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AR Router Maintenance Guide-FAQ(Voice Service)

Latest reply: Aug 30, 2017 06:32:19 3193 1 0 0 0

2.13  Voice Service

2.13.1  What AR Models Support the Voice Function?

The following AR models support the voice function:
  • V200R001C00:

    Only the AR1220V supports the voice function.

  • V200R001C01:

    The AR1220V, AR1220VW, AR2220, AR2240, and AR3260 support the voice function.

  • From V200R002C00:

    The AR207V, AR207V-P, AR1220V, AR1220VW, AR2220, AR2240, and AR3260 support the voice function.

  • V200R005C10:

    The AR157VW, AR158EVW, AR201VW-P, AR207V, AR207V-P, AR207VW, AR1220V, AR1220VW, AR2220, AR2240 and AR3260 support the voice function.

  • V200R005C30:

    The AR157VW, AR158EVW, AR169FVW, AR169FVW-8S, AR169FGVW-L, AR201VW-P, AR207V, AR207V-P, AR207VW, AR1220V, AR1220VW, AR2220, AR2240 and AR3260 support the voice function.

  • V200R006C10:

    AR157VW, AR158EVW, AR169FVW, AR169FVW-8S, AR169FGVW-L, AR201VW-P, AR207V, AR207V-P, AR207VW, AR1220V, AR1220VW, AR1220EV, AR1220EVW, AR2220, AR2240 and AR3260 support the voice function.

2.13.2  Which Voice Service Modes Do AR Series Routers Support?

Currently, AR series routers support the following two voice service modes: Session Initiation Protocol access gateway (SIP AG) ,H.248 access gateway (H248AG) and private branch exchange (PBX). You can run the display voice service-mode command to view the voice service modes of the current AR.

To switch between modes, run the service-mode { sipag | h248ag | pbx } command in the voice view.

6711c418de1945a6a6d6c41745fe9374 NOTE:

Before switching voice modes, delete all configuration data in current mode. Change and save the new configuration, and reboot the device to enable the new mode.

2.13.3  What Are SIP AGs and PBXs, and What Are the Differences Between Them?

Access devices on the IP Multimedia Subsystem (IMS) provide various access modes and convert various services into a uniform format that can be transmitted. On the IMS, an access device is called access gateway (AG). AGs play an important role on the IMS and are connected to users.

AGs and media gateway controllers (MGCs) use Session Initiation Protocol (SIP). SIP-based AGs are called SIP AGs.

A traditional private branch exchange (PBX) manages the incoming and outgoing calls of an enterprise. It connects the enterprise to the Public Switched Telephone Network (PSTN) and provides services for devices such as telephones, fax machines, and modems. It allows users in the enterprise to call each other using extension phones and routes inter-office calls to the PSTN through a trunk line. Traditional PBXs cannot meet the requirements of computer telephony integration (CTI) and voice over IP (VoIP). In addition, these PBXs are expensive and do not use standard and open platforms, creating difficulties in interconnecting PBXs of different vendors. IP PBXs overcome the limitations of traditional PBXs. IP PBXs are based on the IP protocol and provide both local exchange and IP user access functionality. IP PBXs integrate the voice communications system of an enterprise into the enterprise's data network so that the enterprise can build a uniform voice and data network connecting branches, offices, and staff around the world.

The AR1200/2200/3200 can function as a PBX to provide traditional PBX functions and IP PBX functions.

SIP AGs and PBXs have the following differences:
  • PBXs have two functions. PBXs can serve as switching devices on a private network to locate voice users' addresses and process other services. PBXs also allow users on the private network to access the IMS or PSTN through SIP trunks.
  • SIP AGs provide only the second function of PBXs. SIP AGs can be considered to be a plain old telephone service (POTS) that can register with the IMS using SIP trunks. PBXs use SIP trunks to register with other SIP servers. SIP AGs register with the IMS using POTS user numbers. PBXs and SIP AGs provides the same services. However, PBXs provide the services by themselves.

2.13.4  What Are User Identifiers?

User identifiers include device IDs and user IDs.

A device ID is the ID of a device used by users. The devices can be a plain old telephone service (POTS) phone or a user agent device, such as IP phones and software.

A user ID is a unique number assigned to a user. A user ID is the owner of a device ID, and indicates a user's identity. A user ID and a Session Initiation Protocol (SIP) server domain constitute a URL. A device ID can belong to multiple user IDs.

Each user has a fixed user identifier, which is bound to a unique device ID. Usually, a user identifier is the global number or long code for the user. Users can dial long codes to call each other, which is the common dial-up mode. Internal numbers or private numbers are provided to users inside the same group, such as an enterprise, company, group organization, school, or hotel. Users in the same group can dial each other's short numbers or private numbers.

2.13.5  What Are Groups and Enterprises?

The central exchange (Centrex), which is also called group, is a function of telephone exchange. The Centrex groups some users on a private branch exchange (PBX), and provides various functions of a user-specific exchange and other special services and functions to the user group. A user group does not have a dedicated exchange. The internal and external switching of users takes place inside the telephone exchange. Therefore, the Centrex is also called central user exchange or virtual user exchange.

The group function is not only applicable to enterprises or departments with lots of employees, but also applicable to scenarios in which user extensions are widely dispersed. The group function provides flexible networking, and makes it easy to increase or decrease network capacity. Enterprises and departments that use the group function can enjoy specialized services in addition to services on the Public Switched Telephone Network (PSTN).

Group users have two numbers. One is an internal number which is called the private number. The other is an external number which is called the long code. Users in the same group can dial private numbers to call each other. The private numbers have the highest priority. Users inside a group dial private numbers by default. If a user inside a group wants to call an external user, the user must first dial the call prefix of the group that the external user belongs to. The group ID is an important property of group users. You must specify the group ID to configure group users or modify group properties.

The range of the enterprise function is larger than that of the group function. A group can only belong to an enterprise. Users inside an enterprise can belong to any groups, or not belong to a group. Users inside an enterprise can call each other, but cannot call users of another enterprise.

The group and enterprise functions group users on the service layer. The DN set function groups numbers on the switch layer.

2.13.6  What Are DN Sets?

A dial number (DN) set is also called a dial plan. A DN set contains a set of numbers that are processed together. Only numbers, call prefixes, trunks, and routes that belong to the same DN set are processed. Two users can call each other only when the two users belong to the same DN set.

DN sets determine the home of users when associated with country codes and region codes, and determine the dial rules for users when associated with call prefixes. DN sets can be used to divide a physical network into multiple logical networks even in the same device, and are similar to virtual local area networks (VLANs) in a local area network (LAN).

2.13.7  What Are Call Prefixes?

A call prefix is an important property of the call service. The call prefix defines the number rule of the call connection, and reflects the numbering and routing solution of a switch office. The call prefix determines the service type of a dial rule (a basic service or a supplementary service; an office internal service, a local call, or a toll call), and specifies the length of numbers. The call prefix can also be used to control call rights.

A private branch exchange (PBX) system verifies the validity of the numbers dialed by users based on call prefixes, and processes the numbers. Proper call prefixes are critical to a successful service configuration. Users' calls must match call prefixes so that calls can be connected. One or more call prefixes may be matched during a call (a basic service or a supplementary service).

2.13.8  What Are Trunks?

A trunk is a logical link between two exchange offices. Inter-office calls must be transmitted by trunks. The device supports the following trunk modes:
  • PRA trunk

    A PRA trunk is connected to a VE1 interface and is usually used to connect two PBXs.

  • AT0 trunk

    An AT0 trunk is connected to an FXO interface for accessing the PSTN network. The PBX enterprise needs to use the AT0 trunk for making calls to non-enterprise numbers or making long-distance fixed-line calls.

  • SIP trunk

    A SIP trunk is established on an IP link and connects a PBX and an IMS. There are three types of SIP trunks: SIP AT0 trunk, AIP PRA trunk and common SIP trunk.

  • E1R2 trunk

    The E1R2 trunk is bound to VE1 interfaces and is usually used to connect two PBXs. The E1R2 trunk can be connected to a peer device using R2 signaling.

    6711c418de1945a6a6d6c41745fe9374 NOTE:

    The device supports the E1R2 trunk from V200R002C00.

  • H323 trunk

    An H323 trunk is established on an IP link. It connects a PBX and an IMS or connects PBXs in different areas.

    6711c418de1945a6a6d6c41745fe9374 NOTE:

    The device supports the H323 trunk from V200R002C01.

  • BRA trunk

    A BRA trunk is bound to a BRA interface and connects a PBX and an ISDN.

    6711c418de1945a6a6d6c41745fe9374 NOTE:

    The device supports the BRA trunk from V200R002C02.

2.13.9  What Are the Functions of Call Routes?

Call routes bind outgoing call prefixes to trunk groups to map outgoing call prefixes and trunks.

2.13.10  What Are the Properties of the Voice Service, and What Are the Relationships Among These Properties?

The dial number (DN) set and enterprise are important properties of user IDs, trunks, call prefixes, and routes. User IDs manage user information. Trunks, call prefixes, and routes ensure that calls are connected and released properly.

Figure 2-12  Intra-office call
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Figure 2-13  Outgoing call
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  • Intra-office call: Generally, a device is considered to be an exchange office. If both parties of a call are in the same exchange office, the call is an intra-office call.
  • Outgoing call: Voice signals are transmitted over the trunk between two exchange offices.

2.13.11  What Are the Functions of the DSP Module?

DSP is short for digital signal processing. The DSP module supports the following functions: playing voice such as the dial tone and RBT tone, mixing voice, collecting digits, and encoding and decoding voice.

To use the voice feature on the AR2200 and AR3200 series routers, you are advised to install the DSP module.

2.13.12  Are Input Voltages of All FXS Interfaces the Same?

Input voltages of FXS interfaces vary depending on the device.

2.13.13  Can an ISDN Phone Supply Power to Itself?

An ISDN phone's power can be supplied by itself or using the remote PBX. By default, no power is supplied to the AR's 2BST (BRA) board connecting to the ISDN phone. To supply power to the 2BST (BRA) board, run the remote-power command. The detailed configuration is as follows:
<Huawei> system-view 
[Huawei] voice 
[Huawei-voice] port bra 3/0/0 
[Huawei-voice-bra3/0/0] remote-power enable

2.13.14  How Can I View the Usage of the DSP Resources?

  • V200R002C00 and earlier versions

    Run the display voice dsp-dimm stateslot/dsp-index  |  channel  slot/dsp-index/channel   }  command. The idle item in the command output shows the remaining DSP resources. For example:
    <Huawei> display voice dsp-dimm state 15/0 Symbol: 0-idle $-G.711 busy A-All busy W-Wastage X-fault @-IP loopback *-PCM loopback #-prohibited Channel NO.  DSP channel state ---------------------------------------------------------------------- 
    
      0000-0049  00000 00000 00000 00000 00000 00000 00000 00000 00000 00000 
    
      0050-0099  00000 00000 00000 00000 00000 00000 00000 00000 00000 00000 
    
      0100-0149  00000 00000 00000 00000 00000 000                           
    
      Total:                                                                 
    
          128 DSP channel                                                    
    
          128 idle, 0 G.711 Busy, 0 All Busy, 0 Wastage, 0 Fault,            
    
          0   IP loopback, 0 PCM loopback, 0 prohibited
    
    In the preceding information, 128 idle indicates that 128 DSP channels are idle and the current DSP resources are sufficient.
  • Later versions than V200R002C00

    Run the display voice dsp stateslot/dsp-index  |  channel  slot/dsp-index/channel   }  command. The idle item in the command output shows the remaining DSP resources. For example:
    <Huawei> display voice dsp state 15/0 Symbol: 0-idle $-G.711 busy A-All busy W-Wastage X-fault @-IP loopback *-PCM loopback #-prohibited Channel NO.  DSP channel state ---------------------------------------------------------------------- 
    
      0000-0049  00000 00000 00000 00000 00000 00000 00000 00000 00000 00000 
    
      0050-0099  00000 00000 00000 00000 00000 00000 00000 00000 00000 00000 
    
      0100-0149  00000 00000 00000 00000 00000 000                           
    
      Total:                                                                 
    
          128 DSP channel                                                    
    
          128 idle, 0 G.711 Busy, 0 All Busy, 0 Wastage, 0 Fault,            
    
          0   IP loopback, 0 PCM loopback, 0 prohibited
    
    In the preceding information, 128 idle indicates that 128 DSP channels are idle and the current DSP resources are sufficient.

2.13.15  Is There a Possibility that the AR Cannot Detect the DTMF Tone Using G.729?

G.729 is a lossy audio data compression and may cause the AR's failure to detect the DTMF tone.

2.13.16  Does the Voice Feature Require a License?

To use the voice PBX function, apply for and install a commercial license. You can run the display license state command to view the license status.

2.13.17  What Functions Does an FXS Interface Provide in the Voice Feature?

Foreign exchange station (FXS) access is analog access. FXS sets up a connection between a PSTN network and a POTS phone or fax machine.

2.13.18  Why Is an ISDN Phone Unavailable?

An ISDN phone may be unavailable because of the following reasons:
  • The ISDN phone power is not supplied properly.

    An ISDN phone's power can be supplied by itself or using the remote PBX. By default, no power is supplied to the AR's 2BST (BRA) board connecting to the ISDN phone. To supply power to the 2BST (BRA) board, run the remote-power command. The detailed configuration is as follows:
    <Huawei> system-view 
    [Huawei] voice 
    [Huawei-voice] port bra 3/0/0 
    [Huawei-voice-bra3/0/0] remote-power enable
    
  • The cable sequence is incorrect.

    In most cases, ISDN crossover cables are used to connect the AR and an ISDN phone.

If the fault persists, contact Huawei technical support personnel.

2.13.19  Why Is the Busy Tone Played After I Dial a Phone Number?

  • (Versions later than V200R002C00) The possible reasons are:

    • The local call prefix is not correctly configured.

      When you call a local phone number, the busy tone indicates the local call prefix is not configured or incompletely configured. Ensure that the following mandatory parameters of the call prefix are configured correctly: call-type, digit-length, and prefix.

    • The called party's number does not exist.

      You can run the display voice pbxuser or display current-configuration command to check the existence of the called user or configuration integrity. For a POTS user, mandatory pbxuser parameters port and telno must be set correctly. For a SIPUE user, mandatory pbxuser parameters sipue and telno must be set correctly.

    • The called party has configured the call restriction services.

      Run the display voice pbxuser or display current-configuration command to verify that the called party has configured the call restriction services such as call rejection and selective call acceptance.

      If the fault persists, contact Huawei technical support personnel.
  • (V200R002C00 and earlier versions) The possible reasons are:

    • The local call prefix is not correctly configured.

      When you call a local phone number, the busy tone indicates the local call prefix is not configured or incompletely configured. Ensure that the following mandatory parameters of the call prefix are configured correctly: enterprise, dn-set, centrex, prefix, call-type, maximum-length, and minimum-length, and the enterprise, dn-set, and centrex are consistent with planned information about the calling and called parties.

    • The called party's number does not exist.

      Run the display voice dialno or display current-configuration command to verify that the called party exists and the mandatory parameters of dialno including pbxuser, port, and dn-set are configured.

    • The called party has configured the call restriction services.

      Run the display voice dialno or display current-configuration command to verify that the called party has configured the call restriction services such as call rejection and selective call acceptance.

      If the fault persists, contact Huawei technical support personnel.

2.13.20  Why Is the Busy Tone Heard When I Pick up the Phone?

The possible reasons are:
  • The license is not correctly installed.

    To use the voice PBX function, apply for and install a commercial license. Run the display license state command to verify that the license status is normal.

  • No valid PBX user is configured.

    Ensure that the PBX device identifier and PBX user identifier are correctly configured. For details, see the Configure the PBX User in the voice configuration manual.

  • The DSP resources are insufficient.

    Run the display voice dsp-dimm stateslot/dsp-index  |  channel  slot/dsp-index/channel  }  command. The idle item in the command output shows the remaining DSP resources. For example:
    6711c418de1945a6a6d6c41745fe9374 NOTE:

    For V200R002C00 and earlier versions, run the display voice dsp-dimm stateslot/dsp-index  |  channel  slot/dsp-index/channel   } command to view the usage of DSP resources. For versions later than V200R002C00, run the display voice dsp stateslot/dsp-index  |  channel  slot/dsp-index/channel   } command to view the usage of DSP resources. The following uses a version earlier than V200R002C00 as an example:

    <Huawei> display voice dsp-dimm state 15/0 Symbol: 0-idle $-G.711 busy A-All busy W-Wastage X-fault @-IP loopback *-PCM loopback #-prohibited Channel NO.  DSP channel state ---------------------------------------------------------------------- 
    
      0000-0049  00000 00000 00000 00000 00000 00000 00000 00000 00000 00000 
    
      0050-0099  00000 00000 00000 00000 00000 00000 00000 00000 00000 00000 
    
      0100-0149  00000 00000 00000 00000 00000 000                           
    
      Total:                                                                 
    
          128 DSP channel                                                    
    
          128 idle, 0 G.711 Busy, 0 All Busy, 0 Wastage, 0 Fault,            
    
          0   IP loopback, 0 PCM loopback, 0 prohibited
    
    In the preceding information, 128 idle indicates that 128 DSP channels are idle and the current DSP resources are sufficient.

If the fault persists, contact Huawei technical support personnel.

2.13.21  What Is the SIP AT0 Trunk?

The local device uses SIP to connect to the peer device through the SIP AT0 trunk over the IP network.

PBX users can use SIP AT0 trunks to perform SIP calls or other applications through the IMS or softswitch. One SIP AT0 trunk can be used by only one user at a time. Incoming calls over the SIP AT0 trunk are forwarded to the local device or switchboard by default.

2.13.22  What Are Benefits of Using SIP AGs as Voice Gateways

The AR router supports the SIP AG function to provide VoIP services, which brings the following benefits:
  • Lower costs: Traditional call and fax services use the circuit switching mode, in which each call session occupies a circuit. Therefore, fees are high, especially when users make long-distance calls. VoIP services use the packet switching mode, and call fees are much lower.
  • High call quality: SIP AGs ensure call completion rate, voice quality, and service types by configuring QoS.
  • Smooth upgrade/capacity expansion: A VoIP system is compatible with the existing telephony systems and office platforms, and the service capacity can be increased when the enterprise scale expands.

2.13.23  What Is the CID Service?

The Calling Identity Delivery (CID) service or the Calling Line Identification Presentation (CLIP) service enables the AR router to send the calling party's number and call time to the called party and the calling party's number and call time displayed on the terminal device. This service requires phones supporting CID or calling number identifier between a common phone and a user line. The phones supporting CID can save numbers from multiple groups of incoming calls and provide number query and callback.

2.13.24  What Is G.711?

G.711, also known as Pulse Code Modulation (PCM), is a commonly used waveform codec. G.711 uses a sampling rate of 8,000 samples per second. Non-uniform quantization (logarithmic) with 8 bits is used to represent each sample, resulting in a 64 kbit/s bit rate. G.711 defines two main compression algorithms, the µ-law algorithm (used in North America & Japan) and A-law algorithm (used in Europe and the rest of the world). A-law encoding takes a 13-bit signed linear audio sample as input. μ-law encoding takes a 15-bit signed linear audio sample as input.

G.711 is simple, and provides short delay and high audio quality. G.711 has high data rate and is sensitive to errors on channels.

The UMG8900 supports G.711, A/μ conversion, and interworking with the PSTN.

2.13.25  What Trunks Does the AR Router Support?

The AR router supports the following trunks: PRA, BRA, AT0, SIP, E1R2, and H323 trunks.

2.13.26  Why Is Service (Such as Voice) Interrupted After Being Configured with NAT or Firewall

The aging time of session table is shorter than the aging time of the service. The session table is aged out, while the service is not. The service packets sent after session table aging are discarded, so the service is interrupted. Run the firewall-nat session aging-time command to increase the TCP/UDP timeout interval.

2.13.27  Do IP Phones Require External Power Supplies When Connected to AR Series Routers?

When connected to the AR1220V, AR1220W, AR1220VW, IP phones without external power supplies must be connected to FE4 through FE7 interfaces on the main control board. The Routers must be connected to external power over Ethernet (PoE) power supplies.When connected to other AR Series Routers, IP phones must be equipped with external power supplies.

2.13.28  Are There Any Special Configuration Requirements When Connecting IP Phones to an AR1200?

When connecting IP phones to the interfaces on the main control board of an AR1200, configure the virtual local area network (VLAN) priority to ensure the priority of packets.

2.13.29  Is the CT License Required When the IP Phone Connects to the AR?

The CT license is used to implement call transfer over a trunk. That is, calls received over one trunk are forwarded over another trunk. The CT license is not required when the IP phone connects to the AR.

2.13.30  The AR Connects to IP Phones and AT0 Trunk Is Used. Do Value-added Service Packages for Voice Services Need to Be Configured?

The AR must work in PBX mode when the AR connects to the PSTN through the AT0 interface. After the value-added service package for voice services is configured, you need to configure CM&BEST License.

2.13.31  The AR Connects to the TDM PBX VE1. Is the CT License Required?

The CT license is used to implement call transfer over a trunk. That is, calls received over one trunk are forwarded over another trunk. The AR connects to the TDM PBX VE1 as follows:
  • When the AR working in SIP AG mode connects to the TDM PBX VE1, you only need to configure the value-added service package for voice services.
  • When the AR working in PBX mode connects to the TDM PBX VE1, users connected to the TDM PBX communicate with only local users. After the value-added service package for voice services is configured, you need to configure CM&BEST License. The CT license is not required.
  • When the AR working in PBX mode connects to the TDM PBX VE1, users connected to the TDM PBX communicate with external users through the AR. After the value-added service package for voice services is configured, you need to configure CT License.

2.13.32  After an IP Phone Makes an Outgoing Call Through the AT0 Trunk, the Ringback Tone Is Heard After About 8s. Why?

If the trunk function is not used, change the dialup timer value of the AT0 trunk to 0. The AR does not need to spend 5s on detecting the ringback tone. A user needs to wait for only 3s to hear the ringback tone.

Run the pbx number-parameter 40 0 command in the voice view to set parameter 40 to 0.
6711c418de1945a6a6d6c41745fe9374 NOTE:
The modification may cause failures of calls transferred through trunks. The pbx number-parameter 40 0 command can be used only when POTS or SIP phones make outgoing calls through AT0 trunks.

2.13.33  Outgoing Calls Can Be Made, but Incoming Calls Fail to Be Connected on the AR Using a SIP IP Trunk

Check whether the incoming prefix is correctly configured.

  1. Check whether the remote IP address and port number of the SIP IP trunk are correct.
  2. Check whether the configuration is valid. Run the display voice trunk-group trunk-group-name command to check trunk group information.
  3. Verify that the local incoming prefix and incoming number change rules are correct.
  4. Verify that the peer outgoing prefix and route number change rules are correct.

2.13.34  Intra-office Calls on the Device as the IP PBX Are Normal, But Calls Between Two Devices Fail to Be Connected

This fault is often caused by the incorrect configuration.

Perform the following operations:
  1. Check whether the country code and area code at both ends are consistent.
  2. Check whether users, prefixes, enterprises that trunks belong to, and DN sets at both ends are consistent.
  3. Check whether the prefix configuration is correct, whether call rights are correct, and whether the home area attribute of the call prefix is intra-office.
  4. Check whether the inter-office trunk configuration is correct and whether the trunk status is normal.
  5. Check whether the call route configuration is correct and whether the inter-office prefix is bound to the trunk group.

2.13.35  A SIP AT0 Trunk Fails to Be Registered with the IMS

Check whether the registered account and password are correct.

Check the format of the To header field on the SIP AT0 trunk. During registration, number regulation is enabled for SIP registration messages by default. The format of the To header field is Invite "3001;phone-context=+86755@huawei". The IMS may not support this format.

Run the number-parameter 27 0 command in the SIP AT0 trunk view to set parameter 27 to 0. If parameter 27 is set to 0, number regulation is disabled.

2.13.36  A SIP AG User Fails to Call a PBX User

A SIP AG is similar to a voice gateway connected to a PBX, and a SIP AG user is registered with the SIP server on the PBX. That is, the SIP AG user is similar to a SIP UE user on the PBX.

When a SIP AG user calls a PBX user, you need to correctly configure call prefixes. The inter-office trunk does not need to be configured.

Assume that a SIP AG user's number is 2200 and a PBX user's number is 1100. The local call prefixes 11 and 22 need to be set.

Check the SIP AG configuration.
  1. Check whether the SIP AG interface configuration is correct and whether the registration status is normal.
  2. Check whether the user configuration on the SIP AG interface is correct.
  3. Check whether the FXS interface on the SIP AG is normal.
Check the PBX configuration.
  1. Check whether the SIP server configuration is correct and whether the SIP server status is normal.
  2. Check whether the prefix configuration is correct.
  3. Verify that the SIPAG server has registered.
  4. Verify that the SIPAG user on the PBX has been configured correctly.
  5. Verify that the license is valid.

2.13.37  Why the FXO Interface Cannot Identify Onhook Signals of the Remote PSTN User?

Currently, the AR with the voice package cannot identify onhook signals of other countries. By default, the AR can identify only onhook signals of China.

The voice file needs to be customized at each site. Contact the service provider.

2.13.38  Can the AR Connect to the PBX?

A PBX often provides E1 (DB9) and FXO (RJ11) interfaces. E1 and FXO interfaces can connect to the AR.

2.13.39  Does the Device Support Voice and Video?

The device must connect to video phones to support voice and video services. The device supports only the following IP phones:
  • MIEEGXP14001
  • MIEEGXP14501
  • MIEEGXP21001
  • MIEEGXP21101
  • MIEEGXP21201

2.13.40  Voice and Video Services Are Unsmoothly Transmitted When MP Services Are Deployed. How Is This Problem Solved?

When MP services are deployed, QoS LLQ queues are often used in the uplink and special flows enter LLQ queues. Low delay and packet loss ratio are ensured for LLQ queues.

To ensure that voice and video services are rapidly and smoothly transmitted, enable LFI on the MP-group interface.

Run the ppp mp lfi command in the MP-group interface view.

2.13.41  Why Does an IP Phone Fail to Be Called After the RBT Service Is Enabled in an H.323 Trunk Scenario?

In Figure 2-14, RouterA functions as a PBX and is connected to the IP phone of User A. RouterA communicates with the remote PBX through an H.323 trunk. After the RBT service is enabled on RouterA, User B calls User A. The call is unexpectedly released after User A picks up the phone.

6711c418de1945a6a6d6c41745fe9374 NOTE:
If User A uses an analog telephone, calls are not released unexpectedly after User A picks up the phone.
Figure 2-14  RBT service over the H.323 trunk
137bbe1ec0aa4ebf99c6bcc095274492

The cause is that the early media renegotiation function of H.323 is disabled on RouterA. The RBT service negotiation fails, and calls are unexpectedly released.

Run the pbx number-parameter command and set the value of the 7th control point to 1 to enable the H.323 trunk to initiate renegotiation in the early media scenario.

Run the following commands:

<Huawei> system-view
[Huawei] voice
[Huawei-voice] pbx number-parameter 7 1

2.13.42  Why Do Models Supporting Voice Functions Have No Web Configuration?

AR audio modes include PBX, SIP AG, and H.248 AG. Only the PBX mode supports the web configuration.

When the device working mode is switched to SIP AG or H.248 AG, the Voice Management node in the web system is hidden.

You can run the display voice service-mode command to view the current device working mode.

2.13.43  What Are Differences Between V200R003 and V200R005 License Control Items?

V200R003:
<Huawei>display license resource usage 
 Activated License: sd1:/xxxxxxxx.dat
 FeatureName    | ConfigureItemName       | ResourceUsage
 CRFEA1            Test                        0/0
 CRFEA1            LAR0SSLVPN00                0/2
 CRFEA1            LAR0IVR00                   0/24  //Control the number of concurrent IVR users when the IVR service is enabled. 
 CRFEA1            LAR0CT00                    0/128  //Control the number of tandem calls that a trunk can bear. 
 CRFEA1            LAR0CM00                    0/128  //Control the number of local user calls.
V200R005:
[Huawei-voice]display license resource usage 
 Activated License: usb1:/xxxxxxx.dat
 FeatureName    | ConfigureItemName       | ResourceUsage
 CRFEA1            LAR0URLF00                  0/0
 CRFEA1            LAR0IPS00                   0/0
 CRFEA1            LAR0SSLVPN00                0/2
 CRFEA1            Test                        0/0
 CRFEA1            LUCE1VMAIL00                0/64  //Number of voice mailboxes (used/total)
 CRFEA1            LUCE1VCONF00                0/12  //Number of concurrent voice conferences
 CRFEA1            LUCE1ULSCN00                0/64  //Number of SoftConsole users
 CRFEA1            LUCE1ULIPP00                8/64  //Number of SIP UE users
 CRFEA1            LUCE1ULANP00                2/8   //Number of analog users
 CRFEA1            LUCE1SBEST00                0/64  //Number of SRST/BEST users

2.13.44  Does the AR Support the Voice Mailbox Function? How Many Messages Does the AR Support at Most?

The AR supports the build-in voice mailbox function, and saves voice messages to the built-in flash memory, USB flash drive, or SD card. The number of voice messages supported varies according to the capacity of the storage media.

The size of a voice message is calculated as follows:

Size of a voice message of a single user = [Size of PCM linear encoding voice data per second (8 KB) x Maximum duration of a voice message (120s) + File header and index (0.1 KB)] x Maximum number of voice messages for a single user (20) = 19202 KB (that is, about 19 MB)
6711c418de1945a6a6d6c41745fe9374 NOTE:
The preceding formula assumes that the default values are used. By default, the duration of a voice message is 120s, and the maximum number of voice messages for each user is 20.

2.13.45  Does the AR Support SRTP and TLS?

  • In V200R002, the AR does not support the Secure Real-time Transport Protocol (SRTP) or Transport Layer Security (TLS).
  • In V200R003, the AR supports TLS, but not SRTP.
  • In V200R005, V200R006, and V200R007, the AR does not support SRTP or TLS.

2.13.46  What Should I Do If the Hookflash Operation Fails?

  • If a call is released immediately after the hookflash operation, the upper limit of the hookflash detection duration is so small that the hookflash event is converted into a release event. You can increase the flash-hook upper value to increase the hookflash detection duration range.
    [Huawei-voice]flash-hook upper 1000
    [Huawei-voice]
  • If the hookflash operation does not take effect, the lower limit of the hookflash detection duration is so large that the hookflash event of the phone cannot be detected. You can decrease the flash-hook lower value to increase the hookflash detection duration range.
    [Huawei-voice]flash-hook lower 70
    [Huawei-voice]

2.13.47  What Should I Do If I Cannot Enter Voice Commands?

The possible causes are as follows:
  • The AR does not work in the PBX mode in which voice commands cannot be entered.

    You can run the display voice service-mode command to view the current device working mode.

  • The AR model does not support voice functions.

    For models supporting voice functions, see 2.13.1 What AR Models Support the Voice Function?.

2.13.48  Why Does a VoIP Address Pool Need to Be Configured? How Are IP Addresses Selected?

A VoIP address pool defines the IP addresses that can be used in the VoIP scenario, including signaling and media addresses. One feature of VoIP is that signaling and media are separated. Therefore, signaling and media addresses must be configured simultaneously. It is recommended that loopback addresses be used as signaling and media addresses because the loopback addresses are always in the Up state. Using loopback addresses prevents exceptions in some scenarios, for example, an IP phone re-registers after the SIP server address goes Down and Up.

2.13.49  What Are Differences Among G.729, G.729A, G.729B, and G.729AB?

  • G.729B/G.729AB

    If the AnnexB parameter is carried or the AnnexB value is yes, voice activity detection (VAD) of G.729 is enabled.

  • G.729/G.729A

    If the AnnexB value is no, VAD of G.729 is disabled.

By default, the INVITE request initiated by the AR does not carry the AnnexB parameter, and silence suppression is not enabled for DSP; that is, the AR uses G729/G729A at the initial phase. After the peer device returns a codec type supported by the AR, this codec type is used regardless of whether the AnnexB parameter is used.

2.13.50  What Are Differences Between a Registered Trunk and an Unregistered Trunk? What Are Their Advantages and Disadvantages?

  • An unregistered trunk is a P2P trunk. In the unregistered trunk mode, two devices have no registration relationship. In general, only the IP addresses, port numbers, and domains of the two devices need to be specified.
  • In the registered trunk mode, the AR functions as a SIP user of the peer device and is registered with the peer network.

    The registered trunk status can be obtained on the peer network. The registered trunk can be managed because the AR functions as a user who belongs to the peer network. Another advantage of the registered trunk lies in that the core network does not need to be configured, and that only an account needs to be assigned for the AR to register. In the unregistered trunk mode, the trunk needs to be configured on the core network but a user resource can be saved.

From group: Router
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WoodWood
Created Aug 30, 2017 06:32:19

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