HP MSR2000/3000/4000 Router Series

Transcription

1 HP MSR2000/3000/4000 Router Series Voice Configuration Guide (V7) Part number: Software version: CMW710-R0007P02 Document version: 6PW

2 Legal and notice information Copyright 2013 Hewlett-Packard Development Company, L.P. No part of this documentation may be reproduced or transmitted in any form or by any means without prior written consent of Hewlett-Packard Development Company, L.P. The information contained herein is subject to change without notice. HEWLETT-PACKARD COMPANY MAKES NO WARRANTY OF ANY KIND WITH REGARD TO THIS MATERIAL, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE. Hewlett-Packard shall not be liable for errors contained herein or for incidental or consequential damages in connection with the furnishing, performance, or use of this material. The only warranties for HP products and services are set forth in the express warranty statements accompanying such products and services. Nothing herein should be construed as constituting an additional warranty. HP shall not be liable for technical or editorial errors or omissions contained herein.

3 Contents Configuring analog voice interfaces 1 FXS interface 1 FXO interface 1 E&M interface 1 Configuration task list 1 Configuring basic functions 2 Configuring call progress tones 2 Configuring an FXS interface 3 Configuring CID 3 Setting the electrical impedance 4 Configuring the packet loss compensation mode 5 Configuring an FXS interface to send LCFO signals 5 Configuring an FXO interface 6 Configuring CID 6 Configuring busy tone detection 6 Configuring an on-hook delay 9 Configuring the off-hook mode 10 Configuring ring detection parameters 10 Setting the electrical impedance 11 Configuring the packet loss compensation mode 11 Binding an FXS interface to an FXO interface 11 Configuring an E&M interface 12 Configuring the cable type 12 Configuring the signal type 12 Configuring a start mode for E&M signaling 12 Configuring E&M non-signaling mode 15 Enabling E&M control signals pass-through 16 Configuring the output gain of SLIC chip 16 Configuring DTMF 16 Configuring DTMF tone sending 17 Configuring DTMF tone detection 17 Adjusting parameters for voice interfaces 18 Adjusting gains 18 Adjusting timing parameters 18 Configuring the comfortable noise function 19 Configuring echo cancellation 19 Displaying and maintaining analog voice interfaces 21 Analog voice interface configuration examples 21 Two-dial configuration example for the FXO interface 21 FXO interface PLAR configuration example 22 E&M interface configuration example 24 E&M non-signaling mode configuration example 25 FXS&FXO 1:1 binding configuration example 27 Configuring digital voice interfaces 29 E1 and T1 interfaces 29 BSV interfaces 29 Configuration task list 29 Configuring basic parameters for an E1 interface 30 i

4 Configuring a TDM clock source 30 Configuring other parameters 31 Configuring basic parameters for a T1 interface 31 Configuring a TDM clock source 31 Configuring other parameters 32 Configuring a BSV interface 32 Configuring a BRI interface 33 Configuring a digital voice interface 33 Configuring a timeslot set 33 Creating a timeslot set 33 Configuring a logical digital voice interface 34 Configuring R2 signaling 34 Configuring basic R2 signaling parameters 39 Configuring R2 digital line signaling 42 Configuring R2 interregister signaling 43 Configuring the ISDN protocol 44 Binding a digital voice interface to a POTS entity 45 Displaying and maintaining digital voice interfaces 45 Digital voice interface configuration examples 45 R2 signaling configuration example 45 E1 DSS1 signaling configuration example 47 BSV DSS1 signaling configuration example 49 Configuring voice entities 52 Overview 52 Configuring a POTS entity 52 Configuration task list 52 Creating a POTS entity and configuring basic parameters 53 Configuring codecs for a POTS entity 53 Configuring a POTS entity to register with the registrar 54 Configuring DTMF for a POTS entity 55 Enabling VAD for a POTS entity 55 Configuring options related to dial program 56 Configuring a VoIP entity 56 Configuration task list 56 Creating a VoIP entity and configuring basic parameters 57 Configuring codecs for a VoIP entity 57 Configuring DTMF for a VoIP entity 58 Enabling VAD for a VoIP entity 59 Configuring options related to dial program 59 Displaying and maintaining voice entities 60 Configuring dial programs 61 Configuration task list 61 Configuring caller control 61 Procedure 61 Examples 62 Configuring caller group control 63 Configuring a subscriber group 63 Configuring a caller group on a voice entity 63 Enabling private line auto ring-down 64 Configuring a number match mode 64 Procedure 64 Examples 65 Configuring the maximum number of total calls allowed by a voice entity 66 ii

5 Configuring number substitution 66 Number substitution on the calling router 67 Number substitution on the called router 68 Configuring global number substitution 68 Configuring number substitution for a voice entity 69 Examples 70 Configuring a priority for a voice entity 71 Configuring a number sending mode 71 Procedure 71 Examples 72 Configuring a dial prefix 73 Dial program configuration examples 74 Configuring number substitution 74 Configuring caller group control 76 Configuring SIP 79 Overview 79 Terminology 79 SIP functions 79 SIP messages 80 SIP configuration task list 80 Configuring SIP UA registration 81 Configuration prerequisites 81 Configuring SIP credentials 81 Enabling a POTS entity to register with the registrar 84 Configuring registrar information 84 Displaying SIP UA registration status 84 Configuring the call destination address for a VoIP entity 85 Configuring the call destination IP address for a VoIP entity 85 Configuring a VoIP entity to obtain the call destination address from a proxy server 85 Configuring the destination domain name and port number for a VoIP entity 86 Configuring extended SIP functions 86 Configuring out-of-band DTMF 86 Configuring periodic refresh of SIP sessions 87 Configuring PSTN cause-to-sip status mappings 87 Configuring caller privacy 88 Setting the P-Asserted-Identity or P-Preferred-Identity header field 88 Displaying and maintaining SIP 89 SIP UA configuration examples 89 Configuring direct SIP calling 89 Configuring SIP calling through a SIP server 90 Configuring SIP calling through DNS 92 Configuring out-of-band DTMF 93 Configuring SIP trunk 96 Background 96 Features 97 Typical applications 97 Protocols and standards 98 SIP trunk configuration task list 98 Enabling SIP-to-SIP calling 98 Configuring a SIP trunk account 99 Enabling codec transparent transmission 99 Enabling media flow-around 99 Enabling DO-EO conversion 100 iii

6 Displaying and maintaining SIP trunk 100 SIP trunk configuration examples 101 Configuring call services 103 Call waiting 103 Call hold 103 Call forwarding 103 Call transfer 103 Call backup 104 Call services configuration task list 104 Configuring the call hold mode 104 Configuring call forwarding 105 Call services configuration examples 105 Call waiting configuration example 105 Call forwarding configuration example 107 Call transfer configuration example 109 Call backup configuration example 110 Support and other resources 112 Contacting HP 112 Subscription service 112 Related information 112 Documents 112 Websites 112 Conventions 113 Index 115 iv

7 Configuring analog voice interfaces Analog voice interfaces include FXS, FXO, and E&M interfaces. FXS interface A Foreign Exchange Station (FXS) interface connects to a standard telephone, fax machine, or a Private Branch Exchange (PBX) through an RJ-11 connector and a telephone cable. It provides ring, voltage, and dial tone based on level changes on the Tip/Ring line. An FXS interface can only connect to an FXO interface. FXO interface A foreign exchange office (FXO) interface connects to a PBX through an RJ-11 connector and a telephone cable. It provides ring, voltage, and dial tone based on level changes on the Tip/Ring line. An FXO interface can only connect to an FXS interface. E&M interface An ear & mouth or receive & transmit (E&M) interface is a common trunk line that connects to a PBX through an RJ-48 connector. The E&M interface supports E&M signaling. The E signaling receives signals from the peer, and the M signaling sends signals to the peer. An E&M interface can only connect to an E&M interface. Configuration task list Tasks at a glance (Optional.) Configuring basic functions (Optional.) Configuring call progress tones (Optional.) Configuring an FXS interface Configuring CID Setting the electrical impedance Configuring the packet loss compensation mode Configuring an FXS interface to send LCFO signals 1

8 Tasks at a glance (Optional.) Configuring an FXO interface Configuring CID Configuring busy tone detection Configuring an on-hook delay Configuring the off-hook mode Configuring ring detection parameters Setting the electrical impedance Configuring the packet loss compensation mode (Optional.) Binding an FXS interface to an FXO interface (Optional.) Configuring an E&M interface Configuring the cable type Configuring the signal type Configuring a start mode for E&M signaling Configuring E&M non-signaling mode Enabling E&M control signals pass-through (Optional.) Configuring DTMF Configuring DTMF tone sending Configuring DTMF tone detection (Optional.) Adjusting parameters for voice interfaces Adjusting gains Adjusting timing parameters Configuring the comfortable noise function Configuring echo cancellation Configuring basic functions 2. Enter voice interface view. subscriber-line line-number N/A 3. Configure a description for the voice interface. 4. Restore the default settings for the voice interface. description text default By default, the description of a voice interface is interface name Interface. N/A 5. Bring up the voice interface. undo shutdown By default, a voice interface is up. Configuring call progress tones Perform this task to configure call progress tones for all voice interfaces on the device. To configure call progress tones: 2

9 2. Enter voice view. voice-setup N/A 3. Configure call progress tones. 4. Configure the amplitude value for call progress tones. Specify a country: cptone country-type locale Customize call progress tone parameters: cptone custom { busy-tone congestion-tone dial-tone ringback-tone special-dial-tone waiting-tone } comb freq1 freq2 time1 time2 time3 time4 cptone tone-type { all busy-tone congestion-tone dial-tone ringback-tone special-dial-tone waiting-tone } amplitude value By default, the call progress tones of China are used. The default is 1000 for busy tone and congestion tone, 400 for dial tone and special dial tone, and 600 for ringback tone and waiting tone. Configuring an FXS interface This section covers the procedures for configuring an FXS interface. Configuring CID Caller identification (CID) enables called terminals to display the calling information, including the calling number, calling name, date, and time. CID supports two data formats: Single-data-message format Contains date, time, and calling number. Multiple-data-message format Contains date, time, calling number, and calling name. The called telephone displays the calling number differently as follows: The called telephone displays the calling number if the terminating device is enabled with CID and can obtain the calling number. The called telephone displays the character "O" if the terminating device is enabled with CID but fails to obtain the calling number (for example, the originating device does not send the calling number). The called telephone displays the character "P" if CID is disabled on the terminating device. The FXS interface sends the CID to the called telephone through frequency shift keying (FSK) modulation between the first and second rings. For the CID to appear on the called telephone, the called user should pick up the handset after the second ring. The CID function requires configurations on both FXS and FXO interfaces. For configuration on the FXO interface, see "Configuring CID." Configuration guidelines The PBX and called telephones must support CID. 3

10 To ensure correct call time, make sure the router system time transmitted in data-message format stays synchronous with the local standard time. For the CID function to operate correctly, keep the cid send command enabled. Configuration procedure 2. Enter FXS interface view. subscriber-line line-number N/A 3. (Optional.) Configure the calling name for the FXS interface. 4. Enable the FXS interface to send CID to the remote end. 5. Configure the standard mode for sending CID. 6. Configure the data message format. 7. Enable CID on the FXS interface. calling-name text cid send cid standard-type { bellcore brazil } cid type { complex simple } cid display By default, no calling name is configured. The calling name can be sent only in the multiple-data-message format. Use this command on the originating device. Optional. By default, this function is enabled. Use this command on the originating device. By default, the bellcore mode is used. This command is for use on the terminating device, which encapsulates the CID by using the configured standard mode and sends it to the called telephone. The message format configured by using the cid type command takes effect only when the bellcore mode is used. The default is complex. Use this command on the terminating device. When the originating device supports only one message format, you must use that message format on the terminating device. By default, CID is enabled. Use this command on the terminating device. Setting the electrical impedance The electrical impedance setting must be subject to country specifications. Each country corresponds to an impedance value. You specify an impedance value while specifying a country. To set the electrical impedance: 4

11 2. Enter FXS interface view. subscriber-line line-number N/A 3. Set the electrical impedance of a country. impedance { country-name r550 r600 r650 r700 r750 r800 r850 r900 r950 } The default is the electrical impedance of China. You must configure the same electrical impedance value on the originating and terminating devices. Configuring the packet loss compensation mode You can configure the packet loss compensation mode to alleviate the impact of packet loss on voice quality: For discrete packet loss, you can use the general compensation mode to reconstruct lost packets. For continuous packet loss, you can use the voice gateway-specific compensation mode to compensate for lost packets. To configure the packet loss compensation mode: 1. Enter system view. System-view N/A 2. Enter FXS interface view. subscriber-line line-number N/A 3. Configure the packet loss compensation mode for the FXS interface. plc-mode { general specific } By default, the specific algorithm provided by the voice gateway is used for an FXS interface. Configuring an FXS interface to send LCFO signals You can configure an FXS interface to send a loop current feed open (LCFO) signal when the peer goes on-hook. This function is used mainly in North America. To configure an FXS interface to send LCFO signals: 1. Enter system view. System-view N/A 2. Enter FXS interface view. subscriber-line line-number N/A 3. Configure the FXS interface to send an LCFO signal when the peer goes on-hook. 4. Set the duration of the LCFO signal. disconnect lcfo timer disconnect-pulse value By default, no LCFO signal is sent (a busy tone is played to the peer end). The default is 750 milliseconds. 5

12 Configuring an FXO interface This section covers the procedures for configuring an FXO interface. Configuring CID The CID function must be configured on both the FXS and FXO interfaces. For information about configuring this function on the FXS interface, see "Configuring CID." For the CID function to work correctly, enable both CID receiving and CID sending. Enabling CID receiving for an FXO interface The FXO interface receives the CID. By default, it detects the CID from the PBX between the first and second rings. To enable CID receiving: 2. Enter FXO interface view. subscriber-line line-number N/A 3. Configure the time for CID detection and after the CID detection, the number of rings the FXO interface receives before going off-hook. 4. Enable CID receiving for the FXO interface. cid ring { } [ times ] cid receive By default, CID detection is performed between the first and second rings, and the FXO interface goes off-hook as soon as the detection is complete (cid ring 1 0). By default, this function is enabled. Enabling CID sending for an FXO interface 2. Enter FXO interface view. subscriber-line line-number N/A 3. Enable CID sending for the FXO interface. cid send By default, this function is enabled. Configuring busy tone detection PBX switches might use different busy tone standards. If a user on the PBX side hangs up, the router knows the on-hook operation by detecting a busy tone. As shown in Figure 1, Telephone A establishes a call to Telephone B, and then Telephone A goes on-hook. The PBX plays the busy tone to Router A after detecting the on-hook condition of Telephone A. If Router A cannot identify the played busy tone, the FXO interface on Router A will remain seized or fault detection occurs. To solve this problem, you can configure busy tone detection for the FXO interface. 6

13 Figure 1 Busy tone detection You can configure busy tone detection by customizing busy tone parameters or configuring automatic busy tone detection. If the tone that the router receives from the PBX matches the busy tone parameters, the router considers the tone as a busy tone and shuts down the FXO interface. Configuring a busy tone standard Two busy tone standards are available: European standard and North American standard. After you specify a standard, the device uses a set of parameters compliant with that standard for busy tone detection. If the actual busy tone data do not completely match the set of parameters, the device can use customized busy tone parameters or automatic busy tone detection to accurately detect busy tones. To configure a busy tone standard: 2. Enter voice view. voice-setup N/A 3. Specify a busy tone standard. area { europe north-america } By default, the European standard is used. Customizing busy tone parameters 2. Enter voice view. voice-setup N/A 3. Customize busy tone parameters. 4. Enable using customized busy tone parameters. busytone-detect custom area-number index argu f1 f2 p1 p2 p3 p4 p5 p6 p7 area custom N/A By default, the European standard is used. This command takes effect on all FXO interfaces on the device. The customized busy tone parameters take effect only after you configure this command. Configuring automatic busy tone detection Figure 2 Automatic busy tone detection As shown in Figure 2, the process of automatic busy tone detection is as follows: 1. Telephone A establishes a call to Telephone B. 7

14 2. Telephone A first goes on-hook. The PBX plays busy tones to Router A after detecting the on-hook condition. 3. Execute the busytone-detect auto command on Router A to detect the busy tone. To make sure the FXO interface can capture the busy tone sent by the PBX, HP recommends that you execute this command two seconds after Telephone A goes on-hook. 4. The console prompts that busy tone detection is in progress and prompts detection success when the detection is complete. 5. Check whether the detected busy tone parameters are valid by repeating steps 1 and 2. After Telephone A goes on-hook, the PBX plays a busy tone to Router A. If Router A detects the busy tone, it shuts down the FXO interface. To configure automatic busy tone detection: 2. Enter voice view. voice-setup N/A 3. Configure automatic busy tone detection. busytone-detect auto index line-number N/A 4. Return to system view. quit N/A 5. Enter FXO interface view. subscriber-line line-number N/A 6. Configure the number of busy tone detection periods. busytone-detect period value The default is 2. Test multiple values to select the value that can ensure normal on-hook. Configuring busy tone sending If the PBX does not play busy tones, configure the FXO interface to send busy tones. To configure busy tone sending: 2. Enter FXO interface view. subscriber-line line-number N/A 3. Enable busy tone sending. send-busytone enable By default, this function is disabled. 4. Set the busy tone duration. send-busytone time seconds The default is 3 seconds. Configuring silence detection-based automatic on-hook If the device fails to detect busy tones or the PBX does not play busy tones, you can configure this feature to implement automatic on-hook. When the duration of silence (whose volume is smaller than the configured threshold) exceeds the configured silence duration, the FXO interface automatically disconnects the call. Improper configuration of this function can lead to false on-hook. A good practice is to configure multiple sets of parameters and select the set of parameters that can quickly release the FXO interface and does not causes false on-hook. 8

15 To configure silence detection-based automatic on-hook: 2. Enter FXO interface view. subscriber-line line-number N/A 3. Configure silence detection-based automatic on-hook. silence-detect threshold threshold time time-length By default, the silence threshold is 20, and the silence duration f is 7200 seconds (2 hours). Configuring forced on-hook In some countries, PBXs do not play busy tones, or the busy tones only last for a short period of time. When noise is present on a link, even the silence detection-based automatic on-hook function (configured with silence-detect threshold) cannot detect the busy tones and fails to release a call after on-hook. To solve this problem, configure the forced on-hook function. Forced on-hook disconnects a call when the specified time expires, even if the call is ongoing. To configure forced on-hook: 2. Enter FXO interface view. subscriber-line line-number N/A 3. Configure forced on-hook. hookoff-time time By default, forced on-hook is disabled. If you configure this command on an FXO interface of a card, the configuration takes effect on all FXO subscriber lines of the card. Configuring an on-hook delay An FXO interface goes on-hook when detecting a busy tone. This will cause the user of an IP phone connected to the FXO interface to mistake the on-hook as a line problem because the user cannot hear the busy tones. To solve this problem, you can configure a delay time. During the delay time, the FXO interface continues sending the busy tones to the IP phone. To configure an on-hook delay: 2. Enter FXO interface view. subscriber-line line-number N/A 3. Configure the delay time from when the FXO interface detects a busy tone to when the interface goes on-hook. busytone-hookon delay-timer value The delay time is 0 seconds (the FXO interface goes on-hook immediately after detecting a busy tone). 9

16 Configuring the off-hook mode The FXO interface supports two off-hook modes: Immediate mode Upon receiving a call, the FXO interface goes off-hook and sends a dial tone to the calling party. Then, the calling party dials the destination number. Delay mode Upon receiving a call, the FXO interface places a call to the specified private line number. When the called party picks up the phone, the FXO goes off-hook. This mode needs to work with the private line auto ring-down (PLAR) function. For more information about PLAR, see "Configuring dial programs." To configure the off-hook mode: 2. Enter FXO interface view. subscriber-line line-number N/A 3. Configure the off-hook mode. hookoff-mode { delay immediate } By default, the immediate mode is used. Configuring ring detection parameters PBXs from different vendors use different types of ring signals. By setting ring detection parameters, you can enable the FXO interface to detect ring signals of different frequencies and waveforms. To configure ring detection parameters: 2. Enter FXO interface view. subscriber-line line-number N/A 3. Set the debounce time for ring detection on the FXO interface. 4. Set the frequency value for ring detection. ring-detect debounce value ring-detect frequency value The default is 10 milliseconds. Do not set the debounce time during a conversation. Do not set too short a debounce time, because false ring tone recognition might occur under power line interference. If you configure this command on an FXO interface of a card, the configuration takes effect on all FXO subscriber lines of the card. The default is 40 Hz. 10

17 Setting the electrical impedance The electrical impedance setting must be subject to country specifications. Each country corresponds to an impedance value. You specify an impedance value while specifying a country. To set the electrical impedance: 2. Enter FXO interface view. subscriber-line line-number N/A 3. Set the electrical impedance of a country. impedance { country-name r550 r600 r650 r700 r750 r800 r850 r900 r950 } The default is the electrical impedance of China. You must configure the same electrical impedance value on the originating and terminating devices. Configuring the packet loss compensation mode You can configure the packet loss compensation mode to alleviate the impact on voice quality: For discrete packet loss, you can use the general compensation mode to reconstruct lost packets. For continuous packet loss, you can use the voice gateway-specific compensation mode to reconstruct lost packets. To configure the packet loss compensation mode: 1. Enter system view. System-view N/A 2. Enter FXO interface view. subscriber-line line-number N/A 3. Configure the packet loss compensation mode for the FXO interface. plc-mode { general specific } By default, the voice gateway-specific compensation mode is used for an FXO interface. Binding an FXS interface to an FXO interface The one-to-one binding between FXS and FXO interfaces enhances the reliability of voice communication. By working with the PLAR function, this feature enables the telephone connected to the FXS interface to exclusively use the bound FXO interface to place and receive calls. The on-hook/off-hook state of the bound FXS and FXO interfaces is consistent. If the FXS interface receives a call when the bound FXS interface goes off-hook, the calling party will hear busy tones. To bind an FXS interface to an FXO interface: 11

18 2. Enter FXO interface view. subscriber-line line-number N/A 3. Bind an FXS interface to the FXO interface. hookoff-mode delay bind fxs_subscriber_line [ ring-immediately ] By default, no FXS interface is bound. 4. Enable the PLAR function. private-line string 5. Configure the interval between on-hook and off-hook. timer hookoff-interval milliseconds By default, the PLAR function is disabled. For more information about this command, see Voice Command Reference. The default is 500 milliseconds. Configuring an E&M interface Configuring the cable type You must configure the same cable type for the E&M interfaces on the originating and terminating devices. Otherwise, only one-way voice communication can be implemented. To configure the cable type for an E&M interface: 2. Enter E&M interface view. subscriber-line line-number N/A 3. Configure the cable type for the E&M interface. cable { 2-wire 4-wire } The default is 4-wire. Configuring the signal type You must configure the same signal type on the originating and terminating devices. To configure the signal type for an E&M interface: 2. Enter E&M interface view. subscriber-line line-number N/A 3. Configure a signal type for the E&M interface. type { } The default is 5 (corresponding V type signal). Configuring a start mode for E&M signaling Three start modes are available for E&M signaling: 12

19 Immediate start After off-hook, the originating side waits for a specified period of time to send the called number to the terminating side. During this period of time, the originating side does not check whether the terminating side is ready to receive the called number. The terminating side enters off-hook state after receiving the called number. Figure 3 Immediate start Delay start The originating side goes off-hook and seizes the trunk. After detecting the seizure signal from the originating side, the terminating side enters the off-hook state and remains in this state until it is ready to receive the called number. Then, the terminating side enters the on-hook state and sends a signal to indicate that the line is idle. After receiving the idle signal, the originating side sends the called number to the terminating side, which relays the call to the user phone. Figure 4 Delay start Wink start The terminating side remains in on-hook state until it receives the seizure signal from the originating side and then sends a wink to the originating side. After receiving the wink, the originating side sends the called number to the terminating side, which relays the call to the user phone. 13

20 Figure 5 Wink start To configure the immediate start mode: 2. Enter E&M interface view. subscriber-line line-number N/A 3. Configure the immediate start mode for the E&M interface. 4. Configure a delay the originating side waits to send DTMF tones in immediate start mode. signal immediate delay send-dtmf milliseconds The default is the immediate start mode. The default is 300 milliseconds. Use this command on the originating device. To configure the delay start mode: 2. Enter voice interface view. subscriber-line line-number N/A 3. Configure the delay start mode for the E&M interface. 4. Configure the seizure signal duration in delay start mode. 5. Configure the delay time from when the terminating side detects a seizure signal to when it sends the seizure signal in delay start mode. signal delay delay hold milliseconds delay rising milliseconds The default is the immediate start mode. The default is 400 milliseconds. Use this command on the terminating device. The default is 300 milliseconds. Use this command on the terminating device. To configure the wink start mode: 14

21 2. Enter voice interface view. subscriber-line line-number N/A 3. Configure the wink start mode for the E&M interface. 4. Configure the delay time from when the terminating side receives a seizure signal to when it sends a wink signal in the wink start mode. 5. Configure the duration of a wink signal sent by the terminating side in wink start mode. 6. Configure the timeout time for the originating side to wait for a wink signal after sending a seizure signal in the wink start mode. signal wink delay send-wink milliseconds delay wink-hold milliseconds delay wink-rising milliseconds The default is the immediate start mode. The default is 200 milliseconds. Use this command on the terminating device. The default is 500 milliseconds. Use this command on the terminating device. The default is 2000 milliseconds. Use this command on the originating device. Configuring E&M non-signaling mode The E&M non-signaling mode is applied when the E&M interface of the peer device does not provide the M line and E line. In this mode, the E&M interface communicates with the peer device without signaling. You can configure the PLAR function by using the private-line command to form a three-segment E&M virtual private line (E&M-VoIP-E&M). When a subscriber picks up the phone, the originating device directly dials the number specified by using the private-line command. To configure E&M non-signaling mode: 2. Enter voice interface view. subscriber-line line-number N/A 3. Configure the immediate start mode for the E&M interface. 4. Enable E&M non-signaling mode. signal immediate open-trunk { caller [ monitor interval ] called } The default is the immediate start mode. By default, E&M non-signaling mode is disabled. Configure the open-trunk caller [ monitor interval ] command on the originating device, and configure the open-trunk called command on the terminating device. 5. Enable the PLAR function. private-line string Use this command on the originating device. For more information about the PLAR function, see "Configuring dial programs." 15

22 Enabling E&M control signals pass-through This feature operates only when the E&M non-signaling mode is enabled. As shown in Figure 6, an E&M virtual private line is set up between the tone generator and the radio. Enable E&M control signals pass-through for Router A and Router B so that they can send seize and idle signals for the E&M virtual line over the IP network. Figure 6 E&M analog control signals pass-through To enable E&M control signals pass-through: 2. Enter voice interface view. subscriber-line line-number N/A 3. Enable E&M control signals pass-through. passthrough By default, this function is disabled. Configure this command on both the originating and terminating devices. Configuring the output gain of SLIC chip 2. Enter voice interface view. subscriber-line line-number N/A 3. Configure the output gain of the SLIC chip. slic-gain { 0 1 } Optional. The default is 0 (0.8 db). This command controls signal amplification. Configuring DTMF Dual tone multi-frequency (DTMF) uses a mixture of a high frequency tone and a lower frequency tone to represent a key on a keypad. Each column of keys is represented by a high frequency tone and each row of keys is represented by a low frequency tone. For example, as shown in Figure 7, the digit 1 is represented by the combination of a pure 697 Hz signal and a pure 1209 Hz signal. Such DTMF tones have good immunity to interference. 16

23 Figure 7 DTMF keypad frequencies Column Frequency Group 1209Hz 1336Hz 1477Hz 1633Hz 697Hz A 770Hz B 852Hz C 941Hz * 0 # D A DTMF tone must last at least 45 milliseconds. A minimum interval of 23 milliseconds is required between two DTMF tones to make sure DTMF tones are recognizable. Such requirements are roughly the same in all countries. For more information, see the ITU-T Recommendation Q.24. Configuring DTMF tone sending 2. Enter voice view. voice-setup N/A 3. Configure the duration of DTMF tones and the interval between DTMF tones. 4. Configure the amplitude of DTMF tones. dtmf time {interval persist } milliseconds dtmf amplitude value The default duration and interval are both 120 milliseconds. The default is 9.0 dbm. Configuring DTMF tone detection Use the following ways to detect DTMF tones: Energy detection Calculates the spectrum shape of input DTMF signals and matches the spectrum shape with the threshold parameters. A DTMF tone is considered valid if it matches all threshold parameters. Sensitivity detection A higher DTMF detection sensitivity reduces the possibility of missing a true DTMF tone but increases the possibility of false detection. A lower DTMF detection sensitivity reduces the possibility of false detection but increases the possibility of missing a true DTMF tone. To configure DTMF tone detection: 17

24 2. Enter voice interface view. subscriber-line line-number N/A 3. Configure the threshold parameters for DTMF detection. 4. Configure the DTMF detection sensitivity level. dtmf threshold analog index value dtmf sensitivity-level { high low medium [ frequency-tolerance value ] } By default, indexes 0 to 12 correspond to 1400, 458, -9, -9, -9, -9, -3, -12, -12, 30, 300, 3200, and 375, respectively. This command is used by professional personnel to adjust the device in the case of DTMF detection failure. Usually, the default value is adopted. By default, the DTMF detection sensitivity is low. This command applies to only FXS and FXO interfaces Adjusting parameters for voice interfaces The configuration tasks in this section are all optional. Adjusting gains You can adjust gains to control the amount of volume in the input or output direction. IMPORTANT: Gain adjustment might lead to call failures. HP recommends not adjusting the gain. If necessary, do it under the guidance of technical personnel. To adjust gains: 2. Enter voice interface view. subscriber-line line-number N/A 3. Set the input gain. receive gain value The default is 0 db. 4. Set the output gain. transmit gain value The default is 0 db. Adjusting timing parameters 2. Enter voice interface view. subscriber-line line-number N/A 18

25 3. Configure the interval between off-hook and dialing the first digit. 4. Configure the maximum interval for dialing the next digit. 5. Configure the maximum duration for playing ringback tones. timer first-dial seconds timer dial-interval seconds timer ring-back seconds The default is 10 seconds. This command applies only to FXO and FXS interfaces. The default is 10 seconds. The default is 60 seconds. 6. Configure the dial delay time. delay start-dial seconds The default is 1 second. This command applies to only FXO and FXS interfaces. 7. Configure the hookflash time range. 8. Configure the maximum duration the terminating device waits for the first digit. timer hookflash-detect hookflash-range timer wait-digit { seconds infinity } By default, the hookflash time range is 50 to 180 milliseconds; that is, if an on-hook condition lasts for a period that falls within the hookflash time range, it is considered a hookflash. This command applies to only FXS interfaces. The default is 5 seconds. This command applies to only E&M interfaces. Configuring the comfortable noise function You can configure this feature to generate comfortable background noise to replace the silent gaps during a conversation. To configure the comfortable noise function: 2. Enter voice interface view. subscriber-line line-number N/A 3. Enable the comfortable noise function. cng-on By default, this function is enabled. Configuring echo cancellation An echo is the audible leak-through of your own voice into your own receive path. When the voice of a user leaks into the receive path, it is an echo. The echo cancellation function can alleviate the echo problem. The time when an echo occurs and the size of the echo are relatively fixed. As shown in Figure 8, there is a voice signal at time 0, and an echo occurs after about 40 milliseconds (ms). To cancel this echo, set 19

26 the echo cancellation delay (the time between when an interface sends out a signal and when to the interface receives an echo) to 33 ms and the echo cancellation coverage to 16 ms. Figure 8 Echo Configuring the echo cancellation function 2. Enter voice interface view. subscriber-line line-number N/A 3. Enable the echo cancellation function. 4. Configure the echo cancellation delay. 5. Configure the echo cancellation coverage. echo-canceler enable echo-canceler delay milliseconds echo-canceler tail-length milliseconds By default, this function is enabled. The default is 0 milliseconds. The default is 128 milliseconds. Adjusting echo cancellation parameters Table 1 describes how to adjust echo cancellation parameters for different symptoms. Table 1 Adjusting echo cancellation parameters Symptom Parameters adjusted Effect A user hears echoes or loud background noises from the peer when speaking. There are loud environment noises. A user hears echoes when speaking. There are echoes when both parties speak at the same time. Speed up the convergence of comfortable noise amplitudes. Increase the maximum amplitude of comfortable noises. Enlarge the mixture proportion control factor of comfortable noises. Enlarge the two-way judgment threshold. Too fast convergence might make noises uncomfortable. Too large amplitude might make noises uncomfortable. Too high a control factor leads to audio discontinuity. Too high a judgment threshold slows down the convergence of the filter factor. To adjust echo cancellation parameters: 20

27 2. Enter voice view. voice-setup N/A 3. Adjust echo cancellation parameters. echo-canceler { convergence-rate value max-amplitude value mix-proportion-ratio value talk-threshold value } By default, the convergence rate of comfort noise amplitude is 0, the maximum amplitude of comfort noise is 256, the comfort noise mixture proportion control factor is 100, and the two-way judgment threshold is 1. The echo-canceler enable command must be configured for the configured parameters to take effect. Enabling the nonlinear function of echo cancellation After echo cancellation is enabled, nonlinear parts in the line can cause residual echo. The nonlinear function, also called residual echo suppression, can remove the residual echo. To enable the nonlinear function of echo cancellation: 2. Enter voice interface view. subscriber-line line-number N/A 3. Enable the nonlinear function of echo cancellation. nlp-on By default, this function is enabled. This command is available only after you configure the echo-canceler enable command. Displaying and maintaining analog voice interfaces Execute display commands in any view. Task Display information about analog voice interfaces. Command display voice subscriber-line Analog voice interface configuration examples Two-dial configuration example for the FXO interface As shown in Figure 9, Router A and Router B are connected through an IP network and can reach each other. The PBX is configured with a trunk number

28 Configure the two routers to enable Telephone B to establish a call with Telephone A through two dials. Figure 9 Network diagram Eth2/1 Eth2/1 FXS 1/ /24 IP network /24 FXO 1/0 Telephone A Router A Router B PBX Telephone B Configuration procedure 1. On Router A, configure the local number as for POTS entity 1001, and bind FXS interface line1/0 to the POTS entity. <RouterA> system-view [RouterA] voice-setup [RouterA-voice] dial-program [RouterA-voice-dial] entity 1001 pots [RouterA-voice-dial-entity1001] match-template [RouterA-voice-dial-entity1001] line 1/0 2. Configure Router B: # Configure the called number as 010 for VoIP entity 010, and configure the destination IP address as <RouterB> system-view [RouterB] voice-setup [RouterB-voice] dial-program [RouterB-voice-dial] entity 010 voip [RouterB-voice-dial-entity10] match-template [RouterB-voice-dial-entity10] address sip ip [RouterB-voice-dial-entity10] quit # Configure the local number as for POTS entity 2001, and bind FXO interface line1/0 to POTS entity [RouterB-voice-dial] entity 2001 pots [RouterB-voice-dial-entity2001] match-template [RouterB-voice-dial-entity2001] line 1/0 # Configure the number sending mode as all. [RouterB-voice-dial-entity2001] send-number all Verifying the configuration After the user of Telephone B dials , the user hears a dial tone. Then, the user of Telephone B dials , and Telephone A rings. After the user of Telephone A picks up the handset, the two users can establish a conversation. FXO interface PLAR configuration example Network requirements As shown in Figure 10, Router A and Router B are connected through an IP network and can reach each other. The PBX is configured with a trunk number

29 Configure PLAR for the FXO interface of Router B. When the user of Telephone B dials , the FXO interface automatically calls Telephone A. Figure 10 Network diagram Eth2/1 Eth2/1 FXS 1/ /24 IP network /24 FXO 1/0 Telephone A Router A Router B PBX Telephone B Configuration procedure 1. On Router A, configure the local number as for POTS entity 1001, and bind FXS interface line1/0 to the POTS entity. <RouterA> system-view [RouterA] voice-setup [RouterA-voice] dial-program [RouterA-voice-dial] entity 1001 pots [RouterA-voice-dial-entity1001] match-template [RouterA-voice-dial-entity1001] line 1/0 2. Configure Router B: # Configure the called number as 010 for VoIP entity 010, and configure the destination IP address as <RouterB> system-view [RouterB] voice-setup [RouterB-voice] dial-program [RouterB-voice-dial] entity 010 voip [RouterB-voice-dial-entity10] match-template [RouterB-voice-dial-entity10] address sip ip [RouterB-voice-dial-entity10] quit # Configure the local number as for POTS entity 2001, and bind FXO interface line1/0 to the POTS entity. [RouterB-voice-dial] entity 2001 pots [RouterB-voice-dial-entity2001] match-template [RouterB-voice-dial-entity2001] line 1/0 # Configure the number sending mode as all. [RouterB-voice-dial-entity2001] send-number all # Enable the PLAR function and configure the delay off-hook mode for the FXO interface. [RouterB] subscriber-line 1/0 [RouterB-subscriber-line1/0] private-line [RouterB-subscriber-line1/0] hookoff-mode delay Verifying the configuration After the user of Telephone B dials , the FXO interface automatically calls Telephone A, and Telephone A rings. After the user of Telephone A picks up the handset, the two users can establish a conversation. 23

30 E&M interface configuration example Network requirements As shown in Figure 11, Router A and Router B are connected through an IP network and can reach each other. Configure the two routers to enable Telephone A and Telephone B to establish calls. Figure 11 Network diagram Eth2/1 Eth2/1 FXS 1/ /24 IP network /24 E&M 5/0 Telephone A Router A Router B PBX Telephone B Configuration procedure 1. Configure Router A: # Configure the called number as 0755 for VoIP entity 0755, and configure the destination IP address as <RouterA> system-view [RouterA] voice-setup [RouterA-voice] dial-program [RouterA-voice-dial] entity 0755 voip [RouterA-voice-dial-entity755] match-template [RouterA-voice-dial-entity755] address sip ip [RouterA-voice-dial-entity755] quit # Configure the local number as for POTS entity 1001, and bind FXS interface line1/0 to the POTS entity. [RouterA-voice-dial] entity 1001 pots [RouterA-voice-dial-entity1001] match-template [RouterA-voice-dial-entity1001] line 1/0 2. Configure Router B: # Configure the called number as 010 for VoIP entity 010, and configure the destination IP address as <RouterB> system-view [RouterB] voice-setup [RouterB-voice] dial-program [RouterB-voice-dial] entity 010 voip [RouterB-voice-dial-entity10] match-template [RouterB-voice-dial-entity10] address sip ip [RouterB-voice-dial-entity10] quit # Configure the local number as for POTS entity 2001, and bind E&M interface line 5/0 to the POTS entity. [RouterB-voice-dial] entity 2001 pots [RouterB-voice-dial-entity2001] match-template [RouterB-voice-dial-entity2001] send-number all [RouterB-voice-dial-entity2001] line 5/0 [RouterB-voice-dial-entity2001] return 24

31 Verifying the configuration # Enter the view of E&M interface 5/0. The E&M interface must have the same configuration as the connected PBX. <RouterB> system-view [RouterB] subscriber-line 5/0 # Configure the wink start mode. [RouterB-subscriber-line5/0] signal wink # Configure the 4-wire cable type (optional, because the default is the 4-wire cable type). [RouterB-subscriber-line5/0] cable 4-wire # Configure the signal type as 5 (optional, because the default is 5). [RouterB-subscriber-line5/0] type 5 After the user of Telephone A dials , Telephone B rings. When the user of Telephone B picks up the handset, the two users can establish a conversation. After the user of Telephone B dials , Telephone A rings. When the user of Telephone A picks up the handset, the two users can establish a conversation. E&M non-signaling mode configuration example Network requirements As shown in Figure 12, configure the PLAR mode for the E&M interface on Router A. When the user of Telephone A dials picks up the handset, the E&M interface automatically calls Telephone B. Figure 12 Network diagram Configuration procedure 1. Configure Router A: # Configure the called number as 2000 for VoIP entity 2000, and configure the destination IP address as <RouterA> system-view [RouterA] voice-setup [RouterA-voice] dial-program [RouterA-voice-dial] entity 2000 voip [RouterA-voice-dial-entity2000] match-template 2000 [RouterA-voice-dial-entity2000] address sip ip [RouterA-voice-dial-entity2000] quit # Configure the local number as 1000 for POTS entity 1000, and bind E&M interface line 5/0 to the POTS entity. 25

32 [RouterA-voice-dial] entity 1000 pots [RouterA-voice-dial-entity1000] match-template 1000 [RouterA-voice-dial-entity1000] line 5/0 [RouterA-voice-dial-entity1000] return # Enter the view of E&M interface 5/0. <RouterA> system-view [RouterA] subscriber-line 5/0 # Configure the immediate start mode (optional, because the default is the immediate start mode). [RouterA-subscriber-line5/0] signal immediate # Enable the PLAR function. [RouterA-subscriber-line5/0] private-line 2000 # Enable the E&M non-signaling mode. [RouterA-subscriber-line5/0] open-trunk caller # Enable E&M analog control signals pass-through. [RouterA-subscriber-line5/0] passthrough # Disable the echo cancellation function. [RouterA-subscriber-line5/0] undo echo-canceler enable 2. Configure Router B: # Configure the called number as 1000 for VoIP entity 1000, and configure the destination IP address as <RouterB> system-view [RouterB] voice-setup [RouterB-voice] dial-program [RouterB-voice-dial] entity 1000 voip [RouterB-voice-dial-entity10] match-template 1000 [RouterB-voice-dial-entity10] address sip ip [RouterB-voice-dial-entity10] quit # Configure the local number as 2000 for POTS entity 2000, and bind E&M interface line 5/0 to the POTS entity. [RouterB-voice-dial] entity 2000 pots [RouterB-voice-dial-entity2000] match-template 2000 [RouterB-voice-dial-entity2000] line 5/0 [RouterB-voice-dial-entity2000] return # Enter the view of E&M interface 5/0. <RouterB> system-view [RouterB] subscriber-line 5/0 # Configure the immediate start mode (optional, because the default is the immediate start mode). [RouterB-subscriber-line5/0] signal immediate # Enable the E&M non-signaling mode. [RouterB-subscriber-line5/0] open-trunk called # Enable E&M control signals pass-through. [RouterB-subscriber-line5/0] passthrough # Disable the echo cancellation function. [RouterB-subscriber-line5/0] undo echo-canceler enable 26