Page 1 HPE FlexNetwork MSR Router Series Comware 5 Voice Configuration Guide Part number: 5200-2325 Software version: CMW710-R2516 Document version: 6W107-20160831...
Page 2 © Copyright 2016 Hewlett Packard Enterprise Development LP The information contained herein is subject to change without notice. The only warranties for Hewlett Packard Enterprise 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. Hewlett Packard Enterprise shall not be liable for technical or editorial errors or omissions contained herein.
Page 3: Table Of Contents
Contents Voice overview ················································································ 1 Introduction to VoIP ···················································································································· 1 VoIP system ······················································································································· 1 Basic VoIP call flow ············································································································· 1 Hardware compatibility with voice ·································································································· 2 VoIP features ····················································································································· 2 Configuring voice functions ·········································································································· 3 ...
Page 4 Configuring analog voice subscriber lines ············································ 29 Signal tone ····························································································································· 29 FXS voice subscriber line ·········································································································· 29 FXS interface ··················································································································· 29 CID ································································································································ 29 FXO voice subscriber line ·········································································································· 30 FXO interface ··················································································································· 30 ...
Page 5 Configuration guidelines ····································································································· 53 Configuration procedure ····································································································· 53 Configuring global default parameters for voice subscriber lines ························································· 54 Mirroring PCM, RTP, or voice command data on an analog voice subscriber line ·································· 54 Displaying and maintaining analog voice subscriber lines ································································· 54 ...
Page 6 Enabling the AMD function ·································································································· 89 Configuring AMD parameters ······························································································· 89 Configuring reverse charging function ·························································································· 90 Enabling the router to treat DISCONNECT messages with PI 8 as standard DISCONNECT messages ······ 90 Mirroring PCM, RTP packets, or voice command data on a digital voice subscriber line ·························· 91 ...
Page 7 Terminology ··················································································································· 130 SIP functions and features ································································································ 131 SIP messages ················································································································ 132 SIP fundamentals ············································································································ 132 Support for transport layer protocols ·························································································· 135 SIP security ·························································································································· 135 Signaling encryption ········································································································· 135 Media flow encryption ······································································································· 136 ...
Page 8 Failed to set up calls in the proxy server approach to SIP routing ··············································· 175 Failed to register with the registrar ······················································································ 175 Failed to set up point-to-point calls ······················································································ 175 Failed to send register requests ·························································································· 175 ...
Page 9 Configuring H.323 ········································································ 216 H.323 architecture ·················································································································· 217 H.323 fundamentals ··············································································································· 218 Gatekeeper discovery ······································································································ 218 Registration ···················································································································· 218 Address translation ·········································································································· 218 Admission control ············································································································ 218 Call setup ······················································································································ 220 Call proceeding ··············································································································· 220 ...
Page 10 Configuring hunt group ············································································································ 237 Configuration prerequisites ································································································ 237 Enabling hunt group ········································································································· 237 Configuring hunt group priority level ···················································································· 238 Configuration example ······································································································ 238 Configuring incoming call barring ······························································································· 239 Configuration prerequisites ································································································ 239 ...
Page 11 Monitoring local interfaces ································································································· 266 Monitoring remote IP addresses ························································································· 267 Configuring fax over IP ·································································· 270 FoIP protocols and standards ··································································································· 270 Fax flow ······························································································································· 270 Hardware compatibility with FoIP ······························································································· 271 FoIP configuration task list ······································································································· 271 ...
Page 12 Configuring VoFR ········································································· 310 Overview ······························································································································ 310 Fundamental VoFR architecture ························································································· 310 Protocols and standards ··································································································· 310 Call flow in dynamic mode ································································································· 311 Call flow in FRF.11 trunk mode ·························································································· 311 Hardware compatibility with VoFR ····························································································· 312 ...
Page 13 Card number/password process configuration ······································································· 3 42 1 2 2 8 H T roubleshooting voice RADIUS ································································································· 3 45 5 9 8 H 1 2 2 9 H D ocument conventions and icons ···················································· 3 46 5 9 9 H 1 2 3 0 H C onventions ·························································································································...
Page 14: Voice Overview
Voice overview Introduction to VoIP Voice over IP (VoIP) enables IP networks to provide voice services such as plain old telephone service (POTS). In VoIP, the voice gateway encapsulates voice signals into packets to transmit. IP telephony is a typical VoIP application. Interworking between PSTN and IP is implemented through VoIP gateways.
Page 15: Hardware Compatibility With Voice
The session application collects the digits dialed by the user. The session application compares the collected digits with the match template while collecting digits. After finding a match template for the called number, the originating VoIP gateway maps the number to the terminating VoIP gateway. The originating VoIP gateway initiates a VoIP call to the terminating VoIP gateway over the IP network and establishes a logical channel for the call to send and receive voice data.
Page 16: Configuring Voice Functions
PBX to distinguish matching tones. Implement busy-tone detection by configuring these parameters on interfaces. Configuring voice functions Figure 2 shows that voice function configuration includes four parts: voice subscriber line, voice entity, voice protocol, and dial plan. Figure 2 Voice function configuration Voice function configuration Voice...
Page 17 Figure 3 Voice function configuration procedure Start Configure a link connection Is the link available? Configure voice entity Configure number Is number substitution for dial substitution plans necessary? Configure voice subscriber line Configure number application for dial plans Configure voice protocol Is the call Troubleshoot...
Page 18: Voice Subscriber Lines
Operation Reference Configure number application for the dial plan adopted in the network Configuring dial plans diagram. Configure the following voice protocols according to the service and networking environment: • Configuring H.323 H.323 protocol • Configuring SIP SIP protocol • Configuring fax over IP Fax protocol Check whether the network requirements are met:...
Page 19: Voice Protocols
• A VoIP entity relates a call entity with a routing policy. Compared with the POTS entity, the VoIP entity corresponds to the IP network side. VoIP configuration implements the binding between telephone numbers and destination addresses (IP addresses or server addresses). •...
Page 20: Dial Plan
FoIP complies with ITU-T T.30 and T.4 on PSTN and T.38 on the IP network. T.30 defines the procedures necessary for document transmission between facsimile terminals on PSTN. It gives detailed descriptions and stipulations on the communication process, signal format, control signaling, and error correction of Group 3 facsimile terminals on the general switched telephone network.
Page 21: Configuring Voice Entities
Configuring voice entities Overview The voice entity configuration involves: • POTS voice entity configuration • VoIP voice entity configuration According to the position of the caller or callee, a complete telephone-to-telephone connection can be divided into four call segments, each of which corresponds to a voice entity. Figure 4 Two types of voice entities in the VoIP voice communication Figure 4 shows that two types of voice entities are involved in VoIP communication:...
Page 22: Voice Entity Configuration Task List
Voice entity configuration task list Task Remarks Configuring a POTS voice entity Required. Configuring a VoIP voice entity Required. Configuring the timeout interval for RTP streams Optional. Enabling local call identification Optional. Configuring voice call performance-related parameters Optional. Configuring global default parameters for voice entities Optional.
Page 23: Configuring Basic Functions
Step Command Remarks Create a POTS voice entity and entity entity-number pots enter POTS voice entity view. By default, no number template Bind a number template: bind a is bound to the local voice number template to a local subscriber line. If the POTS subscriber line, or if the POTS match-template match-string voice entity serves as a trunk,...
Page 24: Configuring The Local Pots Voice Entity To Play Ringback Tones
Step Command Remarks Create a POTS voice entity and enter POTS voice entity entity entity-number pots view. Optional. By default, the codec with the first priority is g729r8, that with the second priority is g711alaw, that with the third priority is g711ulaw, compression { 1st-level | and that with the fourth priority is 2nd-level | 3rd-level | 4th-level }...
Page 25: Configuring Dtmf Transmission
Step Command Remarks Optional. By default, the local POTS voice entity does not play ringback Enable the local POTS voice tones. send-ring entity to play ringback tones. This command is only applicable for POTS entities bound to a non-FXS or non-FXO voice subscriber lines.
Page 26: Enabling Vad
Enabling VAD The voice activity detection (VAD) discriminates between silence and speech on a voice connection according to their energies. VAD reduces the bandwidth requirements of a voice connection by not generating traffic during periods of silence in an active voice connection. Speech signals are generated and transmitted only when an active voice segment is detected.
Page 27: Configuring The Jitter Buffer
Configuring the jitter buffer Jitter, packet loss, and packet disorder occurs during the transmission of voice packets over IP networks. By using a series of adaptive algorithms, the jitter buffer stores the received IP packets for a period of time and sends the packets in evenly spaced intervals. The jitter buffer reduces packet delay and jitter, and improves the communication quality.
Page 28: Creating A Voip Voice Entity
Creating a VoIP voice entity Step Command Remarks Enter system view. system-view Enter voice view. voice-setup Enter voice dial program dial-program view. Create VoIP voice entity and entity entity-number voip enter its view. Configure a number By default, no called number template on the terminating match-template match-string template is configured for the...
Page 29: Configuring Dtmf Transmission
Step Command Remarks Optional. By default, the codec with the first priority is g729r8, that with the second priority is g711alaw, that with the third priority is g711ulaw, compression { 1st-level | and that with the fourth priority is 2nd-level | 3rd-level | 4th-level } g723r53.
Page 30 before the gateway (GW) receives a Connect message, avoiding H.245 message exchange on TCP connection, and thereby shortening connection time. There is no capability negotiation process in fast connection mode, so the capability of both parties is determined by the terminating GW.
Page 31: Configuring Out-Of-Band Dtmf Transmission In Fast Connection Mode
Step Command Remarks Optional. Enable tunneling on the voip called-tunnel enable terminating GW. Enabled by default. Configuring out-of-band DTMF transmission in fast connection mode Step Command Remarks Enter system view. system-view Enter voice view. voice-setup Enter voice dial program dial-program view.
Page 32: Enabling Vad
Step Command Remarks Optional. Configure the out-of-band outband { h225 | h245 } DTMF transmission. Inband transmission by default. Enabling VAD Step Command Remarks Enter system view. system-view Enter voice view. voice-setup Enter voice dial program dial-program view. Create a VoIP voice entity and enter VoIP voice entity entity entity-number voip view.
Page 33: Setting The Keepalive Interval
Step Command Remarks Enter voice view. voice-setup Enter voice dial program dial-program view. Create a VoIP voice entity and enter VoIP voice entity entity entity-number voip view. Optional. Set the operating mode of By default, the adaptive mode is jitter-buffer mode adaptive the jitter buffer to adaptive.
Page 34: Enabling Local Call Identification
Step Command Remarks Optional. Configure the timeout rtp-detect timeout value By default, the timeout interval for interval for RTP streams. RTP streams is 120 seconds. Enabling local call identification Introduction Figure 6, Telephone A originates a call to Telephone B through a SIP server. If the link As shown in between the SIP server and Router B fails, the SIP server forwards the call back to Router A, which then forwards the call through its FXO interface.
Page 35: Configuring Voice Call Performance-Related Parameters
Configuring voice call performance-related parameters This section describes the configuration procedure for voice call performance-related parameters. Configuration prerequisites You have completed the required configurations for a voice entity. Configuration procedure To configure voice performance-related parameters: Step Command Remarks Enter system view. system-view Enter voice view.
Page 36: Configuring Global Default Parameters For Voice Entities
NOTE: • In voice view, the vqa dscp media command has global significance, and the dscp media command is valid only for the configured voice entity. • The DSP image type setting takes effect after a reboot. Configuring global default parameters for voice entities For each voice entity, if the certain command is not performed to configure a parameter, the system uses the default value of the parameter.
Page 37: Enabling Fxo Monitoring
Enabling FXO monitoring When FXO monitoring is enabled, the router displays the physical state as down for FXO ports that are loosely connected or not connected at all. In environments with instable PSTN lines or poor line quality, FXO monitoring might detect that some lines are disconnected although those line are actually connected, resulting in call failures.
Page 38: Displaying And Maintaining Voice Entity Configuration
Displaying and maintaining voice entity configuration Task Command Remarks display voice call-info { brief | mark number | verbose } [ | Display voice call information. Available in any view. { begin | exclude | include } regular-expression ] display voice cmc { ccb | Display information about the call statistic [ all | em | h323 | iva | management center (CMC)
Page 39: Voice Entity Configuration Examples
Voice entity configuration examples This section provides voice entity configuration examples. Voice entity configuration example for establishing a VoIP call Network requirements Figure 6, two voice gateways (Router A and Router B) communicate with each other As shown in through WAN. For example, the user of tel.
Page 40: Fast Connection
Configure Router B: # Configure the VoIP voice entity to Router A. <RouterB> 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 1.1.1.1 [RouterB-voice-dial-entity10] quit # Configure the POTS voice entity corresponding to the local interface Line 2/0. [RouterB-voice-dial] entity 2001 pots [RouterB-voice-dial-entity1001] match-template 07552001 [RouterB-voice-dial-entity1001] line 2/0...
Page 41: Troubleshooting Voice Entity Configuration
[RouterA-voice-dial-entity755] quit # Configure the local interface and phone number for Telephone A. [RouterA-voice-dial] entity 1001 pots [RouterA-voice-dial-entity1001] match-template 0101001 [RouterA-voice-dial-entity1001] line 1/1 # Enable out-of-band DTMF transmission for the POTS voice entity. [RouterA-voice-dial-entity1001] outband h225 Configure Router B: # Configure the VoIP voice entity. <RouterB>...
Page 42: Configuring Analog Voice Subscriber Lines
Configuring analog voice subscriber lines This chapter covers the configuration of analog FXS, FXO, and E&M voice subscriber lines. Signal tone Call progress tones (CPTone), also called signal tones, are generally composed of several discrete single-frequency tones that are played repeatedly on a make-break ratio basis. Signal tones, including dial tone, ringback tone, and busy tone, are used to inform users of the call progress.
Page 43: Fxo Voice Subscriber Line
after the second ring to make sure that the calling identity information is sent and received correctly. Otherwise, the calling identity information might fail to be displayed. FXO voice subscriber line FXO interface A foreign exchange office (FXO) interface, that is, a two-port loop trunk interface, uses an RJ-11 connector and a telephone cable to connect local calls to a PSTN or PBX.
Page 44: E&M Subscriber Line
Figure 8 Network diagram Perform an automatic busy tone detection test as follows: Dial number 1002 from Telephone A (010-1001). The FXO interface on Router A plays a dial tone to PBX A, which then transmits the tone to Telephone A. Then dial number 07552001 from Telephone A.
Page 45: Hardware Compatibility With Analog Voice Subscriber Lines
Figure 9 Immediate start mode Figure 10, the calling side goes off-hook to seize the trunk and the • Delay start—As shown in called side (PBX) goes off-hook to respond to the seizure. When the called side (PBX) is ready, it goes on-hook.
Page 46: Configuration Task List
• MSR900. • MSR93X. • MSR1000. • MSR30-11E. • MSR30-11F. Configuration task list Task Remarks Configuring call progress tones Required. Configuring basic functions Optional. Configuring FXS voice subscriber line Optional. Configuring FXO voice subscriber line Optional. Binding an FXS voice subscriber line to an FXO voice subscriber line Optional.
Page 47: Customizing Call Progress Tones For A Country
Step Command Remarks Optional. cptone tone-type { all | Configure the level busy-tone | congestion-tone | By default, 1000 for busy tone and parameter for the call dial-tone | ringback-tone | congestion tone, 400 for dial tone progress tones. special-dial-tone | and special dial tone, and 600 for waiting-tone } amplitude value ringback tone and waiting tone.
Page 48: Configuration Procedure
Configuration procedure This section describes the basic functions of voice subscriber lines generated by voice cards. Configurations specific to a voice subscriber line will be introduced individually. To complete the basic functions of voice subscriber lines: Step Command Remarks Enter system view. system-view Enter voice subscriber line subscriber-line line-number...
Page 49: Configuring Packet Loss Compensation Mode
Step Command Remarks Enter FXS voice subscriber-line line-number subscriber line view. Optional. Enable CID on the FXS cid display voice subscriber line. Enabled by default. Enable the FXS voice subscriber line to send Optional. calling identity cid send Enabled by default. information to the remote end.
Page 50: Configuring The Sending Of Lcfo Signals
Configuring the sending of LCFO signals You can configure an FXS voice subscriber line to send a loop current feed open (LCFO) signal to indicate a disconnection to the peer. This feature is used mainly in North America. To configure the sending of LCFO signals: Step Command Remarks...
Page 51: Configuring Busy Tone Detection
Step Command Remarks Enable calling identity information Optional. receiving for the FXO voice cid receive Enabled by default. subscriber line. Enable calling identity information Optional. sending for the FXO voice subscriber cid send Enabled by default. line. Optional. By default, CID check is Configure the time for CID check and performed between the first after the CID check, the number of...
Page 52 NOTE: Before you configure the number of busy tone detection periods, test the new value repeatedly to make sure that the new value does not cause failed or false on-hook. Enabling the busy tone sending If the PBX fails to play a busy tone to a digital telephone, enable the FXO interface to send a busy tone to the PBX, which will transparently send the busy tone to the digital telephone.
Page 53: Configuring The Off-Hook Mode
To configure the duration before a forced on-hook: Step Command Remarks Enter system view. system-view Enter FXO voice subscriber-line line-number subscriber line view. Optional. Forced on-hook is disabled by default. Configure the duration hookoff-time time The configuration will take effect on all before a forced on-hook.
Page 54: Setting Ring Detection Parameters
Step Command Remarks Enter system view. system-view Enter FXO voice subscriber subscriber-line line-number line. Optional. Configure the off-hook hookoff-mode { delay | mode. immediate } Immediate mode by default. Setting ring detection parameters PBXs from different vendors might use different types of ring signals. By setting ring detection parameters, you can detect ring signals of different frequencies and waveforms.
Page 55: Binding An Fxs Voice Subscriber Line To An Fxo Voice Subscriber Line
Binding an FXS voice subscriber line to an FXO voice subscriber line The one-to-one binding between FXS voice subscriber lines and FXO voice subscriber lines enhances the reliability of voice communication, which is particularly important for some industries. After the binding, FXO voice subscriber lines can be used for communication over PSTN when the IP network is unavailable.
Page 56: Configuring E&M Voice Subscriber Line
Step Command Remarks 10. Specify the FXO voice Optional. subscriber line to be By default, no calling number is exclusively used by the caller-permit calling-string configured. That is, incoming bound FXS voice calls are not restricted. subscriber line. Configuring E&M voice subscriber line Configuration prerequisites •...
Page 57 Step Command Remarks Enter voice subscriber line subscriber-line line-number view. Configure the immediate The default is immediate start start mode for the E&M voice em-signal immediate mode by default. subscriber line. Configure a delay before the Optional. originating side sends DTMF delay send-dtmf milliseconds signals in the immediate The default is 300 milliseconds.
Page 58: Enabling E&M Non-Signaling Mode
Enabling 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 end without signaling. You can configure the private line auto ring-down (PLAR) function by using the private-line command to form a three-segment E&M virtual private line (E&M-VoIP-E&M).
Page 59: Configuring Analog E&M Line Failure Tone
NOTE: Configure this feature on the voice gateways of both sides. Configuring analog E&M line failure tone Figure 12, E&M analog control signals pass-through is enabled for the analog E&M line As shown in on Router A. To notify an IP network failure or a peer failure to the Tone Generator, configure Router A to play busy tones for analog E&M line failure by using the em-failure busytone command.
Page 60: Configuring Dtmf Properties
Figure 13 DTMF keypad frequencies Column Frequency Group 1209Hz 1336Hz 1477Hz 1633Hz 697Hz 770Hz 852Hz 941Hz A DTMF signal must last at least 45 milliseconds. A minimum interval of 23 milliseconds is required between two DTMF signals to make sure that DTMF signals are recognizable. Such requirements are roughly the same in all countries.
Page 61: Configuring Options Related To Dial Plan
• Energy detection—DTMF detection is implemented by calculating the frequency spectrum of the input voice signal. The energy threshold limits the spectrum shape of the input signal. A signal is considered a valid DTMF only when all requirements are met. •...
Page 62: Configuring Adjustment Functions
Step Command Remarks Bind a calling/called number By default, no number substitution substitute { called | calling } substitution rule list to a rule list is bound to a voice list-number voice subscriber line. subscriber line. For more information about the operation of the above commands, see "Configuring dial plans." Configuring adjustment functions Configuration task list Task...
Page 63: Configuring Gain Adjustment Function
Symptom Parameters adjusted Effect A user hears echoes when Enlarge the control factor of Too high a control factor leads speaking. mixed proportion of noises. to audio discontinuity. Too high a judgment threshold There are echoes when both Enlarge the judgment threshold slows down the convergence of parties speak at the same time.
Page 64: Configuring Time Adjustment Function
Step Command Remarks… Optional. Set the input gain on the receive gain value voice interface. The default is 0 dB. Optional. Set the output gain on the transmit gain value voice interface. The default is 0 dB. IMPORTANT: Gain adjustment might lead to call failures. Hewlett Packard Enterprise recommends not adjusting the gain.
Page 65: Configuring Comfortable Noise Function
Step Command Remarks Optional. Configure the interval The default is 10 seconds. between off-hook and dialing timer first-dial seconds the first digit. Applicable only in FXO/FXS voice subscriber line view. Configure the maximum Optional. interval for dialing the next timer dial-interval seconds The default is 10 seconds.
Page 66: Rebooting A Voice Card
Configuration restrictions and guidelines • This function is supported on E&M cards only. You must specify an E&M card when you configure this function. • This function takes effect only when G.711 A-Law is used. • You must reboot the specified card after you configure this function. To check whether this function takes effect, use the display device verbose command.
Page 67: Configuring Global Default Parameters For Voice Subscriber Lines
Configuring global default parameters for voice subscriber lines Step Command Remarks Enter system view. system-view Enter voice view. voice-setup Configure the default Optional. default subscriber-line input/output gain for all { receive | transmit } gain value The default is 0. subscriber lines.
Page 68: Analog Voice Subscriber Line Configuration Examples
Analog voice subscriber line configuration examples This section contains examples on configuring the FXO voice subscriber line and one to one binding between FXS and FXO lines. Configuration example for the FXO voice subscriber line Network requirements In the following figure, the FXO voice subscriber line connected to Router B operates in the private-line auto ring-down (PLAR) mode, and the default remote phone number is 010-1001.
Page 69: Configuration Example For One-To-One Binding Between Fxs And Fxo
[RouterB-voice-dial-entity2001] line 1/0 [RouterB-voice-dial-entity2001] send-number all # Configure FXO interface Line 1/0. [RouterB-voice-dial-entity2001] quit [RouterB-voice-dial] quit [RouterB-voice] quit [RouterB] subscriber-line 1/0 [RouterB-subscriber-line1/0] private-line 0101001 Configuration example for one-to-one binding between FXS and FXO Network requirements • Router A and Router B are connected over an IP network and a PSTN. Telephone A attached to Router A can make calls to Telephone B attached to Router B over the IP network or the PSTN.
Page 70 [RouterA-voice-dial] entity 210 voip [RouterA-voice-dial-entity210] match-template 210..[RouterA-voice-dial-entity210] address sip ip 192.168.0.76 [RouterA-voice-dial-entity210] quit # Configure a POTS entity for the FXS voice subscriber line. [RouterA-voice-dial] entity 0101001 pots [RouterA-voice-dial-entity101001] match-template 0101001 [RouterA-voice-dial-entity101001] line 6/24 [RouterA-voice-dial-entity101001] quit # Configure a backup POTS entity on the FXO voice subscriber line, set the match template to .T, enable the sending of all digits of a called number, and configure a calling number permitted to originate calls to the POTS entity.
Page 71: Troubleshooting Analog Voice Subscriber Line Configuration
[RouterB-voice-dial-entity11] line 6/0 [RouterB-voice-dial-entity11] send-number all [RouterB-voice-dial-entity11] caller-permit 2101002 [RouterB-voice-dial-entity11] quit [RouterB-voice-dial] quit [RouterB-voice] quit # Configure the PLAR function and the delay off-hook binding. [RouterB] subscriber-line 6/0 [RouterB-line6/0] private-line 2101002 [RouterB-line6/0] hookoff-mode delay bind 6/24 [RouterB-line6/0] quit # Configure a dial plan where the VoIP entity is preferred. [RouterB] voice-setup [RouterB-voice] dial-program [RouterB-voice-dial] select-rule type-first 2 1 3 4...
Page 72 Figure 16 Network diagram Figure 16, suppose Telephone A hangs up first after conversation, then PBX A plays As shown in busy tones to Router A, which disconnects the line after detecting the busy tones and sends a disconnect message to Router B. Router B sends the message to PBX B that then plays busy tones to Telephone B.
Page 73: Configuring Digital Voice Subscriber Lines
Configuring digital voice subscriber lines This chapter covers the configuration of E1, T1, and BSV voice subscriber lines. Introduction to E1 and T1 Overview Plesiochronous digital hierarchy (PDH) includes two major communications systems: ITU-T E1 system and ANSI T1 system. The E1 system is dominant in Europe. The T1 system is dominant in USA, Canada and Japan.
Page 74: Features Of E1 And T1
You can use an E1 interface as an ISDN PRI or CE1 interface: • As an ISDN PRI interface, the E1 interface adopts DSS1 or QSIG signaling. As TS 0 is used to transfer synchronization information and TS16 is used as a D channel to transfer connection signaling, you can arbitrarily bind any timeslots other than TS0 and TS16 as a logical interface, which is equivalent to an ISDN PRI interface.
Page 75: Hardware Compatibility With Digital Voice Subscriber Lines
• ITU-T R2 signaling—Includes digital line signaling and interregister signaling. Digital line signaling is transmitted in TS16 (ABCD bits) of E1 trunk. It conveys status information about E1 trunks to describe whether the trunks are occupied, released, or blocked. Interregister signaling conveys information about address, language and discriminating digits for internal calls, echo suppressor, caller properties and callee properties in multi-frequency compelled approach (forward and backward) in each timeslot.
Page 76: Configuring Basic Parameters For An E1 Voice Interface
Task Remarks Configuring basic parameters for an E1 voice interface Optional. Configuring basic parameters for a T1 voice interface Optional. Configuring the voice subscriber line for a TS set Required. Binding logical voice subscriber line to POTS entity Required. Configuring basic R2 signaling parameters Optional.
Page 77: Configuring The Framing Format And Line Coding Format
When there is no VCPM on the main board, the configuration of each MIM/FIC is independent, but the clock mode of only one interface can be set to line primary on a device. Suppose that you remove the FIC on an interface whose clock mode is set to line primary without powering it off and that you set the clock mode to line primary for another interface.
Page 78: Set The Physical State Change Suppression Interval On An E1 Interface
Step Command Remarks timeslot-set ts-set-number timeslot-list timeslots-list signal Create a TS set according to { e&m-delay | e&m-immediate | By default, no TS set is the selected signaling mode. e&m-wink | fxo-ground | configured. fxo-loop | fxs-ground | fxs-loop | r2 } Enter CAS view.
Page 79: Configuring The Framing Format And Line Coding Format
Step Command Remarks Enter system view. system-view Enter T1 interface view. controller t1 slot-number Optional. Configure a TDM clock tdm-clock { internal | line By default, the internal clock is source for the T1 interface. [ primary ] } used as the TDM clock source. Configuring the framing format and line coding format Step Command...
Page 80: Restoring Default Settings For A T1 Voice Interface
Step Command Remarks Optional. Set a physical state change By default, the physical state suppression interval on the link-delay delay-time change suppression is disabled T1 interface. on a T1 interface. Restoring default settings for a T1 voice interface Step Command Remarks Enter system view.
Page 81: Configuring The Dtmf Detection Sensitivity
Step Command Remarks Enter voice subscriber line subscriber-line view. slot-number:ts-set-number Optional. Configure a companding law A-law for a VE1 interface card and pcm { a-law | μ-law } for signal quantization. μ-law for a VT1 interface card by default. Optional. Create a description for the The description for a voice description text...
Page 82: Configuring The Echo Adjustment Function
Step Command Remarks… Enter system view. system-view Enter E1/T1 interface view. controller { e1 | t1 } slot-number timeslot-set ts-set-number timeslot-list Create a TS set according to timeslots-list signal { e&m-delay | the selected signaling mode. e&m-immediate | e&m-wink | fxo-ground | fxo-loop | fxs-ground | fxs-loop | r2 } Exit E1/T1 interface view.
Page 83: Configuring The Comfortable Noise Function
Configuring the comfortable noise function Step Command Remarks Enter system view. system-view Enter E1/T1 interface view. controller { e1 | t1 } slot-number timeslot-set ts-set-number timeslot-list Create a TS set according to timeslots-list signal { e&m-delay | the selected signaling mode. e&m-immediate | e&m-wink | fxo-ground | fxo-loop | fxs-ground | fxs-loop | r2 } Exit E1/T1 interface view.
Page 84: Configuring R2 Signaling
Step Command Remarks timeslot-set ts-set-number timeslot-list timeslots-list signal Create a TS set according to { e&m-delay | e&m-immediate | the selected signaling mode. e&m-wink | fxo-ground | fxo-loop | fxs-ground | fxs-loop | r2 } Exit E1/T1 interface view. quit Enter voice view.
Page 85 Table 5 Line signaling bit description Description Vale = 0 Value = 1 Identifies working state of device at the originating side and indicates state of the calling Off-hook, seized On-hook (idle) party line. Indicates fault state from the originating side to Normal Faulty the terminating side.
Page 86 Originating side releases the call. The originating side sends a clear-forward signal 10. When the terminating side recognizes the clear-forward signal, it sends a backward signal 10 (release guard signal or clear-forward acknowledgement signal). After the originating side recognizes the backward signal 10, it releases the circuit.
Page 87 After the terminating side receives a forward signal 11 from the originating side to indicate device fault during conversation, the terminating side releases the line backward. At the same time, it sends a backward signal 11. Then, the circuit is in faulty state. When the device recovers, the originating side sends a forward signal 10, and the terminating side sends back a signal 10.
Page 88 Designation Basic Meaning Congestion in an international exchange; terminate interregister signaling A-15 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 9 Group II forward signals Designation Basic Meaning...
Page 89: Configuring Basic R2 Signaling Parameters
Figure 22 ITU-T R2 interregister signaling exchange process Originating side Terminating side Calling number: 123 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)
Page 90 Step Command Remarks timeslot-set ts-set-number Create a TS set and enable timeslot-list timeslots-list signal R2 signaling for it. Enter R2 CAS view. cas ts-set-number Optional. Configure the trunk direction trunk-direction timeslots for R2 signaling. timeslots-list { dual | in | out } Bidirectional by default.
Page 91 Configuring the connection mode for an R2 call There are two connection modes for an R2 call: terminal-to-terminal (terminal) and segment-to-segment (segment) mode. • In the terminal-to-terminal (terminal) mode, after the called number is received, the R2 protocol module must wait for the real state (busy or idle) of the called party before returning the corresponding register information to the originating side.
Page 92: Configuring R2 Digital Line Signaling
Step Command Remarks Optional. Set the length of called By default, the length numbers that can be posa called-length calledlength of the called numbers received. that can be received is 31 digits. For more information about the POS terminal service, see HPE FlexNetwork MSR Router Series Comware 5 Terminal Access Configuration Guide.
Page 93: Configuring R2 Interregister Signaling
Step Command Remarks Optional. The default is 60,000 milliseconds for answer signal; 10,000 timer dl { answer | clear-back | 13. Set timeout values of line milliseconds for clear-back signal clear-forward | | re-answer | signals. and clear-forward signal and release-guard | seize } time release-guard signal;...
Page 94: Configuring Pri
Step Command Remarks Optional. Adopt Group B signals to complete interregister group-b enable Group B signals are used by signaling exchange. default. 10. Enable this end to send a Optional. number terminator to the final-callednum enable Disabled by default. terminating side. 11.
Page 95: Enabling The Transmission Of Qsig Signaling Over A Sip Network
Step Command Remarks Enter system view. system-view Enter E1/T1 interface view. controller { e1 | t1 } slot-number Bundle timeslots into a PRI pri-set [ timeslot-list range ] group. Exit E1/T1 interface view. quit Enter the view of the serial interface serial number: { 15 | interface created for the PRI 23 }...
Page 96: Configuring Digital E&M Signaling
NOTE: • To enable the QSIG tunneling function, you must also execute the qsig-tunnel enable command on the voice trunks of both the ingress and egress gateways. • When the QSIG tunneling function is enabled, the system will not process or send non-QSIG ISDN calls.
Page 97: Enabling E&M Non-Signaling Mode
Step Command Remarks Enter digital E&M voice subscriber-line subscriber line view. slot-numbe:ts-set-number Configure the maximum Optional. duration of delay signal in delay hold millseconds The default is 400 milliseconds. delay start mode. Configure the delay that the Optional. called party must wait before delay rising millseconds sending a delay signal after it The default is 300 milliseconds.
Page 98: Configuring Receive And Transmit Signaling
Step Command Remarks Enter system view. system-view Enter digital E&M voice subscriber subscriber-line slot-number: line view. ts-set-number Optional. Disabled by default. For more information about the open-trunk { caller monitor Enable E&M non-signaling mode. open-trunk command, see interval | called } HPE FlexNetwork MSR Router Series Comware 5 Voice Command Reference.
Page 99: Configuring The Time Adjustment Function
Configuring the time adjustment function Step Command Remarks Enter system view. system-view Enter E1/T1 interface view. controller { e1 | t1 } slot-number timeslot-set ts-set-number Create a TS set for digital timeslot-list timeslots-list signal E&M signaling and select a { e&m-delay | e&m-immediate | start mode.
Page 100: Querying The Circuits Of A Timeslot Or A Range Of Timeslots
Step Command Remarks timeslot-set ts-set-number Create a TS set, enable timeslot-list timeslots-list signal digital LGS signaling and set { fxo-ground | fxs-ground | Optional. its start mode. fxo-loop | fxs-ground | fxs-loop } Quit digital LGS signaling quit view. Enter voice subscriber line subscriber-line: view.
Page 101: Configuration Prerequisites
Configuration prerequisites The router is equipped with an applicable BSV interface card. Configuration procedure To configure a BSV BRI interface: Step Command Remarks Enter system view. system-view Enter the specified BSV BRI interface bri interface-number interface view. Optional. Configure an interface By default, the description of a description text description.
Page 102: Enabling The Amd Function
Enabling the AMD function Step Command Remarks Enter system view. system-view Enter voice subscriber line subscriber-line view. slot-number:ts-set-number Disabled by default. The AMD detection results are subject to the language Enable the AMD function. amd enable characteristics, answering machines, and background noises and music.
Page 103: Configuring Reverse Charging Function
Configuring reverse charging function Typically, the calling party is charged to place a call to the called party. If you want to charge the called party, you can configure a prefix to match the called number for reverse charging. After you configure this function, the router identifies a reverse-charge call by comparing the called number of an incoming call with a prefix.
Page 104: Mirroring Pcm, Rtp Packets, Or Voice Command Data On A Digital Voice Subscriber Line
Step Command Remarks Enter voice view. voice-setup Enable the router to treat By default, the router does not DISCONNECT messages disconnect a call when it receives call disc-pi-off with PI 8 as standard a DISCONNECT message with PI DISCONNECT messages. Mirroring PCM, RTP packets, or voice command data on a digital voice subscriber line The mirroring function copies the specified PCM, RTP, and voice command data on a digital voice...
Page 105: Digital Voice Subscriber Line Configuration Examples
Digital voice subscriber line configuration examples This section provides configuration examples for digital voice subscriber lines. E1 R2 signaling and digital E&M signaling configuration example Network requirements Figure 23, Telephones in City A and City B communicate with each other through voice As shown in routers (Router A and Router B) across an IP network, as shown in the network diagram.
Page 106 [RouterA-voice-dial] entity 1003 pots # Configure a target match-template for the POTS voice entity. [RouterA-voice-dial-entity1003] match-template 0101003 # Associate the POTS voice entity with FXS subscriber line 3/0. [RouterA-voice-dial-entity1003] line 3/0 [RouterA-voice-dial-entity1003] quit # Create a POTS voice entity for the E1 interface. [RouterA-voice-dial] entity 1001 pots # Configure a target match-template pointing to telephone number 010-1001 for the POTS voice entity.
Page 107: E1 Voice Dss1 Signaling Configuration Example
# Associate the POTS voice entity with FXS subscriber line 1/1:1. [RouterB-voice-dial-entity2001] line 1/1:1 [RouterB-voice-dial-entity2001] send-number all [RouterB-voice-dial-entity2001] quit # Create a POTS voice entity corresponding to telephone number 0755-2002 for the E1 interface. [RouterB-voice-dial] entity 2002 pots # Configure a target match-template for the POTS voice entity. [RouterB-voice-dial-entity2002] match-template 07552002 # Associate the POTS voice entity with subscriber line 1/1:1.
Page 108 [RouterA-Ethernet2/1] ip address 1.1.1.1 255.255.255.0 [RouterA-Ethernet2/1] quit # Create an ISDN PRI group on interface E1 1/1. [RouterA] system-view [RouterA] controller e1 1/1 [RouterA-E1 1/1] pri-set # Create a POTS voice entity for the FXS interface. [RouterA] system-view [RouterA] voice-setup [RouterA-voice] dial-program [RouterA-voice-dial] entity 1003 pots # Configure a target match-template pointing to telephone number 010-1003 for the POTS...
Page 109: Qsig Tunneling Configuration Example
# Create an ISDN PRI group on interface E1 1/1. [RouterB] system-view [RouterB] controller e1 1/1 [RouterB-E1 1/1] pri-set [RouterB-E1 1/1] quit # Create a POTS voice entity for the ISDN PRI interface. [RouterB] voice-setup [RouterB-voice] dial-program [RouterB-voice-dial] entity 2001 pots # Configure a target match-template pointing to telephone number 0755-2001 for the POTS voice entity.
Page 110 Figure 25 Network diagram Configuration procedure Configure Router A: # Configure IP address 1.1.1.1/24 for interface Ethernet 2/1. <RouterA> system-view [RouterA] interface ethernet 2/1 [RouterA-Ethernet2/1] ip address 1.1.1.1 255.255.255.0 [RouterA-Ethernet2/1] quit # Create an ISDN PRI group on interface E1 1/1. [RouterA] system-view [RouterA] controller e1 1/1 [RouterA-E1 1/1] pri-set...
Page 111 # Configure a target match-template for the VoIP voice entity. [RouterA-voice-dial-entity755] match-template 0755..# Configure the target address of the VoIP voice entity. [RouterA-voice-dial-entity755] address sip ip 2.2.2.2 Configure Router B: # Configure the IP address 2.2.2.2/24 for the interface Ethernet 2/1. <RouterB>...
Page 112: Troubleshooting Digital Voice Subscriber Line Configuration
Troubleshooting digital voice subscriber line configuration Failure of call connection from router to PSTN Symptom With R2 signaling adopted, the router cannot establish connection with the subscriber at the switch side. Solution Use the display current-configuration command to check that the trunking mode on the router matches that on the switch.
Page 113: Configuring Dial Plans
Configuring dial plans Overview More requirements on dial plans arise with the wide application of VoIP. A desired dial plan should be flexible, reasonable and operable, and be able to help a voice gateway manage numbers in a unified way, making number management more convenient and reasonable. The dial plan process on the calling side differs from that on the called side.
Page 114: Regular Expression
On the called side Figure 27 shows the dial plan operation process on the called side. Figure 27 Flow chart for dial plan operation process on the called side After receiving a voice call (the called number), the voice gateway on the called side performs global calling/called number substitution.
Page 115: Introduction To Number Substitution
Metacharacter Meaning Wildcard, which can match any valid digit. For example, 555…. can match any Dot (.) number beginning with 555 and ending in four additional characters. Used to connect two numbers (The smaller comes before the larger) to indicate a Hyphen (-) range of numbers, for example, 1-9 inclusive.
Page 116: Hardware Compatibility With Dial Plans
The preferred number substitution rule is matched first. If the match succeeds, the gateway substitutes numbers based on this rule. If the match fails, the gateway matches other number substitution rules in sequence. Once a rule is matched successfully, the gateway stops matching other number substitution rules. Here, the dot represents virtually matched digits.
Page 117: Configuring Call Authority Control
Step Command Remarks Optional. Configure calling numbers By default, no calling number is caller-permit calling-string permitted to call out. configured, and outgoing calls are not restricted. NOTE: The calling-string argument is in the format of { [ + ] string [ $ ] }| $. For specific meanings of these symbols in the format, see HPE FlexNetwork MSR Router Series Comware 5 Voice Command Reference.
Page 118: Enabling Private Line Auto Ring-Down
Step Command Remarks By default, no subscriber group is Bind a subscriber group to the caller-group { deny | permit } bound to the voice entity, that is, voice entity. any calling number is allowed to subscriber-group-list-number originate calls. Enabling private line auto ring-down With the private line auto ring-down (PLAR) function enabled, the voice gateway automatically dials the specified called number (string) as soon as the subscriber picks up the phone.
Page 119: Configuring A Dial Terminator
When a subscriber dials 0106688#, if you configure the router to use longest match mode and the dial terminator "#" on the router, the router will as well ignore the configured longest match mode and use shortest match mode to establish a call connection. To configure a global number match mode: Step Command...
Page 120 Figure 28 Network diagram • Shortest number match a. Configure Router A: <RouterA> system-view [RouterA] voice-setup [RouterA-voice] dial-program # Configure POTS entity 1000. [RouterA-voice-dial] entity 1000 pots [RouterA-voice-dial-entity1000] match-template 10001234$ [RouterA-voice-dial-entity1000] line 1/0 [RouterA-voice-dial-entity1000] quit # Configure VoIP entity 2000 and VoIP entity 2001. [RouterA-voice-dial] entity2000 voip [RouterA-voice-dial-entity2000] match-template 20001234$ [RouterA-voice-dial-entity2000] address ip 1.1.1.2...
Page 121: Configuring Match Order Of Voice Entity Selection Rules
[RouterA-voice-dial] number-match longest After you dial number 20001234 at Telephone A and waits for a period of time (during this period, you can continue dialing), the number matches VoIP entity 2000 and Telephone B is alerted. If you continue to dial 1234 during this period (that is, the dialed number is actually 200012341234), the number matches VoIP entity 2001 and Telephone C is alerted.
Page 122: Configuration Example Of Voice Entity Selection Priority Rules
Step Command Remarks Optional. By default, the match order of Configure match order of select-rule rule-order 1st-rule rules for the voice entity selection voice entity selection rules. [ 2nd-rule [ 3rd-rule ] ] is exact match->voice entity priority->random selection. Configuration example of voice entity selection priority rules Figure 29 and the following describe the voice entity selection priority rule configuration on Router A and Router B.
Page 123: Configuring Voice Entity Type Selection Priority Rules
[RouterB-voice-dial-entity2000] match-template 20001234$ [RouterB-voice-dial-entity2000] line 1/0 Configure different voice entity selection priority rules: Configure voice entities to be selected in sequence of exact match, priority, and random selection. <RouterA> system-view [RouterA] voice-setup [RouterA-voice] dial-program [RouterA-voice-dial] select-rule rule-order 1 2 3 After Telephone dials 20001234, the number will match VoIP entity 2000.
Page 124: Configuration Example Of Voice Entity Type Selection Priority Rules
Configuration example of voice entity type selection priority rules Figure 30 There are an IP connection and a PRI connection between Router A and Router B. and the following describe the configurations for different voice entity type selection priority rules on Router A and Router B.
Page 125: Configuring The Voice Entity Search Function
[RouterB-voice-dial] entity 2000 pots [RouterB-voice-dial-entity2000] match-template 20001234$ [RouterB-voice-dial-entity2000] line 1/0 Configure different voice entity type selection priority rules: Configure the system to select voice entities in order of VoIP->POTS->VoFR->IVR. <RouterA> system-view [RouterA] voice-setup [RouterA-voice] dial-program [RouterA-voice-dial] select-rule type-first 2 1 3 4 After Telephone A dials 20001234, the number will match VoIP entity 2000.
Page 126 [RouterA] voice-setup [RouterA-voice] dial-program # Configure POTS entity 1000. [RouterA-voice-dial] entity 1000 pots [RouterA-voice-dial-entity1000] match-template 10001234$ [RouterA-voice-dial-entity1000] line 1/0 [RouterA-voice-dial-entity1000] quit # Configure VoIP entities 2000, 2001, and 2002. [RouterA-voice-dial] entity 2000 voip [RouterA-voice-dial-entity2000] match-template 20001234$ [RouterA-voice-dial-entity2000] address sip ip 1.1.1.2 [RouterA-voice-dial-entity2000] quit [RouterA-voice-dial] entity2001 voip [RouterA-voice-dial-entity2001] match-template 2000123.$...
Page 127 Voice entities are offered : 2000 2001 Restore the maximum number of voice entities found before a search process stops to the default. <RouterA> system-view [RouterA] voice-setup [RouterA-voice] dial-program [RouterA-voice-dial] undo select-rule search-stop [RouterA-voice-dial] display voice call-info verbose The information table for current calls in detail **************** CALL 0 *************** Call direction : From CS...
Page 128: Configuring A Number Priority Peer
Configuring a number priority peer Configuration prerequisites The required basic configurations have been completed on POTS, VoIP, VoFR, and IVR entities. Configuration procedure To configure a number priority peer: Step Command Remarks Enter system view. system-view Enter voice view. voice-setup Enter voice dial program dial-program view.
Page 129: Configuration Procedure
Configuration procedure To configure a maximum-call-connection set: Step Command Remarks Enter system view. system-view Enter voice view. voice-setup Enter voice dial program dial-program view. Configure a By default, no max-call set-number maximum-call-connection maximum-call-connection set is max-number set. configured. entity entity-number { pots | ivr | Enter voice entity view.
Page 130 Step Command Remarks Enter voice dial program dial-program view. Create a number substitution rule list and enter voice number-substitute list-number number-substitute view. Optional. dot-match { end-only | left-right Configure a dot-match rule. By default, the dot match rule is | right-left } end-only.
Page 131 Step Command Remarks Exit voice number-substitute view and enter voice dial quit program view. entity entity-number { pots | vofr Enter voice entity view. | voip } Optional. 10. Bind a number substitution substitute { called | calling } By default, no number substitution rule list to a voice entity.
Page 132: Configuring A Number Sending Mode
Configuring a number sending mode Configure a number sending mode to control how the originating gateway sends a called number. Three number sending modes are available: • Send the least significant digits (configured by the send-number digit-number command) of a called number.
Page 133: Displaying And Maintaining Dial Plan Configuration
Step Command Remarks Enter system view. system-view Enter voice view. voice-setup Enter voice dial program dial-program view. Enter POTS entity view. entity entity-number pots Configure a dial prefix. dial-prefix string No dial prefix by default. By default, no number template is Bind a number template to match-template match-string bound to the local or trunk voice...
Page 134 Figure 32 Network diagram Configuration consideration The PBX (calling side) at place B changes the called number to an intermediate number. The PBX (called side) at place A changes the received intermediate number to a local number before initiating the call. Configuration procedure The following configuration supports dial plan–based calls from place B to place A only.
Page 135 [RouterB-voice-dial-entity1010] line 1/0 [RouterB-voice-dial-entity1010] send-number all # Configure FXO trunk line 1/1. [RouterB-voice-dial-entity1010] quit [RouterB-voice-dial] entity 2010 pots [RouterB-voice-dial-entity2010] match-template ..[RouterB-voice-dial-entity2010] line 1/1 [RouterB-voice-dial-entity2010] send-number all # Enable hunt group. [RouterB] subscriber-line 1/0 [RouterB-subscriber-line1/0] hunt-group enable [RouterB-subscriber-line1/0] quit [RouterB] subscriber-line 1/1 [RouterB-subscriber-line1/1] hunt-group enable Configure Router A: # Set the address of an Ethernet interface to 1.1.1.1.
Page 136: Configuring The Match Order For Voice Entity Selection
[RouterA-voice-dial-entity2010] line 1/1 [RouterA-voice-dial-entity2010] send-number all # Enable hunt group. [RouterB] subscriber-line 1/0 [RouterB-subscriber-line1/0] hunt-group enable [RouterB-subscriber-line1/0] quit [RouterB] subscriber-line 1/1 [RouterB-subscriber-line1/1] hunt-group enable Configuring the match order for voice entity selection Network requirements Figure 33, two telephones connected to Router A can make PSTN calls through two As shown in trunk voice gateways (Router B and Router C) and the trunk lines of these two trunk voice gateways should be fully utilized.
Page 137 # Configure POTS entities. [RouterA-voice-dial] entity 1000 pots [RouterA-voice-dial-entity1000] match-template 1000 [RouterA-voice-dial-entity1000] line 1/0 [RouterA-voice-dial] entity1001 pots [RouterA-voice-dial-entity1001] match-template 1001 [RouterA-voice-dial-entity1001] line 1/1 [RouterA-voice-dial-entity1001] quit # Configure rules in the match order for voice entity selection. [RouterA-voice-dial] select-rule rule-order 1 4 Configure Router B: # Configure an Ethernet address.
Page 138: Configuring The Maximum-Call-Connection Set
[RouterC-voice-dial-entity1001] send-number all [RouterC-voice-dial-entity1001] quit # Configure rules in the match order for voice entity selection [RouterC-voice-dial] select-rule rule-order 1 4 Configuring the maximum-call-connection set Network requirement Figure 34, there are one trunking voice gateway (Router C) and two subscriber voice As shown in gateways (Router A and Router B) in a city.
Page 139 # Configure the maximum-call-connection set. [RouterA-voice-dial] max-call 1 2 # Bind the maximum-call-connection set to a voice entity. [RouterA-voice-dial] entity 2000 voip [RouterA-voice-dial-entity2000] max-call 1 Configure Router B: # Configure an Ethernet address. <RouterB> system-view [RouterB] interface ethernet 2/1 [RouterB-Ethernet2/1] ip address 1.1.1.2 24 [RouterB-Ethernet2/1] quit # Configure a VoIP entity.
Page 140: Configuring Call Authority Control
[RouterC-voice-dial-entity1001] send-number all [RouterC-voice-dial] entity1002 pots [RouterC-voice-dial-entity1002] match-template 010..[RouterC-voice-dial-entity1002] line 5/2 [RouterC-voice-dial-entity1002] send-number all [RouterC-voice-dial-entity1001] quit [RouterC-voice-dial-entity1002] quit [RouterC-voice-dial] entity1003 pots [RouterC-voice-dial-entity1003] match-template 010..[RouterC-voice-dial-entity1003] line 5/3 [RouterC-voice-dial-entity1003] send-number all [RouterC-voice-dial-entity1003] return # Enable hunt group for voice subscriber lines. <RouterC>...
Page 141 Figure 35 Network diagram Configuration procedure This example does not provide SIP server and digital subscriber line configurations. For more information, see "Configuring SIP" and "Configuring voice subscriber lines." Configure Router A: # Configure two subscriber groups. <RouterA> system-view [RouterA] voice-setup [RouterA-voice] dial-program [RouterA-voice-dial] subscriber-group 1 [RouterA-voice-dial-group1] match-template 1100..
Page 142 # Configure a POTS entity for place C. [RouterA-voice-dial] entity 3100 pots [RouterA-voice-dial-entity3100] line 2/0:15 [RouterA-voice-dial-entity3100] send-number all [RouterA-voice-dial-entity3100] match-template 3... [RouterA-voice-dial-entity3100] caller-group permit 2 Configure Router B: <RouterB> system-view [RouterB] voice-setup [RouterB-voice] dial-program [RouterB-voice-dial] entity 2100 pots [RouterB-voice-dial-entity2100] line 1/0:15 [RouterB-voice-dial-entity2100] send-number all [RouterB-voice-dial-entity2100] match-template 2...
Page 143: Configuring Sip
Configuring SIP Overview The Session Initiation Protocol (SIP) is an application layer control protocol that can establish, modify, and terminate multimedia sessions such as IP phone calls, multimedia session, and multimedia conferences. It is the core component in the multimedia data and control architecture of the IETF (RFC 3261).
Page 144: Sip Functions And Features
For example, when receiving a request from a calling UA, the redirect server searches for the location information of the called UA and returns the location information to the UA. This location can be that of the called UA or another proxy server, to which the UA can initiate the session request again.
Page 145: Sip Messages
• Quick launch—The system can be updated quickly to accommodate new branches and personnel, as well as changes resulting from job rotation or relocation. • Easy to install and maintain—Even non-professional individuals can install and maintain SIP systems. SIP messages SIP messages, consisting of SIP request messages and SIP response messages, are encoded in text mode.
Page 146 Figure 36 Message exchange for a UA to register with a Registrar Call setup SIP operates in the Client/Server mode and sets up calls through communication between UA and proxy server. Figure 37 Network diagram In the above figure, Telephone A will call Telephone B, and Router A and Router B work as SIP endpoints (UAs).
Page 147 Figure 38 Call setup procedures involving a proxy server This is a simplified scenario where only one proxy server is involved and no registrar is present. A complex scenario, however, might involve multiple proxy servers and registrars. Call redirection Figure 39, when a SIP redirect server receives a session request, it sends back a As shown in response indicating the address of the called SIP endpoint instead of forwarding the request.
Page 148: Support For Transport Layer Protocols
Figure 39 Call redirection procedure for UAs Internet User agent User agent Redirect Server INVITE 100 Trying 302 Moved Temporarily INVITE 100 Trying 200 OK This is a common application. Fundamentally, a redirect server can respond with the address of a proxy server as well.
Page 149: Media Flow Encryption
passing authentication. SIP messages are encrypted during SIP over TLS transmissions to prevent your data from being sniffed. This increases the security of voice communications. "Configuring TLS for SIP sessions." For more information about signaling encryption, see SIP over TLS requires the configuration of TLS security policies. For information about how to configure the TLS security policies, see HPE FlexNetwork MSR Router Series Comware 5 Security Configuration Guide.
Page 150: Tls-Srtp Combinations
TLS-SRTP combinations TLS protects control signaling, and SRTP encrypts and authenticates voice media flows. You can Table 15, there are four combinations of TLS and use them separately or together. As shown in SRTP. Table 15 TLS-SRTP combinations SRTP Description Signaling packets are secured.
Page 151: Sip Configuration Task List
For configuration of basic QSIG call supported by the SIP-T protocol, see "Configuring digital voice subscriber lines." VRF-aware SIP VRF-aware SIP enables telephones to call each other in an L3VPN. Figure 41, two VPNs Voice and Data exist on the network. The two VPNs cannot access As shown in each other.
Page 152: Configuring Sip Ua Registration
Task Remarks Configuring call failure-triggered re-registration Optional. Configuring fuzzy telephone number registration Optional. Enabling SIP registration function Required. Configuring SIP server keepalive and backup Optional. Configuring SIP routing Required. Configuring user information Optional. Configuring outbound SIP proxy server information for a SIP UA Optional.
Page 153: Configuring Sip Authentication Information
Configuring SIP authentication information SIP authentication information is required in the following two cases: • UAs use a proxy server and the proxy server needs to authenticate UAs. • The peer SIP device needs to authenticate the local device during SIP message exchange. Authentication information selection rule If the authentication information is configured with the user command, POTS entity view is preferred.
Page 154: Configuring Registrar Information For A Sip Ua
Configuring registrar information for a SIP UA Configuration guidelines The transport layer protocol specified in the registrar command must have been specified with the listen transport command. Otherwise, no register request can be initiated. If TLS is specified in the registrar command, you also need to configure the secure sockets layer (SSL) policy name of the client with the crypto command.
Page 155: Configuring Call Failure-Triggered Re-Registration
To ensure the validity of registration information of a voice entity or an SIP trunk account on the registrar, the voice entity or SIP trunk account must re-register with the registrar at a specified time before the registration expiration interval is reached. Use the timer registration divider or timer registration threshold command to set the time when the voice entity or SIP trunk account re-registers with the registrar.
Page 156: Configuring Fuzzy Telephone Number Registration
Step Command Remarks Enter system view. system-view Enter voice view. voice-setup Enter SIP client view. Optional. Enable call failure-triggered call-fallback register re-registration. Not enabled by default. Configuring fuzzy telephone number registration Fuzzy telephone number registration refers to the use of a wildcard (including the dot . and the character T), rather than a standard E.164 number in the match template of a POTS entity.
Page 157: Configuring Sip Server Keepalive And Backup
Configuring SIP server keepalive and backup The keepalive function detects whether a SIP server is available in either of the following two ways: • Options keepalive—The device periodically sends Options packets to detect the availability of the SIP server. If the response from the server is 408 or 5xx (except 502, 504, 505, and 513), the server is unavailable.
Page 158: Configuration Procedure (1: Destination Ip Address)
Configuration procedure (1: destination IP address) Step Command Remarks Enter system view. system-view Enter voice view. voice-setup Enter voice dial program dial-program view. Enter VoIP entity view. entity entity-number voip Configure the destination IP address sip ip ip-address [ port Not configured by default.
Page 159: Configuration Procedure (4: Enum)
Step Command Remarks Optional. The default DNS lookup method is a-record. If you configure the destination port by using the address sip { dns domain-name [ port Set the DNS lookup mode. dns-type { a-record | srv } port-number ] | enum-group group-number }, proxy dns domain-name [ port port-number ], or mwi-server dns...
Page 160: Configuring Outbound Sip Proxy Server Information For A Sip Ua
To configure user information: Step Command Remarks Enter system view. system-view Enter voice view. voice-setup Enter voice dial program dial-program view. Enter POTS voice entity entity entity-number pots view. Optional. By default, uri user-info [ domain number@SIP-device-domain-n Configure user information. domain-name ] ame or number @SIP-interface-IP-address is...
Page 161 Configuration prerequisites If a SIP server is used to forward calls, you need to complete the configurations on the SIP server. Configuration procedure To specify UDP or TCP as the global transport layer protocol for outgoing SIP calls: Step Command Remarks Enter system view.
Page 162: Configuring Udp Or Tcp For Incoming Sip Calls
Configuring UDP or TCP for incoming SIP calls Configuration prerequisites If a SIP server is used to forward calls, you need to complete the configuration at the SIP server end. Configuration procedure To specify UDP or TCP as the transport layer protocol for incoming SIP calls: Step Command Remarks...
Page 163 Step Command Remarks Enter system view. system-view Enter voice view. voice-setup Enter SIP client view. Specify TLS as the global transport layer protocol for transport tls By default, UDP is adopted. outgoing SIP calls. To specify TLS as the transport layer protocol for outgoing SIP calls on a VoIP voice entity: Step Command Remarks...
Page 164: Configuring Media Flow Protocols For Sip Calls
Step Command Remarks Enter system view. system-view Enter voice view. voice-setup Enter SIP client view. Optional. Set the aging time for TLS timer connection age tls By default, the aging time for TLS connection. tls-age-time connection is 30 minutes. Configuring media flow protocols for SIP calls Configuration prerequisites If a SIP server is used to forward calls, complete the configurations on the SIP server.
Page 165: Configuring Sip Extensions
Configuration procedure To specify the global URL scheme for outgoing SIP calls: Step Command Remarks Enter system view. system-view Enter voice view. voice-setup Enter SIP client view. Optional. By default, the SIP scheme is Specify the global URL adopted. scheme for outgoing SIP url { sip | sips } You can specify the SIPS scheme calls.
Page 166: Configuring Source Ip Address Binding For Sip Messages
Configuration prerequisites If a SIP server is used to forward calls, complete the configurations on the SIP server. To implement bidirectional DTMF out-of-band transmission, configure it in the VoIP entity on the calling UA and in the corresponding POTS voice entity on the called UA. For specific configurations, see "Configuring out-of-band SIP DTMF transmission mode."...
Page 167: Configuring A Domain Name For The Sip Ua
The following table describes how source address binding works in different conditions: Condition Result • A new source address binding for media does not take effect for ongoing SIP media sessions but takes effect for Configure a source address binding when subsequent SIP media sessions.
Page 168 Configuration prerequisites If a SIP server is used to forward calls, complete the configurations on the SIP server. Configuring how to obtain the destination number and address Step Command Remarks Enter system view. system-view Enter voice view. voice-setup Enter SIP client view. Configure the device to By default, the destination obtain the destination...
Page 169: Configuring User-Agent And Server Header Fields
After the T.38 compatibility option is configured, the device can recognize T.38-specific description fields, parameters T38FaxTranscodingJBIG, T38FaxTranscodingMMR, T38FaxFillBitRemoval. These are in the SDP fields of the re-INVITE requests and 200 OK responses, and do not contain :0. By default, the compatibility options are not carried in re-INVITE requests. After the compatibility option command is configured, the compatibility options will be carried in re-INVITE requests.
Page 170: Configuring Sip Extensions For Caller Identity And Privacy
Step Command Remarks Enter system view. system-view Enter voice view. voice-setup Enter SIP client view. By default, the User-Agent header Configure the User-Agent sip-comp agent field in a SIP request is not header field in a SIP request. product-name product-version configured.
Page 171: Configuring Call Release Cause Code Mapping
Configuring call release cause code mapping Configuration prerequisites If a SIP server is used to forward calls, complete the configurations on the SIP server. Configuration procedure No matter whether a voice call is cleared normally or abnormally, a message with the call release cause code will be sent.
Page 172: Enabling Early Media Negotiation
Configuration procedure To configure periodic refresh of SIP sessions: Step Command Remarks Enter system view. system-view Enter voice view. voice-setup Enter SIP client view. Optional. By default,. • The periodic refresh of SIP sessions is not enabled automatically. That is, if Enable periodic refresh of periodic refresh of SIP SIP sessions and set the...
Page 173: Displaying And Maintaining Sip Uas
When you configure VRF-aware SIP, follow these restrictions and guidelines: • Configure this feature when no SIP services are running. This feature takes effect on all SIP services such as SIP calling, registration, and subscription. • To use SIP source binding, the VPN instance associated with the source interface must be the same as the VPN instance specified in this task.
Page 174: Sip Ua Configuration Examples
Task Command Remarks Clear the DNS dynamic cache. reset dns host { naptr | srv } Available in user view. Clear SIP DNS lookup records. reset voice sip dns-record Available in user view. SIP UA configuration examples This section provides SIP UA configuration examples. Configuring direct calling for SIP UAs Network requirements Two routers can directly call each other as SIP UAs.
Page 175: Configuring Proxy Server Involved Calling For Sip Uas
[RouterB-Ethernet2/1] ip address 192.168.2.2 255.255.255.0 # Configure voice entities. <RouterB> system-view [RouterB] voice-setup [RouterB-voice] dial-program [RouterB-voice-dial] entity 2222 pots [RouterB-voice-dial-entity2222] line 1/0 [RouterB-voice-dial-entity2222] match-template 2222 [RouterB-voice-dial-entity2222] quit [RouterB-voice-dial]entity 1111 voip [RouterB-voice-dial-entity1111] address sip ip 192.168.2.1 [RouterB-voice-dial-entity1111] match-template 1111 Configuration verification With the above configuration, you can use telephone 1111 to call telephone 2222, or use telephone 2222 to call telephone 1111.
Page 176: Configure The Ethernet Interface
[RouterA-voice] sip [RouterA-voice-sip] registrar ipv4 192.168.2.3 [RouterA-voice-sip] proxy ipv4 192.168.2.3 [RouterA-voice-sip] user routerA password cipher 1234 [RouterA-voice-sip] register-enable on [RouterA-voice-sip] quit # Configure voice entities. <RouterA> system-view [RouterA] voice-setup [RouterA-voice] dial-program [RouterA-voice-dial] entity 1111 pots [RouterA-voice-dial-entity1111] line 1/0 [RouterA-voice-dial-entity1111] match-template 1111 [RouterA-voice-dial-entity1111] quit [RouterA-voice-dial] entity 2222 voip [RouterA-voice-dial-entity2222] address sip proxy...
Page 177: Configuring Dns Involved Calling For Sip Uas
Configuring DNS involved calling for SIP UAs Network requirements Two routers work as SIP UAs and SIP calls are made through DNS. Figure 44 Network diagram Configuration procedure Configure Router A: # Configure the Ethernet interface. <RouterA> system-view [RouterA] interface ethernet 2/1 [RouterA-Ethernet2/1] ip address 192.168.2.1 255.255.255.0 # Map the IP address 192.168.2.2 to the host name cc.news.com.
Page 178: Configuring Out-Of-Band Sip Dtmf Transmission Mode
[RouterB-voice-dial-entity2222] quit [RouterB-voice-dial]entity 1111 voip [RouterB-voice-dial-entity1111] address sip ip 192.168.2.1 [RouterB-voice-dial-entity1111] match-template 1111 Configuration verification After the above configuration, calls between telephone 1111 and telephone 2222 are made through the DNS server. Configuring out-of-band SIP DTMF transmission mode Network requirements Two routers work as SIP UAs.
Page 179: Configuring Sip Extensions For Caller Identity And Privacy
# Configure the voice entity. <RouterB> system-view [RouterB] voice-setup [RouterB-voice] dial-program [RouterB-voice-dial] entity 1111 voip [RouterB-voice-dial-entity1111] address sip ip 192.168.2.1 [RouterB-voice-dial-entity1111] match-template 1111 [RouterB-voice-dial-entity1111] outband sip [RouterB-voice-dial-entity2222] quit [RouterB-voice-dial] entity 2222 pots [RouterB-voice-dial-entity2222] line 1/0 [RouterB-voice-dial-entity2222] match-template 2222 [RouterB-voice-dial-entity2222] outband sip Configuration verification After the call is established, if one side presses the telephone keys, the DTMF digits are transmitted to the other side using out of band signaling, and the other side hears short DTMF tones from the...
Page 180: Configuring Tcp To Carry Outgoing Sip Calls
[RouterA-voice-dial] entity 1111 pots [RouterA-voice-dial-entity1111] line 1/0 [RouterA-voice-dial-entity1111] match-template 1111 [RouterA-voice-dial-entity1111] quit [RouterA-voice-dial] quit [RouterA-voice] quit # Disable the calling voice subscriber line from sending the calling number to the remote end. [RouterA] subscriber-line 1/0 [RouterA-subscriber-line1/0] undo cid send [RouterA-subscriber-line1/0] quit # Configure the P-Asserted-Identity header field.
Page 181 Confguration procedure Configure Router A: # Configure the IP address of the Ethernet interface. <RouterA> system-view [RouterA] interface ethernet 2/1 [RouterA-Ethernet2/1] ip address 192.168.2.1 255.255.255.0 [RouterA-Ethernet2/1] quit # Specify TCP as the global transport layer protocol for outgoing SIP calls. [RouterA] voice-setup [RouterA-voice] sip [RouterA-voice-sip] transport tcp...
Page 182: Configuring Tls To Carry Outgoing Sip Calls
Configuring TLS to carry outgoing SIP calls Network requirements Two routers Router A and Router B work as SIP UAs. It is required that the SIP calls between the two parties be carried over TLS. Figure 48 Network diagram Confguration procedure The certification authority (CA) server runs RSA Keon in this configuration example.
Page 183 # Retrieve the CA certificate from the certificate issuing server. [RouterA] pki retrieval-certificate ca domain voice # Request a local certificate from the CA. [RouterA] pki request-certificate domain voice # Create an SSL server policy named server and configure the policy to use PKI domain voice. [RouterA] ssl server-policy server [RouterA-ssl-server-policy-server] pki-domain voice # Create an SSL client policy named client and configure the policy to use PKI domain voice.
Page 184: Configuring Sips Url Scheme For Outgoing Sip Calls
[RouterB-pki-domain-voice] certificate request url http://192.168.0.88:446/bd0683e5a369eb4edbb4ef502eaca6ec42d24e97 [RouterB-pki-domain-voice] certificate request from ca [RouterB-pki-domain-voice] certificate request entity aaa [RouterB-pki-domain-voice] quit # Create local RSA key pairs. [RouterB] public-key local create rsa # Retrieve the CA certificate from the certificate issuing server. [RouterB] pki retrieval-certificate ca domain voice # Request a local certificate from the CA.
Page 185 Figure 49 Network diagram Configuration procedure You can use the SIPS URL scheme only when the transport layer protocol is TLS. Therefore, before performing following configurations, you need to configure TLS for SIP as described in "Configuring TLS to carry outgoing SIP calls."...
Page 186: Configuring Srtp For Sip Calls
<RouterB> system-view [RouterB] interface ethernet 2/1 [RouterB-Ethernet2/1] ip address 192.168.2.2 255.255.255.0 [RouterB-Ethernet2/1] quit # Reference the created SSL server and client policies for SIP. Specify TLS as the transport layer protocol for both outgoing and incoming SIP calls, and specify URL scheme for the outgoing SIP call as SIPS.
Page 187: Troubleshooting
# Specify SRTP as the media flow protocol for SIP calls. [RouterA] voice-setup [RouterA-voice] sip [RouterA-voice-sip] media-protocol srtp [RouterA-voice-sip] quit # Specify 1111 as a local number of POTS voice entity 1111. [RouterA-voice-dial] entity 1111 pots [RouterA-voice-dial-entity1111] line 1/0 [RouterA-voice-dial-entity1111] match-template 1111 # Configure VoIP voice entity 2222, and configure the IP address of the peer VoIP gateway as 192.168.2.2, and the called number as 2222.
Page 188: Failed To Register With The Registrar
Failed to set up calls in the proxy server approach to SIP routing Symptom The UA could not set up calls when the proxy server approach was adopted to SIP routing. Solution Do the following: • Execute the display current-configuration command to check for information about the SIP proxy server or the registrar.
Page 189: Failed To Set Up Point-To-Point Sip Calls Over Tls
Failed to set up point-to-point SIP calls over TLS Symptom The UA could not set up point-to-point SIP calls over TLS when the basic voice gateway settings are correct. Solution Check that: • The devices of both sides have certificates. •...
Page 190: Configuring Sip Local Survival
Configuring SIP local survival IP phones have been deployed throughout the headquarters and branches of many enterprises and organizations. Typically, a voice server is deployed at the headquarters to control calls originated by IP phones at the branches. The local survival feature enables the voice router at a branch to automatically detect the reachability to the headquarter voice server, and process calls originated by attached IP phones when the headquarters voice server is unreachable.
Page 191: Configuration Task List
• MSR1000. • MSR30-11E. • MSR30-11F. Configuration task list Task Remarks Configuring an operation mode for the local SIP server Required. Configuring user information Optional. Specifying a trusted node Optional. Configuring call authority control Optional. Optional. Configuring an area prefix Applicable to calls initiated from external users to internal users.
Page 192 Step Command Remarks By default, no IP address is configured, that is, there is no local SIP server. Note that the ipv4-address argument can be the IP address of an interface on Configure the IP the local router, or a loopback address address of the local server-bind ipv4 ipv4-address such as 127.0.0.1.
Page 193: Configuring User Information
Step Command Remarks By default, the local SIP server is disabled. Enable the local SIP server enable The functions of the local SIP server can server. take effect after you execute this command. Configuring user information Step Command Remarks Enter system view. system-view Enter voice view.
Page 194: Configuring Call Authority Control
Step Command Remarks trusted-point ipv4 ipv4-address By default, no trusted node is Specify a trusted node. [ port port-number ] specified. Configuring call authority control This section describes how to configure call authority control. Configuring a call rule The local SIP server supports the call authority control feature. Define different call rules and apply them in different views to control the call authorities of users within the jurisdiction.
Page 195: Configuring An Area Prefix
Step Command Remarks Enter system view. system-view Enter voice view. voice-setup Enter SIP server view. sip-server Create a user to be By default, no user is created to be registered with and enter register-user tag registered. register user view. By default, no call rule set is applied register user view.
Page 196: Feature Configuration
Step Command Remarks Enter voice view. voice-setup Enter SIP server view. sip-server Enter call route view. call-route trunk tag called-number called-pattern By default, no call route is Configure a call route. ipv4 dest-ip-addr [ port port-number ] configured. [ area-prefix prefix ] Displaying and maintaining the SIP local survival feature configuration Task...
Page 197 # Configure authentication information for Phone 1000 and Phone 5000. [RouterC-voice-server] register-user 1000 [RouterC-voice-server-user1000] number 1000 [RouterC-voice-server-user1000] authentication username 1000 password simple 1000 [RouterC-voice-server-user1000] quit [RouterC-voice-server] register-user 5000 [RouterC-voice-server-user5000] number 5000 [RouterC-voice-server-user5000] authentication username 5000 password simple 5000 Configure Router A: # Configure voice entities.
Page 198: Configuring The Local Sip Server To Operate In The Alive Mode (Method 1)
user number status address ----------------------------------------------------------------------1000 1000 online 1.1.1.1:5060 5000 5000 online 2.1.1.1:5060 Phone 1000 and Phone 5000 are successfully registered with the local SIP server Router C and they can communicate with each other. Configuring the local SIP server to operate in the alive mode (method 1) Network requirements Router A and Router B carry out call services through VCX.
Page 199 [RouterA] voice-setup [RouterA-voice] dial-program [RouterA-voice-dial] entity 1000 pots [RouterA-voice-dial-entity1000] match-template 1000 [RouterA-voice-dial-entity1000] line 2/0 [RouterA-voice-dial-entity1000] user 1000 password simple 1000 [RouterA-voice-dial-entity1000] quit [RouterA-voice-dial] entity 5000 voip [RouterA-voice-dial-entity5000] address sip proxy [RouterA-voice-dial-entity5000] match-template 5000 [RouterA-voice-dial-entity5000] quit [RouterA-voice-dial] quit # Configure and enable SIP registration. [RouterA-voice] sip [RouterA-voice-sip] registrar ipv4 3.1.1.1 [RouterA-voice-sip] registrar ipv4 2.1.1.2 slave...
Page 200: Configuring The Local Sip Server To Operate In The Alive Mode (Method 2)
If the link between Router C and VCX is disconnected, Phone 1000 and Phone 5000 can register with the local SIP server again to originate calls. Configuring the local SIP server to operate in the alive mode (method 2) Network requirements Router A and Router B carry out call services through the VCX.
Page 201 <RouterA> system-view [RouterA] voice-setup [RouterA-voice] dial-program [RouterA-voice-dial] entity 1000 pots [RouterA-voice-dial-entity1000] match-template 1000 [RouterA-voice-dial-entity1000] line 8/0 [RouterA-voice-dial-entity1000] quit [RouterA-voice-dial] entity 5000 voip [RouterA-voice-dial-entity5000] address sip proxy [RouterA-voice-dial-entity5000] match-template 5000 [RouterA-voice-dial-entity5000] quit [RouterA-voice-dial] quit # Configure and enable SIP registration. [RouterA-voice] sip [RouterA-voice-sip] registrar ipv4 3.1.1.1 [RouterA-voice-sip] registrar ipv4 2.1.1.2 slave [RouterA-voice-sip] register-enable on...
Page 202: Configuring The Call Authority Control
Configuring the call authority control Network requirements Figure 55, the DNs for Department A in a company are 1000 through 1999, while those As shown in for Department B are 5000 through 5999. The following restrictions need to be implemented: •...
Page 203 [RouterC-voice-server-set-svc0] rule 0 deny outgoing any [RouterC-voice-server-set-svc0] rule 1 permit outgoing 5... [RouterC-voice-server-set-svc0] rule 2 permit outgoing 1... [RouterC-voice-server-set-svc0] quit [RouterC-voice-server-set] service 2 [RouterC-voice-server-set-svc2] rule 0 deny outgoing 1000 [RouterC-voice-server-set-svc2] quit [RouterC-voice-server-set] quit [RouterC-voice-server] # Apply the call rule set. [RouterC-voice-server] srs 0 [RouterC-voice-server] register-user 5000 [RouterC-voice-server-user5000] srs 2...
Page 204: Configuring An Area Prefix
[RouterB-voice-dial-entity5555] line 2/1 [RouterB-voice-dial-entity5555] user 5555 password simple 5555 [RouterB-voice-dial-entity5555] quit [RouterB-voice-dial] entity 1000 voip [RouterB-voice-dial-entity1000] address sip proxy [RouterB-voice-dial-entity1000] match-template 1... [RouterB-voice-dial-entity1000] quit [RouterB-voice-dial] quit # Enable SIP registration. [RouterB-voice] sip [RouterB-voice-sip] registrar ipv4 2.1.1.2 [RouterB-voice-sip] register-enable on Verify the configurations: [RouterC-voice-server] display voice sip-server register-user all user number...
Page 205 # Set Router A to a trusted node. [RouterC-voice-server] trusted-point ipv4 1.1.1.1 # Configure the area prefix 8899. [RouterC-voice-server] area-prefix 8899 # Configure authentication information for Phone 5000. [RouterC-voice-server] register-user 5000 [RouterC-voice-server-user5000] number 5000 [RouterC-voice-server-user5000] authentication username 5000 password simple 5000 Configure Router A: # Configure voice entities.
Page 206: Configuring A Call Route
Configuring a call route Network requirements Figure As shown in • The internal numbers of a company are four-digit long and the area prefix is 8899. • External phone 55665000 attached to Router B is not registered with the local SIP server on Router C.
Page 207 [RouterA-voice-dial] quit # Enable SIP registration. [RouterA-voice] sip [RouterA-voice-sip] registrar ipv4 2.1.1.2 [RouterA-voice-sip] register-enable on Configure Router B: # Configure voice entities. <RouterB> system-view [RouterB] voice-setup [RouterB-voice] dial-program [RouterB-voice-dial] entity 55665000 pots [RouterB-voice-dial-entity55665000] match-template 55665000 [RouterB-voice-dial-entity55665000] line 2/0 [RouterB-voice-dial-entity55665000] quit Verify the configurations: [RouterC-voice-server-user1000] display voice sip-server register-user all user...
Page 208: Configuring Sip Trunk
Configuring SIP trunk This chapter describes how to configure SIP trunk. Background Figure 58, on a typical telephone network, internal calls of the enterprise are made As shown in through the internal PBX, and external calls are placed over a PSTN trunk. Figure 58 Typical telephone network With the development of IP technology, many enterprises deploy SIP-based IP-PBX networks as Figure...
Page 209: Typical Applications
Figure 60 All IP-based network Features SIP trunk has the following features: • Only one secure and QoS guaranteed SIP trunk link is required between a SIP trunk device and the ITSP. The SIP trunk link can carry multiple concurrent calls, and the carrier only authenticates the link instead of each SIP call carried on this link.
Page 210: Protocols And Standards
Figure 61 SIP trunk network diagram Protocols and standards SIP trunk-related protocols and standards are as follows: • RFC 3261 • RFC 3515 • SIPconnect Technical Recommendation v1.1 Hardware compatibility with SIP trunk SIP trunk is not available on the following routers: •...
Page 211: Enabling The Sip Trunk Function
Task Remarks Use one of the three methods. Binding a SIP server group to the VoIP voice entity Specifying the destination address Configuring call routes for outbound calls Specifying the proxy server used for outbound calls Configuring call match rules Optional.
Page 212: Enabling The Real-Time Switching Function
Step Command Remarks Enter system view. system-view Enter voice view. voice-setup Create a SIP server group and enter SIP server-group group-number server group view. By default, a SIP server group has no member server. Add a member address index-number { ipv4 You can add at most five member server to the SIP ip-address | dns dns-name } [ port...
Page 213: Configuring The Keepalive And Redundancy Functions
Step Command Remarks Enable the real-time switching function in hot-swap enable Disabled by default. the SIP server group. NOTE: The real-time switching time is determined by the voip timer voip-to-pots command. For more information about the voip timer voip-to-pots command, see HPE FlexNetwork MSR Router Series Comware 5 Voice Command Reference.
Page 214: Configuring Source Address Binding
Step Command Remarks Enable the Disabled by default. keepalive function If the keepalive function is disabled, the and set the keepalive { options [ interval current server is always the one with keepalive mode for seconds ] | register } the highest priority in the SIP server the SIP server group.
Page 215: Configuring A Sip Trunk Account
The following table describes how source address binding works in different conditions: Condition Result • A new source address binding for media does not take effect for ongoing SIP media sessions but takes effect for Configure a source address binding when subsequent SIP media sessions.
Page 216: Configuring Registration Timers For A Sip Trunk Account
Step Command Remarks Assign the host username allocated assign contact-user user-name Not assigned by default. by the ITSP to the SIP trunk account. Assign the host name Optional. allocated by the ITSP assign host-name host-name to the SIP trunk Not assigned by default. account.
Page 217: Specifying The Destination Address
Step Command Remarks By default, no SIP trunk account is bound to the VoIP Bind a SIP trunk voice entity. account to the VoIP bind sip-trunk account account-index Only an existing SIP trunk voice entity. account can be bound to a VoIP voice entity.
Page 218: Configuring Call Match Rules
Step Command Remarks By default, no SIP trunk account is Bind a SIP trunk bound to the VoIP voice entity. bind sip-trunk account account to the VoIP account-index Only an existing SIP trunk account can voice entity. be bound to a VoIP voice entity. Enable the SIP proxy address sip proxy Not configured by default.
Page 219: Configuring Call Routes For Inbound Calls
Step Command Remarks Optional. Match a source host Not specified by default. In name prefix for the VoIP match source host-prefix host-prefix other words, all source host voice entity. names can be matched. Optional. Match a destination host match destination host-prefix Not specified by default.
Page 220: Enabling Media Flow-Around
Step Command Remarks Optional. By default, codec transparent transfer is disabled and the SIP trunk device is involved in the media negotiation between the calling and Enable codec transparent codec transparent called parties. transfer. Note that to enable codec transparent transfer on the SIP trunk device, execute this command on all VoIP voice entities connected to...
Page 221: Enabling Codec Transcoding
Step Command Remarks Enter VoIP voice entity view. entity entity-number voip Optional. Enable DO-EO conversion. early-offer forced By default, DO-EO conversion is disabled. NOTE: If codec transparent transfer or media flow-around is enabled, the early-offer forced command does not takes effect. Enabling codec transcoding In the scenario where the SIP trunk device controls the results of media capability negotiation, if the SIP trunk device cannot find a common codec for two parties during negotiation, the two parties will...
Page 222: Enabling Call Forwarding
Step Command Remarks Enter voice view. voice-setup Enter SIP client view. Optional. By default, address hiding Enable address hiding. address-hiding enable for SIP-to-SIP calls is disabled. Enabling call forwarding Usually, the SIP trunk device transparently transfers the SIP messages carrying call forwarding information to the endpoints, and the endpoints perform the call forwarding.
Page 223: Enabling Midcall Signaling Pass-Through
Step Command Remarks Optional. supplementary−service sip Enable call transfer. By default, call forwarding call-transfer is disabled. Enabling midcall signaling pass-through When the SIP trunk device forwards the media flows between the calling and called parties, the device can monitor the status of the media flows over a period of time and determine whether the endpoints have hung up abnormally.
Page 224: Sip Trunk Configuration Examples
SIP trunk configuration examples This section provides configuration examples of SIP trunk. Configuring a SIP server group with only one member server Network requirements Figure 62, the enterprise private network has a SIP trunk device deployed. Router A is a As shown in private network device, and Router B is a public network device.
Page 225 [TG-voice] server-group 1 [TG-voice-group1] address 1 ipv4 10.1.1.2 [TG-voice-group1] quit # Create SIP trunk account 1 with the host user name 2000, and associate the account with SIP server group 1. [TG-voice] sip-trunk account 1 [TG-voice-account-1] assign contact-user 2000 [TG-voice-account-1] registrar server-group 1 [TG-voice-account-1] register enable [TG-voice-account-1] quit # Configure the call route for the outbound calls from private network user 2000 to public...
Page 226: Configuring A Sip Server Group With Multiple Member Servers
2000 10.1.1.2:5060 1802 Online The output shows that the private network account 2000 has registered with the server at 10.1.1.2. All calls between the private network and public network are made through the SIP trunk device. Execute the display voice statistics call-active command on the SIP trunk device and you can see that all calls between the private network and public network are made through the SIP trunk device.
Page 227: Configuring Call Match Rules
[TG-voice-group-1] keepalive options [TG-voice-group-1] hot-swap enable [TG-voice-group-1] quit [TG-voice] sip [TG-voice-sip] redundancy mode parking Other configurations on the SIP trunk device and other devices are the same as those described in procedure." "Configuration Configuration verification When the SIP server with IP address 10.1.1.2 fails, the SIP server with IP address 10.1.1.3 takes over communications between the private network and the public network.
Page 228 # Specify that calls with the source IP addresses 1.1.1.1 are permitted on VoIP voice entity 1. [TG-voice-dial-entity1] match source address ipv4 1.1.1.1 Other configurations on the SIP trunk device and on other devices are the same as that described in procedure."...
Page 229: Configuring H.323
Configuring H.323 H.323 is an application-layer control protocol for establishing and terminating multimedia sessions with one or more participants. H.323 can dynamically adjust and modify session attributes, such as required session bandwidth, media type (voice and video), encoding/decoding format of media, and support for broadcast.
Page 230: H.323 Architecture
Table 16 Major RAS messages Category Message RRQ (Registration_Request) Registration RCF (Registration_Confirm) RRJ (Registration_Reject) URQ (Unregister_Request) Unregistration UCF (Unregister_Confirm) URJ (Unregister_Reject) MRQ (Manage_Request) Management MCF (Manage_Confirm) MRJ (Manage_Reject) ARQ (Admission_Request) Admission ACF (Admission_Confirm) ARJ (Admission_Reject) LRQ (Location_Request) Location requests and responses LCF (Location_Confirm) LRJ (Location_Reject) DRQ (Disengage_Request)
Page 231: Gatekeeper Discovery
Figure 66 shows a simple H.323 network. For all calls, a gatekeeper is the call service control and central control unit in its administrative zone. Gateway entities are usually deployed on routers. Configure the IP voice gateway function on the router at CLI.
Page 232 process. This is how the gatekeeper controls admission. In the ARQ message sent to the gatekeeper, Figure 67) or gatekeeper-routed call the calling endpoint might ask for direct call signaling (see Figure 68). The gatekeeper finalizes the way of sending call signaling to the called signaling (see endpoint and notifies the calling endpoint of it through an Admission_Confirm (ACF) message.
Page 233: Call Proceeding
Call setup After receiving the ACF message from the gatekeeper, the calling endpoint sends call signaling to set up a call. In a direct call signaling for example, the calling endpoint first sends a Setup message to the called endpoint requesting for a connection. Call proceeding After receiving the Setup message, the called endpoint replies with a Call Proceeding message.
Page 234: Hardware Compatibility With H.323
Figure 69 Call setup flow and disconnection flow in which gatekeepers are involved Hardware compatibility with H.323 H.323 is not available on the following routers: • MSR900. • MSR93X. • MSR1000. • MSR30-11E. • MSR30-11F. H.323 gateway configuration task list Task Remarks Configuring basic H.323 gateway functions...
Page 235: Configuring Registration Password
Configuration guidelines • The gatekeeper identifies the type of a gateway by its area ID. The gatekeeper and gateways reach an agreement on related gateway types in advance. For example, Area ID 1# represents a voice gateway, and Area ID 2# represents a video gateway. Before a VoIP entity communicates with the gatekeeper, the gatekeeper judges the gateway type by the received area ID.
Page 236: Enabling Security Calling
After the GK Client (router) is configured with the registration password, the password is carried during the whole registration process. Configuration prerequisites Complete the required basic H.323 gateway configurations except for the ras-on command. (This command is used for initiating registration requests to the gatekeeper after all configurations are completed.) Configuration procedure To configure a registration password:...
Page 237: Displaying And Maintaining The H.323 Gateway
Displaying and maintaining the H.323 gateway Task Command Remarks display voice gateway [ | { begin Display the registration state | exclude | include } Available in any view. information of a gateway. regular-expression ] H.323 gateway configuration example Network requirements Telephones in City A and City B can communicate with each other through routers with the voice function across an IP network where a gatekeeper is used for dynamic telephone number to IP address translation.
Page 238 [RouterA-voice-dial-entity755] quit # Create a POTS entity. [RouterA-voice-dial] entity 1001 pots [RouterA-voice-dial-entity1001] match-template 0101001 [RouterA-voice-dial-entity1001] line 1/1 # Specify a loopback interface as the H.323 gateway interface. [RouterA-voice-dial-entity1001] return <RouterA> system-view [RouterA] interface loopback 0 [RouterA-Loopback0] ip address 1.1.1.1 255.255.255.255 # Enter gateway client view.
Page 239: Troubleshooting H.323 Gateway
[RouterB-voice-gk] gk-id gk-center gk-addr 3.3.3.3 1719 # Configure the area ID. [RouterB-voice-gk] area-id 1# # Originate registration to the GK. [RouterB-voice-gk] ras-on Troubleshooting H.323 gateway Registration failure Symptom The gateway failed to register with the gatekeeper. Solution Check that: The gateway and the gatekeeper can communicate with each other at the network layer with the ping command.
Page 240: Configuring Call Services
Configuring call services More and more VoIP-based services are demanded as voice application environments expand. On the basis of basic calls, new features are implemented to meet different application requirements of VoIP subscribers. Call waiting When subscriber C calls subscriber A who is already engaged in a call with subscriber B, the call will not be rejected if call waiting is enabled.
Page 241: Call Backup
subscriber C (final recipient). After Subscriber A hangs up, the call between subscriber B and subscriber C is established. This is call transfer. To perfect the call transfer feature, the device supports the call recovery function after the call transfer fails. In other words, if subscriber C in the previous example is in a conversation with another subscriber and cannot establish a conversation with subscriber B, the call between subscriber A and subscriber B is recovered.
Page 242: Call Services Configuration Task List
three-party conference can be held. In this example, C is called the active participant of the conference, A is the voice mixer, and B is the original participant of the conversation. Silent monitor and barge in services can be considered as the extensions of three-party conference. To distinguish them with traditional three-party conference, these two services are called three-party conference in active participation mode.
Page 243: Configuring Call Waiting
• The system administrator performs configurations by using command lines on the device. This method allows for more configuration options. • Terminal subscribers perform configurations on telephones for convenience of use and for relieving the burden on the system administrator. By default, the device supports the following functions, without any configuration: •...
Page 244: Enabling And Disabling Call Waiting By Using Keys
Enabling and disabling call waiting by using keys The device allows you to configure call waiting for FXS voice subscriber lines through telephones. The call waiting configuration made on a telephone is also valid for the device. For example, after you enable call waiting on a telephone, you can view the operation result by using the display this command in corresponding voice subscriber line view.
Page 245: Configuration Example
Configuration example Enable call waiting for the voice subscriber line of Telephone A. Place a call from Telephone C to Telephone A, which is in a conversation with Telephone B. The subscriber at Telephone A hears call waiting tones, and Telephone C is not rejected but waits for Telephone A to answer the new call. # Enable call waiting for the voice subscriber line of Telephone A.
Page 246: Configuring Call Forwarding
trunk." Before specifying the MoH number, make sure that the playing mode of the call hold initiator is inactive (silent mode). To configure tone playing mode of the call hold feature: Step Command Remarks Enter system view. system-view Enter voice view. voice-setup call-hold-format { inactive | Configure the tone playing...
Page 247: Configuring Call Forwarding By Using Command Lines
Enable keys Disable keys Remarks Enable and disable call forwarding busy, where "number" *40*number# #40# represents a forwarded-to number. Enable and disable call forwarding no reply, where *41*number# #41# "number" represents a forwarded-to number. Enable and disable call forwarding unavailable, where *60*number# #60# "number"...
Page 248: Configuration Example
Configuring call forwarding priority level A priority level applies to only the features of call waiting, call forwarding, and hunt group. By default, the priority levels for hunt group, call forwarding, and call waiting are 1, 2, and 3 respectively. The smaller the value is, the higher the priority level is. When you change the priority level of a feature, make sure that different features have different priority levels.
Page 249: Configuring Call Transfer
<Sysname> system-view [Sysname] subscriber-line 2/0 [Sysname-subscriber-line2/0] call-forwarding unavailable enable forward-number 3000 Configuring call transfer This section describes the procedure for configuring call transfer by using command lines. Configuration prerequisites • The router is equipped with an FXS voice interface card. •...
Page 250: Configuring Hunt Group
Configuring call backup By default, the call backup function is enabled on the device. The system supports two types of call backup: • Strict call backup One of the following three conditions will trigger strict call backup: The device does not receive any reply from the peer after sending out a call request. The device fails to initiate a call to the IP network side.
Page 251: Configuring Hunt Group Priority Level
Step Command Remarks Enable hunt group for the hunt-group enable Disabled by default. voice subscriber line. NOTE: To use the hunt group feature, you need to configure the hunt-group enable command on all involved voice subscriber lines. Configuring hunt group priority level A priority level applies to the features of call waiting, call forwarding, and hunt group only.
Page 252: Configuring Incoming Call Barring
Configuring incoming call barring When you do not want to receive any incoming call, you can enable incoming call barring (that is, the Do Not Disturb feature). The device supports two incoming call barring configuration methods: • Subscribers perform configurations by using keys on a telephone terminal. •...
Page 253: Configuring Outgoing Call Barring
Configuring outgoing call barring When subscribers do not want others to use their telephones, they can set a password to lock their telephones. Outgoing call barring can achieve this purpose. When they want to make calls, they can disable outgoing call barring. The device supports two outgoing call barring configuration methods: •...
Page 254: Configuring Mwi
Configuring MWI This section describes configuration procedures for enabling, disabling, displaying, and maintaining MWI. Configuration prerequisites The router is equipped with an FXS voice interface card. Enabling and disabling MWI Configure MWI using command lines to enable or disable the feature, and to set the message waiting tone duration.
Page 255: Displaying And Maintaining Mwi
Displaying and maintaining MWI Task Command Remarks display voice ss mwi { all | number number } [ | { begin | Display the information of MWI. Available in any view. exclude | include } regular-expression ] display voice sip subscribe-state [ | { begin | Display subscription information.
Page 256: Configuration Example
Configuring the three-party conference service in voice subscriber line view will invalidate the local call identification function. For more information about the configuration of the local call identification function, see the distinguish-localtalk command in HPE FlexNetwork MSR Router Series Comware 5 Voice Command Reference. To configure three-party conference by using command lines: Step Command...
Page 257: Configuring Three-Party Conference In Active Participation Mode By Using Command Lines
To configure three-party conference in active participation mode on a telephone: Enable keys Disable keys *34# #34# Configuring three-party conference in active participation mode by using command lines Step Command Remarks Enter system view. system-view Enter voice subscriber line subscriber-line line-number view.
Page 258: Configuring Feature Service
Configuring door opening control Step Command Remarks Enter system view. system-view Enter voice subscriber line subscriber-line line-number view. Enable the door opening control service and set a Optional. door-relay password [ simple | password for opening the cipher ] password [ time Door opening control is disabled door and the door open seconds ]...
Page 259 Feature name Enable keys Disable keys Remarks A subscriber can dial the feature code plus a destination mailbox number to transfer incoming calls to the voice Transfer To *441*destination# None mailbox. The destination mailbox Voicemail number must be an existing one. Applied only once In conversation, a subscriber can dial the feature code to park the call to a...
Page 260 Feature name Enable keys Disable keys Remarks A subscriber can dial the feature code to retrieve the voice mailbox. The subscriber can perform operations Retrieve Voice Mail *600 or *600*user# None according to the prompts. Applied only once. A subscriber can dial the feature code to enable or disable the feature.
Page 261: Configuring Feature Service By Using Command Lines
Configuring Feature service by using command lines The feature service indicates the service that is used together with the VCX. When you need to interact with the VCX by using telephone keys, you need to adopt out-of-band named telephone event (NTE) transmission to send the DTMF digits to the VCX. The execution of the feature permit command does not enable out-of-band NTE transmission, and you need to execute the outband nte command on the called entity to enable it.
Page 262: Call Services Configuration Examples
Call services configuration examples This section provides call services configuration examples. Call waiting Network requirements Figure 71, place a call from Telephone C to Telephone A which is already engaged in a As shown in call with Telephone B, and the call will not be rejected. Just like a normal call, the subscriber at Telephone C will hear ringback tones, while the subscriber at Telephone A will hear call waiting tones, as a reminder that another call is waiting on the line.
Page 263: Call Forwarding Busy
Configure Router B: <RouterB> system-view [RouterB] voice-setup [RouterB-voice] dial-program [RouterB-voice-dial] entity 1000 voip [RouterB-voice-dial-entity1000] address sip ip 10.1.1.1 [RouterB-voice-dial-entity1000] match-template 1000 [RouterB-voice-dial-entity1000] quit [RouterB-voice-dial] entity 2000 pots [RouterB-voice-dial-entity2000] line 1/0 [RouterB-voice-dial-entity2000] match-template 2000 Configure Router C: <RouterC> system-view [RouterC] voice-setup [RouterC-voice] dial-program [RouterC-voice-dial] entity 3000 pots [RouterC-voice-dial-entity3000] line 1/0...
Page 264 Figure 72 Network diagram Router A Router B Router C Eth1/1 Eth1/1 10.1.1.1/24 20.1.1.2/24 Eth1/2 Eth1/1 1000 10.1.1.2/24 20.1.1.1/24 3000 Telephone A Telephone C 2000 Telephone B Configuration procedure Before performing the following configuration, make sure that Router A, Router B and Router C are routable to each other.
Page 265: Call Transfer
[RouterC-voice-dial-entity3000] line 1/0 [RouterC-voice-dial-entity3000] match-template 3000 Call transfer Network requirements Figure 73, call transfer enables Telephone A to transfer Telephone B to Telephone C. As shown in After the call transfer is completed, Telephone B and Telephone C are in a conversation. The whole process is as follows: Call Telephone B from Telephone A, so that Telephone B and Telephone A are in a conversation.
Page 266: Hunt Group
<RouterA> system-view [RouterA] subscriber-line 1/0 [RouterA-subscriber-line1/0] call-hold enable [RouterA-subscriber-line1/0] call-transfer enable Configure voice entities on Router B. <RouterB> system-view [RouterB] voice-setup [RouterB-voice] dial-program [RouterB-voice-dial] entity 2000 pots [RouterB-voice-dial-entity2000] line 1/0 [RouterB-voice-dial-entity2000] match-template 2000 Configure Router C. <RouterC> system-view [RouterC] voice-setup [RouterC-voice] dial-program [RouterC-voice-dial] entity 3000 pots [RouterC-voice-dial-entity3000] line 1/0...
Page 267 Configure Router A: # Configure the voice entity with a higher priority. <RouterA> system-view [RouterA] voice-setup [RouterA-voice] dial-program [RouterA-voice-dial] entity 1000 pots [RouterA-voice-dial-entity1000] line 1/0 [RouterA-voice-dial-entity1000] match-template 1000 [RouterA-voice-dial-entity1000] quit # Configure the voice entity with a lower priority. [RouterA-voice-dial] entity 1001 pots [RouterA-voice-dial-entity1001] line 1/1 [RouterA-voice-dial-entity1001] match-template 1000 [RouterA-voice-dial-entity1001] priority 4...
Page 268 Network requirements Figure 75, Telephone A and Telephone B registered with the VCX through Router A and As shown in Router B respectively. Configure a voice mailbox for Telephone A on the voice server, configure the address and operation mode of the MWI server on Router A, and enable MWI on the voice subscriber line of Telephone A.
Page 269 Figure 77 Configuration page of call processing server (2) Configure unified messaging server # Configure 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. You can see the Figure Configuration Option box, as shown in Figure 78 Configuration page of unified messaging server...
Page 270 # Configure the voice mailbox of Telephone A Click the Edit A Mailbox link, enter the mailbox number 1000 of Telephone A, and then check that if the mailbox is created successfully. If you are prompted that the mailbox is not present, select the Create/Delete Mailboxes link to create the mailbox of Telephone A, with the mailbox number as 1000.
Page 271: Three-Party Conference
[RouterB-voice] quit # Configure the SIP server. [RouterB-voice] sip [RouterB-voice-sip] registrar ipv4 100.1.1.101 [RouterB-voice-sip] register-enable on After the above configuration, if a call is placed from Telephone B to Telephone A which is not picked up within the ringing timeout interval, the call will be forwarded to the voice mailbox. Then, the subscriber of Telephone B can leave a message and hang up.
Page 272 Configure Router B: # Configure the voice entity. <RouterB> 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 [RouterB-voice-dial] entity 1000 voip [RouterB-voice-dial-entity1000] address sip ip 10.1.1.1 [RouterB-voice-dial-entity1000] match-template 1000 [RouterB-voice-dial-entity1000] quit [RouterB-voice-dial] entity 2000 pots [RouterB-voice-dial-entity2000] line 1/0...
Page 273: Silent Monitor And Barge In
Silent monitor and barge in Network requirements • Configure silent monitor for Telephone C to monitor the conversation between Telephone A and Telephone B. Figure 81, Telephone A and Telephone B are in a conversation. Dial the feature As shown in code *425*Number of Telephone A# at Telephone C to monitor the conversation between Telephone A and Telephone B.
Page 274 Figure 82 Telephone configuration page # Configure the monitoring authority Click Features of number 1000 to enter the feature configuration page, and then click Edit Figure Feature of the Silent Monitor and Barge In feature to enter the page as shown in Figure 83 Silent monitor and barge in feature configuration page (1) Click Assign External Phones to specify that number 3000 has the authority to monitor Figure 84...
Page 275 After the above configuration, Telephone C with the number 3000 can monitor and barge in the conversations of Telephone A with the number 1000. Configure Router A: # Configure VoIP voice entities to Router B and Router C. <RouterA> system-view [RouterA] voice-setup [RouterA-voice] dial-program [RouterA-voice-dial] entity 3000 voip...
Page 276 [RouterB-voice-dial] entity 2000 pots [RouterB-voice-dial-entity2000] line 1/0 [RouterB-voice-dial-entity2000] match-template 2000 [RouterB-voice-dial-entity2000] quit [RouterB-voice-dial] quit # Specify the IP address 100.1.1.101 as the registrar and proxy server and enable the registrar. [RouterB-voice] sip [RouterB-voice-sip] registrar ipv4 100.1.1.101 [RouterB-voice-sip] proxy ipv4 100.1.1.101 [RouterB-voice-sip] register-enable on Configure Router C: # Configure VoIP voice entities to Router A and Router B.
Page 277: Configuring Call Watch
Configuring call watch The call watch function is only applicable to voice E1/T1 interfaces. The E1/T1 interfaces mentioned in this document are all voice interfaces. Overview Call watch enables a voice device to decide whether an E1/T1 interface is available for setting up calls for a callee by monitoring the state of the local interface or the IP connectivity to the remote interface connected to the callee.
Page 278: Call Watch Mode
• If local interfaces are monitored, the E1/T1 interface is set to watch-out state when all the monitored local interfaces are down. • If IP connectivity to remote interfaces is monitored, the E1/T1 interface is set to watch-out state when all the monitored remote IP addresses are unreachable. Call watch mode A call watch group can operate in hard or soft mode on an E1/T1 interface.
Page 279: Associating The E1/T1 Interface With The Call Watch Group
Step Command Remarks • Specify a local interface in the rule call-watch rule watch-number local Use either command. interface interface-type Create a rule in a call To create multiple rules in the interface-number watch group. call watch group, repeat the •...
Page 280: Monitoring Remote Ip Addresses
• Configure a call watch group to monitor interfaces Ethernet 1/1 and Ethernet 1/2. • Apply the call watch group to interface E1 1/0 and configure the call watch group to work in hard mode. Figure 86 Network diagram Configuration procedure Configure device Voice A: # Configure an IP address for each interface.
Page 281 Figure 87 Network diagram Configuration procedure Configure device Voice A: # Enable NQA server, configure two NQA test groups to monitor remote IP addresses 10.1.1.2 and 10.1.2.2, and associate the NQA test groups each with a track object. <VoiceA> system-view [VoiceA] nqa server enable [VoiceA] nqa entry admin test1 [VoiceA-nqa-admin-test1] type icmp-echo...
Page 282 Configure an IP address for each interface on Router A, Router B and Voice B. (Details not shown.)
Page 283: Fax Flow
Configuring fax over IP Traditional fax machines transmit and receive faxes over PSTN. Fax has gained wide acceptance due to its many advantages, such as high transmission speed and simple operations. By far, G3 fax machines dominant fax communications. A G3 fax machine adopts the signal digitizing technology. Image signals are digitized and compressed internally, then converted into analog signals through a modem, and finally transmitted into the PSTN switch through common subscriber lines.
Page 284: Hardware Compatibility With Foip
A real-time fax process consists of five phases: Fax call setup phase. This phase is similar to the process of a telephone call setup. The difference is that the fax tones identifying the sending/receiving terminals are included. Prior-messaging phase. During this phase, fax faculty negotiation and training are performed. Messaging phase.
Page 285: Configuring Fax Interworking Protocol
Configuring fax interworking protocol The device supports two fax protocols: T.38 protocol and standard T.38 protocol. The standard T.38 protocol should be selected for interworking with leading fax terminals in the industry. Since most leading fax terminals in the industry do not support the local training mode, the end-to-end training mode must be selected for interworking with them.
Page 286 You can implement the fax pass-through function on the voice gateway (router) in the following ways: • Configure the fax to work in the pass-through mode on both sides. • Negotiate the codec as G.711 and set the fax rate to disable on both sides. Then, disable the VAD function to avoid fax failures.
Page 287: Enabling Ecm For Fax
Step Command Remarks Optional. Configure the NTE By default, the value of the payload type for the NTE payload type is 100. modem compatible-param payload-type compatible-switching This command is valid only mode. for the NTE-compatible switching mode. Enabling CNG fax switchover Configuration prerequisites •...
Page 288: Configuring Fax Faculty Transmission Mode
Configuration procedure To configure ECM for fax: Step Command Remarks Enter system view. system-view Enter voice view. voice-setup Enter voice dial program dial-program view. entity entity-number { pots | vofr | Enter voice entity view. voip } By default, ECM is disabled on the gateway.
Page 289: Configuring Maximum Fax Rate
Configuring maximum fax rate You can configure the maximum fax rate according to the fax protocols. If the baud rate is set to a value other than disable and voice, the configured value is adopted as the allowed maximum fax rate.
Page 290: Configuring Threshold Of Local Training
Configuration prerequisites VoIP configuration is completed, IP calls can be made successfully, and fax machines are connected correctly. Configuration procedure To configure the fax training mode: Step Command Remarks Enter system view. system-view Enter voice view. voice-setup Enter voice dial program dial-program view.
Page 291: Configuring Transmit Energy Level Of Gateway Carrier
NOTE: When the local training mode is adopted, use the fax local-train threshold command to configure the threshold in percentage. When the PPP training mode is adopted, the gateway does not participate in rate training and the threshold of local training is not applicable. Configuring transmit energy level of gateway carrier Usually, the default transmit energy level of the gateway carrier is acceptable.
Page 292: Configuring Global Default Parameters For Fax
NOTE: Because NetMeeting does not support T.38 faculty description parsing, you must disable the voice gateway in H.323 slow connection mode from containing the T.38 faculty description in its faculty set to interwork with NetMeeting. Configuring global default parameters for fax Configuration prerequisites VoIP configuration is completed, IP calls can be made successfully, and fax machines are connected correctly.
Page 293: Displaying And Maintaining Foip Configuration
Step Command Remarks Optional. Configure the fax negotiation default entity fax nsf-on By default, the fax negotiation is faculty globally. based on the standard faculty. Optional. Configure the fax training default entity fax train-mode By default, the ppp training is mode globally.
Page 294 • The IP addresses 1.1.1.1/24 and 2.2.2.2/24 are respectively assigned to the interfaces through which the routers in City A and City B access the Internet. • Router A in City A and Router B in City B are connected to fax terminals through an FXS voice subscriber line.
Page 295: Configuring Sip Modem Pass-Through
Configuring SIP modem pass-through Network requirements Figure 90, enable SIP modem pass-through on Router A and Router B to realize data As shown in communication between PC1 and PC2. Figure 90 Network diagram Configuration procedure Configure Router A: # Set the switching mode to Re-Invite switching and the codec type to g711alaw for SIP modem pass-through.
Page 296 [RouterB-voice-dial] entity 2000 pots [RouterB-voice-dial-entity2000] match-template 2000 [RouterB-voice-dial-entity2000] line 1/0...
Page 297: Configuring Customizable Ivr
Configuring customizable IVR Overview Interactive voice response (IVR) is extensively used in voice communications. The IVR system enables you to customize interactive operations and humanize other services. If a subscriber dials an IVR access number, the IVR system plays the prerecorded voice prompts to direct the subscriber about how to proceed, for example, dial a number.
Page 298: Successive Jumping
Successive jumping The IVR process can realize successive jumping up to eight times from node to node. Error processing methods The IVR system provides three error processing methods: terminate the call, jump to a specified node, and return to the previous node. To handle errors, select an error processing method for a Call node, for a Jump node, or globally.
Page 299: Configuring An Ivr Voice Entity
Task Remarks Configuring a Call node Required. Use one of, two of or Configuring a Jump node Creating an IVR node all of the configurations as Configuring a Service node needed. Configuring an IVR voice entity This section describes the procedures for creating and configuring an IVR voice entity. Creating an IVR voice entity Step Command...
Page 300 Step Command Remarks Create an IVR voice entity and enter IVR voice entity entity entity-number ivr view. Configure a target By default, no number template is match-template for the match-template match-string configured for an IVR voice entity. IVR voice entity. Optional.
Page 301: Specifying The Id For A Media Resource
Step Command Remarks Optional. By default, no maximum-call-connection set is 13. Set the bound to an IVR voice entity (that maximum-call-connection max-call set-number is, the IVR voice entity does not number to the IVR voice belong to any entity. maximum-call-connection set and there is no limitation on the number of call connections).
Page 302: Creating An Ivr Node
To configure IVR processing methods globally: Step Command Remarks Enter system view. system-view Enter voice view. voice-setup Enter IVR management view. ivr-system Optional. By default, the maximum number Configure IVR global of times permitted for input errors ivr-input-error { media-play processing method for is 3.
Page 303: Configuring A Call Node
Configuring a Call node Use Call nodes to configure the secondary call function. You can configure two kinds of dial plans for a Call node: normal secondary call and extension secondary call. If you configure both dial plans for a Call node, the extension secondary call plan takes precedence over the normal secondary call plan, that is, after the subscriber dials the secondary call number, the system matches it with the extension secondary call numbers first, and then matches it with the normal secondary call numbers.
Page 304: Configuring A Jump Node
NOTE: If you do not configure error processing and timeout processing methods for a node, the system applies global processing methods. If you configure the processing methods both for the node and globally, the configurations made for the node takes precedence. Configuring a Jump node You can configure the following functions for a Jump node: playing audio files, jumping to another node, and terminating a call.
Page 305: Displaying And Maintaining Customizable Ivr
You can configure at most three functions for a Service node and use the select-rule operation-order command to specify the execution order of the functions. For example, if you specify the order by using the select-rule operation-order 2 1 3 command, the execution order of the functions is 2->1->3.
Page 306: Call Node Configuration Example 1: Dial Terminator Match, Normal Secondary Call
Call node configuration example 1: dial terminator match, normal secondary call Network requirements Figure 91, configure an IVR access number and customize Call node functions on As shown in Router B to meet the following requirements. • After the subscriber originates a SIP call by dialing 300 (the IVR access number of Router B) from Telephone A, Router B plays the audio file welcome.wav.
Page 307 [RouterB-voice] dial-program [RouterB-voice-dial] entity 500 pots [RouterB-voice-dial-entity500] match-template 500 [RouterB-voice-dial-entity500] line 1/0 [RouterB-voice-dial-entity500] quit [RouterB-voice-dial] entity 50 pots [RouterB-voice-dial-entity50] match-template 50 [RouterB-voice-dial-entity50] line 1/1 [RouterB-voice-dial-entity50] quit # Configure IVR voice entity 300 and specify node 10 as the root node. [RouterB-voice-dial] entity 300 ivr [RouterB-voice-dial-entity300] match-template 300 [RouterB-voice-dial-entity300] ivr-root 10...
Page 308: Call Node Configuration Example 2: Number Length Match, Normal Secondary Call
Call node configuration example 2: number length match, normal secondary call Network requirements Figure 92, configure the IVR access number and customize Call node functions on As shown in Router B. • After the subscriber originates a call through a SIP server by dialing 300 (the IVR access number of Router B) from Telephone A, Router B plays the audio file welcome.wav.
Page 309: Call Node Configuration Example 3: Number Match, Normal Secondary Call
Call node configuration example 3: number match, normal secondary call Network requirements Figure 93, configure the IVR access number and customize Call node functions on As shown in Router B. • After the subscriber originates a call through a SIP server by dialing 300 (the IVR access number of Router B) from Telephone A, Router B plays the audio file welcome.wav.
Page 310: Call Node Configuration Example 4: Extension Secondary Call
Call node configuration example 4: extension secondary call Network requirements Figure 94, configure the IVR access number and customize Call node functions on As shown in Router B. • After the subscriber originates a call through a SIP server by dialing 300 (the IVR access number of Router B) from Telephone A, Router B plays the audio file welcome.wav.
Page 311: Jump Node Configuration Example
[RouterB-voice-dial-entity300] ivr-root 10 [RouterB-voice-dial-entity300] quit [RouterB-voice-dial] quit # Specify media resource IDs for media resource files: Specify 10001 for the file cfa0:/wav/g729r8/welcome.wav Specify 10002 for the file cfa0:/wav/g729r8/timeout.wav Specify 10003 for the file cfa0:/wav/g729r8/input_error.wav [RouterB-voice] ivr-system [RouterB-voice-ivr] media-file g729r8 [RouterB-voice-ivr-g729r8] set-media 10001 file cfa0:/wav/g729r8/welcome.wav [RouterB-voice-ivr-g729r8] set-media 10002 file cfa0:/wav/g729r8/timeout.wav [RouterB-voice-ivr-g729r8] set-media 10003 file cfa0:/wav/g729r8/input_error.wav [RouterB-voice-ivr-g729r8] quit...
Page 312 Figure 95 Network diagram Configuration procedure Configure Router A: # Configure POTS voice entity 100. <RouterA> system-view [RouterA] voice-setup [RouterA-voice] dial-program [RouterA-voice-dial] entity 100 pots [RouterA-voice-dial-entity100] match-template 100 [RouterA-voice-dial-entity100] line 1/0 [RouterA-voice-dial-entity100] quit # Configure VoIP voice entity 300 to Router B. [RouterA-voice-dial] entity 300 voip [RouterA-voice-dial-entity300] match-template 300 [RouterA-voice-dial-entity300] address sip ip 1.1.1.2...
Page 313: Service Node Configuration Example 1
[RouterB-voice-ivr-g729r8] quit # Configure global error processing and timeout processing methods: If the timeout value expires before the subscriber dials at Telephone A, Router B plays the audio file timeout.wav. If the timeout value expires for four times, Router B terminates the call.
Page 314 [RouterA-voice-dial-entity100] line 1/0 [RouterA-voice-dial-entity100] quit # Configure VoIP voice entity 300 to Router B. [RouterA-voice-dial] entity 300 voip [RouterA-voice-dial-entity300] match-template 300 [RouterA-voice-dial-entity300] address sip ip 1.1.1.2 [RouterA-voice-dial-entity300] outband sip Configure Router B: # Configure POTS voice entity 500. <RouterB> system-view [RouterB] voice-setup [RouterB-voice] dial-program [RouterB-voice-dial] entity 500 pots...
Page 315: Service Node Configuration Example 2
Verify configurations: If the subscriber dials the number 300 at Telephone A, Telephone B will ring. The subscriber at Telephone A will not hear the voice prompts of the audio file welcome.wav. Service node configuration example 2 Network requirements Figure 97, configure the IVR access number and customize Service node functions on As shown in Router B.
Page 316: Configuration Example For Three Types Of Nodes
# Configure IVR voice entity 300 and specify node 10 as the root node. [RouterB-voice-dial] entity 300 ivr [RouterB-voice-dial-entity300] match-template 300 [RouterB-voice-dial-entity300] ivr-root 10 [RouterB-voice-dial-entity300] quit [RouterB-voice-dial] quit # Specify media resource IDs for media resource files: Specify 10002 for the file cfa0:/wav/g729r8/timeout.wav Specify 10003 for the file cfa0:/wav/g729r8/input_error.wav Specify 10004 for the file cfa0:/wav/g729r8/bye.wav [RouterB-voice] ivr-system...
Page 317 Figure 98 Network diagram Configuration procedure Configure Router A: # Configure POTS voice entity 100. <RouterA> system-view [RouterA] voice-setup [RouterA-voice] dial-program [RouterA-voice-dial] entity 100 pots [RouterA-voice-dial-entity100] match-template 100 [RouterA-voice-dial-entity100] line 1/0 [RouterA-voice-dial-entity100] quit # Configure VoIP voice entity 300 to Router B. [RouterA-voice-dial] entity 300 voip [RouterA-voice-dial-entity300] match-template 300 [RouterA-voice-dial-entity300] address sip ip 1.1.1.2...
Page 318 [RouterB-voice-ivr-g729r8] set-media 10001 file cfa0:/wav/g729r8/welcome.wav [RouterB-voice-ivr-g729r8] set-media 10002 file cfa0:/wav/g729r8/timeout.wav [RouterB-voice-ivr-g729r8] set-media 10003 file cfa0:/wav/g729r8/input_error.wav [RouterB-voice-ivr-g729r8] set-media 10004 file cfa0:/wav/g729r8/bye.wav [RouterB-voice-ivr-g729r8] set-media 10005 file cfa0:/wav/g729r8/call.wav [RouterB-voice-ivr-g729r8] quit # Configure global error processing and timeout processing methods: If the timeout value expires before the subscriber dials at Telephone A, Router B plays the audio file timeout.wav.
Page 319: Troubleshooting Ivr
Troubleshooting IVR Invalid node Symptom The subscriber dials the IVR access number 300 and the voice prompt is played. According to the voice prompt, the subscriber presses the # key to originate the secondary call, but the call is terminated. Following are the configurations: [Sysname-voice-dial] entity 300 ivr [Sysname-voice-dial-entity300] match-template 300 [Sysname-voice-dial-entity300] ivr-root 1...
Page 320: Node Depth Exceeds Eight Levels
[Sysname-voice-ivr-node11] operation 1 goto-node 12 [Sysname-voice-ivr-node11] quit [Sysname-voice-ivr] node 12 service [Sysname-voice-ivr-node12] operation 1 goto-node 10 Analysis After the # key is pressed, the IVR process jumps to node 10, and then jumps to node 11, node 12, and jumps back to node 10 again. Node 10 is the loopback node that causes the termination of the call.
Page 321: Matching Mistake 1
[Sysname-voice-ivr] node 9 call [Sysname-voice-ivr-node9] extension 1 call 300 [Sysname-voice-ivr-node9] quit Analysis When the IVR process jumps to node 8, it has already jumped for more than eight nodes, the eight-level limit is reached, and the call is terminated. Solution Avoid the configuration of making the IVR process jump from node to node for more than eight times.
Page 322 need to change the extension number 7745231 to *7, and do not configure the matching terminator of the normal secondary call as *.
Page 323: Configuring Vofr
Configuring VoFR Overview Voice over frame relay (VoFR) enables a router to transmit voice and voice-band data (for example, fax data and analog data from a modem) over a frame relay network. When voice traffic is sent over frame relay, it is segmented and encapsulated for transmission across a frame relay network. VoFR supports permanent virtual circuit (PVC) statistical multiplexing to carry multiple channels of voice, data, and fax over one PVC.
Page 324: Call Flow In Dynamic Mode
Figure 100 Protocols and standards that VoFR complies with Call flow in dynamic mode The following shows the call flow in the dynamic mode: The calling party picks up the phone. The voice interface card detects the off-hook action, plays dial tones to the calling party, and waits for the calling party to dial a number.
Page 325: Configuring Vofr Entity
Upon the completion of dialing, the voice gateway matches the called number against voice entities. VoFR processes the call if a VoFR entity is matched. The processing method depends on the call mode configured for the VoFR entity. In the FRF.11 trunk mode, the originating voice gateway directly goes into the conversation state and sends voice packets to the terminating voice gateway over the voice channel in the FRF.11 trunk.
Page 326: Configuring Basic Functions
Step Command Remarks Create a VoFR entity and entity entity-number vofr enter VoFR entity view. Configure a match template By default, no match template is match-template match-string for the VoFR entity. configured for the VoFR entity. Optional. Configure an area ID. area-id string By default, no area ID is configured.
Page 327: Configuring Vofr Voice Bandwidth
Step Command Remarks Create a VoFR entity and entity entity-number vofr enter VoFR entity view. Configure a match template By default, no match template is match-template match-string for the VoFR entity. configured for the VoFR entity. Configure the out-of-band By default, the inband DTMF outband vofr DTMF transmission mode.
Page 328: Configuring Dynamic Mode
Step Command Remarks Create a frame relay class and enter frame relay class fr class class-name view. By default, no Reserve a bandwidth for voice bandwidth reserved-bps [ reserved ] bandwidth is reserved voice. for voice. Exit frame relay class view. quit Enter frame relay interface interface serial interface-number...
Page 329 Configuring Huawei-compatible mode Configuration prerequisites Before configuring Huawei-compatible VoFR, complete the following tasks: • Configure POTS entities. • Configure VoFR entities. Configuring a call mode Step Command Remarks Enter system view. system-view Enter voice view. voice-setup Enter voice dial program dial-program view.
Page 330 Configuring VoFR packets to carry a timestamp Step Command Remarks Enter system view. system-view Enter voice view. voice-setup Enter voice dial program dial-program view. Enter VoFR entity view. entity entity-number vofr Configure a match template By default, no match template is match-template match-string for the VoFR entity.
Page 331: Configuring Call Mode
However, the use of sequence numbers increases the required network bandwidth. Therefore, determine whether to use sequence numbers based on actual conditions. To configure VoFR packets to carry a sequence number: Step Command Remarks Enter system view. system-view Enter voice view. voice-setup Enter voice dial program dial-program...
Page 332: Configuring Call Control Protocol
Step Command Remarks Enter voice view. voice-setup Enter voice dial program dial-program view. Enter VoFR entity view. entity entity-number vofr Configure a PSTN-dialed By default, no PSTN-dialed number in the FRF.11 trunk trunk-id string number is configured in the mode. FRF.11 trunk mode.
Page 333: Configuring Vofr Packets To Carry Sequence Number
Step Command Remarks Enter voice view. voice-setup Configure the trunk wait Optional. timer length in the FRF.11 vofr frf11-timer time The default is 30 seconds. trunk mode. NOTE: All FRF.11 trunks use the same trunk wait timer length. Configuring VoFR packets to carry sequence number "Configuring VoFR packets to carry a sequence number."...
Page 334 # Enter interface Serial 2/0 view and configure the encapsulation format and interface type. [RouterA] interface serial 2/0 [RouterA-Serial2/0] link-protocol fr ietf [RouterA-Serial2/0] fr interface-type dce # Enter DLCI 100 view and set the frame relay class to VoFR for DLCI. [RouterA-Serial2/0] fr dlci 100 [RouterA-fr-dlci-Serial2/0-100] fr-class vofr # Specify the call control protocol to be used on DLCI 100 as Huawei-compatible (DCE).
Page 335 [RouterB-voice-dial-entity2001] match-template 07552001 [RouterB-voice-dial-entity2001] line 2/0 Nonstandard-compatible VoFR Network requirements Figure 102, Telephone A (010-1001) attached to voice Router A in City A communicates As shown in with Telephone B (0755-2001), which is attached to voice Router B in City B over a frame relay network.
Page 336 [RouterA-voice-dial-entity755] entity 1001 pots [RouterA-voice-dial-entity1001] match-template 0101001 [RouterA-voice-dial-entity1001] line 3/0 Figure 102. Configure Router B according to the network requirements and network diagram in FRF.11 trunk Network requirements Figure 103, Telephone A (010-1001) attached to voice Router A in City A communicates As shown in with Telephone B (0755-2001), which is attached to voice Router B in City B over a frame relay network.
Page 337: Concurrent Transmission Of Voice And Data
[RouterA-voice-dial] entity 9 vofr [RouterA-voice-dial-entity9] match-template 9 [RouterA-voice-dial-entity9] call-mode static [RouterA-voice-dial-entity9] address vofr-static serial 2/0 100 6 [RouterA-voice-dial-entity9] trunk-id 0101001 [RouterA-voice-dial-entity9] quit # Configure the local POTS entity (0101001). [RouterA-voice-dial] entity 1001 pots [RouterA-voice-dial-entity1001] match-template 0101001 [RouterA-voice-dial-entity1001] line 3/0 Configure Router B: # Create a new frame relay class VoFR and set the maximum amount of voice bandwidth.
Page 338 The IP address of the interface Serial 1/0 on Router A is 1.1.1.1/24, and that of the interface Serial 1/0 on Router B is 2.2.2.2/24. Figure 104 Network diagram Configuration procedure Configure Router A: # Create a new frame relay class VoFR. <RouterA>...
Page 339 [RouterA-voice-dial] entity 1001 pots [RouterA-voice-dial-entity1001] match-template 0101001 [RouterA-voice-dial-entity1001] line 2/0 Configure Router B: # Create a new frame relay class VoFR. <RouterB> system-view [RouterB] fr class vofr # Configure the voice bandwidth. [RouterB-fr-class-vofr] voice bandwidth 32000 reserved # Configure the fragment size of data packets in the case that the voice function is disabled. [RouterB-fr-class-vofr] fragment 200 data-level # Configure the fragment size of data packets in the case that the voice function is enabled.
Page 340: Troubleshooting Vofr
Troubleshooting VoFR Call failure in Huawei-compatible mode Symptom Calls cannot be connected in Huawei-compatible mode. Analysis In the Huawei-compatible mode, calls can be connected only when the frame relay is normal, voice entities are configured correctly, and a sufficient bandwidth is reserved for voice. Solution •...
Page 341 Configuring voice RADIUS Remote authentication dial-in user service (RADIUS) is a protocol standard developed for implementing authentication, authorization and accounting (AAA) for access users, who can be PPP users or voice users. The voice RADIUS function provided by the voice gateway (RADIUS client) is suitable for small- and medium-sized network operators or enterprises to control voice calls and perform voice call accounting statistics.
Page 342: Radius Provided By Voice Gateway
the originating gateway, the terminating gateway sends a VoIP_Accounting_Start request (call segment 3) to the RADIUS server. After receiving a VoIP_Accounting_Start acknowledgment (call segment 3) from the RADIUS server, the terminating gateway originates a call to the called party over PSTN. After receiving an Alert from PSTN, the terminating gateway sends a Notification to the originating gateway and the originating gateway then plays ringback tones to the calling party.
Page 343 • stop-only—The voice gateway sends an Accounting_Stop request to the RADIUS server only when the call ends, and it releases the call only after receiving an Accounting_Stop acknowledgment. Voice dialing process There are two types of dialing process for voice calls: one-stage dialing and two-stage dialing. Two-stage dialing consists of caller number process, caller number process with IVR, and card number/password process.
Page 344: Voice Radius Configuration Task List
Recording and querying detailed voice call information This function records detailed information of each voice call. Use the cdr command to set the lifetime and number of records. The following call information is recorded: • Calling number. • Called number. •...
Page 345 • Configure access numbers. • Configure a two-stage dialing process. • Enable the AAA functions for two-stage dialing users. • Configure the method of collecting digits of called numbers. • Configure the number of digits in a card number and that in a password. •...
Page 346: Configuring Accounting Method
Task Remarks Enabling the authorization function for two-stage dialing users Optional. Configuring the method of collecting the digits of called number Optional. Configuring the timeout interval between two digits for two-stage dialing users Optional. Configuring the number of digits in a card number/password Optional.
Page 347 • If you want to prohibit users from making calls when their account balance in the RADIUS server which is a comprehensive access management server (CAMS) is 0, and allow them to make calls when there is sufficient account balance, configure the authentication, authorization, and accounting functions on the voice gateway.
Page 348: Configuring Rule For Saving Cdrs
Before enabling the authorization function, you must make sure that the authentication function is enabled on the RADIUS client, that the RADIUS server and RADIUS client can communicate with each other at the network layer, and that a list of one-stage dialing users as well as authorization policies has been configured on the RADIUS server.
Page 349 Step Command Remarks Enter system view. system-view Enter voice view. voice-setup Enter voice AAA client aaa-client view. The default is 50 for the size-number cdr { buffer size-number | Configure a rule for argument, 86,400 seconds (namely, 24 duration timer-length | saving CDRs.
Page 350 Configuration procedure To configure a two-stage dialing process: Step Command Remarks Enter system view. system-view Enter voice view. voice-setup Enter dial program view. dial-program gw-access-number Enter access number view. access-number Configure a two-stage process-config { callernumber | Caller number process with IVR dialing process.
Page 351 Configuration prerequisites You have configured an access number. Configuration procedure To enable the authentication function for two-stage dialing users: Step Command Remarks Enter system view. system-view Enter voice view. voice-setup Enter dial program view. dial-program gw-access-number Enter access number view. access-number Enable the authentication function for authentication...
Page 352 Step Command Remarks gw-access-number Enter access number view. access-number Enable the authorization function for two-stage dialing authorization Disabled by default. users. Configuring the method of collecting the digits of called number Use the callednumber receive-method command to configure the device to originate a call immediately after all digits of a called number are collected or after a dial terminator # is collected.
Page 353 Step Command Remarks Enter voice view. voice-setup Enter dial program view. dial-program Enter access number view. gw-access-number access-number Configure the timeout interval The default is 10 for a user to dial the next digit in timer two-stage dial-interval seconds seconds. a two-stage dialing process.
Page 354 Configuring the number of redial attempts The redialtimes command applies only to the card number/password process and the caller number process with IVR. This command is unavailable in the case of the caller number process. For the card number/password process, a user first dials an access number, then selects a language option, next enters a prepaid card number and password, and finally dials a called number.
Page 355: Displaying And Maintaining Voice Radius
Step Command Remarks Enter system view. system-view Enter voice view. voice-setup Enter dial program view. dial-program gw-access-number Enter access number view. access-number Specify the two-stage dialing process as caller number process-config voice-caller process with IVR. Chinese by default. After a user dials the access number, the Configure the language selectlanguage { enable |...
Page 356 When making an IP call, users first dial the access number 12345, then select a language option and enter a card number and password as prompted, and finally dial the called number if the card number/password authentication succeeds. Figure 106 Network diagram Configuration procedure The configuration procedure does not cover the configurations of route, voice subscriber line, or called number on the routers.
Page 357 [RouterA-voice-dial] gw-access-number 12345 [RouterA-voice-dial-anum12345] process-config cardnumber # Configure the number of digits in a card number and password. [RouterA-voice-dial-anum12345] card-digit 10 [RouterA-voice-dial-anum12345] password-digit 4 # Enable the authentication function. [RouterA-voice-dial-anum12345] authentication # Enable the authorization function. [RouterA-voice-dial-anum12345] authorization # Enable the accounting function. [RouterA-voice-dial-anum12345] accounting # Set the number of redial attempts to 3.
Page 358: Troubleshooting Voice Radius
[RouterB-voice-dial-anum12345] authorization # Enable the accounting function. [RouterB-voice-dial-anum12345] accounting # Set the number of redial attempts to 3. [RouterB-voice-dial-anum12345] redialtimes 3 NOTE: • The RADIUS scheme for voice RADIUS must be applied to the default domain named system. • If a voice call is originated from an IP network to the voice gateway, the voice gateway does not support the dialing of access numbers.
Page 359: Command Conventions
Document conventions and icons Conventions This section describes the conventions used in the documentation. Port numbering in examples The port numbers in this document are for illustration only and might be unavailable on your device. Command conventions Convention Description Boldface Bold text represents commands and keywords that you enter literally as shown.
Page 360 Network topology icons Convention Description 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.
Page 361: Support And Other Resources
Hewlett Packard Enterprise Support Center More Information on Access to Support Materials page: www.hpe.com/support/AccessToSupportMaterials IMPORTANT: Access to some updates might require product entitlement when accessed through the Hewlett Packard Enterprise Support Center. You must have an HP Passport set up with relevant entitlements.
Page 362: Documentation Feedback
Websites Website Link Networking websites Hewlett Packard Enterprise Information Library for www.hpe.com/networking/resourcefinder Networking Hewlett Packard Enterprise Networking website www.hpe.com/info/networking Hewlett Packard Enterprise My Networking website www.hpe.com/networking/support Hewlett Packard Enterprise My Networking Portal www.hpe.com/networking/mynetworking Hewlett Packard Enterprise Networking Warranty www.hpe.com/networking/warranty General websites Hewlett Packard Enterprise Information Library www.hpe.com/info/enterprise/docs Hewlett Packard Enterprise Support Center...
Page 363 part number, edition, and publication date located on the front cover of the document. For online help content, include the product name, product version, help edition, and publication date located on the legal notices page.
Page 364 Index H.323, architecture (fundamental VoFR), AAA function, area prefix AAA voice calls, configuration example, access number configuring, configuring, SIP local survival configuration, two-stage dialing, associating accounting function, call watch group, enabling for one-stage dialing users, authentication enabling for two-stage dialing users, SIP trunk credentials, accounting method (configuring), SIP UA registration,...
Page 365 call barring, call transfer, call control call transfer configuration, 236, dial plan, 100, call waiting, max-call-connection, call waiting configuration, 230, SIP local survival configuration, calling party control, call control protocol calling party control configuration, door opening control, configuring (FRF.11 trunk), configuring (Huawei-compatible), door opening control configuration, configuring (nonstandard-compatible),...
Page 366 caller number (two-stage dialing process), configuring local SIP server to operate in alone mode, caller number with IVR (two-stage dialing process), configuring out-of-band SIP DTMF transmission mode, caller privacy configuring proxy server involved calling for SIP INVITE message, UAs, Remote-Party-ID, configuring SIP extensions for caller identity and calling party control, privacy,...
Page 367 cable type, dial terminator, call authority control, 181, digital E&M signaling, call backup, digital E&M signaling start mode, call control protocol (FRF.11 trunk), digital voice subscriber line, 60, call control protocol (Huawei-compatible), direct calling for SIP UA, call control protocol DNS involved calling for SIP UA, (nonstandard-compatible), domain name for SIP UA,...
Page 368 gain adjustment function for voice subscriber options related to dial plan for subscriber line line, (digital), gateway carrier transmit energy level options related to dial plan for VoIP, (FoIP), outbound SIP proxy server, global default parameter, outbound SIP proxy server information for SIP global default parameters for voice subscriber line, outgoing call barring,...
Page 369 SIP compatibility for fax/modem TLS (SIP), pass-through, TLS for SIP session, SIP compatibility for modem transport layer protocol for SIP calls, pass-through, trunk timer length in FRF.11 trunk mode, SIP extensions, two-stage dialing process, SIP extensions for caller identity and UDP for outgoing SIP calls, privacy, user-agent and server header fields,...
Page 370 flexible node configuration, number match mode, hardware compatibility, number match mode configuration, 105, jump node configuration, 291, number priority peer configuration, media resource ID specification, number sending mode, node creation, number substitution, 100, 102, 116, 120, processing method configuration, number substitution regular expression, secondary call type, operation process, service node configuration,...
Page 371 displaying voice RADIUS, subscriber line, DO-EO conversion E&M start mode SIP trunk configuration, delay start, DO-EO conversion (enabling), immediate start, domain name wink start, configuring for SIP UA, SIP routing, binding FXS voice subscriber line to POTS entity, door opening control, configuring basic function for voice subscriber DTMF line,...
Page 372 signaling mode, silence detection-based automatic on-hook, TDM clock source configuration, SIP trunk function, TS set creation, trap function, voice DSS1 signaling configuration, VAD for POTS, voice function, VAD for VoIP, voice subscriber line configuration for TS enabling VAD, set, encryption E1 interface media flow, call watch,...
Page 373 fax training mode configuration, gateway carrier transmit energy level analog voice subscriber line configuration, 29, configuration, binding FXS voice subscriber line to FXO voice global default parameterconfiguration, subscriber line, 42, hardware compatibility, CID, local training threshold configuration, interface, maintaining, signal tone, maximum fax rate configuration, voice subscriber line, protocols and standards,...
Page 374 call service, configuration, call watch, customizable IVR configuration, customizable IVR, IVR service node, dial plan, configuration, digital voice subscriber line, configuration (call termination), FoIP, configuration (secondary call), H.323, IVR voice entity SIP, configuration, SIP local survival, creation, SIP trunk, VoFR, join voice, SIP INVITE message,...
Page 375 PSTN SIP, out-of-band SIP DTMF transmission (configuring), max-call-connection set out-of-band transmission (configuring for SIP dial plan, DTMF), media flow encryption (security), parking, media flow protocol parking backup, configuring for SIP calls, REGISTER, RTCP, modem pass-through RTP, SIP compatibility, SIP security configuration, modem pass-through (configuring SIP media flow-around compatibility),...
Page 376 enabling authorization function for users, enabling fast connection and tunneling on originating GW, voice dialing process, enabling fast connection and tunneling on operation mode terminating GW, alive, enabling FXO monitoring, alone, enabling local call identification, configuring for local SIP server, enabling trap function, OPTIONS (keepalive mode), enabling VAD,...
Page 377 configuring basic function, configuring CNG fax switchover (FoIP), configuring basic function for POTS voice configuring comfortable noise function entity, (analog), configuring basic function for voice subscriber configuring comfortable noise function (digital), line, configuring delay time before on-hook, configuring basic function for VoIP voice configuring destination number/address obtain entity, method,...
Page 378 configuring global default parameters for voice configuring PSTN-dialed number, subscriber line, configuring redundancy function (SIP trunk), configuring global number match mode, configuring registration timers, configuring H.323 registration password, configuring registration timers for SIP trunk configuring H.323 security calling, account, configuring how to obtain destination number configuring reverse charging function (analog), and address, configuring sending of LCFO signals,...
Page 379 configuring timeout interval for RTP enabling fast connection and tunneling on stream, originating GW, configuring TLS for SIP session, enabling fast connection and tunneling on terminating GW, configuring trunk timer length in FRF.11 trunk mode, enabling FXO monitoring, configuring user-agent and server header enabling local call identification, fields, enabling media flow-around, 207,...
Page 380 troubleshooting point-to-point SIP call configuring digital line signaling, failure, configuring interregister signaling, troubleshooting SIP call in proxy server enabling DTMF mode to receive and send R2 method, signaling, troubleshooting SIP UA register request enabling terminating side to send busy tones to sending failure, calling subscriber, troubleshooting SIP UA registration...
Page 381 registration password Session Initiation Protocol. Use H.323 gateway configuration, session management registration timer SIP, configuring, session refresh configuring for SIP trunk account, BYE message, regular expression INVITE message, SIP, matching pattern, matching pattern metacharacter, UPDATE message, request session setup SIP client, SIP, re-registration (configuring call setting...
Page 382 displaying local survival, troubleshooting, 174, displaying UAs, troubleshooting gateway registration failure, extended functions, troubleshooting point-to-point call failure, features, troubleshooting SIP call in proxy server method, functions, 131, troubleshooting TLS-based point-to-point SIP call fundamentals, 132, failure, fuzzy telephone number registration, troubleshooting UA register request sending hardware compatibility, failure, local server operation mode,...
Page 383 parking backup, SIP extension SIP server group configuring, binding to VoIP voice entity, configuring for caller identity and privacy, configuring, SIP configuration, creating, SIP functions SIP server keepalive multimedia, options keepalive, SIP fundamental register keepalive, call redirection, SIP session call setup, early media negotiation, registration, periodic refresh,...
Page 384 SIP trunk account subscriber group control configuring, calling side, configuring for registration, dial plan, configuring registration timers, subscriber group configuration, credentials, subscriber group control binding SIP trunk applications calling side, IP-PBX, subscriber group control template PSTN, calling side, SIP trunk features subscriber line IP-PBX, private line auto ring-down,...
Page 385 restoring default settings for E1 voice failed to send register requests, interface, failed to set up calls in the proxy server approach setting length of called numbers that can be to SIP routing, received, failed to set up point-to-point calls, setting physical state change suppression failed to set up point-to-point SIP calls over interval on T1 interface,...
Page 386 caller number with IVR, configuring dynamic mode, card number/password, configuring entity, configuring FRF.11 trunk mode, configuring Huawei-compatible mode, UA. See configuring nonstandard-compatible mode, location server, configuring packets to carry sequence proxy, number, redirect, configuring PSTN-dialed number, SIP call redirection, configuring trunk timer length (FRF.11 trunk SIP call setup, mode), SIP registration,...
Page 387 call hold tone playing mode, configuring off-hook mode, call node configuration, configuring options related to dial plan for POTS, call node configuration (dial terminator match), configuring options related to dial plan for subscriber line (analog), call node configuration (extension secondary call), configuring options related to dial plan for subscriber line (digital),...
Page 388 enabling busy tone sending, H.323 call setup, enabling DTMF mode to receive and send R2 H.323 capability negotiation, signaling, H.323 complete release, enabling E&M analog control signals H.323 configuration, 216, pass-through, H.323 connection, enabling E&M non-signaling mode, H.323 disconnection, enabling fast connection and tunneling on H.323 fundamental, originating GW, H.323 gatekeeper discovery,...
Page 389 overview, SIP server backup, packet loss compensation mode SIP server keepalive, configuration, SIP support for basic QSIG call, PCM pass-through function configuration SIP support for transport layer protocol, (analog), SIP terminology, POTS voice entity configuration, SIP transport layer protocol configuration, PRI configuration, SIP trunk, 195, private line auto ring-down,...
Page 390 troubleshooting TLS-based point-to-point SIP creating VoIP voice entity, call failure, customizable IVR configuration, troubleshooting voice entity, dial plan process (called side), troubleshooting voice subscriber line, dial plan process (calling side), trusted node, dial prefix, user agent, displaying, VoFR, DTMF transmission configuration for POTS, VoFR configuration, DTMF transmission configuration for VoIP, voice entity,...
Page 391 dial plan, troubleshooting, voice gateway voice gateway, 328, access numbers, voice setup accounting, dial plan, 105, 116, accounting methods, dial plan configuration, CDRs, dial prefix, fails to connect calls (troubleshooting), DNS involved calling configuration (SIP), RADIUS provided by, number sending mode, voice RADIUS, proxy server involved calling configuration, voice over frame relay.
Page 392 enabling VAD, fast connection and tunneling configuration, fax faculty transmission mode configuration (FoIP), fax interworking protocol configuration (FoIP), features, FoIP, FoIP configuration, 271, FoIP fax training mode configuration, FoIP global default parameter configuration, FoIP maximum fax rate configuration, gateway carrier transmit energy level configuration (FoIP), jitter buffer configuration, local training threshold configuration...

HP FlexNetwork MSR Series Configuration Manual

Voice Overview

Configuring Voice Entities

Configuring Analog Voice Subscriber Lines

Configuring Digital Voice Subscriber Lines

Configuring Dial Plans

Configuring SIP

Configuring SIP Local Survival

Configuring SIP Trunk

Quick Links

Troubleshooting

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Summary of Contents for HP FlexNetwork MSR Series