HP MSR Router Series Voice Configuration Guide(V7) Part number: 5998-5683 Software version: CMW710-R0106 Document version: 6PW100-20140607
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Contents Configuring analog voice interfaces ·························································································································· 1 FXS interface ······································································································································································ 1 FXO interface··················································································································································
Configuring a TDM clock source ························································································································· 30 Configuring other parameters ······························································································································ 31 Configuring basic parameters for a T1 interface ······································································································· 31 Configuring a TDM clock source ···········
Examples ································································································································································ 65 Configuring the maximum number of total calls allowed by a voice entity····························································· 67 Configuring number substitution ··································································································································· 67 Number substitution on the calling r
Specifying a URL scheme ············································································································································ 101 Specifying a global URL scheme for outgoing SIP calls ·················································································· 102 Specifying a URL scheme for outgoing SIP calls on a VoIP entity ·································································· 102 Setting the global DSCP value ·····································
Enabling CNG fax switchover ··························································································································· 139 Enabling ECM······················································································································································ 139 Configuring an NSF code for nonstandard capabilities negotiation ···························································· 140 Configuring the maximum fax rate for rate training ········
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.
Tasks at a glance (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.
Step Command Remarks • Specify a country: cptone country-type locale • Customize call progress tone Configure call progress tones. parameters: cptone custom { busy-tone | congestion-tone | dial-tone | ringback-tone | special-dial-tone | waiting-tone } comb freq1 freq2 time1 time2 time3 time4 By default, the call progress tones of China are used. Defaults are: 3. Configure the amplitude value for call progress tones.
Configuration guidelines • The PBX and called telephones must support CID. • 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 Step Command Remarks 1. Enter system view. system-view N/A 2. Enter FXS interface view. subscriber-line line-number N/A By default, no calling name is configured. 3.
Step Command Remarks 1. Enter system view. system-view N/A 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 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.
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 renders the FXO interface on-hook. Configuring a busy tone standard Two busy tone standards are available: European and North American.
2. Telephone A first goes on-hook, and the PBX plays a busy tone 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.
Step Command Remarks 1. Enter system view. system-view N/A 2. Enter FXO interface view. subscriber-line line-number N/A 3. Configure detection-based on-hook. silence-detect threshold threshold time time-length By default, the silence threshold is 20, and the silence duration is 7200 seconds (2 hours). silence automatic 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.
• 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 set the electrical impedance: Step Command Remarks 1. Enter system view. system-view N/A 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.
Step Command Remarks By default, the PLAR function is disabled. 4. Enable the PLAR function. private-line string For more information about this command, see Voice Command Reference. 5. Configure between off-hook. timer hookoff-interval milliseconds The default is 500 milliseconds. the interval on-hook and 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.
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.
Step Command Remarks 1. Enter system view. system-view N/A 2. Enter E&M interface view. subscriber-line line-number N/A 3. Configure the immediate start mode for the E&M interface. signal immediate The default is the immediate start mode. 4. Configure a delay for the originating side to wait to send DTMF tones in immediate start mode. delay send-dtmf milliseconds The default is 300 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.
Step Command Remarks By default, this function is disabled. 3. Enable E&M control signals pass-through. Configure this command on both the originating and terminating devices. passthrough Configuring the output gain of SLIC chip Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice interface view. subscriber-line line-number N/A Optional. 3. Configure the output gain of the SLIC chip. slic-gain { 0 | 1 } The default is 0 (0.8 dB).
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 Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Configure the duration of DTMF tones and the interval between DTMF tones.
Adjusting parameters for voice interfaces All configuration tasks in this section are 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 engineers. To adjust gains: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice interface view. subscriber-line line-number N/A 3.
Step Command 8. Configure the maximum duration the terminating device waits for the first digit. timer wait-digit { seconds | infinity } Remarks 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: Step Command Remarks 1. Enter system view. system-view N/A 2.
Step 4. Configure the cancellation delay. echo Command Remarks echo-canceler delay milliseconds The default is 0 milliseconds. The default is 128 milliseconds. 5. Configure the cancellation coverage. echo echo-canceler tail-length milliseconds For routers installed with the interface module SIC VE1 or SIC VT1, the value range is 32, 48, 64, 80, 96, 112, and 128. For routers installed with any other interface module type, the value can only be 128.
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: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice interface view. subscriber-line line-number N/A By default, this function is enabled. 3.
[RouterA-voice] dial-program [RouterA-voice-dial] entity 1001 pots [RouterA-voice-dial-entity1001] match-template 0101001 [RouterA-voice-dial-entity1001] line 2/1/1 2. Configure Router B: # Configure the called number as 010 for VoIP entity 010, and configure the destination IP address as 1.1.1.1. system-view [RouterB] voice-setup [RouterB-voice] dial-program [RouterB-voice-dial] entity 010 voip [RouterB-voice-dial-entity10] match-template 0101001. [RouterB-voice-dial-entity10] address sip ip 1.
[RouterA-voice-dial] entity 1001 pots [RouterA-voice-dial-entity1001] match-template 0101001 [RouterA-voice-dial-entity1001] line 2/1/1 2. Configure Router B: # Configure the called number as 010 for VoIP entity 010, and configure the destination IP address as 1.1.1.1. system-view [RouterB] voice-setup [RouterB-voice] dial-program [RouterB-voice-dial] entity 010 voip [RouterB-voice-dial-entity10] match-template 0101001. [RouterB-voice-dial-entity10] address sip ip 1.1.1.
system-view [RouterA] voice-setup [RouterA-voice] dial-program [RouterA-voice-dial] entity 0755 voip [RouterA-voice-dial-entity755] match-template 07552001 [RouterA-voice-dial-entity755] address sip ip 2.2.2.2 [RouterA-voice-dial-entity755] quit # Configure the local number as 0101001 for POTS entity 1001, and bind FXS interface line 2/1/1 to the POTS entity. [RouterA-voice-dial] entity 1001 pots [RouterA-voice-dial-entity1001] match-template 0101001 [RouterA-voice-dial-entity1001] line 2/1/1 2.
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 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 2.2.2.2.
2. Configure Router B: # Configure the called number as 1000 for VoIP entity 1000, and configure the destination IP address as 1.1.1.1. system-view [RouterB] voice-setup [RouterB-voice] dial-program [RouterB-voice-dial] entity 1000 voip [RouterB-voice-dial-entity1000] match-template 1000 [RouterB-voice-dial-entity1000] address sip ip 1.1.1.1 [RouterB-voice-dial-entity1000] quit # Configure the local number as 2000 for POTS entity 2000, and bind E&M interface line 2/3/1 to the POTS entity.
Figure 13 Network diagram IP 192.168.0.71/24 FXS2/1/1 Telephone A 0101001 192.168.0.76/24 Router B Router A FXS2/1/1 FXO2/2/1 FXO2/2/1 Telephone B 2101002 PSTN Configuration procedure 1. Configure Router A: # Configure the called number template as 210…. for VoIP entity 210, and configure the destination IP address as 192.168.0.76. system-view [RouterA] voice-setup [RouterA-voice] dial-program [RouterA-voice-dial] entity 210 voip [RouterA-voice-dial-entity210] match-template 210....
system-view [RouterB] voice-setup [RouterB-voice] dial-program [RouterB-voice-dial] entity 010 voip [RouterB-voice-dial-entity10] match-template 010.... [RouterB-voice-dial-entity10] address sip ip 192.168.0.71 [RouterB-voice-dial-entity10] quit # Configure the local number as 2101002 for POTS entity 2101002, and bind the FXS interface line 2/1/1 to the POTS entity.
Configuring digital voice interfaces Digital voice interfaces include E1, T1, and BSV interfaces. E1 and T1 interfaces E1 interfaces (also called VE1 interfaces) and T1 interfaces (also called VT1 interfaces) can connect to the PSTN as trunk interfaces. An E1 interface provides 32 timeslots and 2.048 Mbps bandwidth. A T1 interface provides 24 timeslots and 1.544 Mbps bandwidth. ITU-T E1 is used mainly in Europe and China. ANSI T1 is used mainly in America, Canada, and Japan.
Tasks at a glance (Required.) Perform one of the following tasks: • Configuring R2 signaling { Configuring basic R2 signaling parameters { Configuring R2 digital line signaling { Configuring R2 interregister signaling • Configuring the ISDN protocol (Required.) Binding a digital voice interface to a POTS entity To configure BSV interfaces, perform the following tasks: Tasks at a glance (Required.
Step Command Remarks 2. Enter E1 interface view. controller e1 number N/A 3. Configure a TDM clock source for the E1 interface. tdm-clock { internal | line [ primary ] } The default is internal. Configuring other parameters Step Command Remarks 1. Enter system view. system-view N/A 2. Enter E1 interface view. controller e1 number N/A 3. Configure a description. description text The default is interface name Interface. 4. Configure the framing format.
Configuring other parameters Step Command Remarks 1. Enter system view. system-view N/A 2. Enter T1 interface view. controller t1 number N/A 3. Configure a description. description text The default is interface name Interface. 4. Configure the framing format. frame-format { esf | sf } The default is esf. 5. Set the line coding format. code { ami | b8zs } The default is b8zs. 6. Set the cable type. cable { long | 0db | -7.5db | -15db | -22.
Step Command Remarks 2. Enter BRI interface view. interface bri interface-number N/A 3. (Optional.) Configure a description for the interface. description text The default is interface name Interface. 4. (Optional.) Enable loopback detection for B channels. loopback { b1 | b2 | both } By default, loopback detection is disabled. 5. (Optional.) Configure the expected bandwidth of the interface. bandwidth bandwidth-value The default is 0 kbps. 6. Set the MTU of the interface.
Configuring a logical digital voice interface The system automatically creates a logical voice interface in the form of E1/T1 interface number:timeslot set number for a timeslot set. The description, shutdown, receive gain, transmit gain, and cng-on commands for a logical digital voice interface have the same functions as those for analog voice interfaces. For more information about those commands, see related sections in "Configuring analog voice interfaces.
Digital line signaling sets the line to be idle or seized according to the state of the trunk. This signaling is transmitted through timeslot 16. The two transmission directions of each line have four bits (A, B, C and D) as flag bits, with C and D bits fixed to 01.
Figure 16 Call establishment • Call release at the originating side. The originating side sends a clear-forward signal 10. When the terminating side recognizes the clear-forward signal, it sends a backward signal 10. When the originating side recognizes the backward signal 10, it releases the call. Figure 17 Call release at originating side • Call release at the terminating side.
it sends a backward signal 10 to indicate that the line is idle. The originating side maintains the forward signal 10 and unblocks the local-end circuit for the next call. ITU-T interregister signaling Interregister signaling transmits and requests calling and called numbers. It adopts the multifrequency compelled (MFC) mode and includes forward signaling and backward signaling. Forward signaling exchange includes Group I and Group II, and backward signaling exchange includes Group A and Group B.
Designation Definition A-15 Congestion in an international exchange. Terminate interregister signaling interaction. Group II forward signals—Identify the calling party category. The system looks at the calling party category to decide whether the calling party can perform forced release or break-in. • Table 6 Group II forward signals Designation Definition II-1 Subscriber without priority. II-2 Subscriber with priority. II-3 Maintenance equipment. II-4 Spare for national use. II-5 Operator.
Figure 19 ITU-T R2 interregister signaling exchange process Originating side Calling number: 123 Terminating side Called number 789 Line signaling exchange Send called number digit 7 (I-7) Request next digit (A-1) Send called number digit 8 (I-8) Request calling party information (A-5) Send calling accounting category 2 (II-7) Request calling party information (A-5) Send calling number digit 1 (I-1) Request calling party information (A-5) Send calling number digit 2 (I-2) Request calling party information
Configuring the trunk direction and routing mode Step Command Remarks 1. Enter system view. system-view N/A 2. Enter E1 or T1 interface view. controller { e1 | t1 } number N/A 3. Create a timeslot set and enable R2 signaling for it. timeslot-set ts-set-number timeslot-list timeslots-list signal r2 By default, no timeslot set is created. 4. Enter R2 CAS view. cas ts-set-number N/A The default is dual. 5. Configure the trunk direction.
Configuring the duration for the terminating side to play ringback tones Step Command Remarks 1. Enter system view. system-view N/A 2. Enter E1 or T1 interface view. controller { e1 | t1 } number N/A 3. Create a timeslot set and enable R2 signaling for it. timeslot-set ts-set-number timeslot-list timeslots-list signal r2 By default, no timeslot set is created. 4. Enter R2 CAS view. cas ts-set-number N/A 5. Configure the duration for playing ringback tones.
• segment—After receiving the called number, the terminating side directly returns the "called party idle" interregister signaling, without waiting for the real state of the terminating side. If the called party is busy, the terminating side plays busy tones to the originating side. To configure the connection mode: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter E1 or T1 interface view. controller { e1 | t1 } number N/A 3. Create a timeslot set and enable R2 signaling for it.
Step Command Remarks metering enable By default, metering signal processing is disabled. 9. Configure the originating side to require the terminating side to send seizure acknowledgement signals. seizure-ack enable By default, the originating side requires the terminating side to send seizure acknowledgement signals. 10. Configure the ABCD bit pattern for line signals.
Step 8. Configure the originating side to send a number terminator to the terminating side after sending the called number. Command Remarks final-callednum enable By default, the originating side does not send a number terminator to the terminating side after sending the called number. By default, no signal code is configured for a special character. 9. Configure a signal code for a special character. special-character character number 10. Configure interregister signal values.
Step Command 1. Enter system view. system-view 2. Enter E1 or T1 interface view. controller { e1 | t1 } number 3. Bundle timeslots into a PRI set. pri-set [ timeslots-list range ] Binding a digital voice interface to a POTS entity Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter voice dial program view. dial-program N/A 4. Create a POTS entity and enter POTS entity view.
Figure 20 Network diagram Configuration procedure 1. Configure Router A: # Configure the IP address 1.1.1.1/24 for interface GigabitEthernet 1/0/1. system-view [RouterA] interface gigabitethernet 1/0/1 [RouterA-GigabitEthernet1/0/1] ip address 1.1.1.1 255.255.255.0 [RouterA-GigabitEthernet1/0/1] quit # Create a timeslot set on interface E1 2/4/1.
[RouterB-GigabitEthernet1/0/1] quit # Create a timeslot set on interface E1 2/4/1. [RouterB] controller e1 2/4/1 [RouterB-E1 2/4/1] timeslot-set 1 timeslot-list 1-31 signal r2 [RouterB-E1 2/4/1] quit # Configure the local number 07552001 for POTS entity 2001, and bind the digital voice interfaces line 2/4/1:1 to the POTS entity.
Figure 21 Network diagram Configuration procedure 1. Configure Router A: # Configure the IP address 1.1.1.1/24 for interface GigabitEthernet 1/0/1. system-view [RouterA] interface gigabitethernet 1/0/1 [RouterA-GigabitEthernet1/0/1] ip address 1.1.1.1 255.255.255.0 [RouterA-GigabitEthernet1/0/1] quit # Bundle the timeslots on interface E1 2/4/1 into a PRI set.
[RouterB] interface gigabitethernet 1/0/1 [RouterB-GigabitEthernet1/0/1] ip address 2.2.2.2 255.255.255.0 [RouterB-GigabitEthernet1/0/1] quit # Bundle the timeslots on interface E1 2/4/1 into a PRI set. [RouterB] controller e1 2/4/1 [RouterB-E1 2/4/1] pri-set [RouterB-E1 2/4/1] quit # Configure the local number 07552001 for POTS entity 2001, and bind the digital voice interfaces line 2/4/1:15 to the POTS entity.
Figure 22 Network diagram Configuration procedure 1. Configure Router A: # Configure the IP address 1.1.1.1/24 for interface GigabitEthernet 1/0/1. system-view [RouterA] interface gigabitethernet 1/0/1 [RouterA-GigabitEthernet1/0/1] ip address 1.1.1.1 255.255.255.0 [RouterA-GigabitEthernet1/0/1] quit # Configure the local number 0101001 for POTS entity 1001, and bind the digital voice interfaces line 2/5/1 to the POTS entity.
# Configure the local number 07552001 for POTS entity 2001, and bind the digital voice interfaces line 2/5/1 to the POTS entity.
Configuring voice entities Overview Voice entities include POTS and VoIP entities: • POTS entity—A local POTS entity connects to a local telephone and maintains local number information. A trunk POTS entity connects to the PSTN and maintains call destination information. • VoIP entity—Connects to the IP side and maintains the called information such as the called number and call destination. A VoIP entity can use SIP to make VoIP calls.
Creating a POTS entity and configuring basic parameters Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter dial program view. dial-program N/A 4. Create a POTS entity and enter POTS entity view. entity entity-number pots By default, no POTS voice entities exist. 5. (Optional.) Configure a description for the POTS entity. description string By default, no description is configured.
Step Command Remarks 5. Configure a codec for the POTS entity. codec { g711alaw | g711ulaw | g723r53 | g723r63 | g726r16 | g726r24 | g726r32 | g726r40 | g729a | g729br8 | g729r8 } [ bytes payload-size ] By default, no codecs are configured. To configure codecs for a POTS entity (method 2): Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Create a codec template. voice class codec tag By default, no codec templates exist. 4.
Configuring DTMF for a POTS entity There are two ways to transmit DTMF tones: inband signaling and out-of-band signaling. Inband signaling sends DTMF tones in RTP packets, and out-of-band signaling sends DTMF tones in SIP messages, or RFC 2883-compliant RTP messages (NTE mode). To use NTE mode, configure the outband nte command and the same payload type value on the originating and terminating sides. Otherwise, DTMF tones might fail to be transmitted.
Enabling VAD for a POTS entity The voice activity detection (VAD) discriminates between silence and speech on a voice connection according to their energies. With VAD, a POTS entity does not generate traffic during periods of silence. To enable VAD for a POTS entity: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter dial program view. dial-program N/A 4. Create a POTS entity and enter POTS entity view.
Configuring a VoIP entity This section covers the procedures for creating and configuring a VoIP entity. Configuration task list To configure a VoIP entity, perform the following tasks: Tasks at a glance (Required.) Creating a VoIP entity and configuring basic parameters (Optional.) Configuring codecs for a VoIP entity (Optional.) Configuring DTMF for a VoIP entity (Optional.) Setting the DSCP value for a VoIP entity (Optional.) Enabling VAD for a VoIP entity (Optional.
Configuring codecs for a VoIP entity By default, a VoIP entity has four codecs g729r8, g711alaw, g711ulaw, and g723r53 in descending order of priority. You can use one of the following methods to configure codecs for a VoIP entity: • Method 1: Configure codecs directly for a VoIP entity. • Method 2: Create a codec template, assign priorities to codecs in the codec template, and bind the codec template to a VoIP entity. Two parties must have the same codecs to communicate with each other.
Configuring DTMF for a VoIP entity There are two ways to transmit DTMF tones: inband signaling and out-of-band signaling. Inband signaling sends DTMF tones in RTP packets, and out-of-band signaling sends DTMF tones in SIP messages, or RFC 2883-compliant RTP packets (NTE mode). To use the NTE mode, configure command outband nte and the same payload type value on both the originating and terminating sides. Otherwise, DTMF tones might fail to be transmitted.
Enabling VAD for a VoIP entity The voice activity detection (VAD) discriminates between silence and speech on a voice connection according to their energies. With VAD, a POTS entity does not generate traffic during periods of silence in an active voice connection. To enable VAD: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter dial program view. dial-program N/A 4. Create a VoIP entity and enter VoIP entity view.
Task Command Display voice call information. display voice call-info { tag | all } Display the configuration of voice entities.
Configuring dial programs This chapter describes using dial programs to define call control policies such as caller control, number substitution, call sending, and number match policies. Configuration task list Tasks at a glance (Optional.) Configuring caller control (Optional.) Configuring caller group control (Optional.) Enabling private line auto ring-down (Optional.) Configuring a number match mode (Optional.) Configuring the maximum number of total calls allowed by a voice entity (Optional.
Examples As shown in Figure 24, configure caller control to permit only the number 1000 to call the number 2000. Figure 24 Caller control diagram You can configure caller control either on the calling side or on the called side to meet the requirement. • Method 1: Configure the calling side to permit only the calling number 1000 to call the number 2000.
Configuring a subscriber group Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter dial program view. dial-program N/A 4. Create a subscriber group and enter subscriber group view. subscriber-group group-id 5. (Optional.) Configure a description for the subscriber group. description text 6. Configure a match template for the subscriber group. match-template match-string By default, no subscriber group exists.
Configuring a number match mode Two number match modes are available: longest match and shortest match. Suppose you have configured match-template 0106688 on a voice entity and match-template 01066880011 on another voice entity. When a subscriber dials 01066880011: • If shortest match is used, the router matches and calls the number 0106688. • If longest match is used, the router matches and calls the number 01066880011.
Figure 25 Network diagram FX S2 /1 /1 Telephone B 20001234 1.1.1.1/24 FXS2/1/1 / /1 S2 FX Router B Router A 2 Telephone A 10001234 1.1.1.2/24 Telephone C 200012341234 Using shortest match 1. Configure Router A: # Configure POTS entity 1000.
After you dial number 20001234 at Telephone A, the number matches VoIP entity 2000 and Telephone B rings because the device uses shortest match mode by default. Configuring longest match # Configure the longest match mode on Router A. [RouterA-voice-dial] number-match longest After you dial number 20001234 at Telephone A and waits for a period of time, the number matches VoIP entity 2000 and Telephone B rings.
Number substitution on the calling router As shown Figure 26, the calling router performs number substitution for a number as follows: 1. The router matches the number against number substitution rules on the voice interface. If a match is found, the router replaces the number based on the matching rule. 2. If the match fails, the router matches the number against global number substitution rules. If a match is found, the router replaces the number based on the matching rule. 3.
2. If no match is found, the router selects a matching voice entity, and matches the number against rules on the voice entity. If a match is found, the router replaces the number based on the matching rule. 3. If no match is found, the router calls the called number if the callee is a local subscribe line, or initiates a call to the PSTN if the callee is in the PSTN.
Step Command Remarks 6. Configure a substitution rule. rule id input-template output-template [ number-type input-number-type output-number-type | numbering-plan input-numbering-plan output-numbering-plan ] * By default, no number substitution rule is configured. 7. (Optional.) Configure the preferred number substitution rule. first-rule id By default, the preferred number substitution rule is not configured. 8. Return to dial program view. quit N/A 9.
Configuring number substitution for a voice interface Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter dial program view. dial-program N/A 4. Create a number substitution rule list and enter number-substitute view. number-substitute list-number N/A 5. Configure a dot match rule. dot-match { end-only | left-right | right-left } By default, the dot match rule is end-only. 6. Configure a substitution rule.
[RouterA-voice-dial-substitute1] rule 0 ^010....$ .... [RouterA-voice-dial-substitute1] quit # Configure the destination address as 1.1.1.2 and the called number as 1001 for VoIP entity 1001. [RouterA-voice-dial] entity 1001 voip [RouterA-voice-dial-entity1001] address sip ip 1.1.1.2 [RouterA-voice-dial-entity1001] match-template 1001 [RouterA-voice-dial-entity1001] quit # Apply number substitution rule list 1 to the called number of outgoing calls.
[RouterA-voice-line2/1/1] substitute called 1 When you use Telephone A to call number 0101001, FXS interface 2/1/1 replaces 0101001 with 1001 and sends the call to the destination address 1.1.1.2. Then Telephone B rings. Configuring a priority for a voice entity If a number matches multiple voice entities, the router selects the voice entity with the highest priority. To configure a priority for a voice entity: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view.
Sends a truncated called number. When the match-template command configured for a voice entity contains dots ".", only the digits that match the dots are sent. • Procedure To configure a number sending mode: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter dial program view. dial-program N/A 4. Enter POTS voice entity view. entity entity-number pots N/A 5. Configure a number sending mode.
[RouterA-voice-dial-entity1001] line 2/4/1:0 # Configure a local number 2000 for POTS entity 2000, and bind line 2/2/1 to the entity. [RouterA-voice-dial] entity 2000 pots [RouterA-voice-dial-entity2000] match-template 2000 [RouterA-voice-dial-entity2000] line 2/2/1 2. Configure Router B: # Configure a timeslot set using R2 signaling for controller E1 2/4/1.
Configuring a dial prefix After you configure a dial prefix, the router adds the prefix before each called number. If the called number exceeds 31 digits, the router sends only the first 31 digits. To configure a dial prefix: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter dial program view. dial-program N/A 4. Enter POTS entity view. entity entity-number pots N/A 5. Configure a dial prefix.
# Create number substitution rule list 21101 and add rules to the list. system-view [RouterB] voice-setup [RouterB-voice] dial-program [RouterB-voice-dial] number-substitute 21101 [RouterB-voice-dial-substitute21101] rule 1 ^0101688$ 0001 [RouterB-voice-dial-substitute21101] rule 2 ^0103366$ 0002 [RouterB-voice-dial-substitute21101] rule 3 ^0102323$ 0003 [RouterB-voice-dial-substitute21101] quit # Create number substitution rule list 21102 and add rules to the list.
[RouterA-voice-dial] substitute incoming-call called 101 # Apply the number substitution rule list 102 to the calling numbers of incoming calls. This will change the called numbers 0210001, 0210002, and 0210003 into 0211234, 0216788, and 0216565, respectively. [RouterA-voice-dial] substitute incoming-call calling 102 # Configure the match template …. for POTS entity 1010, bind line 2/2/1 to the entity, and configure the entity to send all the digits of a called number.
Configuration procedure This example does not provide SIP server and digital voice interface configurations. For information about configuring SIP servers and digital voice interfaces, see "Configuring SIP" and "Configuring voice interfaces." 1. Configure Router A: # Configure subscriber group 1 that matches calling numbers starting with 1100 and configure subscriber group 2 that matches calling numbers starting with 1200.
[RouterA-voice-dial-entity2100] line 2/4/1:15 [RouterA-voice-dial-entity2100] send-number all [RouterA-voice-dial-entity2100] match-template 1..... 2. Configure Router B: # Configure POTS entity 2100 to match local numbers and bind line 2/4/1:15 to the entity.
Configuration procedure 1. Configure Router A: # Configure the called number template as 010…. for VoIP entity 2000, and configure the destination IP address as 1.1.1.3. system-view [RouterA] voice-setup [RouterA-voice] dial-program [RouterA-voice-dial] entity 2000 voip [RouterA-voice-dial-entity2000] match-template 010.... [RouterA-voice-dial-entity2000] address sip ip 1.1.1.
[RouterB-voice-dial-entity2001] line 2/1/2 [RouterB-voice-dial-entity2001] quit # Configure the maximum number of total calls allowed by VoIP entity 1000 as 2. [RouterB-voice-dial] entity 1000 voip [RouterB-voice-dial-entity1000] max-conn 2 3. Configure Router C: # Configure the local number template as 010…. for POTS entity 1000, bind FXO interface line 2/2/1 to the POTS entity, and configure the number sending mode as all.
Configuring SIP Overview The Session Initiation Protocol (SIP) is an application layer control protocol that can create, modify, and terminate multimedia sessions such as voice and video calls over IP networks. Terminology User agent A user agent (UA) is a SIP endpoint such as a phone, a gateway, or a router. There are two types of UAs: user agent client (UAC) and user agent server (UAS). A UAC sends SIP requests, and a UAS receives SIP requests and returns SIP responses.
SIP messages SIP is a text-based protocol. A SIP message is either a request from a client to a server, or a response from a server to a client. SIP requests include INVITE, ACK, OPTIONS, BYE, CANCEL, and REGISTER. • INVITE—Invites a user to join a call. • ACK—Acknowledges the response to a request. • OPTIONS—Queries for the capabilities. • BYE—Releases an established call. • CANCEL—Gives up a call attempt. • REGISTER—Registers with the SIP registrar.
Media authentication and encryption SIP supports two media stream protocols: Real-Time Transport Protocol (RTP) and Real-Time Transport Control Protocol (RTCP). RTP provides end-to-end transmission for real-time data, such as interactive voice and video. RTCP monitors transmission quality and provides congestion control and flow control. RTP and RTCP work together to achieve optimal transmission efficiency by providing efficient feedback and minimizing overheads.
Tasks at a glance (Required.) Use one of the following methods to specify a call destination address for a VoIP entity: • Configuring the call destination IP address for a VoIP entity • Configuring a VoIP entity to obtain the call destination address from a proxy server • Configuring the destination domain name and port number for a VoIP entity (Optional.) Enabling keepalive function for a VoIP entity (Optional.) Specifying a trusted node (Optional.
3. The UA sends a REGISTER request to the registrar. 4. The registrar returns a 401/407 response, challenging the originator to provide credentials. 5. The UA sends a REGISTER request that includes credentials to the registrar. 6. The registrar returns a 200 OK response to the UA if the authentication succeeds.
SIP UA responds with the username 1000 and password 3000 because no credentials binding contains the realm server4. Credentials selection for a phone number that exists on multiple voice entities Upon receiving a 401/407 response for a phone number that exists on multiple voice entities, the SIP UA considers the phone number to belong to the voice entity with the smallest ID. The SIP UA selects the credentials for the phone number in the following order: 1.
configured by using the credentials command, and the credentials pass the authentication. The output from the display voice sip register-status command shows that the phone number 1000 belongs to voice entity 1. display voice sip register-status Number Entity Registrar Server Expires Status -------------------------------------------------------------------------------1000 1 192.168.4.240:5060 2877 Online Configuring global SIP credentials Step Command Remarks 1. Enter system view.
Configuring registrar information Perform this task to specify a registrar. The expires keyword sets the registration expiration time, and the refresh-ratio keyword sets the refresh percentage. When the registration time reaches the registration expiration time multiplied by the refresh percentage, a voice entity or SIP trunk re-registers the number with the registrar to avoid expiration. To configure registrar information: Step Command Remarks 1. Enter system view. system-view N/A 2.
Configuring a VoIP entity to obtain the call destination address from a proxy server Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter SIP view. sip N/A 4. Specify the proxy server. proxy { dns domain-name port port-number | ip ip-address [ port port-number ] } By default, no proxy server is specified. 5. Return to voice view. quit N/A 6. Enter dial program view. dial-program N/A 7. Enter VoIP entity view.
The keepalive function does not take effect for a VoIP entity that has been shut down by using the shutdown command. To enable keepalive function for a VoIP entity: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter dial program view. dial-program N/A 4. Enter VoIP entity view. entity entity-number voip N/A 5. Enable keepalive function for the VoIP entity.
Configuring source interface binding for outgoing SIP messages or media packets Perform this task to specify the source interface for outgoing SIP messages and media packets. The IP address of the specified source interface is used as the source address. If the source interface obtains its IP address from a DHCP or PPPoE server, you do not need to manually re-configure the source address when the IP address of the source interface is changed.
Configuring source interface binding on a VoIP entity Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter SIP view. sip N/A 4. Enter VoIP entity view. entity entity-number voip N/A 5. Configure source interface binding for outgoing SIP messages or media packets. voice-class sip bind { control | media } source interface interface-type interface-number By default, the default global source interface is used.
To configure periodic refresh of SIP sessions: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter SIP view. sip N/A session refresh By default, session refresh is disabled if the device acts as a UAC, and is enabled if the device acts as a UAS. 4. Enable session refresh. Configure this command on the UAC. 5. Set the maximum session expiration time and minimum session expiration time.
receives the INVITE request preferably obtains caller information from this field regardless of whether the local Remote-Party-ID header field is set. NOTE: The Remote-Party-ID header field cannot coexist with the P-Preferred-Identity or P-Asserted-Identity header field. To configure caller privacy: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter SIP view. sip N/A 4. Add the Privacy header field into INVITE requests.
If the UAS fails to receive the PRACK message from the UAC within a specified time, the UAS retransmits the 18x response. 4. The UAS returns a 200 OK. For more information about reliable provisional responses, see RFC 3262. You must enable reliable provisional responses and configure the same value for the value argument on both the UAC and UAS.
Configuring UDP or TCP for outgoing SIP calls Specifying UDP or TCP as the global transport protocol Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter SIP view. sip N/A 4. Specify UDP or TCP as the global transport protocol for outgoing SIP calls. session transport { tcp | udp } By default, UDP is used as the global transport protocol. Specifying UDP or TCP as the transport protocol on a VoIP entity Step Command Remarks 1.
Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter SIP view. sip N/A 4. Set the aging time for TCP connections. timers connection aging tcp tcp-age-time By default, the aging time for TCP connections is 5 minutes. Configuring SIP security This section describes how to configure TLS for outgoing SIP calls and media flow protocols for SIP calls.
Step Command Remarks 2. Enter voice view. voice-setup N/A 3. Enter SIP view. sip N/A 4. Specify TLS as the global transport protocol for outgoing SIP calls. session transport tcp [ tls ] By default, UDP is used as the global transport protocol. Specifying TLS as the transport protocol for outgoing SIP calls on a VoIP entity Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter dial program view. dial-program N/A 4.
Configuring SRTP for SIP calls The differences between the srtp and srtp fallback commands are as follows: • With the srtp command configured: { { • The device includes crypto and RTP/SAVP parameters in outgoing INVITE requests and disconnects the call after receiving a 488 response. The device can only accept calls using SRTP.
VoIP entity view takes precedence over the global configuration. A VoIP entity uses the global configuration only when no URL scheme is configured in VoIP entity view. Specifying a global URL scheme for outgoing SIP calls Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter SIP view. sip N/A 4. Specify a global URL scheme for outgoing SIP calls. url { sip | sips } By default, the SIP scheme is used.
Displaying and maintaining SIP Use display commands in any view. Task Command Display SIP UA registration status information. display voice sip register-status Display PSTN cause-to-SIP status mappings. display voice sip map { pstn-sip | sip-pstn } Display SIP calling information. display voice sip call Display information about SIP connections. display voice sip connection { tcp | tls } Disconnect a specific SIP connection.
[RouterA-voice-dial-entity1111] line 2/2/1 [RouterA-voice-dial-entity1111] match-template 1111 2. Configure Router B: # Configure an IP address for GigabitEthernet 2/1/1. system-view [RouterB] interface gigabitethernet 2/1/1 [RouterB-GigabitEthernet2/1/1] ip address 192.168.2.2 255.255.255.0 [RouterB-GigabitEthernet2/1/1] quit # Configure local number 2222 for POTS entity 1111 and bind FXS interface 2/2/1 to the POTS entity.
Configuration procedure 1. Configure Router A: # Configure an IP address for GigabitEthernet 2/1/1. system-view [RouterA] interface gigabitethernet 2/1/1 [RouterA-GigabitEthernet2/1/1] ip address 192.168.2.1 255.255.255.0 [RouterA-GigabitEthernet2/1/1] quit # Specify the SIP registrar and proxy server. [RouterA] voice-setup [RouterA-voice] sip [RouterA-voice-sip] registrar 1 ip 192.168.2.3 [RouterA-voice-sip] proxy ip 192.168.2.3 # Configure user name routerA and plaintext password 1234.
# Configure user name routerB, plaintext password 7890, and domain name server1. [RouterB-voice-dial-entity2222] user routerB password simple 7890 realm server1 # Configure VoIP entity 2222 to get the call destination address from the proxy server and to match number 1111.
# Configure local number 1111 for POTS entity 1111 and bind FXS interface 2/2/1 to the POTS entity. [RouterA-voice-dial] entity 1111 pots [RouterA-voice-dial-entity1111] line 2/2/1 [RouterA-voice-dial-entity1111] match-template 1111 2. Configure Router B: # Configure an IP address for GigabitEthernet 2/1/1. system-view [RouterB] interface gigabitethernet 2/1/1 [RouterB-GigabitEthernet2/1/1] ip address 192.168.2.2 255.255.255.
[RouterA-GigabitEthernet2/1/1] quit # Configure TCP as the global transport protocol for outgoing SIP calls. [RouterA] voice-setup [RouterA-voice] sip [RouterA-voice-sip] session transport tcp [RouterA-voice-sip] quit # Configure the called number as 2222 for VoIP entity 2222, and configure the destination IP address as 192.168.2.2. [RouterA-voice] dial-program [RouterA-voice-dial] entity 2222 voip [RouterA-voice-dial-entity2222] address sip ip 192.168.2.
Configuring SIP to use TLS as the transport protocol Network requirements As shown in Figure 39, configure SIP to use TLS as the transport protocol on Router A and Router B, so phone 1111 can call phone 2222 over TLS. Figure 39 Network diagram Configuration procedure In this example, the CA server runs RSA Keon. To make sure the certificate on the device is valid, the device system time must be earlier than the expiration time of the certificate. 1.
# Obtain the CA certificate and save it locally. [RouterA] pki retrieve-certificate domain voice ca # Submit a certificate request manually. [RouterA] pki request-certificate domain voice # Create an SSL server policy, and specify a PKI domain for the SSL server policy. [RouterA] ssl server-policy server [RouterA-ssl-server-policy-server] pki-domain voice # Create an SSL client policy, and specify a PKI domain for the SSL client policy.
# Configure the URL of the registration server in the form of http://host:port/Issuing Jurisdiction ID, where Issuing Jurisdiction ID is a hexadecimal string generated on the CA server. [RouterB-pki-domain-voice] certificate request url http://192.168.2.88:446/bd0683e5a369eb4edbb4ef502eaca6ec42d24e97 # Specify the CA for accepting certificate requests. [RouterB-pki-domain-voice] certificate request from ca # Specify the PKI entity name as voice.
Configuring out-of-band DTMF Network requirements As shown in Figure 40, configure Router A and Router B to use out-of-band signaling to transmit DTMF tones so the phones 1111 and 2222 can call each other. Figure 40 Network diagram Configuration procedure 1. Configure Router A: # Configure an IP address for GigabitEthernet 2/1/1. system-view [RouterA] interface gigabitethernet 2/1/1 [RouterA-GigabitEthernet2/1/1] ip address 192.168.2.1 255.255.255.
[RouterB-voice-dial-entity1111] match-template 1111 # Configure out-of-band DTMF for VoIP entity 1111. [RouterB-voice-dial-entity1111] outband sip [RouterB-voice-dial-entity1111] quit # Configure local number 2222 for POTS entity 2222 and bind FXS interface 2/2/1 to the POTS entity. [RouterB-voice-dial] entity 2222 pots [RouterB-voice-dial-entity2222] line 2/2/1 [RouterB-voice-dial-entity2222] match-template 2222 # Configure out-of-band DTMF for POTS entity 2222.
Configuring SIP trunk This chapter describes how to configure SIP trunk. Background As shown in Figure 41, in a typical telephone network, a PBX forwards internal calls among enterprise phones and forwards outbound calls over a PSTN trunk. Figure 41 Typical telephone network With the development of IP technology, many enterprises deploy SIP-based IP-PBX networks as shown in Figure 42. All internal calls are placed using SIP, and external calls are still placed over a PSTN trunk.
Figure 43 All IP-based network Features SIP trunk has the following features: • The SIP trunk device and the ITSP only establish one secure and QoS guaranteed SIP trunk link. The SIP trunk link can carry multiple concurrent calls, and the ITSP only authenticates the link instead of each SIP call carried on this link. • The enterprise IP-PBX forwards internal calls. The SIP trunk device forwards outbound calls to the ITSP, and then the devices in the ITSP forward the calls to the PSTN.
Figure 44 SIP trunk network diagram Protocols and standards • RFC 3261 • RFC 3515 • SIP connect Technical Recommendation v1.1 SIP trunk configuration task list Tasks at a glance (Required.) Enabling SIP-to-SIP calling (Required.) Configuring a SIP trunk account (Optional.) Enabling codec transparent transmission (Optional.) Enabling media flow-around (Optional.) Enabling DO-EO conversion Enabling SIP-to-SIP calling After you enable SIP-to-SIP calling, the device works as a SIP trunk device.
Configuring a SIP trunk account A SIP trunk account contains a phone number, credentials, and realms assigned by the service provider. SIP can send a REGISTER request for the phone number to up to six registrars specified by using the registrar command. SIP uses the realm value in 401/407 responses from the registrars to identify the matching credentials. You can configure up to 12 realm values for a phone number, and up to 128 SIP trunk accounts on the device.
Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter dial program view. dial-program N/A 4. Enter VoIP entity view. entity entity-number voip N/A media flow-around By default, the SIP trunk device changes the media address of a media packet to its own address before forwarding the media packet. 5. Enable media flow-around.
SIP trunk configuration examples Network requirements As shown in Figure 45, configure the SIP trunk device to forward calls between the private network and public network. Figure 45 Network diagram ITSP-A Enterprise private network FXS2/1/1 Public network 1.1.1.1/24 1.1.1.2/24 2.1.1.1/24 IP 2.1.1.2/24 SIP trunk Router A SIP trunk device FXS2/1/1 Router B 1000 2000 SIP server 10.1.1.2/24 Configuration procedure 1. Configure Router A: # Specify local number 2000 on POTS entity 2000.
[TG-voice-sip] registrar 1 ip 10.1.1.2 [TG-voice-sip] quit # Configure the destination address as 2.1.1.2 for outbound calls from private phone 2000 to public phone 1000. [TG-voice] dial-program [TG-voice-dial] entity 1 voip [TG-voice-dial-entity1] address sip ip 2.1.1.2 [TG-voice-dial-entity1] match-template 1000 [TG-voice-dial-entity1] quit # Configure the destination address as 1.1.1.1 for inbound calls from public phone 1000 to private phone 2000.
Configuring call services Call services include call waiting, call hold, call forwarding, call transfer, call backup, MWI, and three-party conference. Call waiting When subscriber C calls subscriber A who is in a conversation with subscriber B, the call is not rejected. Like a normal call, subscriber C hears ringback tones, and subscriber A hears call waiting tones. This is call waiting. Subscriber A can answer the new call by using either of the following methods • Pressing hookflash.
C (the final recipient). After Subscriber A hangs up, the call between subscriber B and subscriber C is established. This is call transfer. There are three types of call transfer: • Unattended transfer—The originator hangs up before receiving ringback tones from the final recipient. • Half-attended transfer—The originator hangs up after receiving ringback tones from the final recipient and before speaking with the final recipient.
3. Presses hookflash and presses 2. Each party restores its state before entering the three-party conference. The call between Telephone B and Telephone A (the other party in the original active call) is held, and the call between Telephone B and Telephone C can continue. 4. Presses hookflash again and presses 3. The three parties re-enter the three-party conference. A participant of the three-party conference can invite another party to join the conference to implement conference chaining.
Configuring the call hold mode There are two call hold modes: • Silent mode (inactive)—During call hold, the held party hears silence. This mode is configured on the holding party (the initiator of call hold) to signal the held party to close the transmit and receive media channels of the held party. • Unidirectional playing mode (sendonly)—During call hold, the held party hears tones or music played by a third-party music server.
Configuring MWI There are two MWI types: • Solicited—The SIP UA has subscribed to a voice mail server during registration and can receive NOTIFY messages from the server without sending SUBSCRIBE messages. • Unsolicited—The SIP UA needs to subscribe to a voice mail server by sending SUBSCRIBE messages before it can receive NOTIFY messages from the server.
Step Command Remarks 4. Configure the voice mail server information for the SIP UA. mwi-server { dns domain-name | ip ip-address } [ port port-number ] [ expires seconds ] [ transport { tcp [ tls ] | udp } ] [ scheme { sip | sips } ] unsolicited By default, no voice mail server information is configured. 5. Configure the registrar information for the SIP UA.
Configuration procedure Before performing the following configuration, make sure Router A, Router B and Router C can reach each other. 1. Configure Router A: # Create VoIP entity 2000, configure the destination IP address as 10.1.1.2, and configure the called number as 2000. system-view [RouterA] voice-setup [RouterA-voice] dial-program [RouterA-voice-dial] entity 2000 voip [RouterA-voice-dial-entity2000] address sip ip 10.1.1.
[RouterC-voice-dial-entity3000] line 2/1/1 [RouterC-voice-dial-entity3000] match-template 3000 [RouterC-voice-dial-entity3000] quit # Create VoIP entity 1000, configure the destination IP address as 10.1.1.1, and configure the called number as 1000. [RouterB-voice-dial] entity 1000 voip [RouterB-voice-dial-entity1000] address sip ip 10.1.1.
[RouterA-voice] dial-program [RouterA-voice-dial] entity 2000 voip [RouterA-voice-dial-entity2000] address sip ip 10.1.1.2 [RouterA-voice-dial-entity2000] match-template 2000 [RouterA-voice-dial-entity2000] quit # Configure the local number as 1000 for POTS entity 1000, and bind FXS interface line 2/1/1 to the POTS entity. [RouterA-voice-dial] entity 1000 pots [RouterA-voice-dial-entity1000] line 2/1/1 [RouterA-voice-dial-entity1000] match-template 1000 2. Configure Router B: # Create VoIP entity 3000.
Figure 49 Network diagram Configuration procedure Before performing the following configuration, make sure Router A, Router B and Router C can reach each other. 1. Configure Router A: # Create VoIP entity 2000. Configure the destination IP address as 10.1.1.2, and configure the called number as 2000. system-view [RouterA] voice-setup [RouterA-voice] dial-program [RouterA-voice-dial] entity 2000 voip [RouterA-voice-dial-entity2000] address sip ip 10.1.1.
[RouterC] voice-setup [RouterC-voice] dial-program [RouterC-voice-dial] entity 3000 pots [RouterC-voice-dial-entity3000] line 2/1/1 [RouterC-voice-dial-entity3000] match-template 3000 Verifying the configuration Verify that Telephone B and Telephone C can establish a call by using call transfer. Call backup configuration example Network requirements As shown in Figure 50, Router A and Router B are connected through two IP links. One of the two links has a higher priority and serves as the primary link.
system-view [RouterB] voice-setup [RouterB-voice] dial-program [RouterB-voice-dial] entity 2000 pots [RouterB-voice-dial-entity2000] line 2/1/1 [RouterB-voice-dial-entity2000] match-template 2000 Verifying the configuration When the primary link (corresponding to VoIP entity 2000) fails, place a call from Telephone A to Telephone B to verify that Telephone A can establish a call with Telephone B through the backup link (corresponding to VoIP entity 3000).
# Configure the destination IP address as 20.1.1.2 for VoIP entity 3000, and configure the called number as 3000. system-view [RouterB] voice-setup [RouterB-voice] dial-program [RouterB-voice-dial] entity 3000 voip [RouterB-voice-dial-entity3000] address sip ip 20.1.1.2 [RouterB-voice-dial-entity3000] match-template 3000 [RouterB-voice-dial-entity3000] quit # Configure the destination IP address as 10.1.1.1 for VoIP entity 1000, and configure the called number as 1000.
Figure 52 Network diagram Configuration procedure 1. Configure the VCX server: { Configure the call processing server: Open the Web interface of the server and select Central Management Console. Configure the telephone information of Telephone A and Telephone B, with the subscriber passwords as 1000 and 2000, respectively. Figure 53 uses Telephone A as an example.
Figure 54 Configuration page of call processing server (2) { Configure the unified messaging server: # Configure the mailbox access number as 9000. Open the Web interface of the server, select IP Messaging Web Provisioning to log in to the unified messaging server, and click the Configuration link. The Configuration Option box appears, as shown in Figure 55. Figure 55 Configuration page of unified messaging server Select 9000 from the Main Voicemail Access Number List, as shown in Figure 56.
# Configure the voice mailbox of Telephone A Click the Edit A Mailbox link, enter the mailbox access number 9000 of Telephone A, and then check whether the mailbox is created successfully. If you are prompted that the mailbox is not present, click the Create/Delete Mailboxes link to create the mailbox of Telephone A, with the mailbox number as 9000. 2.
# Configure the local number as 2000 for POTS entity 2000, and bind FXS interface line 2/1/1 to the POTS entity. [RouterB-voice-dial] entity 2000 pots [RouterB-voice-dial-entity2000] line 2/1/1 [RouterB-voice-dial-entity2000] match-template 2000 [RouterB-voice-dial-entity2000] quit [RouterB-voice-dial] quit [RouterB-voice] quit [RouterB-voice] quit # Specify the IP address for the registrar and proxy server. [RouterB-voice] sip [RouterB-voice-sip] registrar 1 ip 100.1.1.
Configuring fax over IP Fax over IP (FoIP) transmits and receives faxes over the Internet. Figure 57 FoIP network diagram PSTN Internet Fax PSTN Fax Fax transmission flow Fax transmission operates in the following phases: 1. Call establishment—The calling terminal sends a Calling Tone (CNG) to identify itself as a fax machine. The called terminal sends a Called Station Identifier (CED) tone to identify itself as a fax machine. 2.
Configuring the fax protocol Configuring the standard T.38 protocol The standard T.38 protocol refers to the ITU-T T.38 protocol. It converts T.30-compliant fax signals to T.38 fax packets. To configure the standard T.38 protocol: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter dial program view. dial-program N/A 4. Enter voice entity view. entity entity-number { pots | voip } N/A By default, the standard T.38 protocol is used. 5.
Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter dial program view. dial-program N/A 4. Enter voice entity view. entity entity-number { pots | voip } N/A 5. Enable ECM. fax ecm By default, ECM is disabled. Use this command on both the calling and called devices. Configuring an NSF code for nonstandard capabilities negotiation Perform this task to match the nonstandard capabilities of the peer fax machine.
If you specify the disable keyword, the fax function is disabled. To configure the maximum fax rate for rate training: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter dial program view. dial-program N/A 4. Enter voice entity view. entity entity-number { pots | voip } N/A 5. Configure the maximum fax rate for rate training.
Adjusting the transmit energy level Perform this task if you cannot establish fax calls when all other settings are correct. To adjust the transmit energy level: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter dial program view. dial-program N/A 4. Enter voice entity view. entity entity-number { pots | voip } N/A 5. Adjust the transmit energy level. fax level level The default value is –15 dBm.
Step Command Remarks 3. Enter dial program view. dial-program N/A 4. Enter voice entity view. entity entity-number { pots | voip } N/A 5. Configure modem pass-through. modem passthrough { nse [ payload-type number ] | protocol } codec { g711alaw | g711ulaw } By default, modem pass-through is not used. FoIP configuration examples This section provides FoIP configuration examples. Configuring FoIP Network requirements As shown in Figure 58, configure Router A and Router B to use the standard T.
# Configure the called fax number as 1000 for VoIP entity 1000, and configure the destination IP address as 1.1.1.1. system-view [RouterB] voice-setup [RouterB-voice] dial-program [RouterB-voice-dial] entity 1000 voip [RouterB-voice-dial-entity1000] match-template 1000 [RouterB-voice-dial-entity1000] address sip ip 1.1.1.1 # Configure the standard T.38 protocol for VoIP entity 1000, with a low redundancy value of 4.
[RouterA-voice-dial-entity2000] quit # Configure the local number as 1000 for POTS entity 1000, and bind FXS interface line 2/1/1 to the POTS entity. [RouterA-voice-dial] entity 1000 pots [RouterA-voice-dial-entity1000] match-template 1000 [RouterA-voice-dial-entity1000] line 2/1/1 # Configure modem pass-through for POTS entity 1000. [RouterA-voice-dial-entity1000] modem passthrough protocol codec g711alaw 2.
Support and other resources Contacting HP For worldwide technical support information, see the HP support website: http://www.hp.
Conventions This section describes the conventions used in this documentation set. Command conventions Convention Description Boldface Bold text represents commands and keywords that you enter literally as shown. Italic Italic text represents arguments that you replace with actual values. [] Square brackets enclose syntax choices (keywords or arguments) that are optional. { x | y | ... } Braces enclose a set of required syntax choices separated by vertical bars, from which you select one.
Network topology icons Represents a generic network device, such as a router, switch, or firewall. Represents a routing-capable device, such as a router or Layer 3 switch. Represents a generic switch, such as a Layer 2 or Layer 3 switch, or a router that supports Layer 2 forwarding and other Layer 2 features. Represents an access controller, a unified wired-WLAN module, or the switching engine on a unified wired-WLAN switch. Represents an access point. Represents a mesh access point.
Index ABCDEFOPRST Configuring MWI,125 A Configuring number substitution,67 Adjusting parameters for voice interfaces,18 Configuring R2 signaling,34 Analog voice interface configuration examples,21 Configuring SIP security,99 B Configuring SIP UA registration,86 Background,114 Configuring the call destination address for a VoIP entity,90 Binding a digital voice interface to a POTS entity,45 Configuring the call hold mode,124 Binding an FXS interface to an FXO interface,11 Configuring the fax pr
FoIP configuration examples,143 S FoIP configuration task list,138 Setting the global DSCP value,102 FXO interface,1 SIP configuration task list,85 FXS interface,1 SIP trunk configuration examples,119 O SIP trunk configuration task list,116 SIP UA configuration examples,103 Overview,52 Specifying a trusted node,92 Overview,83 Specifying a URL scheme,101 P T Protocols and standards,116 Typical applications,115 R Related information,146 150
