picks up the phone and answers the call, the terminating point should send a backward answer signal. The call is established successfully after the originating point recognizes the backward answer signal.

Figure 3-3 R2 digital line signaling – call establishment

l Originating point releases the call: The originating point sends a clear-forward signal 10. When the terminating point recognizes the clear-forward signal, it sends a backward signal 10 (release guard signal or clear-forward acknowledgement signal). When the originating point recognizes the backward signal 10, the corresponding trunk circuit is released.

Figure 3-4 R2 digital line signaling – originating point releases the call

l Terminating point releases the call: The terminating point sends a clear-back signal 11. When the originating point receives the clear-back signal, it sends a clear-forward signal 10. When the terminating point recognizes the forward signal 10 sent by the originating point, it sends a backward signal 10. When the originating point recognizes the backward signal 10, the corresponding trunk circuit is released.

Figure 3-5 R2 digital line signaling – terminating point releases the call

l Line released by forced release signal: If the terminating point supports metering signal, the system can use the forced release signal 00 to replace the clear-back signal 11, so as to avoid collision of metering signal and clear-back signal, which is sent when the called party releases the call.

l Blocking in idle state or during conversation: If the originating point receives the blocking signal 11 from the terminating point when the trunk circuit is idle or during conversation, it sends the forward signal 10. At this time, the corresponding trunk circuit is blocked. When the terminating point unblocks the trunk circuit, it sends the backward signal 10 in the corresponding line to indicate that the line is idle. The originating point should maintain the forward signal 10 and unblock the corresponding trunk circuit for a next call connection.

l Troubleshooting originating point in idle state: If the terminating point receives the forward signal 11 from the originating point to indicate device fault when the trunk circuit is idle, the terminating point sends the backward signal 11. Then, the trunk circuit is in faulty state. When the device recovers, the originating point sends the forward signal 10, and the terminating point responds with the signal 10. At this time the trunk circuit regains normal state.

l Troubleshooting originating point during conversation: If the terminating point receives the forward signal 11 from the originating point to indicate device fault during conversation, the terminating point releases the line backward. At the same time, it sends the backward signal 11. Then, the trunk circuit is in faulty state. When the device recovers, the originating point sends the forward signal 10, and the terminating point sends back the signal 10. At this time, the trunk circuit regains normal state.

2) ITU-T interregister signaling

Interregister signaling mainly functions to control automatic connection of the circuit. It adopts MFC mode and is divided into forward signaling and backward signaling. Forward signaling exchange is divided into Group I and Group II, while backward signaling exchange is divided into Group A and Group B. When the originating point recognizes the seizure-ack signal, the register begins to send the first digit of the called number, and waits for the response of Group A signaling from the terminating point.

l Group I forward signaling: It is made up of connection control signaling and digital signaling.

Table 3-8 Forward Group I signaling

Signal	Basic Meaning
I-1 -- I-10	Digital signaling, corresponds to 1, 2, 3, 4, 5, 6, 7, 8, 9, 0 in turn, responsible for sending number information to the terminating point
I-11	Reserved
I-12	Request refused
I-13	Connected to test device
I-14	Reserved
I-15	Address identification terminator and pulse terminator (used in international call)

l Group A backward signaling: Control signaling of Group I forward signaling, which controls and acknowledges Group I signaling.

Table 3-9 Group A backward signaling

Signal	Basic Meaning
A-1	Digit control signal, which requests a next digit.
A-2	Digit control signal, which requests retransmission starting from the last digit
A-3	Digits of the number are received completely and transferred to the exchange process of Group II forward signaling and Group B backward signaling.
A-4	Congestion on internal network (sent from the internal switch), which terminates interregister signaling exchange.
A-5	Requests calling party information.
A-6	Digits of the number are received completely, which terminates interregister signaling exchange. Charging and conversation begin.
A-7	Digit control signal, which requests retransmission starting from the last second digit.
A-8	Digit control signal, which requests for retransmission starting from the last third digit.
A-9	Reserved.
A-10	Reserved.
A-11	Requests country code flag.
A-12	Requests language bit or identification bit.
A-13	Requests circuit category.
A-14	Requests echo canceller information.
A-15	Congestion on international network (sent from the foreign exchange office), which terminates interregister signaling exchange.

l Group II forward signaling: calling category. According to different calling categories, the system determines whether forced release and break-in can be implemented on or by the calling party.

Table 3-10 Group II forward signaling

Signal	Basic Meaning
II-1	Subscriber without priority
II-2	Subscriber with priority
II-3	Maintenance device
II-4	Reserved
II-5	Operator
II-6	Data transmission
II-7	Global use: calling party does not support forward transfer
II-8	Global use: data transmission
II-9	Global use: calling party is the subscriber with priority
II-10	Global use: used in international aid; calling party supports forward transfer
II-11 -- II-15	Reserved

l Group B backward signaling: state of the called party, acknowledges Group II signaling and controls connection.

Table 3-11 Group B backward signaling

Signal	Basic Meaning
B-1	Reserved
B-2	Requests special tone to the calling party
B-3	Subscriber line busy
B-4	Congestion
B-5	Unallocated number
B-6	Subscriber idle, charging
B-7	Subscriber idle, no charging
B-8	Subscriber line faulty
B-9 -- B-15	Reserved

Description of typical R2 interregister signaling exchange. The following figure shows the exchange process requesting calling party information.

Figure 3-6 Interregister signaling exchange process (R2)

Configuration tasks R2 signaling parameters include (all optional):

l Access R2 CAS view

l Configure country and region mode

l Configure and maintain trunk circuit

l Configure R2 digital line signaling

l Configure R2 interregister signaling

l Configure DTMF signal

l Configure signal tones

II. Accessing R2 CAS view

Perform the following configuration in CE1 interface view.

Table 3-12 Enter R2 CAS view

Operation	Command
Enter R2 CAS view	cas ts-set-number

III. Configuring country or region mode

The implementation and parameter values of R2 signaling vary in different countries. To make it possible for your VG to exchange R2 signaling with devices in other countries and regions, you need to adjust the country or region mode. The system automatically selects the appropriate subscriber-line status, service type, metering signal, and values of C and D signal bits. Currently, the modes of Brazil, Mexico, Argentina, India, New Zealand, Thailand, Bengal, Hong Kong, Indonesia, and the countries and regions that comply with ITU-T standard are supported.

Perform the following configuration in R2 CAS view.

Table 3-13 Configure country or region mode

Operation

Command

Configure country or region mode

mode zone-name { default-standard | custom }

mode custom

By default, the parameter zone-name is set to itu-t, that is, ITU-T mode is adopted.

IV. Configuring and maintaining trunk circuit

The circuit is used to carry incoming and outgoing calls. You can unblock, block, query, and reset the trunk circuit in the specified time slot. Blocking and unblocking are reverse processes.

Perform the following configuration in R2 CAS view.

Table 3-14 Configure and maintaining trunk circuit

Operation	Command
Set E1 trunk direction	trunk-direction timeslots timeslot-list { in \| out \| dual }
Restore the default value	undo trunk-direction timeslots timeslot-list
Set E1 trunk routing mode	select-mode [ max \| maxpoll \| min \| minpoll ]
Maintain the trunk circuit in the specified time slot	ts { block \| open \| query \| reset } timeslots timeslots-list

By default, dual direction is adopted in E1 trunk, and minimum routing mode is adopted in E1 trunk routing.

V. Configuring R2 digital line signaling

1) Enable/disable R2 digital line signaling exchange

Perform the following configuration in R2 CAS view.

Table 3-15 Enable/disable R2 digital line signaling exchange

Operation	Command
Specify whether the originating point requests the peer device to send answer signal	answer { enable \| disable }
Enable/disable the originating point to process re-answer signal	re-answer { enable \| disable }
Enable/disable the terminating point to send clear-forward acknowledgement signal (clear-back signal)	clear-forward-ack { enable \| disable }
Enable/disable metering signal of R2 signaling	force-metering { enable \| disable }
Specify whether the originating point requests the terminating point to send seizure-ack signal	seizure-ack { enable \| disable }

By default, the originating point requests the peer device to send answer and seizure-ack signals, does not process re-answer or clear-forward acknowledgement signals, and disables metering signal of R2 signaling.

2) Configure signal value of R2 digital line signaling

Perform the following configuration in R2 CAS view.

Table 3-16 Configure signal value of R2 digital line signaling

Operation	Command
Configure values of ABCD bits of all line signals	dl-bits { answer \| blocking \| clear-back \| clear-forward \| idle \| seizure \| seizure-ack \| release-guard } rx-bits ABCD tx-bits ABCD
Restore the default values of ABCD bits of all line signals	undo dl-bits { answer \| blocking \| clear-back \| clear-forward \| idle \| seizure \| seizure-ack \| release-guard }
Configure values of C and D signals	renew A-bit B-bit C-bit D-bit
Restore the default values	undo renew
Configure reverse mode of line signals	reverse A-bit B-bit C-bit D-bit
Restore the default values	undo reverse

By default, the reverse mode of line signal is 0 0 0 0, that is to say, reverse conversion function is disabled. C and D signals are set to 1 1 1 1 in transmission. The following table shows the default values of ABCD bits of line signals.

Table 3-17 Default values of signals of R2 digital line signaling

Signal	rx-bits ABCDdefault value	tx-bits ABCDdefault value
Answer	0101	0101
Blocking	1101	1101
Clear-back	1101	1101
Clear-forward	1001	1001
Idle	1001	1001
Seize	0001	0001
Seizure-ack	1101	1101
Release-guard	1001	1001

3) Configure time parameters of R2 digital line signaling

Perform the following configuration in R2 CAS view.

Table 3-18 Configure time parameters of R2 digital line signaling

Operation	Command
Set effect-time of digital line signaling	effect-time number
Restore the default value	undo effect-time
Set timeout value of line signals of R2 digital line signaling	timer dl { answer \| clear-back \| clear-forward \| seize \| re-answer \| release-guard } value
Restore the default value	undo timer dl { answer \| clear-back \| clear-forward \| seize \| re-answer \| release-guard }

By default, the effect-time of digital line signaling is 10 ms. In R2 digital line signaling, the timeout values of answer signal, clear-back signal, clear-forward signal, seizure signal, re-answer signal, release guard signal are 60,000 ms, 10,000 ms, 10,000 ms, 1,000 ms, 1,000 ms and 10,000 ms respectively.

VI. Configuring R2 interregister signaling

1) Enable/disable R2 interregister signaling exchange

Perform the following configuration in R2 CAS view.

Table 3-19 Enable/disable R2 interregister signaling exchange

Operation	Command
Enable to request calling number from the peer end	ani
Disable to request calling number from the peer end	undo ani
Configure how many digits to be collected to receive calling party flag	ani-offset number
Restore the default value	undo ani-offset
Enable/disable to use Group B signal to complete interregister signaling exchange	group-b { enable \| disable }
Enable/disable to send number terminator to the terminating point	finale-callednum { enable \| disable }
Restore the default value	undo req-category-offset

By default, calling number request is disabled; 1 digit is requested to receive calling party flag; Group B signal is enabled to complete interregister signaling exchange; called number terminator is disabled; calling category is required when receiving the first digit of the called number.

2) Configure signal values of R2 interregister signaling

Perform the following configuration in R2 CAS view.

Table 3-20 Configure signal values of R2 interregister signaling

Operation	Command
Configure the special characters that are supported during interregister signaling exchange	special-character character value
Configure register signal value of R2 signaling	register-value { billingcategory \| callcreate-in-groupa \| callingcategory \| congestion \| demand-refused \| digit-end \| nullnum \| req-billingcategory \| req-callingcategory \| req-currentdigit \| req-firstcallingnum \| req-firstdigit \| req-nextcallednum \| req-nextcallingnum \| req-lastfirstdigit \| req-lastseconddigit \| req-lastthirddigit \| req-switch-groupb \| subscriber-busy \| subscriber-idle \| subscriber-idle-nocharge \| req-firstcallednum-groupc \| req-currentcallednum-groupc \| req-callednum-switchgroupa } value
Configure the terminating point whether to send answer signal of the calling number	respond-reqcallernum
Restore the default value	undo register-value { all \| billingcategory \| callcreate-in-groupa \| callingcategory \| congestion \| demand-refused \| digit-end \| nullnum \| req-billingcategory \| req-callingcategory \| req-currentdigit \| req-firstcallingnum \| req-firstdigit \| req-nextcallednum \| req-nextcallingnum \| req-lastfirstdigit \| req-lastseconddigit \| req-lastthirddigit \| req-switch-groupb \| subscriber-busy \| subscriber-idle \| subscriber-idle-nocharge \| respond-req-callernum \| req-firstcallednum-groupc \| req-currentcallednum-groupc \| req-callednum-switchgroupa }

By default, no special character is configured. By default, in R2 interregister signaling, billingcategory signal is set to 1, callcreate-in-groupa signal to 6, callingcategory signal 3, congestion signal 4, demand-refused signal 12, digit-end signal 15, nullnum signal 5, req-billingcategory signal 6, req-callingcategory signal 3, req-currentdigit signal 16, req-firstcallingnum signal 1, req-firstdigit signal 2, req-nextcallednum signal 1, req-nextcallingnum signal 1, req-lastfirstdigit signal 16, req-lastseconddigit signal 16, req-lastthirddigit signal 16, req-switch-groupb signal 3, subscriber-busy signal 3, and subscriber-idle signal 6, subscriber-idle-nocharge signal 7, respond-req-callernum signal req-firstcallednum-groupc signal 16, req-currentcallednum-groupc signal 16, and req-callednum-switchgroupa signal 1.

3) Configure time parameters of R2 interregister signaling

Perform the following configuration in R2 CAS view.

Table 3-21 Configure time parameters of R2 interregister signaling

Operation	Command
Configure the persistence time and identification time of register pulse signals (A3, A4, and A6, etc.) of R2 signaling	timer register-pulse persistence time-value
Restore the default values	undo timer register-pulse
Configure the timeout value of register signals of R2 signaling	timer register-complete { group-b } value
Restore the default values	undo timer register-complete { group-b }

By default, the persistence time of register pulse signals of R2 signaling is 150±30 ms, and the timeout value in which the terminating point waits for Group B signal is 30,000 ms.

4) Manage MFC channels

MFC channels are used to carry R2 interregister signaling. You can open, block, and query MFC channels in the specified time slot. Blocking and unblocking are reverse processes.

Perform the following configuration in R2 CAS view.

Table 3-22 Manage MFC channels

Operation	Command
Maintain MFC channels in the specified time slot	mfc { block \| open \| query } timeslots timeslots-list

By default, all MFC channels are in idle state.

VII. Configuring DTMF signal

1) Configure R2 signaling reception and transmission modes

At present, R2 signaling reception and transmission modes include: Multiple Frequency Control (MFC) and Dual Tone Multi-Frequency (DTMF).

Perform the following configuration in R2 CAS view.

Table 3-23 Configure number reception and transmission modes of R2 signaling

Operation	Command
Enable/disable to use DTMF mode to receive and send R2 signaling	dtmf { enable \| disable }

By default, DTMF mode is not used to receive and send R2 signaling, that is to say, MFC mode is used.

2) Configure the time interval in which DTMF signal is sent

At present, R2 signaling reception and transmission modes include: MFC and DTMF.

Perform the following configuration in R2 CAS view.

Table 3-24 Configure the time interval in which DTMF signal is sent

Operation	Command
Configure the time interval in which DTMF signal is sent	timer dtmf time-value
Restore the default value	undo timer dtmf

By default, the time interval in which R2 signaling sends DTMF signal is 0 ms.

VIII. Configuring signal tones

1) Configure the called side to send ring-back tone or ring-busy tone to the caller

Perform the following configuration in R2 CAS view.

Table 3-25 Configure the called side to send ring-back tone or ring-busy tone to the caller

Operation	Command
Enable/disable the called side to send ring-back tone or ring-busy tone to the caller	sendring { ring-back \| ring-busy } { enable \| disable }

By default, a ring-busy tone, rather than a ring-back tone, is sent.

Caution:

The PBXs as the terminating point (the calling side) in some places may not send back the ring-back tone to the caller during the call process if no fast connection is used (the peer does not send back the ring-back tone signal). To avoid call failure due to that the caller does not hear the corresponding signal tone, the sendring command can be manually configured to send the ring-back tone signal to the determining point.

2) Configure the timeout value of signal tones

Perform the following configuration in R2 CAS view.

Table 3-26 Configure the timeout value of signal tones

Operation	Command
Configure the timeout value of signal tones	timer ring { ringback \| ringbusy } value
Restore the default value	undo timer ring

By default, the timeout value of ring-back tone is 60,000 ms and that of ring-busy tone is 30,000 ms.

IX. Configuring the R2 call connection mode

Perform the following configuration in R2 signaling view.

Table 3-27 Configure the R2 call connection mode

Operation	Command
Configure the R2 call connection mode	callmode { segment \| terminal }
Restore the default setting of the R2 call connection mode	undo callmode

By default, the “terminal” mode is used.

3.3 E1 Voice Digital E&M Signaling Configuration

E1 voice Digital E&M signaling configuration includes the following items (all optional):

l Create TS set

l Configure voice subscriber line corresponding to TS set

l Configure POTS voice entity

l Configure VoIP voice entity

l Configure basic parameters of E1 interface

l Configure relevant parameters of digital E&M signaling

3.3.1 Creating TS set

Same as R2 signaling, digital E&M signaling also needs the help of TS set to configure relevant parameters of digital E&M signaling. After successfully configuring TS set, the system will generate a voice subscriber-line corresponding to the TS set according to the current E1 interface number and TS set number. The voice subscriber-line number is “E1 interface number:TS set number”. One E1 interface (digital E&M interface) can only define one TS set.

Similar to analog E&M signaling, digital E&M signaling also provides three kinds of start mode: immediate start, wink start and delay start. The digital E&M signaling time sequence of each state mode is consistent with that of analog E&M signaling completely. The only difference is that analog E&M signaling transmits signaling information through the level change of Tip and Ring line but digital E&M signaling adopts 4 bits of TS16 time slot to transmit signaling information (similar to R2 signaling).

In system view, use the controller e1 command to perform configuration, and use other commands in CE1/PRI interface view.

Table 3-28 Create TS set (start mode)

Operation	Command
Enter CE1/PRI interface view	controller e1 e1-number
Establish TS set of digital E&M type and select start mode	timeslot-set ts-set-number timeslots-list signal { e&m-delay \| e&m-immediate \| e&m-wink }
Delete specified TS set of digital E&M type	undo timeslot-set ts-set-number

By default, the new established TS set of digital E&M type adopts immediate start mode, that is to say, selecting the parameter e&m-immediate.

3.3.2 Configuring Voice Subscriber Line Corresponding to TS set

The detailed configuration steps are the same as those described in “3.2.2 Configuring Voice Subscriber Line Corresponding to TS set” in the previous section “E1 Voice R2 Signaling Configuration”.

3.3.3 Configuring POTS Voice Entity

The detailed configuration steps are the same as those described in section 3.2.5 Configuring POTS Voice Entity” in section 3.2 “E1 Voice R2 Signaling Configuration”.

3.3.4 Configuring VoIP Voice Entity

The detailed configuration steps are the same as those described in section 3.2.6 Configuring VoIP Voice Entity” in section 3.2 “E1 Voice R2 Signaling Configuration”.

3.3.5 Configuring Basic Parameter of E1 Interface

The detailed configuration steps are the same as those described in section 3.2.7 Configuring Basic Parameter of E1 Interface” in section 3.2 “E1 Voice R2 Signaling Configuration”.

3.3.6 Configuring Relevant Parameter of Digital E&M Signaling

I. Configuring timeout parameter of digital E&M signaling

To improve the interoperability of digital E&M signaling between the voice gateway and the peer device (e.g., PBX) and enhance communication performance, you can adjust time parameters in delay and wink start mode of digital E&M signal, and adjust DTMF signal width and the interval for sending signal.

Perform the following configuration in digital E&M voice subscriber-line view.

Table 3-29 Configure timeout parameter of digital E&M signaling

Operation	Command
Configure waiting time of call clearing	delay call-interval milliseconds
Configure maximum hold duration of delay signal in delay start mode	delay hold milliseconds
Configure waiting time of sending delay signal in delay start mode	delay rising milliseconds
Configure delay time of sending called number in immediate start mode	delay send-dtmf milliseconds
Configure hold time of sending DTMF number	delay dtmf milliseconds
Configure interval for sending DTMF number	delay dtmf-interval milliseconds
Configure maximum time of waiting for wink signal after calling side sends seizure signal in wink start mode	delay wink-rising milliseconds
Configure maximum hold duration of sending wink signal by called side in wink start mode	delay wink-hold milliseconds
Configure maximum delay time of sending wink signal by called side in wink start mode	delay send-wink milliseconds
Restore corresponding parameters of digital E&M signaling to default value	undo delay { call-interval \| hold \| rising \| send-dtmf \| dtmf \| dtmf-interval \| wink-rising \| wink-hold \| send-wink }

By default, the waiting time of call clearing (call-interval) is 200 ms, the maximum hold duration of delay signal (hold) is 400 ms, the delay time of checking called line status (rising) is 300 ms, the delay time of sending called number (send-dtmf) is 300ms, the hold duration of sending DTMF number (dtmf) is 120 ms, the interval for sending DTMF number (dtmf-interval) is 120 ms, the maximum time of waiting for wink signal (wink-rising) is 2000 ms, the maximum hold duration of wink signal (wink-hold) is 500 ms and the maximum delay time before sending wink signal (send-wink) is 200 ms.

II. Configuring timeout of digital E&M signaling call connection signal

You can configure the timeout that calling end waits for called ringback response on the digital E&M voice subscriber-line. If called response signal is not received after specifying time, then the called end is regarded as not being connected. You also can configure the timeout that called end waits for calling end to send called number. If called end does not receive called number within specified time, the call connection fails.

Perform the following configuration in digital E&M voice subscriber-line view.

Table 3-30 Configure timeout of digital E&M signaling call connection signal

Operation	Command
Configure the maximum dial-interval	timer dial-interval seconds
Configure timeout for calling end to wait for called ringback response	timer ring-back { seconds \| infinity }
Configure timeout for called end to wait for calling end to send called number	timer wait-digit { seconds \| infinity }
Restore the timeout of digital E&M signal call connection signal to default value	undo timer { dial-interval \| ring-back \| wait-digit }

By default, the maximum dial-interval (dial-interval) is 4s, the timeout of waiting for called ringback response (ring-back) is 60s and the timeout of waiting for calling end to send called number (wait-digit) is 5s.

III. Configuring digital E&M signaling ABCD bit value

To make it possible for the calling and called PBXs (or voice gateways) to communicate using the digital E&M signaling, the two sides must reach agreement on bit values of receiving/transmitting idle signaling and receiving/transmitting seizure signaling. Therefore you need to define the bit values of these signals.

Perform the following configuration in digital E&M voice subscriber-line view.

Table 3-31 Configure digital E&M signaling bit value

Operation	Command
Configure ABCD bit value of digital E&M signaling to receive idle signaling	signal-value received idle A-bit B-bit C-bit D-bit
Configure ABCD bit value of digital E&M signaling to receive seizure signaling	signal-value received seize A-bit B-bit C-bit D-bit
Configure ABCD bit value of digital E&M signaling to send idle signaling	signal-value transmit idle A-bit B-bit C-bit D-bit
Configure ABCD bit value of digital E&M signaling to send seizure signaling	signal-value transmit seize A-bit B-bit C-bit D-bit
Restore corresponding signaling bit value of digital E&M signaling to default value	undo signal-value { received idle \| received seize \| transmit idle \| transmit seize }

By default, all the ABCD bit values of digital E&M signaling for receiving/transmitting idle signaling are 1101, and those for receiving/transmitting seizure signaling are 0101.

IV. Configuring and maintain trunk circuit

Trunk circuit is configured to carry specific incoming and outgoing calls. You can also query the trunk circuit with the specified timeslot.

Perform the following configuration in digital E&M signaling view.

Table 3-32 Configure and maintain trunk circuit

Operation	Command
Set E1 trunk routing mode	select-mode [ max \| maxpoll \| min \| minpoll ]
Query the trunk circuit with the specified timeslot	ts query timeslots timeslots-list

By default, E1 trunk routing mode adopts the minimum routing (min).

V. Configuring whether the VG as the originating point sends the dial tone to the terminating point

As the originating point, the VG sends the dial tone to the terminating point when preparing for receiving numbers, prompting the caller to dial the number.

Perform the following configuration in digital E&M signaling view.

Table 3-33 Configure whether the VG as the originating point sends the dial tone to the terminating point

Operation	Command
Configure the VG as the terminating point to send the dial tone to the originating point	dialtone-generate
Configure the VG as the terminating point not to send the dial tone to the originating point	undo dialtone-generate

By default, the VG as the terminating point does not send the dial tone to the originating point.

3.4 E1 Voice DSS1 and QSIG Subscriber Signaling Configuration

& Note:

For the detailed configuration of the ISDN part, refer to Chapter 4 “Configuring ISDN Protocol”.

E1 voice DSS1 and QSIG subscriber signaling configuration includes:

l Create PRI Set

l Configure the voice subscriber line corresponding to PRI set

l Configure POTS voice entity

l Configure VoIP voice entity

l Configure basic parameters of PRI interface

3.4.1 Creating PRI Set

When DSS1 or QSIG subscriber signaling of PRI interface is adopted between the voice gateway and switch, only voice transmission is supported.

After successfully configuring the PRI set, the system will generate a voice subscribe-line corresponding to this PRI set according to the number of E1 interface where current interface lies in. The voice subscriber-line number is “E1 interface number: 15”.

On the other hand, the logical serial interface corresponding to PRI set is generated according to current serial interface number. All logical serial interfaces and physical serial interfaces are numbered uniformly (i.e., logical serial interfaces are numbered from the maximum physical serial interface plus 1). The new logical serial interface is numbered as “(number of serial interface where PRI set lies + total physical serial interfaces):15”.

& Note:

The logic serial interfaces generated by E1 interface are numbered uniformly, so the total number of new serial interfaces will be attached the total logic serial interfaces generated by E1 interface.

I. Configuring basic parameters of PRI interface on CE1/PRI interface

In system view, use controller e1 command to perform configuration and use other commands to perform in CE1/PRI interface view.

Table 3-34 Configure basic parameters of PRI interface on CE1/PRI interface

Operation	Command
Enter CE1/PRI interface view	controller e1 e1-number
Create PRI set	pri-set [ timeslots-list timeslots-list ]
Create PRI set	pri-set [ timeslots-list timeslots-list ]
Configure work mode of CE1/PRI interface	using { e1 \| ce1 }
Configure clock source of CE1/PRI interface	clock { master \| slave }
Configure frame format of CE1/PRI interface	frame-format { crc4 \| no-crc4 }
Configure line code format of CE1/PRI interface	code { ami \| hdb3 }
Start loopback/dial-back of CE1/PRI interface	loopback
Enter serial interface corresponding to PRI set	interface serial serial-number :15

Be default, the work mode of CE1/PRI interface adopts interface channelized mode (ce1), the clock source adopts line clock (slave), frame format is non-CRC4 frame (no-crc4), line code format is HDB3 format (hdb3) and loopback/dial-back is forbidden (undo loopback).

3.4.2 Configuring the Voice Subscriber Line Corresponding to PRI Set

Before entering voice subscriber-line (PRI interface) view, you must establish a PRI set first. Then the system will automatically establish a voice subscriber-line corresponding to this PRI set.

Voice subscriber-line corresponding to PRI set configuration includes the following optional items:

l Enter voice subscriber-line view

l Configure description information of voice subscriber-line

l Enable/disable voice subscriber-line

l Enable comfort noise function

l Configure automatic ring function of leased line

l Configure dial-back cancellation function and echo hold duration

l Configure voice input gain and output gain

l Configure sensitivity level of checking DTMF code

& Note:

Most of above configurations are the same as “3.2.2 Configuring Voice Subscriber Line Corresponding to TS set. The following subsection only introduces the unique configuration of voice subscriber-line corresponding to PRI set.

3.4.3 Entering voice subscriber-line view

After successfully configuring a PRI set, the system will generate a voice subscribe-line corresponding to this PRI set according to the current E1 interface number. The voice subscriber-line number is “E1 interface number:15”.

Perform the following configuration in any voice view except voice entity view.

Table 3-35 Enter voice subscriber-line view

Operation	Command
Enter voice subscriber-line view	subscriber-line e1-number :15

3.4.4 Configuring POTS Voice Entity

The detailed configuration steps are similar to those described in section 3.2.5 Configuring POTS Voice Entity” in section 3.2 “E1 Voice R2 Signaling Configuration”. The following table lists their difference.

Perform the following configuration in POTS voice entity view.

Table 3-36 Configure POTS voice entity

Operation	Command
Associate a POTS voice entity with the logical voice subscriber-line of PRI set	line e1-number :15
Cancel the association between a POTS voice entity and a logical subscriber-line	undo line

3.4.5 Configuring VoIP Voice Entity

The detailed configuration steps are same as those described in 3.2.6 Configuring VoIP Voice Entity.

3.4.6 Configuring Basic Parameters of PRI Interface

I. Configuring CE1/PRI interface

Perform the following configurations in CE1/PRI interface view.

Table 3-37 Configure basic parameters of PRI interface on CE1/PRI interface

Operation	Command
Enter CE1/PRI interface view	controller e1 e1-number
Create PRI set	pri-set [ timeslots-list timeslots-list ]
Configure work mode of CE1/PRI interface	using { e1 \| ce1 }
Configure clock source of CE1/PRI interface	clock { master \| slave }
Configure frame format of CE1/PRI interface	frame-format { crc4 \| no-crc4 }
Configure line code format of CE1/PRI interface	code { ami \| hdb3 }
Start loopback/dial-back of CE1/PRI interface	loopback
Enter serial interface corresponding to PRI set	interface serial serial-number :15

3.5 Displaying and Debugging E1 Voice

3.5.1 Displaying and Debugging E1 Voice

After the above configuration, execute the display command in any view to display the running status of the E1 voice and to verify the effect of the configuration.

Execute the following command in any view.

Table 3-38 Display and debug E1 voice

Operation	Command
Reset the call statistics of R2 signaling	reset voice r2
Display features of CE1/PRI interface	display controller e1 e1-number
Display call statistics of E&M signaling	display voice em call-statistics
Display call control module information of E&M signaling	display voice em ccb
Display call statistics information of R2 signaling in RCV software module	display voice rcv statistic r2
Display call statistics information of R2 signaling	display voice r2 call-statistics
Display voice subscriber-line configuration	display voice subscriber-line e1-number : { ts-set-number \| 15 }
Display relevant information of VoIP	display voice voip { down-queue e1vi-no \| phy-statistic e1vi-no \| up-queue e1vi-no }
Enable output switch of corresponding debug information of E&M subscriber-line	debugging voice vas em
Enable output switch of corresponding debug information of R2 software module	debugging voice r2 { all \| ccb controller e1-number timeslots-list \| dl \| dtmf \| error \| mfc \| msg \| rcv \| warning }
Enable debug switch between RCV software module and lower R2 module	debugging voice rcv r2
Enable debug switch between VPP software module and lower R2 module	debugging voice vpp r2

3.6 Updating Functional Program of E1 Voice Card

You can live-update the functional program of the E1 voice card by means of the following command.

Perform the following configuration in system view.

Table 3-39 Update the functional program of the E1 voice card

Operation	Command
Update the functional program of the E1 voice card	update slot slot-number ftpserver { ip-address \| host-name } filename filename [ username name \| password password \| port port ]*

3.7 Examples of Typical E1 Voice Configuration

3.7.1 Voice Gateways Connecting PBXs through E1 Voice Subscriber-Lines (R2 Signaling)

I. Network requirement

The subscribers in city A and city B can use voice gateways on the IP network to talk to each other. The voice gateway in city A is connected to the PBX through the E1 voice subscriber-line using R2 signaling. The voice gateway in city B is connected to the PBX through E1 voice subscriber-line using digital E&M signaling (with delay start mode). The dialing mode is one-stage dialing.

II. Network diagram

Figure 3-7 Voice gateways are connected to PBXs through E1 mode

III. Configuration procedure

1) Configure parameters for the voice gateway in city A.

# Configure a TS set.

[VG] controller e1 0

[VG-E1-0] timeslot-set 1 1-31 signal r2

# Establish POTS voice entity 1001 on the E1 subscriber-line.

[VG-voice-dial] entity 1001 pots

[VG-voice-dial-entity1001] match-template 0101001

[VG-voice-dial-entity1001] line 0:1

[VG-voice-dial-entity1001] send-number all

[VG-voice-dial-entity1001] quit

# Establish POTS voice entity 1002 on the E1 subscriber-line.

[VG-voice-dial-entity1001] entity 1002 pots

[VG-voice-dial-entity1001] match-template 0101002

[VG-voice-dial-entity1002] line 0:1

[VG-voice-dial-entity1002] send-number all

[VG-voice-dial-entity1002] quit

# Establish VoIP voice entity 0755.

[VG-voice-dial] entity 0755 voip

[VG-voice-dial-entity755] match-template 0755

[VG-voice-dial-entity755] address ip 2.2.2.2

2) Parameter configuration for the voice gateway in city B is similar to that in city A basically

# Configure a TS set.

[VG] controller e1 0

[VG-E1-0] timeslot-set 1 1-31 signal e&m-delay

# Establish POTS voice entity 2001 on the E1 subscriber-line.

[VG] voice-setup

[VG-voice] dial-program

[VG-voice-dial] entity 2001 pots

[VG-voice-dial-entity2001] match-template 07552001

[VG-voice-dial-entity2001] line 0:1

[VG-voice-dial-entity2001] send-number all

[VG-voice-dial-entity2001] quit

# Establish POTS voice entity 2002 on the E1 subscriber-line.

[VG-voice-dial-entity2001] entity 2002 pots

[VG-voice-dial-entity2002] match-template 07552002

[VG-voice-dial-entity2002] line 0:1

[VG-voice-dial-entity2002] send-number all

[VG-voice-dial-entity2002] quit

# Establish VoIP voice entity 010.

[VG-voice-dial] entity 010 voip

[VG-voice-dial-entity10] match-template 010....

[VG-voice-dial-entity10] address ip 1.1.1.1

3.7.2 Voice Gateways Connecting PBXs with ISDN PRI Mode through E1 Lines

I. Network requirements

The subscribers in city A and city B can use voice gateways on the IP network to talk to each other. The voice gateway in city A is connected to the PBX through the E1 voice subscriber-line. The voice gateway in city B is connected to the PBX through the E1 voice subscriber-line. Both voice gateways are connected to the PBXs through ISDN PRI interface using DSS1 signaling. The dialing mode is one-stage dialing.

II. Network diagram

Figure 3-8 The PBX is connected to the voice gateway through the ISDN PRI interface using DSS1 signaling

III. Configuration procedure

1) Configure parameters for the voice gateway in city A.

# Configure an ISDN PRI set.

[VG] controller e1 0

[VG-E1-0] pri-set

# Establish POTS voice entity 1001 on the ISDN PRI.

[VG-voice-dial] entity 1001 pots

[VG-voice-dial-entity1001] match-template 0101001

[VG-voice-dial-entity1001] line 0:15

[VG-voice-dial-entity1001] send-number all

[VG-voice-dial-entity1001] quit

# Configure POTS voice entity 1002 on the ISDN PRI subscriber-line.

[VG-voice-dial] entity 1002 pots

[VG-voice-dial-entity1002] match-template 0101002

[VG-voice-dial-entity1002] line 0:15

# Establish VoIP voice entity 0755.

[VG-voice-dial] entity 0755 voip

[VG-voice-dial-entity755] match-template 0755....

[VG-voice-dial-entity755] address ip 2.2.2.2

2) Parameter configuration of the voice gateway in city B is similar to that in city A basically.

# Configure an ISDN set.

[VG] controller e1 0

[VG-E1-0] pri-set

# Configure POTS voice entity 2001 on the ISDN PRI subscriber-line.

[VG] voice-setup

[VG-voice] dial-program

[VG-voice-dial] entity 2001 pots

[VG-voice-dial-entity2001] match-template 07552001

[VG-voice-dial-entity2001] line 0:15

[VG-voice-dial-entity2001] send-number all

[VG-voice-dial-entity2001] quit

# Configure POTS voice entity 2002 on the ISDN PRI subscriber-line.

[VG-voice-dial] entity 2002 pots

[VG-voice-dial-entity2002] match-template 07552002

[VG-voice-dial-entity2002] line 0:15

[VG-voice-dial-entity2002] send-number all

[VG-voice-dial-entity2002] quit

# Establish VoIP voice entity 010.

[VG-voice-dial] entity 010 voip

[VG-voice-dial-entity10] match-template 010....

[VG-voice-dial-entity10] address ip 1.1.1.1

3.8 E1 Voice Troubleshooting

Fault 1: the call connection cannot be established between the switch and the voice gateway.

Troubleshooting:

l First, use the display current-configuration command to check whether the signaling uses all time slots. The time slots used by switch must be consistent with those configured on the voice gateway. If the switch uses the time slots that are not configured on the voice gateway in outbound calls, the connection cannot be established.

l If there is no dialing tone during calling, you should check whether the switch gives the outgoing exchange number to the voice gateway and whether the phone number is configured with exchange number or access number on the voice gateway. When the switch sends the outgoing exchange number, the connection cannot be established if no exchange number or access number is configured to the phone number on the voice gateway. So, you should delete the outgoing exchange number on the switch or set the access number on the voice gateway.

Fault 2: the voice gateway cannot use R2 signaling to establish connection with a subscriber connected to the switch.

Troubleshooting:

First, use the display current-configuration command to check whether the trunk mode of voice gateway corresponds with that configured on the switch. That is to say, when the trunk mode is outgoing trunk at the switch, it must be incoming trunk or bi-directional trunk on the voice gateway. When the trunk mode is incoming trunk at the switch, it must be outgoing trunk or bi-directional on the voice gateway. When incoming trunk is used at the voice gateway, the voice gateway permits incoming calls only and forbids outgoing calls.

Chapter 4 Configuring ISDN Protocol

4.1 Overview

ISDN (Integrated Services Digital Network), developed from integrated digital network (IDN), provides end-to-end digital connection, so as to support wide range of services (including voice and non-voice services).

ISDN provides the user with a group of standard multifunctional user-network interfaces. In ITU-T I.412 protocols, two types of user-network interfaces are specified: Basic Rate Interface (BRI) and Primary Rate Interface (PRI). The bandwidth of BRI is 2B+D, and that of PRI is 30B+D or 23B+D. Here:

l B channel is a user channel, used to transmit the voice, data and other user information with the transmission rate 64 kbps.

l D channel is a control channel and used to transmit the common channel signaling, controlling the calls on B channels of the same interface. The rate of D channel is 64 kbps (PRI) or 16 kbps (BRI).ITU-T Q.921, the data link layer protocol of D channel, defines the rules by which the information is exchanged between layer-2 entities on the user-network interface through D channel. Meanwhile, it supports the access of layer-3 entity. ITU-T Q.931, the network layer protocol of D channel, provides methods to establish, maintain, and terminate the network connection between communication application entities. Call Control (CC) is a further encapsulation of Q931, which forwards the message from the network side to CC for CC to perform information interchange with higher layer applications such as DDR.

Figure 4-1 ISDN D channel protocol stack

The ISDN protocols proposed by ITU-T provides different services in different areas, forming the ISDN protocols that are suitable for different regions, such as NTT (Nippon Telegraph and Telephone Corporation) in Japan, ETSI (European Telecommunications Standards Institute) in Europe, NI (National ISDN) in North America, AT&T 5ESS, and ANSI (American National Standard Institute). The H3C voice gateways support the basic calling functions of NTT, ETSI, ATT, ANSI and NI, but do not support the supplementary functions or network-side functions of these protocols.

4.2 Configuring ISDN

ISDN configuration includes:

l Configuring ISDN Signaling Type

l Configuring the Negotiation Parameters of ISDN Layer 3 Protocol

l Setting to Check the Called Number or the Sub-address in an ISDN Incoming Call

l Configuring the Service Type Accepted by ISDN Interface

l Configuring the Calling Number Carried in an ISDN Message

l Configuring ISDN Specially for Telecom Italia

4.2.1 Configuring ISDN Signaling Type

Perform the following configurations in system view or interface view.

Table 4-1 Configure ISDN signaling type

Operation	Command
Set ISDN signaling type	isdn protocol-type { dss1 \| qsig }

By default, DSS1 signaling is used on ISDN PRI interfaces.

Using the isdn protocol-type command in system view will not affect the existing ISDN PRI interface, and it will only change the default type of signaling on the newly created ISDN PRI interface.

& Note:

l You cannot configure the command when there is active call on the ISDN interface. Or you can use the shutdown command to disable the ISDN interface to configure the command and then use the undo shutdown command to enable the interface. This operation will disconnect the active calls on the interface.

l The DSS1 ISDN protocol can be configured on PRI interfaces.

l The QSIG protocol can be configured only on PRI interfaces.

l Other protocols are made up by the negotiation commands of Layer 3 protocol under DSS1 protocol.

4.2.2 Configuring the Negotiation Parameters of ISDN Layer 3 Protocol

If ISDN interface supports different protocols, the configurations differ in the negotiation parameters of Layer 3 protocol and whether to configure SPID parameter. The following explains the way to choose the negotiation parameters of Layer 3 protocol.

Perform the following configuration in interface view.

I. DSS1 protocol

Table 4-2 Optional commands of DSS1 protocol

Operation	Command
Configure the SETUP message not to carry high-level compatibility information unit when the ISDN initiates data call.	isdn ignore hlc
Restore the SETUP message to carry high-level compatibility information unit.	undo isdn ignore hlc
Configure the SETUP message not to carry low-level compatibility information unit when the ISDN initiates call.	isdn ignore llc
Restore the SETUP message to carry low-level compatibility information unit.	undo isdn ignore llc
Disable the VG from checking the SETUP message of incoming calls for complete information unit.	isdn ignore callednum
Restore the default setting.	undo isdn ignore callednum
Set the VG to switch to the ACTIVE state and start data and voice service communications until receiving the CONNECT ACK messages for the CONNECT messages it has sent during the communications between the VG and the connected exchange.	isdn waitconnectack
Set the VG to get ACTIVE directly to start data and voice service communications rather than waiting for the CONNECT ACK messages	undo isdn waitconnectack
Enable a VG to carry the Sending-Complete Information Element in the SETUP message sent to PBX	isdn sending-complete
Disable the VG to carry the Sending-Complete Information Element in the SETUP message sent to PBX	undo isdn sending-complete
Configure the interval for a Q931 signaling timer	isdn q931-timer timer-name time-interval
Restore the default interval values of Q931 signaling timers	undo isdn q931-timer { timer-name \| all }

By default, SETUP message carries high-level compatibility, low-level compatibility, sends complete information unit and waits for CONNECT ACK message.

II. QSIG protocol

Table 4-3 Optional commands of QSIG protocol

Operation	Command
Set the length of call reference to be used when the ISDN interface initiates calls	isdn crlength { 1 \| 2 }
Configure the SETUP message not to carry high-level compatibility information unit when the ISDN initiates calls.	isdn ignore hlc
Restore the SETUP message to carry high-level compatibility information unit.	undo isdn ignore hlc
Configure the SETUP message not to carry low-level compatibility information unit when the ISDN initiates call.	isdn ignore llc
Restore the SETUP message to carry low-level compatibility information unit.	undo isdn ignore llc
Disable the VG from sending the SETUP ACK message if the received SETUP message does not carry the called number in a data service call	isdn ignore callednum
Restore the default setting	undo isdn ignore callednum
Set the VG to switch to the ACTIVE state and start data and voice service communications until receiving the CONNECT ACK messages for the CONNECT messages it has sent during the communications between the VG and the connected exchange	isdn waitconnectack
Set the VG to get ACTIVE directly to start data and voice service communications rather than waiting for the CONNECT ACK messages	undo isdn waitconnectack
Enable a VG to carry the Sending-Complete Information Element in the SETUP message sent to PBX	isdn sending-complete
Disable the VG to carry the Sending-Complete Information Element in the SETUP message sent to PBX	undo isdn sending-complete
Configure an ISDN PRI interface, cE1 PRI interface to receive dialed numbers in overlap-receiving mode	isdn overlap-receiving
Configure the ISDN PRI interface, cE1 PRI interface to receive dialed numbers in complete receiving mode	undo isdn overlap-receiving
Configure the interval for a QSIG signaling timer	isdn qsig-timer timer-name time-interval
Restore the default interval values of QSIG signaling timers	undo isdn qsig-timer { timer-name \| all }
Enable passthrough transmission of the facility information field of QSIG when the VG interoperates with a PBX through the QSIG signaling	isdn facility-passthrough
Disable passthrough transmission of the facility information field of QSIG	undo isdn facility-passthrough

By default, the SETUP message carries high-level compatibility, low-level compatibility, sends complete information unit and waits for CONNECT ACK message.

By default, the length of call reference of E1 PRI interface is 2 bytes. ISDN PRI interface receives called numbers in overlap-receiving mode.

Caution:

Commands that are neither optional nor required commands cannot be configured.

4.2.3 Setting to Check the Called Number or the Sub-address in an ISDN Incoming Call

Perform the following configuration in interface view.

Table 4-4 Set to check the called number or the sub-address in an ISDN incoming call

Operation	Command
Set to check the called number or the sub-address in an ISDN incoming call	isdn check-called-number check-index called-party [ :subaddress ]
Remove the check of the called number or the sub-address in an ISDN incoming call	undo isdn check-called-number check-index

By default, no called number or sub-address is configured.

The commands are used to set the items to be checked in the digital incoming call. If the sub-address is set, the opposite will be rejected when the sub-address is not sent or is sent incorrectly.

4.2.4 Configuring the Service Type Accepted by ISDN Interface

ISDN interface can accept three types of service: data, speech and audio. You can set it to accept some of these types and deny others.

Perform the following configuration in ISDN interface view.

Table 4-5 Configure the service type accepted by ISDN interface

Operation	Command
Configure the service type accepted by ISDN interface	isdn service [audio \| speech ]
Configure the ISDN to deny all types of data	undo isdn service

By default, ISDN interface accepts speech service.

4.2.5 Configuring the Calling Number Carried in an ISDN Message

Perform the following configuration in ISDN interface view.

Table 4-6 Configure the calling number carried in an ISDN message

Operation	Command
Configure the calling number carried in an ISDN message	isdn callingnum calling-number
Delete the calling number carried in an ISDN message	undo isdn callingnum

By default, no calling number is carried by the ISDN calling message.

& Note:

If the calling party has configured with this command on the ISDN interface, it can send the calling number to the called party who can get the calling number from history record.

4.2.6 Configuring ISDN Specially for Telecom Italia

As Italy has strict standards on network service and information element, use the command isdn communicate italy to ensure interconnection when interconnecting with the European ISDN network.

Perform the following configuration in cE1/PRI interface view.

Table 4-7 Configure ISDN specially for Telecom Italia

Operation	Command
Enable ISDN specially for Telecom Italia	isdn communicate italy
Disable ISDN specially for Telecom Italia	undo isdn communicate italy

By default, ISDN special for Telecom Italia is disabled.

4.3 Displaying and Debugging ISDN Configuration

Carry out the following commands in any view.

Table 4-8 Display and debug ISDN

Operation	Command
Display the current activated call information of ISDN interface	display isdn active-channel [ interface type number ]
Display the value of ISDN DSS1 signaling timer	display isdn q931-timer
Display the value of ISDN QSIG signaling timer	display isdn qsig-timer [ interface type number ]
Display the current status of ISDN interface	display isdn call-info [ interface type number ]
Display the value of the ISDN timer	display isdn dss1-parameters
Display the related information of SPID on the BRI interface running the NI protocol	display isdn spid [interface type number]
Enable the debugging for ISDN CC	debugging isdn cc [ interface type number ]
Enable the debugging for ISDN q921 protocol	debugging isdn q921 [ interface type number ]
Enable the debugging for ISDN q931 protocol	debugging isdn q931 [ interface type number ]
Enable the debugging for ISDN QSIG signaling	debugging isdn qsig { alarm \| call-state \| error \| information \| message \| all } [ interface type number ]

4.4 Typical Configuration Example

4.4.1 Interconnect VGs for Data Transmission via ISDN PRI Line

I. Network requirements

As shown in the following diagram, VG A and VG B implement VoIP calls over an IP network. VG B is connected to a PBX over an ISDN PRI line to implement voice calls over the local PSTN.

II. Network diagram

Figure 4-2 Network diagram of ISDN protocol configuration example

III. Configuration procedure

1) Configure VG A:

# Configure voice entities.

[VG] voice-setup

[VG-voice] dial-program

[VG-voice-dial] entity 1001 pots

[VG-voice-dial-entity1001] match-template 1001

[VG-voice-dial-entity1001] line 0

[VG-voice-dial] entity 1 voip

[VG-voice-dial-entity1] match-template 88..

[VG-voice-dial-entity1] address ip 192.168.80.50

[VG-voice-dial-entity1] fast-connect

2) Configure VG B:

# Create an ISDN PRI interface.

[VG] controller e1 0

[VG-E1-0]pri-set

[VG-E1-0]quit

# Configure the ISDN PRI interface.

[VG] interface serial 0:15

[VG-Serial0:15] link-protocol ppp

[VG-Serial0:15] isdn service data

[VG-Serial0:15] isdn service speech

[VG-Serial0:15] dialer enable-circular

# Configure voice entities.

[VG] voice-setup

[VG-voice] dial-program

[VG-voice-dial] entity 88 pots

[VG-voice-dial-entity88] match-template 88..

[VG-voice-dial-entity88] line 0:15

[VG-voice-dial-entity88] dial-prefix 88

[VG-voice-dial-entity88] quit

[VG-voice-dial] entity 1 voip

[VG-voice-dial-entity1] match-template 10..

[VG-voice-dial-entity1] address ip 192.168.80.30

[VG-voice-dial-entity1] fast-connect

Chapter 5 Voice AAA Configuration

5.1 Voice AAA Overview

Voice gateways support local and RADIUS AAA modes. Through local users configured on the voice gateways, authentication, authorization and dial-up flow control can be implemented. When used as RADIUS client, a voice gateway can work with the RADIUS server to implement accounting and advanced call control for voice calls.

5.1.1 Fundamentals of Voice AAA

I. Introduction to Voice RADIUS

As specified in RFC2865/2866 or newer, Remote Authentication Dial-In User Service (RADIUS) is a protocol standard for implementing Authentication, Authorization, and Accounting (AAA) on the login users. RADIUS is suitable for not only PPP dial users but also voice users. The VG series provides voice Radius intended for the use in the networks of small-to-medium-sized service providers. Enterprise-level users can use this function to make control and management in voice calls and accounting in charges.

In one word, voice RADIUS is a RADIUS client module that provides voice user AAA services on a VG. It is a part of the whole voice functionality module. When a voice call occurs, the VG will interact with the calling party based on the configuration parameters, and encapsulate the obtained user information and statistics into the RADIUS AAA packets and send them to the RADIUS server. Then, the VG will make call connection decision based on the response message received from the RADIUS server. When the call ends, the VG will report the connection duration, number of packets and bytes, as well as other statistics information of the call to the RADIUS server to complete call accounting and so on.

II. Call flow in voice RADIUS

The following figure provides a voice RADIUS application scenario:

Figure 5-1 Network diagram for the voice RADIUS application

As the figure shows, a call can be set as follows:

1) The calling party (assigned the number 1000) originates a call through a PSTN and puts through the voice subscriber-line of a local VG (O_Gateway), requesting for a long-distance call to 1001. From the perspective of PSTN, the voice subscriber-line of O_Gateway is the called party at this given time. So, O_Gateway sends a PSTN-side (Call segment1) accounting start request to the RADIUS Sever as the PSTN-side called party.

2) After receiving the RADIUS server’s response to the PSTN-side (call segment 1) accounting start request, O_Gateway requests the RADIUS server for user ID authentication by sending a RADIUS packet encapsulating the calling user ID information obtained from the calling party. After receiving the ACK to this authentication request, O_Gateway requests the RADIUS server for authorization by sending a RADIUS packet encapsulating the called number input by the calling user. Upon the receipt of the ACK to this authorization request, O_Gateway sends a VoIP-side (call segment2) accounting start request to the RADIUS server as the calling party. (From the perspective of IP, O_Gateway will be the calling party in the subsequent IP call process).

3) After receiving the RADIUS server’s response to the VoIP-side (call segment 2) accounting start request, O_Gateway calls the T_Gateway specified by the called number across an IP network and establishes an IP-side voice channel. Upon the receipt of the connection request, T_Gateway calls the called number across a PSTN. After receiving the ringing message of the PSTN-side called party, T_Gateway notifies O_Gateway and sends a VoIP-side (Call segment3) accounting start request to the RADIUS server as the IP-side called party. Then, O_Gateway will send the ringback tone to its PSTN-side calling party.

4) After receiving the RADIUS server’s response to the VoIP-side accounting start request, T_Gateway begins to seize the called voice resources. As soon as the called party picks up the phone, T_Gateway sends the PSTN-side (call segment4) accounting start request as the calling party. From the perspective of PSTN, the T_Gateway voice subscriber-line is the calling party at this given time. Upon receiving the RADIUS server’s response to the accounting start request of the PSTN-side (call segment4), T_Gateway notifies O_Gateway and proceeds to H.245 facilities negotiation together with it.

5) After transiting to the communication state from the call process, O_Gateway sets the communication duration timers according to the communication duration setting in the received authorization response. If the timer times out or the calling/called party hangs up before the timer times out, the call will be normally terminated.

6) Suppose it is the calling party hangs up first, then O_Gateway will release the occupied voice subscriber-line resources and request the RADIUS server to stop its PSTN-side accounting (Call segment1). At the same time, it will send a voice channel release message to T_Gateway and then, request the RADIUS server to stop its VoIP-side (call segment 2) accounting. Upon the receipt of the IP-side voice channel release message, T_Gateway will request the RADIUS server to stop the VoIP-side (call segment3) accounting stop request and release its PSTN-side channel resources. Upon the receipt of the channel release ACK from the PSTN side, T_Gateway will send a PSTN-side (call segment 4) accounting stop request to the RADIUS server. The accounting stop requests of these four call segments do not require a response.

& Note:

l In the call process, the sending process of accounting request is divided into four segments for the sake of easy settlement between involved service providers and the segment-based call channel control.

l The procedure described above is just a brief description on the call setup and release processes shown in the figure above. In fact, the processes and messages are quite different from call to call due to the complexity of message interoperations of the entire call setup/release process, differences among call connection processes, failure of AAA operations, call termination due to errors, or the called party’s action of hanging up the phone, and so on.

As you can see, the support of RADIUS server is indispensable to the normal working of voice RADIUS. The authentication, authorization, and accounting packets that the VGs and the RADIUS server interact are compliant with RFC2865/2866 provisions. At the same time, in order to accommodate the various requirements in voice call user information recording, the structure and value assignment of the proprietary attribute field Vendor_Specific in the packets are compliant with the private protocol proposed by H3C and AsiaInfo Company. Before using this system, you must ensure that the attribute dictionary of the RADIUS server has included the definition of H3C in this attribute option. In addition, make sure that the RADIUS server has been configured with a voice call user list associated with the VG system.

5.1.2 Voice AAA Functions Provided by VG

I. Voice-call-supported AAA

The VG first implements local authentication and authorization for voice calls. If the local authentication and authorization fail, the VG implements authentication and authorization through the RADIUS server. If no local voice users have been configured on a VG, the VG will implement authentication and authorization directly through the RADIUS server. The accounting on voice calls can be implemented only through the RADIUS server.

Voice call authentication and authorization can be made in two modes: authentication and authorization by card IDs/passwords and by calling numbers. RADIUS authorization is made by authorizing the called numbers and can be made only after the RADIUS authentication is passed.

Voice call accounting packets can be processed in three approaches, start-no-ack, stop-only, and start-ack, described as follows:

l start-ack: The VG sends an accounting request to the RADIUS server when the call is set up and ends. The VG waits for the RADIUS server’s response to the accounting start request before it can make a call connection, and it can release the call without receiving the response from the RADIUS server for the end-accounting request.

l start-no-ack: The VG sends an accounting request message to the RADIUS server when the call setup starts and when the call ends. However, it directly initiates and releases the voice call no matter whether it has received a response from the RADIUS server at the call setup and at the end of the call.

l stop-only: The VG sends an end-accounting request to the RADIUS server only when the call ends and directly releases the call no matter whether it has received a response from the RADIUS server.

l In any of these three modes, a response of the RADIUS server is not required for releasing the call connection.

II. Voice call processing procedure

AAA operations and details recording on an IP voice call can be done only if the identification information of the calling party is available. The ID information can be the calling number or a pre-defined card ID/password pair. In the latter case, the IP telephone user has to input a number comprising digits (e.g., access number) to let the IP telephone system know that he/she is about to input the card ID/password pair. (Otherwise, the system will resolve the card ID/password pair as a called number.)

In fact, a user uses calling number for authentication can use the access number as well for the sake of charge centralization (for example, the users of a group can use the same access number) or for the convenience of maintaining call rights. (For example, you can only place local calls on a telephone if there is no access number but will be able to place long-distance calls, national or international, on the same telephone if there is an access number). Thus, there are three types of dial processes, of which the first one is a one-stage dial process and the rest two are two-stage dial processes.

l Directly dial the called number

l Dial the access number first and then the called number

l Dial the access number first and then enter the card ID/password pair and the called number

Depending on the user's configuration, voice RADIUS can provide any basic access procedures described above and set the process attribute parameters (like the redial times, the number of digits of card ID/password, and so on). Moreover, the user can configure the access number to be a Private Line Automatic Ringdown (PLAR) number (the number that can be dialed automatically) on the voice subscriber-line, and customize the details of the access process as needed.

& Note:

l In current release, the VG does not support dialing of nested access numbers.

III. Playing voice prompts

In a two-stage dial process, voice prompts are provided.

IV. Providing voice CDR and query

Call Detail Record (CDR) retains the details of each voice call, and the user can set the number of the entries that can be saved in the CDR as well as the duration that they can be retained by configuring command lines. Following are the main contents of CDR:

l Calling number

l Called number

l IP address of the peer VG

You can retrieve the entries listed above for the retained call information.

5.2 Basic Configuration of Voice AAA

To implement voice AAA, you should first enable the AAA function and then select the AAA mode according to users' requirement.

The basic configuration of voice AAA includes:

l Enable AAA

l Configure local voice users

l Configure the RADIUS client

l Configure the access number

l Configure dial process and related parameters

l Configure the CDR storage rule

5.2.1 Enabling AAA

Perform the following configurations in system view.

Table 5-1 Enable/disable AAA

Operation	Command
Enable AAA.	aaa-enable
Disable AAA.	undo aaa-enable

By default, AAA is enabled.

& Note:

The radius command is available only if the AAA function is enabled.

5.2.2 Configuring Local Voice Users

I. Entering voice AAA service view

Before you can configure the voice AAA service, you must enter voice AAA service view by carrying the aaa-client command.

Perform the following configuration in voice view.

Table 5-2 Enter voice AAA service view

Operation	Command
Enter voice AAA service view	aaa-client

II. Configuring local voice users

To simplify user authentication for medium- and small-sized enterprises, you can configure a local voice user database on a VG so that the system can implement local authentication first. If passing the local authentication, the user will not need to undergo the subsequent RADIUS authentication; if failing the local authentication, the user will undergo authentication on the RADIUS server. If RADIUS authentication succeeds, voice communication can start; otherwise, the user will be rejected. For larger enterprises, it is recommended to directly adopt RADIUS authentication rather than configuring local voice users. So far, up to 200 local users are supported.

If only the user name is configured but the password is not, the local voice user is equivalent to the calling number in a voice call.

With a password configured, the local voice user is equivalent to the card number and password of the card number/password process in a dial process.

Perform the following configurations in voice AAA service view.

Table 5-3 Configure a local user and its password

Operation	Command
Configure a local user and its password.	local-user username [ password password ]
Remove a specified local voice user.	undo local-user username

By default, no local voice user has been configured.

& Note:

Instead of applying to all the service applications, the local voice users configured using the local-user username command can only apply to voice communications. The local-user command discussed here is different from the one used for configuring a system user. You should distinguish them carefully when making the configurations.

5.2.3 Configuring RADIUS Client

The basic RADIUS client configuration includes:

l Configure the IP address, authentication port number and accounting port number of the RADIUS server host

l Configure the RADIUS server key

l Configure the RADIUS client protocol type

The advanced RADIUS client configuration tasks include:

l Configure the interval for sending a request to the RADIUS server

l Configure the authentication request retry times

l Configure the query interval after the RADIUS server fails

l Configure the interval for sending real-time accounting packets to the RADIUS server

l Configure the maximum number of attempts of sending the accounting stop packet to the RADIUS server

l Configure the source IP address for sending RADIUS packets

I. Configuring IP address of server host and authentication and accounting ports

The user is allowed to configure up to three RADIUS servers.

Following are the rules observed in authentication and accounting server selection in RADIUS.

l The servers are selected depending on the order in which they are configured by the user, i.e., the one configured first is used first.

l If the first RADIUS server in use gives no response, the second server takes over the role, and if the second server fails, the third server takes over the role.

l If the authentication/accounting port number is set to 0 on a RADIUS server, the clients will not use the authentication/accounting services provided by the server.

Table 5-4 Configure the IP address of the server host and the authentication and accounting port numbers

Operation	Command
Configure the IP address of the RADIUS server, authentication and authorization port numbers	radius server ip-address [authentication-port port-number ] [ accounting-port port-number ]
Disable the host address of the RADIUS server	undo radius server ip-address

By default, authentication port number and accounting port number are respectively set to 1812 and 1813. Setting authentication/accounting port number to 0 indicates that the server does not provide the authentication/accounting server service.

II. Configuring RADIUS server key

A shared key is used for encrypting a user password and generating the response authenticator. A RADIUS server can secure the transmission of authentication information on a network by encrypting its crucial information (such as password) with MD5. The VG and the RADIUS server participating in an authentication therefore can be regarded valid and pass the authentication only when the same shared key has been configured on them.

Table 5-5 Configure the RADIUS server key

Operation	Command
Configure the RADIUS server key.	radius shared-key string
Delete the RADIUS server key.	undo radius shared-key

By default, the RADIUS server key is not configured. Configure the password first before using the RADIUS server.

III. Setting the RADIUS Client Protocol Type

RADIUS client and RADIUS server suppliers support almost all the public attributes in their implementations of RADIUS authentication, authorization, and accounting on voice calls, and they have also defined many private attributes. In order to accommodate these different RADIUS servers, you are allowed to change the voice RADIUS packet format by setting the protocol type at the RADIUS client end.

So far, the H3C voice gateways support four RADIUS client protocol types that allow flexible interoperation with most RADIUS servers. They are:

l overload-nonstandard: supports the interoperation with non-standard RADIUS servers. It encapsulates the extending accounting information in the RFC standard attribute field “Acct-Session-Id”.

l vsa-nonstandard: supports the interoperation with the non-standard RADIUS servers. It encapsulates the extending accounting information in the RFC standard attribute field “Vendor-Specific Attributes” and supports more private attributes.

l private: supports the interoperation with the RADIUS servers of private protocol proposed by H3C and AsiaInfo Company.

l ietf-rfc: supports the interoperation with the standard RADIUS servers, i.e., the servers in strict compliance with RFC2865 and RFC2866.

Perform the following configurations in voice AAA service view.

Table 5-6 Set the RADIUS client protocol type

Operation	Command
Set the RADIUS client protocol type	clienttype { overload-nonstandard \| vsa-nonstandard \| private\| ietf-rfc }
Restore the default protocol type applied on the RADIUS client	undo clienttype

The default protocol type is private.

IV. Configuring the interval for sending a request to a RADIUS server

In order to determine whether a RADIUS server has failed, the VG regularly sends authentication request packets to RADIUS server.

Perform and the following configuration in system view.

Table 5-7 Configure the interval at which the VG sends authentication request packets to the RADIUS server

Operation	Command
Configure the interval at which the VG sends authentication request packets to the RADIUS server.	radius timer response-timeout seconds
Restore the default interval at which the VG sends authentication request packets to the RADIUS server.	undo radius timer response-timeout

Authentication request packets are sent to a RADIUS server at the interval in the range of 1 to 65535 seconds and at the interval of 10 seconds by default.

V. Configuring authentication request retry times

In order to determine whether a RADIUS server has failed, the system will periodically send authentication request packets to the RADIUS server. In case the system does not receive the response from the RADIUS server before the specified timeout time expires, it needs to retransmit the authentication packet. The user can set a limit on the request retry times. Thus, when the request retry attempts exceed the allowed retry times, the system will assume that the server has failed and will set its state to the FAILED state.

Table 5-8 Configure authentication request retry times

Operation	Command
Configure the allowed authentication request retry times.	radius retry times
Restore the default authentication request retry times.	undo radius retry

The authentication request retry times can be set in the range of 1 to 255 and the system retries to send an authentication request to the RADIUS server for 3 times by default.

VI. Configuring the query interval after the RADIUS server fails

If the first RADIUS server cannot operate normally due to the failure of the line between the Net Access Server (NAS) and the server or due to the fault on the RADIUS process for example, the system will set RADIUS server's state to INVALID and replace it with some other RADIUS server selected following the specified rule. At the same time, the system will set a timer on the INVALID state of the first RADIUS server. Upon the expiration of the timer, the state of this server will be set to VALID again, regardless whether the server can provide services or not. Once the current server fails to provide services, the system will query the servers that are in valid state, beginning with the first RADIUS server. Only if it is unavailable will the system turn to the next valid RADIUS server.

Table 5-9 Configure the query interval after the RADIUS server fails

Operation	Command
Configure the interval at which the VG queries a RADIUS server after it fails.	radius timer quiet minutes
Restore the default query interval.	undo radius timer quiet

The interval at which the VG queries a failed RADIUS server can be set in the range of 1 to 255 minutes and defaults to 5 minutes.

VII. Configuring the interval for sending real-time accounting packets to the RADIUS server

After a user passes the authentication, the RADIUS client (VG) sends real-time accounting information to the RADIUS server at a regular interval. If the real-time accounting request fails, the RADIUS client will allow the user to continue using the network services.

In normal cases, the accounting packets that the RADIUS client (VG) sends involve only the accessing time and the disconnection time. However, For the sake of reliability, you can configure the interval for sending real-time accounting packets to the RADIUS server.

Table 5-10 Configure the interval for sending real-time accounting packets to the RADIUS server

Operation	Command
Configure the interval for sending real-time accounting packets to the RADIUS server.	radius timer realtime-accounting minutes
Restore the default interval for sending real-time accounting packets to the RADIUS server.	undo radius timer realtime-accounting

A RADIUS Client (VG) is allowed to send real-time accounting packets to a RADIUS server at an interval of 0 to 32767 minutes and this interval is 0 by default, i.e., there is no real-time accounting.

VIII. Configuring the maximum times that a VG is allowed to send voice accounting stop packets to RADIUS

When a VG sends a voice accounting stop packet to RADIUS, it will retransmit this accounting packet in the event that it does not receive the desired response packet from the RADIUS server. In case the RADIUS server does not make a response at all, the VG will stop the retransmission after making several attempts as specified. The retransmission attempt times are user-configurable.

Perform the following configuration in system view.

Table 5-11 Configure the maximum times that a VG is allowed to send a voice accounting stop packet to RADIUS

Operation	Command
Configure the maximum times that a VG is allowed to send a voice accounting stop packet to RADIUS.	radius stop-resend times
Restore the default maximum times that the VG is allowed to send a voice accounting stop packet to RADIUS.	undo radius stop-resend

By default, a VG is allowed to send a voice accounting stop packet to the RADIUS server up to 100 times.

IX. Specifying a source IP address for the RADIUS packets to be transmitted

You may specify a source IP address for the RADIUS packets sent from different interfaces on the VG. In this way, the RADIUS server will contact the VG only at that IP address.

An RADIUS server requires the administrator to register all the RADIUS clients, which are determined on the basis of source IP address. Therefore, the interfaces with different IP addresses on the same VG will be regarded by the RADIUS server as different clients. Whenever the RADIUS server receives a packet carrying an unregistered source IP address, it will regarded the packet as illegal and hence make no processing on it. For this reason, you may configure a source IP address for the transmitted RADIUS packets on the VG to get free of the work of registering the IP addresses of all the interfaces on the VG with the server.

Caution:

You must make sure that the specified source IP address is the IP address of an interface on the VG, and the route to that IP address exists on the server. You can also configure a loopback interface on the VG, specify an IP address for it, and take this address as the source IP address of the RADIUS packets.

Perform the following configuration in system view.

Table 5-12 Specify the source IP address for the transmitted RADIUS packets

Operation	Command
Configure the source IP address for the transmitted RADIUS packets	radius source-ip ip-address
Remove the source IP address specified for the RADIUS packets to be transmitted	undo radius source-ip

By default, no specified source IP address is configured for sending RADIUS packets.

5.2.4 Access Number Configuration

It is necessary to note the order and application ranges of the voice RADIUS configurations. As the prerequisite for the authorization function to be available, the authentication function must be enabled before the authorization function. For one-stage dial users and two-stage dial users, different enabling methods and application ranges apply. The enabling of accounting function and its accounting-scheme setting are independent of those of authentication/authorization function and are applicable to all the dial users. Of course, you must also ensure that the attribute dictionary of the RADIUS server has included the proprietary attribute of H3C and information of the users using the VG system has been configured. Read through the following subsections carefully and acquaint yourself with the views, steps and application ranges of all commands.

Voice RADIUS configuration task list includes:

l Access voice AAA view

l Configure access number

I. Accessing Voice AAA Service View

Before you can configure voice AAA services, you must access the voice AAA view by executing the aaa-client command.

Perform the following configuration in voice view.

Table 5-13 Access the voice AAA view

Operation	Command
Access the voice AAA view.	aaa-client

II. Configuring Access Number

A two-stage dial user must dial a particular access number before he can enjoy the IP telephony service. Therefore, a VG must be configured with the access numbers for the final users before it can provide the two-stage service to them. So far, a VG system can support up to 100 access numbers.

Perform the following configurations in voice dial program view.

Table 5-14 Configure an access number

Operation	Command
Set an access number or access the access number view.	gw-access-number access-number
Delete the configured one or all the access numbers.	undo gw-access-number [ access-number ]

By default, no access number has been configured.

5.2.5 Configuring Dial Process and Related Parameters

I. Configuring dial process

Access number by itself is only the code of a dial process. To obtain a complete dial process, you must set a set of process and attribute parameters for it.

Two-stage dial has three dial process approaches: calling number dial process (calling number authentication), voice calling number process (calling number authentication), and card number dial process (card number/PIN authentication). You must specify the dial process appropriate to each access number. When the user changes the current dial process, the system will restore the involved parameters to the defaults automatically.

This is how the voice calling number process differs from the calling number dial process:

In the voice calling number process, the voice gateway plays voice messages to prompt you to select language and input the called number after you dial the access number; but in the calling number dial process, it only plays dial tone (long tone).

In the voice calling number process, you can enable or disable language selection prompt. If it is disabled, the voice gateway directly plays the voice message that prompts users to dial the called numbers after they dial the access number.

You can proceed to configure the allowed dial times, the digits of card number/PIN and so on only after you have selected the card dial number process (card number/PIN authentication). If you have selected the calling number dial process or voice calling number process, these two configuration tasks will be omitted.

Perform the following configuration in access number view.

Table 5-15 Specify a dial process

Operation	Command
Specify a dial process.	process-config { callernumber \| cardnumber \| voice-caller }

By default, card number dial process is used.

II. Specifying dial times allowed in a card number dial process

This configuration task can be omitted if the calling number dial process is adopted. As for the card number dial process, the dial order is access number, prepaid card number, PIN, and finally the called number. During the dial process, even a digit input mistake can cause the failure of the entire dial process. In order to give a chance for users to correct the errors, it is necessary to set a maximum number of dial attempts that the users are allowed to make at each dial stage.

Perform the following configuration in access number view.

Table 5-16 Specify dial times allowed in a card number dial process

Operation	Command
Specify the allowed dial attempts.	redialtimes redialtimes-number

By default, three dial attempts (that is, two redial attempts) are allowed at each stage in a card number dial process.

& Note:

Before you set the maximum number of dial attempts, you should note that:

l The setting of allowed dial attempts is also significant to a calling number dial process. You should specify the attempts that a user is allowed to enter the correct destination number after entering the correct access number.

l This setting takes the same effect at each dial stage. For example, in a card number dial process, the attempts that you are allowed to enter the card number, PIN, and destination number are the same.

l The redialtimes command literally means the redial attempts. But it actually refers to the total dial times. Therefore, if you want to redial n times, you must make redialtimes-number equals n+1. For example, if you want to give three redial chances, set redialtimes-number to 4.

III. Specifying the number of card number/PIN digits for the card number dial process

This configuration task can be omitted if the calling number dial process is adopted. As for the card number dial process, it is necessary to provide for a fixed number of digits for card numbers/PINs to facilitate user management and access control.

Perform the following configurations in access number view.

Table 5-17 Specify the card number/PIN digits allowed in the card number dial process

Operation	Command
Specify the number of card number digits.	card-digit card-digit
Specify the PIN digits.	password-digit password-digit

By default, card-digit is set to 12 and password-digit to 6.

IV. Configuring the method to receive the called number

Using this command, you can either immediately initiate a call or input the # after all the digits of the called number have been collected.

Perform the following configuration in access number view.

Table 5-18 Configure the method to receive the called number

Operation	Command
Configure the method to receive the called number	callednumber receive-method { immediate \| terminator }
Restore the default method	undo callednumber receive-method

In the two-stage dial process, you need to input a # to end the called number by default.

V. Configuring the language selection for the voice calling number process

With the language selection enabled, the voice gateway will prompt you to select a language and then input the called number after you dial the access number.

Perform the following configuration in access number view.

Table 5-19 Configure the language selection

Operation	Command
Configure the language selection	selectlanguage{ disable \| enable }

By default, the language selection prompt is disabled. After dialing the access number, you hear the voice prompt for inputting the called number.

5.2.6 Setting the Rules for Saving Call Detail Record (CDR)

The system will generate a CDR entry about the call process upon the termination of each call regardless whether the result is normal or not. A VG is impossible to retain infinite CRD entries due to its limited memory, so it is necessary to put a limitation on the record saving space. You can put the restriction by limiting the number of record entries and limiting the live duration of each record.

Perform the following configurations in voice AAA service view.

Table 5-20 Set a CDR entry retaining rule

Operation	Command
Set a CDR entry retaining rule.	cdr { buffer [ size-number ] \| duration [ timer-number ] \| threshold [ percentage ] }
Restore the default CDR entry retaining rule.	undo cdr

By default, the restriction is made by limiting the number of retained CDR entries (i.e., by selecting the buffer approach), the value of size-number is 50, the value of timer-number is 86,400 seconds (namely 24 hours), and the value of percentage is 80.

& Note:

Each VG can save up to 500 CDR entries. In other words, even if you have specified the longest live duration of CDR entries, the system cannot retain CDR entries more than 500. In case more than 500 CDR entries that are compliant with the configured live duration are created as a result of the generation of enormous traffic during a period of time, those terminated earlier and beyond 500 will be deleted, despite they are complied with the retaining rule.

5.3 AAA Configuration

5.3.1 Enabling Authentication

I. Enabling authentication on one-stage dial users

One-stage dial users are those who can directly place a call without dialing an access number. In this case, it is impossible to enable the user authentication specific to a personal ID as a result of the lack of access number. The only feasible approach is to perform an authentication enabling operation on all the one-stage dial users.

If the RADIUS server is used to authenticate one-stage dial users, the RADIUS server and the RADIUS Client (VG) must be interoperable at the network layer and the user list containing all the one-stage dial users must have been configured on the RADIUS server.

Perform the following configurations in voice AAA service view.

Table 5-21 Enable the authentication on all the one-stage dial users

Operation	Command
Enable the authentication on all the one-stage dial users.	authentication-did
Disable the authentication on all the one-stage dial users.	undo authentication-did

By default, authentication is not enabled on one-stage users.

& Note:

“did” is the abbreviation of “direct inward dial”.

II. Enabling the authentication on two-stage dial users

In the case of two-stage dial users, you can enable the authentication after setting the appropriate access number by following the steps described earlier in this chapter.

If the RADIUS server is used to authenticate two-stage dial users, make sure that the RADIUS server and the RADIUS Client are interoperable at the network layer and the list containing all the one-stage users has already been configured on the RADIUS server.

Perform the following configurations in access number view.

Table 5-22 Enable the authentication on two-stage dial users

Operation	Command
Enable the authentication on two-stage dial users.	authentication
Disable the authentication on two-stage dial users.	undo authentication

By default, authentication of two-stage dial users (users requiring an access number) is not enabled.

& Note:

Following are the differences of two-stage and one-stage dial approaches in enabling the authentication function:

l The authentication on two-stage dial users is enabled specific to a particular access number;

l The authentication on one-stage dial users is enabled for all the one-stage dial users in voice AAA view, and is valid to all one-stage dial users.

5.3.2 Enabling Authorization

Authentication is a prerequisite to authorization. You must enable authentication before you can enable desired authorization. On the contrary, enabling authentication does not necessarily require the enabled authorization function. In the command configuration procedure, authorization is enabled after authentication. In fact, users cannot view the authorization enable command at all before enabling authentication.

I. Enabling the authorization on one-stage dial users

Similar to enabling authentication, as there is no access number for one-stage dial users, it is impossible to enable the user authorization specific to an individual ID. As a result, if needed, your only choice is to perform an authentication enabling operation on all the one-stage dial users.

If the RADIUS server is to be used for authorization of one-stage dial users, the RADIUS server and the RADIUS Client (VG) must be interoperable at the network layer, the appropriate access number has been configured and the authentication function has been enabled on the RADIUS Client, and the user right list containing all the one-stage dial users has been configured on the RADIUS server.

Perform the following configuration in voice AAA service view.

Table 5-23 Enable the authorization on all the one-stage dial users

Operation	Command
Enable the authorization on all the one-stage dial users.	authorization-did
Disable the authorization on all the one-stage dial users.	undo authorization-did

By default, one-stage dial user authorization is not enabled.

II. Enabling the authorization on two-stage dial users

If the RADIUS server is to be used for authorization of two-stage dial users, you must make sure that the RADIUS server and the RADIUS Client (VG) are interoperable at the network layer, the appropriate access number has been configured and the authentication function has been enabled on the RADIUS Client, and the corresponding user right list has been configured on the RADIUS server.

Perform the following configurations in access number view.

Table 5-24 Enable the authorization on two-stage dial users

Operation	Command
Enable the authorization on two-stage dial users.	authorization
Disable the authorization on two-stage dial users.	undo authorization

By default, authorization of two-stage dial users (users requiring an access number) is not enabled.

& Note:

The differences between two-stage and one-stage dial approaches in enabling the authorization function are the same as those in enabling the authentication function.

5.3.3 Configure Authorization of Local Voice Users by the RADIUS server

By default, if local users are configured in voice AAA service view, call authorization will be implemented locally, with no authorization requests sent to the RADIUS server.

After this command is executed, local users will request the RADIUS server for authorization, and the RADIUS server will respond with the authorization result. The information of voice RADIUS users needs to be configured on the RADIUS server, which performs authorization after authentication succeeds; otherwise the RADIUS will reject the authorization request.

Perform the following configuration in AAA service view.

Table 5-25 Configure authorization of local voice users by the RADIUS server

Operation	Command
Enable local users to request the RADIUS server for authorization	server-authorization
Disable local users from requesting the RADIUS server for authorization	undo server-authorization

By default, local users do not request the RADIUS server for authorization.

5.3.4 Enabling RADIUS Accounting

Although authentication and authorization on two-stage and one-stage dial users are enabled separately, their accounting is enabled together. After enabling accounting on a RADIUS Client (VG), the system will generate the accounting information for the calls from all the one-stage and two-stage dial users and send the information to the specified RADIUS server where the accounting is performed appropriate to the configured accounting policy.

Perform the following configurations in voice AAA service view.

& Note:

If you have configured accounting, you must configure authentication and authorization in corresponding access number for two-stage dial (card number dial process and voice calling number process).The accounting only occurs after the authentication and authorization have succeeded.

Table 5-26 Enable the RADIUS accounting

Operation	Command
Enable the RADIUS accounting on all the dial users.	accounting
Disable the RADIUS accounting on all the dial users.	undo accounting

By default, the RADIUS accounting on dial users is not enabled.

5.3.5 Configuring RADIUS Accounting Information Sending Mode

A RADIUS Client (VG) can process the RADIUS accounting request/response packets in several approaches. At the system level, these approaches are different in the sense that the accounting information is sent to the RADIUS server at different time points and in different ways. You can select the desired accounting approach by setting the method for processing RADIUS accounting request/response packets.

Perform the following configurations in voice AAA service view.

Table 5-27 Configure the accounting information sending method

Operation	Command
Configure an accounting information sending method.	acct-method { none \| start-no-ack \| stop-only \| start-ack }
Restore the default accounting information sending method.	undo acct-method

By default, the start-no-ack method is adopted to process the accounting request/response packets.

& Note:

If the accounting method is set to none, the system will not charge any IP telephone users, no matter whether the accounting function has been enabled by the accounting command. So after enabling the accounting function, you must make sure that the accounting command does not contain the parameter none so that the system can perform accounting for the IP telephone users.

5.4 Displaying and Debugging Voice RADIUS

After completing the configurations described above, you can view the operations of voice RADIUS by executing the display commands in any view to verify the configuration effect.

Perform the following configuration in any view.

Table 5-28 Display and debug the voice RADIUS information

Operation	Command
Clear the state statistics related to the RADIUS server.	reset voice aaa-client statistic
Clear the information related to the VCC software module.	reset voice vcc { all \| call-record \| statistics }
Display the CDRs not successfully sent.	display aaa unsent-h323-call-record
Display the call history.	display voice call-history-record { callednumber called-number \| callernumber caller-number \| cardnumber card-number \| remote-ip-addr ip-address \| last [ last-number ] } [ brief ]
Display the call information of the specified voice subscriber line	display voice call-history-record line line-number
Display the statistics related to the RADIUS server state.	display voice aaa-client statistic
Display information of the local voice user database.	display voice aaa-client local-user
Display the current VoIP RADIUS configurations of the system.	display current-configuration voice [ aaa \| access-number \| cdr \| acct-method ]
Display the call channel state information and the call statistics.	display voice vcc { channel [ channel-number ] \| statistic { all \| error \| ipp \| proc \| rcv \| rds \| timeouts \| vpp } }
Enable the debugging of the VCC module at different levels.	debugging voice vcc { all \| error \| ipp \| proc \| radius \| rcv \| timer \| vpp \| channel channel-number }

5.5 Voice RADIUS Configuration Example

5.5.1 Configuring the Calling Procedure in Two-stage Dialing

I. Network requirements

VG A and VG B communicate over an IP network. VG A is connected to a telephone device via the FXS (POTS) port, and VG B is connected to a telephone device via the FXS (POTS) port. The users at VG A side adopt the two-stage dial approach to call the users at VG B side. They dial “600” first and then called numbers in turn. The configuration of VG B is similar to that of VG A.

II. Network diagram

Figure 5-2 Network for a typical two-stage dial application

III. Configuration procedure

& Note:

This example is discussed assuming that the route between VG A and VG B is reachable.

1) Configurations on VG A

# Create a POTS voice entity (tel. number 010-1001) on the FXS port.

[VGA] voice-setup

[VGA-voice] dial-program

[VGA-voice-dial] entity 1001 pots

[VGA-voice-dial-entity1001] match-template 0101001

[VGA-voice-dial-entity1001] line 0

[VGA-voice-dial-entity1001] quit

# Create a POTS voice entity (tel. number 010-1002) on the FXS port.

[VGA-voice-dial] entity 1002 pots

[VGA-voice-dial-entity1002] match-template 0101002

[VGA-voice-dial-entity1002] line 1

[VGA-voice-dial-entity1002] quit

# Create a VoIP voice entity.

[VGA-voice-dial] entity 0755 voip

[VGA-voice-dial-entity755] match-template 0755....

[VGA-voice-dial-entity755] address ip 2.2.2.2

[VGA-voice-dial-entity755] quit

# Configure the two-stage dialing with the access number 600 and specify it as the calling number procedure.

[VGA-voice-dial] gw-access-number 600

[VGA-voice-dial-anum600] process-config callernumber

[VGA-voice-dial-anum600] return

[VGA] aaa-enable

# Configure a default route and the IP address of the Ethernet interface.

[VGA]ip route-static 0.0.0.0 0 1.1.1.2

[VGA]interface ethernet 0

[VGA-Ethernet0]ip address 1.1.1.1 24

[VGA-Ethernet0] return

2) Configurations on VG B

# Create a POTS voice entity (tel. number 0755-2001) on the FXS port.

[VGB] voice-setup

[VGB-voice] dial-program

[VGB-voice-dial] entity 2001 pots

[VGB-voice-dial-entity2001] match-template 07552001

[VGB-voice-dial-entity2001] line 0

[VGB-voice-dial-entity2001] quit

# Create a POTS voice entity (tel. number 0755-2002) on the FXS port.

[VGB-voice-dial] entity 2002 pots

[VGB-voice-dial-entity2002] match-template 07552002

[VGB-voice-dial-entity2002] line 1

[VGB-voice-dial-entity2002] quit

# Create a VoIP voice entity.

[VGB-voice-dial] entity 010 voip

[VGB-voice-dial-entity10] match-template 010....

[VGB-voice-dial-entity10] address ip 1.1.1.1

[VGB-voice-dial-entity10] quit

# Configure the two-stage dialing with the access number 600 and specify it as the calling number procedure.

[VGB-voice-dial] gw-access-number 600

[VGB-voice-dial-anum600] process-config callernumber

[VGB-voice-dial-anum600] return

[VGB] aaa-enable

# Configure a default route and the IP address of the Ethernet interface.

[VGB]ip route-static 0.0.0.0 0 2.2.2.1

[VGB]interface ethernet 0

[VGB-Ethernet0]ip address 2.2.2.2 24

[VGB-Ethernet0] return

5.5.2 Configuring Card Number Dial Process in Two-Stage Dial Approach

I. Network requirements

In a VoIP voice network that the VG establishes, use the RADIUS server to perform the authentication, authorization and accounting on users that dial the specific access numbers. The route between the VG and the RADIUS server is reachable.

In its attribute dictionary, the attribute field of the RADIUS server contains the proprietary attributes of H3C. The configured access number is “18901”, and the calling users that use this access number are card users. The user list includes such information as card number, password, etc. All the card numbers are ten digits in length and all the passwords are four digits in length.

Configure the access number “18901” and its user attribute parameters on the current VG, perform AAA on the users dialing that access number, the start-ack method is required for accounting, the users are allowed to make four attempts (redial three times) at each dial stage. Up to 200 CDR entries can be retained.

II. Network diagram

Figure 5-3 Network for the application of the two-stage card number dial process

III. Configuration procedure

& Note:

This example is discussed assuming that the route between VG A and VG B is reachable.

1) Configure the VG A to serve as the RADIUS Client

# Configure the IP address of the Ethernet interface

[VGA] interface ethernet 0

[VGA-Ethernet0] ip address 192.168.80.30 24

[VGA-Ethernet0] return

# Use H3C private protocol as the voice RADIUS client protocol type (default configuration).

[VGA] aaa-enable

[VGA] voice-setup

[VGA-voice] aaa-client

[VGA-voice-aaa] clienttype private

[VGA-voice-aaa] return

# Set the RADIUS server address.

[VGA] radius server 192.168.80.100

# Set the RADIUS server key.

[VGA] radius shared-key win

# Set the source IP address of RADIUS packets as when the packets are sent.

[VGA] radius source-ip 192.168.80.30

# Set the access number 18901 and apply the card number dial process on it.

[VGA] voice-setup

[VGA-voice] dial-program

[VGA-voice-dial] gw-access-number 18901

[VGA-voice-dial-anum18901] process-config cardnumber

# Specify the digit numbers of card number and password of the users using the access number respectively to 10 and 4.

[VGA-voice-dial-anum18901] card-digit 10

[VGA-voice-dial-anum18901] password-digit 4

# Allow the dial users to dial for four times at each dial phase, i.e., redial for three times.

[VGA-voice-dial-anum18901] redialtimes 4

# Enable authentication and authorization.

[VGA-voice-dial-anum18901] authentication

[VGA-voice-dial-anum18901] authorization

# Set the accounting method to start-ack.

[VGA-voice-dial-anum18901] quit

[VGA-voice-dial] quit

[VGA-voice] aaa-client

[VGA-voice-aaa] acct-method start-ack

# Enable accounting.

[VGA-voice-aaa] accounting

# Allow the system to retain up to 200 CDR entries.

[VGA-voice-aaa] cdr buffer 200

[VGA-voice-aaa] return

2) Configure the voice entity on VG A

# Create a POTS voice entity (tel. number 010-1001) on the FXS port.

[VGA] voice-setup

[VGA-voice] dial-program

[VGA-voice-dial] entity 1001 pots

[VGA-voice-dial-entity1001] match-template 0101001

[VGA-voice-dial-entity1001] line 0

[VGA-voice-dial-entity1001] quit

# Create a POTS voice entity (tel. number 010-1002) on the FXS port.

[VGA-voice-dial] entity 1002 pots

[VGA-voice-dial-entity1002] match-template 0101002

[VGA-voice-dial-entity1002] line 1

[VGA-voice-dial-entity1002] quit

# Create a VoIP voice entity.

[VGA-voice-dial] entity 0755 voip

[VGA-voice-dial-entity755] match-template 0755....

[VGA-voice-dial-entity755] address ip 2.2.2.2

[VGA-voice-dial-entity755] return

3) Configure the VG B to serve as the RADIUS Client

# Configure the IP address of the Ethernet interface

[VGB] interface ethernet 0

[VGB-Ethernet0] ip address 192.168.80.40 24

[VGB-Ethernet0] return

# Use H3C private protocol as the voice RADIUS client protocol type (default configuration).

[VGB] aaa-enable

[VGB] voice-setup

[VGB-voice] aaa-client

[VGB-voice-aaa] clienttype private

[VGB-voice-aaa] return

# Set the IP address of the RADIUS server.

[VGB] radius server 192.168.80.100

# Set the RADIUS server key.

[VGB] radius shared-key win

# Set the source IP address of RADIUS packets as when the packets are sent.

[VGB] radius source-ip 192.168.80.40

# Set the access number 18901 and apply the card number dial process on it.

[VGB] voice-setup

[VGB-voice] dial-program

[VGB-voice-dial] gw-access-number 18901

[VGB-voice-dial-anum18901] process-config cardnumber

# Specify the digit numbers of card number and password of the users using the access number respectively to 10 and 4.

[VGB-voice-dial-anum18901] card-digit 10

[VGB-voice-dial-anum18901] password-digit 4

# Allow the dial users to dial for four times at each dial phase, i.e., redial for three times.

[VGB-voice-dial-anum18901] redialtimes 4

# Enable authentication and authorization.

[VGB-voice-dial-anum18901] authentication

[VGB-voice-dial-anum18901] authorization

# Set the accounting method to start-ack.

[VGB-voice-dial-anum18901] quit

[VGB-voice-dial] quit

[VGB-voice] aaa-client

[VGB-voice-aaa] acct-method start-ack

# Enable accounting.

[VGB-voice-aaa] accounting

# Allow the system to retain up to 200 CDR entries.

[VGB-voice-aaa] cdr buffer 200

[VGB-voice-aaa] return

4) Configure the voice entity on VG B

# Create a POTS voice entity (tel. number 0755-2001) on the FXS port.

[VGB] voice-setup

[VGB-voice] dial-program

[VGB-voice-dial] entity 2001 pots

[VGB-voice-dial-entity2001] match-template 07552001

[VGB-voice-dial-entity2001] line 0

[VGB-voice-dial-entity2001] quit

# Create a POTS voice entity (tel. number 0755-2002) on the FXS port.

[VGB-voice-dial] entity 2002 pots

[VGB-voice-dial-entity2002] match-template 07552002

[VGB-voice-dial-entity2002] line 1

[VGB-voice-dial-entity2002] quit

# Create a VoIP voice entity.

[VGB-voice-dial] entity 010 voip

[VGB-voice-dial-entity10] match-template 010....

[VGB-voice-dial-entity10] address ip 1.1.1.1

[VGB-voice-dial-entity10] return

# Configure the RADIUS server.

Take the operation in winradius software for example,

# Configure the authentication port, accounting port, and NAS key of the RADIUS server.

Choose [Settings/System] to set the NAS key the same as the one of the RADIUS server configured on the VG.

Set 1812 as the authentication port and 1813 as the accounting port.

# Set the authentication mode.

Choose [Setting/Accounting method/Pre-paid charging] to set the amount of deposit, 100000 cents for example.

# Set the accounting mode.

Choose [Setting/Accounting Method/By Duration] to set the basic charge rate, 30 cents per 60 seconds for example.

# Add an account.

Choose [Operation/Add an Account] to add an account, ensuring that the number of digits of the card number and that of the password are compliant with the configurations on the gateway. For example, take 8801356729 as the card number, and 4477 as the password. Set the prepaid amount to 10,000,000 cents, and the accounting method to “By Duration”

5.5.3 Configuring Local Authentication (No Accounting)

I. Network requirements

Configure the access number 18901 and establish a local voice user database on the VG A with the username as 1234567890 and password as 1234.

II. Network diagram

Figure 5-4 Network diagram for no-accounting configuration

III. Configuration procedure

& Note:

This example is discussed assuming that the route between the VGs is reachable.

1) Configure the VG A to serve as the RADIUS Client

# Configure the IP address of the Ethernet interface

[VGA] interface ethernet 0

[VGA-Ethernet0] ip address 192.168.80.30 24

[VGA-Ethernet0] return

# Set the access number to 18901, and apply the card number dial process on it.

[VGA] voice-setup

[VGA-voice] dial-program

[VGA-voice-dial] gw-access-number 18901

[VGA-voice-dial-anum18901] process-config cardnumber

# Specify the numbers of digit of the card number and password of the users using the access number respectively to 10 and 4.

[VGA-voice-dial-anum18901] card-digit 10

[VGA-voice-dial-anum18901] password-digit 4

# Allow the dial users to dial four times at each dial phase, i.e., redial three times.

[VGA-voice-dial-anum18901] redialtimes 4

# Enable authentication and authorization.

[VGA-voice-dial-anum18901] authentication

[VGA-voice-dial-anum18901] authorization

[VGA-voice-dial-anum18901] return

# Enable local authentication.

[VGA] aaa-enable

[VGA] voice-setup

[VGA-voice] aaa-client

[VGA-voice-aaa] local-user 1234567890 password 1234

# Specify that the system can retain up to 200 CDR entries.

[VGA-voice-aaa] cdr buffer 200

[VGA-voice-aaa] return

2) Configure the voice entity on VG A

# Create a POTS voice entity (tel. number 010-1001) on the FXS port.

[VGA] voice-setup

[VGA-voice] dial-program

[VGA-voice-dial] entity 1001 pots

[VGA-voice-dial-entity1001] match-template 0101001

[VGA-voice-dial-entity1001] line 0

[VGA-voice-dial-entity1001] quit

# Create a POTS voice entity (tel. number 010-1002) on the FXS port.

[VGA-voice-dial] entity 1002 pots

[VGA-voice-dial-entity1002] match-template 0101002

[VGA-voice-dial-entity1002] line 1

[VGA-voice-dial-entity1002] quit

# Create a VoIP voice entity.

[VGA-voice-dial] entity 0755 voip

[VGA-voice-dial-entity755] match-template 0755....

[VGA-voice-dial-entity755] address ip 2.2.2.2

[VGA-voice-dial-entity755] return

3) Configure the VG B

# Configure the IP address of the Ethernet interface.

[VGB] interface ethernet 0

[VGB-Ethernet0] ip address 192.168.80.40 24

[VGB-Ethernet0] return

# Set the access number to 16000, and apply the card number dial process on it.

[VGB] voice-setup

[VGB-voice] dial-program

[VGB-voice-dial] gw-access-number 16000

[VGB-voice-dial-anum16000] process-config cardnumber

# Set the number of digits of card numbers to 10, and set the number of digits of passwords to 4.

[VGB-voice-dial-anum16000] card-digit 10

[VGB-voice-dial-anum16000] password-digit 4

# Allow the dial users to dial four times at each dial phase, i.e., redial three times.

[VGB-voice-dial-anum16000] redialtimes 4

# Enable authentication and authorization.

[VGB-voice-dial-anum16000] authentication

[VGB-voice-dial-anum16000] authorization

[VGB-voice-dial-anum16000] return

# Enable local authentication.

[VGB] aaa-enable

[VGB] voice-setup

[VGB-voice] aaa-client

[VGB-voice-aaa] local-user 1122334455 password 4321

# Specify that the system can retain up to 200 CDR entries.

[VGB-voice-aaa] cdr buffer 200

[VGB-voice-aaa] return

4) Configure the voice entity on VG B

# Create a POTS voice entity (tel. number 0755-2001) on the FXS port.

[VGB] voice-setup

[VGB-voice] dial-program

[VGB-voice-dial] entity 2001 pots

[VGB-voice-dial-entity2001] match-template 07552001

[VGB-voice-dial-entity2001] line 0

[VGB-voice-dial-entity2001] quit

# Create a POTS voice entity (tel. number 0755-2002) on the FXS port.

[VGB-voice-dial] entity 2002 pots

[VGB-voice-dial-entity2002] match-template 07552002

[VGB-voice-dial-entity2002] line 1

[VGB-voice-dial-entity2002] quit

# Create a VoIP voice entity.

[VGB-voice-dial] entity 010 voip

[VGB-voice-dial-entity10] match-template 010....

[VGB-voice-dial-entity10] address ip 1.1.1.1

[VGB-voice-dial-entity10] return

5.6 Voice RADIUS Troubleshooting

Symptom 1: The system cannot make call connection after the accounting function is enabled and the accounting method is set to start-ack, or after the authentication/authorization function is enabled.

Troubleshooting:

l First, use the display current-configuration command to check that all the configurations are correct.

l If there is no problem with the configurations, use the display aaa unsent-h323-call-record command to locate the call disconnection cause.

l If it is the problem of the RADIUS server, execute the display voice aaa-client statistics command to check that the RADIUS server has not gone down and can be successfully pinged, and the configurations of RADIUS server’s address, port number, and key on the VG are consistent with those of the RADIUS server.

l If no errors are found, check that the proprietary attributes of H3C have been included in the attribute dictionary of the RADIUS server.

l Use the radius server command to configure the RADIUS server again. The system may assume that the RADIUS server is already unavailable because the communication with the RADIUS server has just failed. In addition, as the radius timer quiet command is not configured or the quiet period is set to a relatively long time (5 minute by default), the system may assume that the RADIUS server has not resumed its work yet. In this case, use the undo radius server command to delete the configured RADIUS server and the radius server command to configure a new RADIUS server. The configuration of the new RADIUS server can take effect immediately.

Symptom 2: The RADIUS authentication/authorization on a voice call user is always rejected.

Troubleshooting:

l First take the steps in the troubleshooting procedure described above to make the check.

l Then check that the correct user name, password and the assigned service rights have been set on the RADIUS server for the user, making sure that they have taken effect.

l Use the debugging voice vcc command to enable the debugging of the VCC software module and review the message interaction in the entire authentication/authorization process. You will find detailed descriptions about voice call disconnection causes.

Chapter 6 GK Client Configuration

6.1 GK Client Overview

6.1.1 Introduction to GK and GK Client

The IP telephony technology makes use of the Internet to carry the voice information. IP telephone GW is located between a PSTN and an Internet access workstation. It functions to compress audio signals from the PSTN network and transmit them to the remote IP telephone GW across the Internet, as well as receive the IP packets from the Internet and decompress them into the voice signals for the PSTN network.

As defined by ITU-T, GateKeepers (GKs) are H.323 entities that provide such functions as address translation, admission control, bandwidth control and management, zone management, security, call control signaling, and call management for H.323 endpoints, GW, and Multipoint Control Units (MCUs) on a LAN or WAN. Sometimes, they also provide routing control and accounting functions. For all the calls within a GK’s zone of control, it not only delivers call service control but also functions as a focal control point.

The entity components for the complete GK functionality implementation can be divided into Client and Server. GK Client entity usually uses a VG as the hardware bearer. It configures the VG via the CLI, and interacts with the GK Server by sending the ITU-T H.225.0 Registration, Admission, and Status (RAS) messages. Thus, it helps the VG obtain from the GK Server the services of address translation, admissions control, bandwidth management, and gateway management.

So far the GK Server functionality is always delivered on SUN workstations/servers and the GK Client functionality on the VG. For the sake of reliability, a GK Server is always required to provide backup for its GK Clients. Thus, when the communications with the primary GK Server is abnormal, timeout for example, or when the primary GK Server is unavailable, GK Clients can initiate registration requests to the secondary GK Server and carry out the RAS communications.

6.1.2 Communication Protocol Used between GK Server and GK Client

The ITU-T RAS protocol primarily complies with H.323v2 and is used for the information interaction between a GW (GK Client) and a GK Server. Normally, in RAS, it is the GW (GK Client) initiates a request to the GK Server, and the GK Server returns an accept or reject message depending on the actual situation. The GK Server uses the port 1719 as the default port for RAS communications. The following table lists the major RAS messages and their description.

Table 6-1 Major RAS messages

Message type	Message
Registration message	RRQ, RCF, RRJ
Unregistration message	URQ, UCF, URJ
Modification message	MRQ, MCF, MRJ
Admissions message	ARQ, ACF, ARJ
Location message	LRQ, LCF, LRJ
Disengage message	DRQ, DCF, DRJ
Status message	IRQ, IRR, IACK, INAK
Bandwidth message	BRQ, BCF, BRJ
GW resource availability message	RAI, RAC
RAS timer modification message	RIP

6.2 GK Client Configuration

The GK Client configuration task list includes:

l Access voice gatekeeper view

l Enable/Disable the GK Client function

l Configure H.323 GW area ID

l Configure GW alias

l Configure multiple GW-ID mode

l Configure GW source address

l Configure the alias and address of the GK for the GW

l Configure password for GK Client’s registry to GK Server

l Configure security call

6.2.1 Accessing Voice Gatekeeper View

Before you can perform any other GK Client configurations task, you must access voice gatekeeper view first.

Perform the following configuration in voice view.

Table 6-2 Access voice gatekeeper view

Operation	Command
Access voice gatekeeper view.	gk-client

6.2.2 Enabling and Disabling GK Client

Only when an interface has been configured as the H.323 GW interface successfully, can the GK Client function be enabled on it. If a new H.323 GW interface is specified or other related GW parameters (e.g. GW alias and the name of the associated GK) are changed, the GK Client function should be re-enabled so as to update the Client information saved in the GK Server in time.

Perform the following configurations in voice gatekeeper view.

Table 6-3 Enable/disable the GK Client function

Operation	Command
Enable the GK Client function.	ras-on
Disable the GK Client function.	undo ras-on

By default, the GK Client function is disabled.

6.2.3 Configuring Area ID of H.323 GW

The primary purpose of configuring area ID of H.323 in VoIP voice entity view rather than any other views is to facilitate the GK Server to identify the voice entity type. The VoIP voice entity and the GK Server can reach an agreement on how to distinguish voice entity types beforehand. For example, they can agree to use the area ID “1#” to represent audio entities, “2#” to represent the video entities, etc. Thus, when a VoIP voice entity and GK Server carry out communications between them, the GK Server can determine the voice entity type by the received area ID of H.323 GW.

Perform the following configurations in VoIP voice entity view.

Table 6-4 Configure area ID for an H.323 GW

Operation	Command
Configure area ID for an H.323 GW.	area-id string
Delete the area ID of the H.323 GW.	undo area-id

By default, area ID of the H.323 GW is not configured.

& Note:

Up to 30 area ides can be configured.

6.2.4 Configuring GW Alias

GW alias is used for the registration with a GK Server and GW identification. One GW is allowed to have only one alias.

& Note:

After switching between the non-gw-id mode and multi-gwid mode, the configured gw-id features will be lost.

Perform the following configurations in GK Client view.

Table 6-5 Configure an alias for the GW

Operation	Command
Configuring a GW alias.	gw-id namestring
Delete the GW alias.	undo gw-id [ namestring ]

By default, the parameter namestring is null, i.e., no GW alias is configured.

6.2.5 Configuring Multiple GW-ID Mode

In the multiple GW-ID mode, every analog port supports one independent GW-ID number.

& Note:

l If you have enabled the multi-gwid feature, a parameter will be added into the gw-id command to specify the analog port for configuration. You need to configure different gw-ids for every port so that one VG can register to the GK as several VGs. VG 80-20 does not support the multi-gwid feature.

l After switching between the non-gw-id mode and multi-gwid mode, the configured gw-id features will be lost.

l In the multiple GW-ID mode, only POTS voice entities attached to voice subscriber lines configured with GW-ID can register with the GK.

Perform the following configuration in voice GK Client view.

Table 6-6 Configure to enable multiple GW-ID feature mode

Operation	Command
Enable multiple GW-ID mode	multi-gwid
Disable multiple GW-ID mode	undo multi-gwid

By default, GW-ID mode is disabled.

6.2.6 Configuring Source IP Address for GW

Every VG can be configured with a loopback interface. A loopback interface is a virtual (logical) interface that is always UP. The VG that has multiple physical interfaces can make VoIP calls so long as it still has an UP interface.

On a VG, before the address of the loopback interface is specified to be the source address, the voice IP packets sent out from the VG may carry different source addresses, as they can be sent out from different physical interfaces and may adopt different protocols (H.225.0, H.245, or RTP) for communications. Such uncertainty significantly restrains the power of the firewall.

If the loopback interface is configured with the voice interface parameters and its address is specified to be the source IP address of the VG (the so-called source IP address binding), all the voice IP packets sent out from the VG will carry this address as their source address regardless from which physical interface they are sent out and use what communications protocol. The traffic filtering becomes a simple task as a result of the adoption of the same source address that is definite. More important, the firewall can insert its forceful control over the traffic.

In practice, binding source IP address and configuring voice communications parameters on the loopback interface must be used in conjunction and the cooperation of the VGs at the two sides is also required. To put it more specific,

l In order to make the voice IP packets sent from a VG carry the same source IP address, you must bind the address of the loopback interface with the VG as the source IP address and configure the communications parameters on the loopback interface, such as the firewall parameters, the RAS communications parameters for the interaction between the GK Client and the GK Server.

l To allow the VG’s peer to initiate VoIP calls to the VG configured with Loopback source IP address binding, you can configure the source IP address (the address of the loopback interface) bound with the VG as the address of the called number on the VG's peer by using the address command.

Caution:

l You must ensure that the gateway address and remote H.323 entity (gatekeeper, terminal, or MCU) address in a bind are accessible to each other to avoid call failures.

l Only VG 10-40 and VG 10-41 support PPPoE Client, that is, support the dial-bundle-number parameter.

Perform the following configurations in voice GK Client view.

Table 6-7 Configure the source IP address of the GW

Operation	Command
Configure the source IP address of the GW.	gw-address { dial-bundle-number dial-number \| ethernet interface-number \| ip address }
Delete the source IP address of the GW.	undo gw-address { dial-bundle-number \| ethernet \| ip }

By default, the GW is not bound with a source IP address.

6.2.7 Configuring the Alias and IP Address of the GK Controlling the VG

If the GK Client function of the VG is activated, the VG will register the information of the VG with the master GK Server automatically In order to make the VG find the appropriate GK Server device, you must configure the IP address and alias of the master GK Server on the VG.

In case there is communications problem (such as timeout) between the GK Client and the master GK Server or the GK Server is unavailable, the VG can initiate its registration request to the secondary GK Server.

Perform the following configurations in voice GK Client view.

Table 6-8 Configure the alias and address of the GK controlling the GW

Operation	Command
Configure the alias and the IP address of the primary GK Server for the GW.	gk-id gk-name gk-addr gk-ipaddress [ ras-port ]
Delete the alias and IP address of the primary GK Server for the GW.	undo gk-id
Configure the alias and IP address of the secondary GK Server for the GW.	gk-2nd-id gk-name gk-addr gk-ipaddress [ ras-port ]
Delete the alias and IP address of the secondary GK Server for the GW.	undo gk-2nd-id

By default, no primary or secondary GK Server has been specified for the GW. If one has been configured, its RAS communications port will default to 1719.

& Note:

Before you can configure the name and address of the secondary GK Server, you must use the gk-id command to configure the name and address of the primary GK Server.

6.2.8 Configuring Password for GK Client’s Registry to GK Server

In the GK Client’s registry to GK Server, you can configure the password in the RRQ message. In this case, the GK Server checks the password that is in the received request packets. It accepts this request when the password in the message is the same as in the GK Server configuration and returns the RCF packets.

Once you configure the password for GK Client’s registry to GK Server, this password involves in all the registry process.

Perform the following configuration in voice GK Client view.

Table 6-9 Configure password for GK Client’s registry to GK Server

Operation	Command
Configure the password for GK Client’s registry to GK Server	gk-security register-pwd { cipher \| simple } password
Delete the Password for GK Client’s registry to GK Server	undo gk-security register-pwd

By default, no password is configured on GK Client, therefore no password in the registry.

6.2.9 Configuring Security Call

Configuring the GK Client security call is to make the VG of GK Client to pass the GK Server call token. When calling, the calling VG obtains the call token from the calling GK Server and transparently transmits to the called VG which transmits the call token to the called GK Server. The simultaneous token exists among GK Servers. Only the GK Server accepts the call token, can it returns the call accept message to the called VG.

Perform the following configuration in voice GK view.

Table 6-10 Configure security call

Operation	Command
Enable the security call	gk-security call enable
Disable the security call	gk-security call disable

By default, the security call is enabled.

6.3 Displaying and Debugging GK Client

After completing the configurations described above, you can execute the display commands to view the registration status of the GK Client, and thus verify the configuration effect.

Perform the following configurations in any view.

Table 6-11 Display and debug the GK Client information

Operation	Command
Display the registration state information of the GW.	display voice gateway
Enable the debugging on the RAS messages interacted between the GK Client and the GK Server.	debugging voice ras event

6.4 GK Client Configuration Example

6.4.1 Registering the VG as the GK Client with the GK

I. Network requirements

VG A and VG B communicate across an IP network and make use of a GK to dynamically translating telephone numbers into IP addresses.

On VG A, the IP address of the Ethernet interface is 192.168.80.30, and the GK ID is gw01.

On VG B, the IP address of the Ethernet interface is 192.168.80.40, and the GK ID is gw02.

The GK ID for VG A and VG B is gkserver, the GK IP address is 192.168.80.50 and RAS port number is 1719.

II. Network diagram

Figure 6-1 GWs networked with GK

III. Configuration procedure

& Note:

In this scenario, all the discussions are made assuming that the route between VG A and VG B is reachable.

1) Configure VG A

# Configure the Ethernet interface.

[VGA] interface ethernet 0

[VGA-Ethernet0] ip address 192.168.80.30 255.255.255.0

[VGA-Ethernet0] quit

# Create a POTS voice entity.

[VGA] voice-setup

[VGA-voice] dial-program

[VGA-voice-dial] entity 1001 pots

[VGA-voice-dial-entity1001] match-template 1001

[VGA-voice-dial-entity1001] line 0

[VGA-voice-dial-entity1001] return

# Create a VoIP voice entity.

[VGA] voice-setup

[VGA-voice] dial-program

[VGA-voice-dial] entity 2001 voip

[VGA-voice-dial-entity2001] match-template 2001

[VGA-voice-dial-entity2001] address ras

[VGA-voice-dial-entity2001] return

# Enter voice GK view.

[VGA] voice-setup

[VGA-voice] gk-client

# Configure an alias of the GW (gw01), binding of the source IP address of the GW and the Ethernet interface 0, the GK identifier (gkserver) and its IP address (192.168.80.50) as well as the RAS port number (1719).

[VGA-voice-gk] gw-id gw01

[VGA-voice-gk] gw-address ethernet 0

[VGA-voice-gk] gk-id gkserver gk-addr 192.168.80.50 1719

[VGA-voice-gk] ras-on

2) Configure VG B

# Configure the Ethernet interface.

[VGB] interface ethernet 0

[VGB-Ethernet0] ip address 192.168.80.40 255.255.255.0

[VGB-Ethernet0] quit

# Create a POTS voice entity.

[VGB] voice-setup

[VGB-voice] dial-program

[VGB-voice-dial] entity 2001 pots

[VGB-voice-dial-entity2001] match-template 2001

[VGB-voice-dial-entity2001] line 0

[VGB-voice-dial-entity2001] return

# Create a VoIP voice entity.

[VGB] voice-setup

[VGB-voice] dial-program

[VGB-voice-dial] entity 1001 voip

[VGB-voice-dial-entity1001] match-template 1001

[VGB-voice-dial-entity1001] address ras

[VGB-voice-dial-entity1001] return

# Enter voice GK view.

[VGB] voice-setup

[VGB-voice] gk-client

# Configure an alias of the GW (gw02), binding of the source IP address of the GW and the Ethernet interface 0, the GK identifier (gkserver) and its IP address (192.168.80.50) as well as the RAS port number (1719).

[VGB-voice-gk] gw-id gw02

[VGB-voice-gk] gw-address ethernet 0

[VGB-voice-gk] gk-id gkserver gk-addr 192.168.80.50 1719

[VGB-voice-gk] ras-on

3) Configure the GK

Configure the GK according to the actual network environments. The details are omitted here.

6.4.2 Configuring the VG as the GK Client with Multi-GK-ID

I. Network requirements

VG A and VG B communicate across an IP network and make use of a GK to dynamically translating telephone numbers into IP addresses. Here, VG A is configured with multi-GW-ID, and every subscriber line corresponds to one GW-ID.

On VG A, the IP address of the Ethernet interface is 192.168.80.30, the GK for VG A is identified as “gkserver” and its IP address is 192.168.80.50. The RAS port is 1719.

The multi-GW-ID function is enabled on VG A, the interface of the subscriber line 0 corresponds to the alias vga-1001, and the interface of the subscriber line 1 corresponds to the alias vga-1002.

On VG B, the IP address of the Ethernet interface is 192.168.80.40. Other configurations of VG B are the same as those of VG A. The multi-GW-ID function is not enabled on VG B.

II. Network diagram

Figure 6-2 Network of the GK Client with multi-GW-ID

III. Configuration procedure

& Note:

In this scenario, all the discussions are made assuming that the route between VG A and VG B is reachable.

1) Configure VG A

# Configure the Ethernet interface.

[VGA] interface ethernet 0

[VGA-Ethernet0] ip address 192.168.80.30 255.255.255.0

[VGA-Ethernet0] quit

# Create a POTS voice entity.

[VGA] voice-setup

[VGA-voice] dial-program

[VGA-voice-dial] entity 1001 pots

[VGA-voice-dial-entity1001] match-template 1001

[VGA-voice-dial-entity1001] line 0

[VGA-voice-dial-entity1001] quit

[VGA-voice-dial] entity 1002 pots

[VGA-voice-dial-entity1002] match-template 1002

[VGA-voice-dial-entity1002] line 1

[VGA-voice-dial-entity1002] return

# Create a VoIP voice entity.

[VGA] voice-setup

[VGA-voice] dial-program

[VGA-voice-dial] entity 2001 voip

[VGA-voice-dial-entity2001] match-template 2001

[VGA-voice-dial-entity2001] address ras

[VGA-voice-dial-entity2001] return

# Enter voice GK view.

[VGA] voice-setup

[VGA-voice] gk-client

# Enable multi-GW-ID.

[VGA-voice-gk] multi-gwid

# Configure the interface of the subscriber line 0 to correspond to the alias vga-1001, and the interface of subscriber line 1 to correspond to the alias vga-1002.

[VGA-voice-gk] gw-id 0 vga-1001

[VGA-voice-gk] gw-id 1 vga-1002

# Configure the binding of the source IP address of the GW and the Ethernet interface 0, the GK identifier (gkserver) and its IP address (192.168.80.50) as well as the RAS port number (1719).

[VGA-voice-gk] gw-address ethernet 0

[VGA-voice-gk] gk-id gkserver gk-addr 192.168.80.50 1719

[VGA-voice-gk] ras-on

2) Configure VG B

For the configuration on VG B, refer to section 6.4.1 III. 2)“Configure VG B”.

3) Configure the GK

Configure the GK according to the actual network environments. The details are omitted here.

6.5 GK Client Troubleshooting

Symptom: GW fails to register with the GK Server.

Troubleshooting:

l Check that the GK Server and the GW are interoperable at the network layer by executing the ping command;

l Check that the ras-on command has been enabled by executing the display current-configuration command;

l Check that the GK functions of the GK Server have been enabled;

l Check that the area information of the GW has been configured on the GK Server.

l Execute the ras-on command on the GK if the GW unregisters this node.

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