HP MSR Router Series Voice Configuration Guide(V5) Part number: 5998-2029 Software version: CMW520-R2511 Document version: 6PW103-20140128
Legal and notice information © Copyright 2014 Hewlett-Packard Development Company, L.P. No part of this documentation may be reproduced or transmitted in any form or by any means without prior written consent of Hewlett-Packard Development Company, L.P. The information contained herein is subject to change without notice.
Contents Voice overview ····························································································································································· 1 Introduction to VoIP ··························································································································································· 1 VoIP system ·········································································································································
Voice entity configuration example for establishing a VoIP call ······································································ 26 Fast connection ······················································································································································ 28 Troubleshooting voice entity configuration ·················································································································· 29 Busy tone heard immediately after number di
Configuring DTMF detection ································································································································ 49 Configuring options related to dial plan ····················································································································· 50 Configuring adjustment functions ································································································································· 50 Configuration task list ·
Enabling the transmission of QSIG signaling over a SIP network ··································································· 82 Configuring digital E&M signaling ······························································································································ 83 Configuring a start mode ····································································································································· 83 Enabling E&M non-signaling mode ····················
Configuration procedure ···································································································································· 115 Configuring number substitution ································································································································· 115 Configuration prerequisites ································································································································ 115 Configuration procedure
Specifying the URL scheme for outgoing SIP calls ···································································································· 152 Configuring SIP extensions·········································································································································· 153 Strict SIP routing··················································································································································· 153 Configuring out-of-
SIP trunk configuration task list ··································································································································· 198 Enabling the SIP trunk function ··································································································································· 199 Configuring a SIP server group ·································································································································· 199 Creating
Configuring call services ········································································································································ 228 Call waiting·························································································································································· 228 Call hold ······························································································································································· 228
Configuring three-party conference ··························································································································· 243 Configuration prerequisites ································································································································ 243 Enabling three-party conference by using keys································································································ 243 Enabling three-party conference by using c
Displaying and maintaining FoIP configuration········································································································ 283 FoIP configuration examples ······································································································································· 283 Configuring FoIP ·················································································································································· 283 Configuring SIP modem
Configuring VoFR voice bandwidth ··························································································································· 316 Configuring dynamic mode ········································································································································ 317 Configuring Huawei-compatible mode ············································································································· 317 Configuring nonstandard-compatible m
Websites······························································································································································· 349 Conventions ·································································································································································· 350 Index ··························································································································································
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. VoIP meets the commercial requirements for PC-to-telephone, telephone-to-PC, and telephone-to-telephone technologies. H.
4. The user hears dial tones played by the session application and begins dialing before the dial tone timer expires. 5. The session application collects the digits dialed by the user. 6. The session application compares the collected digits with the match template while collecting digits. 7. After finding a match template for the called number, the originating VoIP gateway maps the number to the terminating VoIP gateway. 8.
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 subscriber line Voice entity Voice protocol Dial plan Configuration procedure Figure 3 shows the voice function configuration procedure of the router. For more information, see Table 1.
Figure 3 Voice function configuration procedure Start Configure a link connection Is the link available? No Yes Configure voice entity Is number substitution necessary? Yes Configure number substitution for dial plans No Configure voice subscriber line Configure number application for dial plans Configure voice protocol Troubleshoot No Is the call established? Yes End Table 1 Description of the voice function configuration procedure Operation Reference 1.
Operation Reference 6. Configure number application for the dial plan adopted in the network diagram. 7. Configure the following voice protocols according to the service and networking environment: { H.323 protocol { SIP protocol { Fax protocol Configuring dial plans • Configuring H.323 • Configuring SIP • Configuring fax over IP Check whether the network requirements are met: 8. { If so, the configuration is completed. { If not, check the fault and perform re-configuration.
• A POTS entity corresponds to the local telephone (or PSTN) side. POTS entity configuration associates a voice subscriber line on the VoIP gateway with a local telephone. The POTS entity configuration also implements the binding between telephone numbers and voice subscriber lines. • 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.
messages. The originating SIP endpoints directly re-originate a session request message to the terminating SIP endpoints. The terminating SIP endpoints also directly return a reply message to the originating SIP endpoints. As a SIP endpoint, the voice router needs to exchange information with the servers to accomplish functions such as registration. For more information, see "Configuring SIP." 3. Fax protocol FoIP complies with ITU-T T.30 and T.4 on PSTN and T.38 on the IP network. T.
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.
Task Remarks Configuring voice call performance-related parameters Optional. Configuring global default parameters for voice entities Optional. Enabling the trap function Optional. Configuring a POTS voice entity This section covers the procedures for creating and configuring a POTS voice entity. POTS voice entity configuration task list Task Remarks Creating a POTS voice entity Required. Configuring basic functions Required.
Step Command Remarks 14. Bind the local voice subscriber line to the POTS voice entity. line line-number By default, no voice subscriber line is bound to the POTS voice entity. Optional. 15. Enable the VoIP gateway to register numbers of the POTS voice entity with the H.323 gatekeeper or SIP server. register-number 16. Set the DSCP value in the ToS field in the IP packets that carry the RTP stream of the voice entity.
Step Command Remarks Optional. 5. Specify the codecs and their priority levels for the POTS voice entity. compression { 1st-level | 2nd-level | 3rd-level | 4th-level } { g711alaw | g711ulaw | g723r53 | g723r63 | g726r16 | g726r24 | g726r32 | g726r40 | g729a | g729br8 | g729r8 } By default, the codec with the first priority is g729r8, that with the second priority is g711alaw, that with the third priority is g711ulaw, and that with the fourth priority is g723r53.
Step Command Remarks Optional. Enable the local POTS voice entity to play ringback tones. 7. By default, the local POTS voice entity does not play ringback tones. send-ring This command is only applicable for POTS entities bound to a non-FXS or non-FXO voice subscriber lines.
Step 6. Command Configure the value of the payload type used by NTE. Remarks Optional. rtp payload-type nte value The default is 101. 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.
Step 8. Command Configure the number sending mode. send-number { digit-number | all | truncate } Remarks Optional. By default, the truncate mode is used, that is, numbers matching the wildcard dot (.) at the end will be sent. For more information about the above commands, see "Configuring dial plans." Configuring the jitter buffer Jitter, packet loss, and packet disorder occurs during the transmission of voice packets over IP networks.
Task Remarks Configuring out-of-band DTMF transmission in fast connection mode Optional. Configuring out-of-band DTMF transmission with tunneling enabled Optional. Enabling VAD Optional. Configuring options related to dial plan Optional. Configuring the jitter buffer Optional. Setting the keepalive interval Optional. Creating a VoIP voice entity Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter voice dial program view.
Step 4. Create a VoIP voice entity and enter VoIP voice entity view. Command Remarks entity entity-number voip N/A Optional. 5. Specify the codecs and their priority levels for the POTS voice entity.
Configuring fast connection and tunneling Fast connection and tunneling • According to the specification of H.225.0 recommendation, fast connection means that a Setup, CallProceeding, Alerting, or Connect message carries a standard H.245 message (for example, Open Logical Channel message) so that an RTP/RTCP voice channel can be established before the gateway (GW) receives a Connect message, avoiding H.245 message exchange on TCP connection, and thereby shortening connection time.
Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter voice dial program view. dial-program N/A 4. Configure the call initialization method for the terminating GW. voip called-start { fast | normal } Enable tunneling on the terminating GW. voip called-tunnel enable 5. Optional. Fast connection by default. Optional. Enabled by default. Configuring out-of-band DTMF transmission in fast connection mode Step Command Remarks 1.
Step Command Remarks 4. Enter VoIP voice entity view. entity entity-number voip N/A 5. Enable fast connection. fast-connect Disabled by default. Disabled by default. This command is applicable only after the fast connection is enabled. 6. Enable the tunneling. tunnel-on 7. Configure the out-of-band DTMF transmission. outband { h225 | h245 } Optional. Inband transmission by default. Enabling VAD Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view.
Configuring the jitter buffer Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter voice dial program view. dial-program N/A 4. Create a VoIP voice entity and enter VoIP voice entity view. entity entity-number voip N/A 5. Set the operating mode of the jitter buffer to adaptive. 6. Set the operating parameters for the jitter buffer that operates in the adaptive mode. Optional.
Step Command Remarks N/A 2. Enter voice view. voice-setup 3. Configure the timeout interval for RTP streams. rtp-detect timeout value Optional. By default, the timeout interval for RTP streams is 120 seconds. Enabling local call identification Introduction As shown in Figure 6, Telephone A originates a call to Telephone B through a SIP server.
Step 3. Command Enable local call identification. Remarks distinguish-localtalk Optional. Not enabled by default. 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 1. Enter system view. system-view N/A 2.
Step Command Remarks The default is the general version. 6. Configure the type of the DSP image. vi-card dsp-image { ms | general } 7. Configure the maximum number of call history records that can be stored. call-history max-count number The general version supports the G.723 codec, but it cannot meet voice quality requirements of Microsoft. The ms version can meet voice quality requirements of Microsoft, but it does not support the G.723 codec. Optional. 50 by default.
Step 4. Command Enable VAD globally as the default. default entity vad-on Remarks Optional. Not enabled by default. Optional. 5. Specify the default global codecs and their priority levels.
To enable the trap function globally: Step Command Remarks N/A 1. Enter system view. system-view 2. Enable the trap function globally. snmp-agent trap enable voice dial Optional. Disabled by default. To enable the trap function for an entity: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter voice dial program view. dial-program N/A 4. Create a POTS or VoIP voice entity and enter its view.
Task Command Remarks Display the statistics of the active calls. display voice statistics call-active { all | calling calling-number | called called-number } [ | { begin | exclude | include } regular-expression ] Available in any view. Display the call statistics of voice entities after the system starts up. display voice statistics entity { all | mark entity-index } [ | { begin | exclude | include } regular-expression ] Available in any view. Display the statistics of the IPP module.
Figure 6 Network diagram Configuration procedure Routing-related configurations are beyond the scope of this example. This example assumes that Router A and Router B are reachable to each other. 1. Configure Router A: # Configure the VoIP voice entity to Router B. system-view [RouterA] voice-setup [RouterA-voice] dial-program [RouterA-voice-dial] entity 0755 voip [RouterA-voice-dial-entity755] match-template 0755.... [RouterA-voice-dial-entity755] address sip ip 2.2.2.
# Configure the POTS voice entity corresponding to the local interface Line 2/1. [RouterB-voice-dial] entity 2002 pots [RouterB-voice-dial-entity1002] match-template 07552002 [RouterB-voice-dial-entity1002] line 2/1 Fast connection Network requirements As shown in Figure 7, phone users of Router A and Router B communicate with each other over WAN. The connection from Router A to Router B adopts fast connection and DTMF H.225 out-of-band transmission.
[RouterB-voice-dial] entity 010 voip [RouterB-voice-dial-entity10] match-template 010.... [RouterB-voice-dial-entity10] address ip 1.1.1.1 # Enable out-of-band DTMF transmission for the VoIP voice entity. [RouterB-voice-dial-entity10] outband h225 [RouterB-voice-dial-entity10] quit # Configure the local interface and phone number for Telephone B.
Configuring analog voice subscriber lines This chapter covers the configuration of analog FXS, FXO, and E&M voice subscriber lines. The HP MSR900 and MSR93X routers do not support voice functions. 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.
• O if the terminating PBX fails to obtain the calling name (for example, the originating PBX end does not send it) The FXS voice subscriber line sends the calling identity information to the called telephone. The calling identity information is sent to the called telephone through frequency shift keying (FSK) modulation between first and second rings.
Figure 8 Network diagram Perform an automatic busy tone detection test as follows: 1. 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. Telephone B rings. After Telephone B is picked up, the call is connected. 2. If you hang up Telephone A, PBX B plays a busy tone to Router A. 3. Use the vi-card busy-tone-detect command in voice view to start the detection.
side receives the digits, it rings to alert the called party of the call. If the called party picks up the phone, the call is connected. Figure 9 Immediate start mode • Delay start—As shown in Figure 10, the calling side goes off-hook to seize the trunk and the called side (PBX) goes off-hook to respond to the seizure. When the called side (PBX) is ready, it goes on-hook. The off-hook interval is the delay dial signal.
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. Configuring E&M voice subscriber line Optional. Configuring DTMF Optional. Configuring options related to dial plan Optional. Configuring adjustment functions Optional.
Customizing call progress tones for a country You can customize the call progress tones of a country if necessary. To customize call progress tones for a country: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Customize call progress tone parameters.
Step Command Optional. Set the expected bandwidth for the voice subscriber line. 3. Remarks bandwidth bandwidth-value Support for the command depends on your device model. Optional. 4. Configure a voice subscriber-line description. description string By default, the description string of a voice subscriber line is interface-name+interface. 5. Restore the default configuration of the voice subscriber line. default Optional. Tear down the voice subscriber line. shutdown 6. Optional.
Step 5. 6. 7. Command Configure the calling name for the FXS voice subscriber line. Remarks calling-name text Optional. Not configured by default. Optional. Configure the format of messages (which carry the calling number information) transmitted over the FXS voice subscriber line. cid type { complex | simple } When the peer end supports only one message format, you must use the same message format at the local end as the one used at the peer end.
Step Command Remarks Optional. 3. Enable the sending of pulse signal at hangup. disconnect lcfo 4. Configure the time duration for the sending of the pulse signal at hangup. timer disconnect-pulse milliseconds By default, the sending of pulse signals at hangup is disabled, and the system plays busy tones to the other end. Optional. 750 milliseconds by default. Configuring FXO voice subscriber line This section covers the prerequisites and procedures for configuring an FXO voice subscriber line.
Step Command Remarks Optional. 5. Configure the time for CID check and after the CID check, the number of rings the FXO line receives before going off-hook. cid ring { 0 | 1 | 2 } [ times ] By default, CID check is performed between the first and the second rings and the FXO line goes off-hook as soon as the check completes, that is, cid ring 1 0. Configuring busy tone detection This section describes how to configure busy tone detection for FXO voice subscriber lines.
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. To enable the busy tone sending: Step Command Remarks 1. Enter system view. system-view N/A 2.
Configuring the duration before a forced on-hook In some countries, PBXs do not play busy tones, or the busy tones played only last for a short period of time. When noise is present on a transmission link, the silence-th-span command might fail to release the FXO interface after on-hook. In this case, use the hookoff-time command to solve the problem. To configure the duration before a forced on-hook: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter FXO voice subscriber line view.
• Immediate mode—The calling party dials the number of the trunk. Upon receiving the call, the FXO goes off-hook and sends a dial tone to the calling party. Then, the calling party dials the destination number. • Delay mode—The number of the private line is configured in subscriber line view on the system. When the calling party dials the destination number, the call is routed to its destination based on the configured private line number.
Configuring other functions Step Command Remarks 1. Enter system view. system-view N/A 2. Enter FXO voice subscriber line view. subscriber-line line-number N/A 3. Set the electrical impedance. impedance { country-name | r550 | r600 | r650 | r700 | r750 | r800 | r850 | r900 | r950 } 4. Configure the packet loss compensation mode. plc-mode { general | specific } Optional. By default, the electrical impedance conforms to Chinese standards. Optional. The default is specific.
Step Command Remarks 3. Bind an FXS voice subscriber line to the FXO voice subscriber line. hookoff-mode delay bind fxs_subscriber_line [ ring-immediately ] By default, no FXS voice subscriber line is bound. 4. Enable the automatic dialing of the bound FXS voice subscriber line. private-line string Disabled by default. Configure an interval between on-hook and off-hook. timer hookoff-interval milliseconds 6. Exit FXO voice subscriber line view. quit N/A 7. Enter voice view.
Configuring signal type Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice subscriber line view. subscriber-line line-number N/A 3. Configure a signal type for the E&M voice subscriber line. type { 1 | 2 | 3 | 5 } The default is 5 (corresponding V type signal). Configuring start mode To configure the immediate start mode: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice subscriber line view. subscriber-line line-number N/A 3.
Step Command Remarks The default is immediate start mode. 3. Configure the wink start mode for the E&M voice subscriber line. em-signal wink 4. Configure the delay time from when the terminating side receives a seizure signal to when it sends a wink signal in the wink start mode. delay send-wink milliseconds 5. 6. Configure the maximum amount of time the originating side waits for a wink signal after sending a seizure signal in the wink start mode.
Figure 12 E&M analog control signals pass-through To enable E&M analog control signals pass-through: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice subscriber line view. subscriber-line line-number N/A 3. Enable E&M analog control signals pass-through. em-passthrough Optional. Disabled by default. NOTE: Configure this feature on the voice gateways of both sides.
Configuring DTMF Introduction to DTMF Dual tone multi-frequency (DTMF) uses a mixture of a high frequency tone and a lower frequency tone to represent a key on a keypad. Each column of keys is represented by a high frequency tone and each row of keys is represented by a low frequency tone. For example, as shown in Figure 13, the digit 1 is represented by the combination of a pure 697 Hz signal and a pure 1209 Hz signal. Such DTMF signals have good immunity to interference.
NOTE: The dtmf time and dtmf amplitude commands in voice view have global significance. Once you carry out either of the two commands, the configuration will take effect on the whole device. Configuring DTMF detection Introduction to DTMF Use the following ways to detect DTMF: • 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.
Configuring options related to dial plan Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice subscriber line view. subscriber-line line-number N/A 3. Enable the private line auto ring function. private-line string Bind a calling/called number substitution rule list to a voice subscriber line. substitute { called | calling } list-number 4. Optional. Disabled by default. By default, no number substitution rule list is bound to a voice subscriber line.
Table 3 Adjust echo cancellation parameters Symptom Parameters adjusted Effect A user hears echoes or loud background noises from the peer when speaking. Speed up the convergence of comfortable noise amplitudes Too fast convergence might make noises uncomfortable. There are loud environment noises. Increase the maximum amplitude of comfortable noises. Too large amplitude might make noises uncomfortable. A user hears echoes when speaking. Enlarge the control factor of mixed proportion of noises.
Configuring gain adjustment function To configure the gain adjustment function on a voice subscriber line: Step Command Remarks… 1. Enter system view. system-view N/A 2. Enter voice subscriber line view. subscriber-line line-number N/A 3. Set the input gain on the voice interface. receive gain value Set the output gain on the voice interface. transmit gain value 4. Optional. The default is 0 dB. Optional. The default is 0 dB. IMPORTANT: Gain adjustment might lead to call failures.
Step Command Remarks Optional. Configure the time range for the duration of an on-hook condition that will be detected as a hookflash. 8. timer hookflash-detect hookflash-range By default, the time range is 50 to 180 milliseconds, that is, if an on-hook condition that lasts for a period that falls within the hookflash duration range is considered a hookflash. Applicable only in analog FXS voice subscriber line view.
Step Command Remarks 1. Reset the voice card in user view. reboot slot slot-number Optional. 2. Enter system view. system-view N/A 3. Enter voice view. voice-setup N/A 4. Reboot a voice card. vi-card reboot slot-number Optional. Configuring global default parameters for voice subscriber lines Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Configure the default input/output gain for all subscriber lines.
Displaying and maintaining analog voice subscriber lines Task Command Remarks Display analog voice subscriber-line information. display voice subscriber-line [ line-number ] [ brief ] [ | { begin | exclude | include } regular-expression ] Available in any view. 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.
[RouterA-voice-dial] entity 1001 pots [RouterA-voice-dial-entity1001] match-template 0101001 [RouterA-voice-dial-entity1001] line 1/0 2. Configure Router B: 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.
Configuration outline • Configure one-to-one binding between FXS and FXO voice subscriber lines. • When the IP network is available, the VoIP entity is preferably used to make calls over the IP network. • When the IP network is unavailable, the POTS entity is used to make calls through the bound FXO voice subscriber line over the PSTN. Configuration procedure Routing-related configurations are beyond the scope of this example.
[RouterB] voice-setup [RouterB-voice] dial-program # Configure a VoIP entity for IP calls and set the match template to 010…. [RouterB-voice-dial] entity 010 voip [RouterB-voice-dial-entity10] match-template 010.... [RouterB-voice-dial-entity10] address sip ip 192.168.0.71 [RouterB-voice-dial-entity10] quit # Configure a POTS entity on the FXS voice subscriber line.
Troubleshooting analog voice subscriber line configuration Failed to hang up Symptom The FXO voice subscriber line cannot detect busy tone signals sent from the PBX, so the line is in connection even if the remote end hangs up. Figure 16 Network diagram As shown in Figure 16, suppose Telephone A hangs up first after conversation, then PBX A plays busy tones to Router A, which disconnects the line after detecting the busy tones and sends a disconnect message to Router B.
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. E1 and T1 use the same sampling frequency (8 kHz), PCM frame length (125 μs), bits per code word (8 bits) and timeslot bit rate (64 kbps).
E1 interface An E1 interface is logically divided into timeslots (TSs) with TS 16 being a signaling channel. On E1 interfaces, you can create PRI groups or TS sets. 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.
Signaling modes E1/T1 interfaces support these types of signaling: • DSS1/QSIG user signaling—Adopted on the D channel between ISDN user and network interface (UNI). It comprises a data link layer protocol and a Layer 3 protocol used for basic call control. • ITU-T R2 signaling—Includes digital line signaling and interregister signaling. Digital line signaling is transmitted in TS16 (ABCD bits) of E1 trunk.
Task Remarks 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 R2 signaling Configuring basic R2 signaling parameters Optional. Configuring R2 digital line signaling Optional. Configuring R2 interregister signaling Optional. Configuring PRI Required. Configuring digital E&M signaling Optional. Configuring digital LGS signaling Optional.
powered off and the clock mode of interface 6/0 is set to line primary, the clock mode of interface 5/0 will be restored to internal, instead of being line primary, if the FIC is re-inserted. NOTE: The line clock source and internal clock source are referred to as slave and master clock modes in some features. Configure a TDM clock source To configure a TDM clock source for E1 interfaces: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter E1 interface view.
Step Command Remarks Optional. 5. Configure an E1 trunk routing mode. select-mode { max | maxpoll | min | minpoll } By default, the timeslot with the lowest number is selected from all available timeslots for routing. Set the physical state change suppression interval on an E1 interface Step Command Remarks 1. Enter system view. system-view N/A 2. Enter E1 interface view. controller e1 slot-number N/A 3. Set the physical state change suppression interval on the E1 interface. Optional.
Step 3. Command Configure a TDM clock source for the T1 interface. tdm-clock { internal | line [ primary ] } Remarks Optional. By default, the internal clock is used as the TDM clock source. Configuring the framing format and line coding format Step Command Remarks 1. Enter system view. system-view N/A 2. Enter T1 interface view. controller t1 slot-number N/A 3. Configure the framing format. frame-format { esf | sf } 4. Configure the line coding format. code { ami | b8zs } Optional.
Step 3. Command Set a physical state change suppression interval on the T1 interface. Remarks Optional. link-delay delay-time By default, the physical state change suppression is disabled on a T1 interface. Restoring default settings for a T1 voice interface Step Command Remarks 1. Enter system view. system-view N/A 2. Enter T1 interface view. controller t1 slot-number N/A 3. Restore the default settings on the T1 voice interface. default Optional.
Step Command Remarks 4. Exit E1/T1 interface view. quit N/A 5. Enter voice subscriber line view. subscriber-line slot-number:ts-set-number N/A 6. Configure a companding law for signal quantization. 7. Create a description for the voice subscriber line. description text 8. Shut down the voice subscriber line. shutdown Optional. pcm { a-law | μ-law } A-law for a VE1 interface card and μ-law for a VT1 interface card by default. Optional.
Configuring the volume adjustment function To configure the volume adjustment function: Step Command Remarks… 1. Enter system view. system-view N/A 2. Enter E1/T1 interface view. controller { e1 | t1 } slot-number N/A 3. Create a TS set according to the selected signaling mode. timeslot-set ts-set-number timeslot-list timeslots-list signal { e&m-delay | e&m-immediate | e&m-wink | fxo-ground | fxo-loop | fxs-ground | fxs-loop | r2 } N/A 4. Exit E1/T1 interface view. quit N/A 5.
Step Command Remarks Optional. 8. Enable the nonlinear function of echo cancellation. Enabled by default. nlp-on This command takes effect only after the echo-canceller enable command is issued. Configuring the comfortable noise function Step Command Remarks 1. Enter system view. system-view N/A 2. Enter E1/T1 interface view. controller { e1 | t1 } slot-number N/A 3. Create a TS set according to the selected signaling mode.
Binding logical voice subscriber line to POTS entity Step Command Remarks 1. Enter system view. system-view N/A 2. Enter E1/T1 interface view. controller { e1 | t1 } slot-number N/A 3. Create a TS set according to the selected signaling mode. timeslot-set ts-set-number timeslot-list timeslots-list signal { e&m-delay | e&m-immediate | e&m-wink | fxo-ground | fxo-loop | fxs-ground | fxs-loop | r2 } N/A 4. Exit E1/T1 interface view. quit N/A 5. Enter voice view. voice-setup N/A 6.
off-hook and answers the call, calling party releases the call, and called party releases the call. Accordingly, it sets the line to be idle or seized. This signaling is transmitted in the 16th multiframe timeslot of PCM system. The two transmission directions of each line have 4 bits (A, B, C and D) as flag bits, with C and D bits fixed to 01.
Figure 19 R2 digital line signaling – call establishment 2. 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. Figure 20 R2 digital line signaling – originating side releases the call 3. Terminating side releases the call.
When the terminating side supports metering signals, the system might send a forced release signal 00 instead of a clear-back signal 11 to release the line. This is to avoid collision between the clear-back signal sent by the called party and the metering signal. 5. Blocking in idle state or during conversation. After the originating side receives a blocking signal 11 from the terminating side when the circuit is idle or during conversation, it sends a forward signal 10.
Designation Basic Meaning A-2 Send last but one digit A-3 Address-complete; changeover to reception of Group B signals A-4 Congestion in the national network; terminate interregister signaling exchange A-5 Send calling party’s category A-6 Address-complete; terminate interregister signaling exchange, charge, and set up speech conditions A-7 Send last but two digits A-8 Send last but three digits A-9 Spare for national use A-10 Spare for national use A-11 Send country code indicator A-1
Table 10 Group B backward signals Designation Basic Meaning B-1 Spare for national use B-2 Send special information tone B-3 Subscriber line busy B-4 Congestion B-5 Unallocated number B-6 Subscriber line free, charge B-7 Subscriber line free, no charge B-8 Subscriber line out of order B-9 through B-15 Spare for national use The following figure shows the exchange process requesting calling party information, which is typical of R2 interregister signaling.
Configuring basic R2 signaling parameters Configuring the country or region mode Step Command Remarks 1. Enter system view. system-view N/A 2. Enter E1/T1 interface view. controller { e1 | t1 } slot-number N/A 3. Create a TS set and enable R2 signaling for it. timeslot-set ts-set-number timeslot-list timeslots-list signal r2 N/A 4. Enter R2 CAS view. cas ts-set-number N/A 5. Configure the country or region mode. mode zone-name [ default-standard ] Optional.
Enabling the DTMF mode to receive and send R2 signaling Step Command Remarks 1. Enter system view. system-view N/A 2. Enter E1/T1 interface view. controller { e1 | t1 } slot-number N/A 3. Create a TS set and enable R2 signaling for it. timeslot-set ts-set-number timeslot-list timeslots-list signal r2 N/A 4. Enter R2 CAS view. cas ts-set-number N/A 5. Enable the receiving and sending of R2 signaling in the DTMF mode. Optional.
Step Command Remarks 3. Create a TS set and enable R2 signaling for it. timeslot-set ts-set-number timeslot-list timeslots-list signal r2 N/A 4. Enter R2 CAS view. cas ts-set-number N/A 5. Maintain the circuits of a timeslot or a range of timeslots. ts { block | open | query | reset } timeslots timeslots-list Optional. Configuring R2 digital line signaling Step Command Remarks 1. Enter system view. system-view N/A 2. Enter E1/T1 interface view.
Step Command Remarks Optional. 13. Set timeout values of line signals. timer dl { answer | clear-back | clear-forward | | re-answer | release-guard | seize } time The default is 60,000 milliseconds for answer signal; 10,000 milliseconds for clear-back signal and clear-forward signal and release-guard signal; 1,000 milliseconds for seizure signal and re-answer signal.
Step 9. Command Configure the special characters that are supported during interregister signaling exchange. special-character character number Remarks Optional. No special character is configured by default. 10. Configure register signal values in R2 signaling.
Step Command Remarks 3. Bundle timeslots into a PRI group. pri-set [ timeslot-list range ] N/A 4. Exit E1/T1 interface view. quit N/A 5. Enter the view of the serial interface created for the PRI group. interface serial number: { 15 | 23 } N/A 6. Enable DSS1 or QSIG signaling. isdn protocol-type { dss1 | qsig } By default, DSS1 signaling is enabled. 7. Configure a signaling protocol mode. isdn protocol-mode { network | user } The default is user.
Configuring digital E&M signaling Configuring a start mode Similar to analog E&M signaling, digital E&M signaling also provides three start modes: immediate-start, wink-start and delay-start. The time sequence of digital E&M signaling is the same as that of analog E&M signaling for all three start modes.
To configure the wink start mode: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter E1/T1 interface view. controller { e1 | t1 } slot-number N/A 3. Create a TS set for digital E&M signaling and select the immediate start mode. timeslot-set ts-set-number timeslot-list timeslots-list signal e&m-wink N/A 4. Exit E1/T1 interface view. quit N/A 5. Enter digital E&M voice subscriber line view. subscriber-line slot-numbe:ts-set-number N/A 6.
Step Command Remarks Optional. 3. Enable E&M non-signaling mode. Disabled by default. open-trunk { caller monitor interval | called } For more information about the open-trunk command, see Voice Command Reference. Configuring receive and transmit signaling The ABCD bit pattern of the receive idle signal from the local end must be the same as that of the transmit idle signal from the remote end. Seized signals and idle signals are processed in the same way.
Step Command Remarks Create a TS set for digital E&M signaling and select a start mode. timeslot-set ts-set-number timeslot-list timeslots-list signal { e&m-delay | e&m-immediate | e&m-wink } N/A 4. Exit E1/T1 interface view. quit N/A 5. Enter digital E&M voice subscriber line view. subscriber-line slot-number:ts-set-number N/A 6. Configure the maximum interval between any two digits of a dialed number.
Step Command Remarks N/A 5. Enter voice subscriber line view. subscriber-line: ts-set-number:ts-set-number 6. Configure the dial delay. delay start-dial value 7. Configure the maximum time waiting for the subscriber to dial the first digit. timer first-dial seconds Configure the maximum interval between any two dialed digits. timer dial-interval seconds Configure the maximum time that the calling party waits for a ringback response. timer ring-back seconds 8. 9. Optional.
Step Command Remarks 1. Enter system view. system-view N/A 2. Enter the specified BSV BRI interface view. interface bri interface-number N/A 3. Configure an interface description. description text 4. Enable B channel loopback detection. loopback { b1 | b2 | both } 5. Set the MTU of the BSV BRI interface. mtu size 6. Set the interval for sending keepalive packets. timer hold seconds 7. Enable the interface to generate linkUp/linkDown traps upon link changes.
Step Command Remarks Disabled by default. 3. Enable the AMD function. amd enable The AMD detection results are subject to the language characteristics, answering machines, and background noises and music. NOTE: Support for the AMD function depends on the voice card. Configuring AMD parameters The accuracy of AMD detection results depends on the settings of detection parameters. You can increase the accuracy of the detection by adjusting AMD parameters.
To mirror PCM, RTP, or voice command data on a digital voice subscriber line to a specified interface or destination: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter digital voice subscriber line view. subscriber-line line-number N/A • Mirror PCM or RTP data based on a calling number: mirror number number { pcm | { in | out | all } data } calling calling-number to { local-interface interface-type interface-number [ mac H-H-H ] | remote-ip ip-address [ port port ] } 3.
E1 R2 signaling and digital E&M signaling configuration example Network requirements As shown in Figure 23, Telephones in City A and City B communicate with each other through voice routers (Router A and Router B) across an IP network, as shown in the network diagram. • In City A, Router A is connected to a PBX with an E1 subscriber line on which R2 signaling travels, and to the telephone at 0101003 with an FXS voice subscriber line.
[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. [RouterA-voice-dial-entity1001] match-template 0101001 # Associate the POTS voice entity with subscriber line 1/1:1.
[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. [RouterB-voice-dial-entity2002] line 1/1:1 [RouterB-voice-dial-entity2002] send-number all [RouterB-voice-dial-entity2002] quit # Create a VoIP voice entity. [RouterB-voice-dial] entity 010 voip # Configure a target match-template for the VoIP voice entity.
[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 voice entity. [RouterA-voice-dial-entity1003] match-template 0101003 # Associate the POTS voice entity with FXS subscriber line 3/0.
[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. [RouterB-voice-dial-entity2001] match-template 07552001 # Associate the POTS voice entity with subscriber line 1/1:15.
Figure 25 Network diagram Configuration procedure 1. Configure Router A: # Configure IP address 1.1.1.1/24 for interface Ethernet 2/1. 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 [RouterA-E1 1/1] quit # Set the ISDN protocol type and protocol mode for the ISDN interface serial 1/1:15.
# 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 2. Configure Router B: # Configure the IP address 2.2.2.2/24 for the interface Ethernet 2/1. system-view [RouterB] interface ethernet 2/1 [RouterB-Ethernet2/1] ip address 2.2.2.2 255.255.255.0 [RouterB-Ethernet2/1] quit # Create an ISDN PRI group on interface E1 1/1.
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. When the switch adopts outgoing trunking mode, the router must adopt incoming or bidirectional trunking mode.
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. The following discusses these two dial plan processes.
4. The gateway initiates a call to the called side and sends the calling and called numbers. 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 1. After receiving a voice call (the called number), the voice gateway on the called side performs global calling/called number substitution. 2.
Table 12 Metacharacters Metacharacter Meaning 0-9 Digits 0 through 9. Pound sign (#) and asterisk (*) Each indicates a valid digit. Dot (.) Wildcard, which can match any valid digit. For example, 555…. can match any number beginning with 555 and ending in four additional characters. Hyphen (-) Used to connect two numbers (The smaller comes before the larger) to indicate a range of numbers, for example, 1-9 inclusive. Brackets ([ ]) Delimits a range for matching.
Introduction to number substitution According to the network requirements, you can first configure a number substitution rule list, and then define specific number substitution rules, dot-match rules, and preferred number substitution rules for the list. Finally, you can apply these substitution rules globally or to voice entities and voice subscriber lines to substitute calling/called numbers flexibly.
Step Command Remarks Optional. 5. Configure calling numbers permitted to call out. caller-permit calling-string By default, no calling number is configured, and outgoing calls are not restricted. NOTE: The calling-string argument is in the format of { [ + ] string [ $ ] }| $. For specific meanings of these symbols in the format, see Voice Command Reference.
Step Bind a subscriber group to the voice entity. 5. Command Remarks caller-group { deny | permit } subscriber-group-list-number By default, no subscriber group is bound to the voice entity, that is, any calling number is allowed to originate calls. Enabling private line auto ring-down With the private line auto ring-down (PLAR) function enabled, the voice gateway automatically dials the specified called number (string) as soon as the subscriber picks up the phone.
If the router is configured to use longest match mode, it will wait for further digits. After the dial timer expires, the router will ignore the configured longest match mode and automatically use shortest match to establish a call connection. • When a subscriber dials 0106688#, if you configure the router to use longest match mode and the dial terminator "#" on the router, the router will as well ignore the configured longest match mode and use shortest match mode to establish a call connection.
Figure 28 Network diagram • Shortest number match a. Configure Router A: system-view [RouterA] voice-setup [RouterA-voice] dial-program # Configure POTS entity 1000. [RouterA-voice-dial] entity 1000 pots [RouterA-voice-dial-entity1000] match-template 10001234$ [RouterA-voice-dial-entity1000] line 1/0 [RouterA-voice-dial-entity1000] quit # Configure VoIP entity 2000 and VoIP entity 2001.
# Configure the longest match mode on Router A. The other steps are the same as those for the shortest match mode. [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.
Step 5. 6. Command Configure the voice entity priority. priority priority-order Exit voice entity view. quit Remarks Optional. 0 by default. N/A Optional. 7. Configure match order of voice entity selection rules. select-rule rule-order 1st-rule [ 2nd-rule [ 3rd-rule ] ] By default, the match order of rules for the voice entity selection is exact match->voice entity priority->random selection.
[RouterB] voice-setup [RouterB-voice] dial-program # Configure POTS entity 2000. [RouterB-voice-dial] entity 2000 pots [RouterB-voice-dial-entity2000] match-template 20001234$ [RouterB-voice-dial-entity2000] line 1/0 3. { Configure different voice entity selection priority rules: Configure voice entities to be selected in sequence of exact match, priority, and random selection.
Configuration example of voice entity type selection priority rules There are an IP connection and a PRI connection between Router A and Router B. Figure 30 and the following describe the configurations for different voice entity type selection priority rules on Router A and Router B. Figure 30 Network diagram 1. Configure Router A: # Configure PRI signaling.
# Configure POTS entity 2000. [RouterB-voice-dial] entity 2000 pots [RouterB-voice-dial-entity2000] match-template 20001234$ [RouterB-voice-dial-entity2000] line 1/0 3. { Configure different voice entity type selection priority rules: Configure the system to select voice entities in order of VoIP->POTS->VoFR->IVR.
1. Configure Router A: 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 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.
Module ID : CMC Reference Numbers : 1 Current used voice entity : 2000 Voice entities are offered : 2000 2001 # End 4. Restore the maximum number of voice entities found before a search process stops to the default.
Voice entities are offered : 2000 # End 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 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter voice dial program view. dial-program N/A Optional. 4. Configure 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 maximum-call-connection set: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter voice dial program view. dial-program N/A 4. Configure a maximum-call-connection set. max-call set-number max-number By default, no maximum-call-connection set is configured. 5.
Configuration procedure To configure global number substitution: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter voice dial program view. dial-program N/A 4. Create a number substitution rule list and enter voice number-substitute view. number-substitute list-number N/A Configure a dot-match rule. dot-match { end-only | left-right | right-left } 6. Configure a number substitution rule.
Step Command Remarks 6. Configure a number substitution rule. rule rule-tag input-number output-number [ number-type input-number-type output-number-type | numbering-plan input-numbering-plan output-numbering-plan ] * 7. Configure the preferred number substitution rule. first-rule rule-number By default, the preferred number substitution rule is not configured. 8. Exit voice number-substitute view and enter voice dial program view. quit N/A Enter voice entity view.
Step Command Remarks quit N/A 10. Exit voice view and enter system view. quit N/A 11. Enter voice subscriber line view. subscriber-line line-number N/A Exit voice dial program view and enter voice view. 9. 12. Configure a number substitution rule list for a subscriber line. Optional. substitute { called | calling } list-number By default, no number substitution rule list is configured for a subscriber line.
Configuring a dial prefix The following section provides information about configuring a dial prefix. Configuration prerequisites The required basic functions have completed on POTS and VoIP entities. Configuration procedure You can configure a prefix for dialed PSTN numbers. When a POTS entity originates a call, the dial prefix will be added to the called number. To configure a dial prefix: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3.
Configuring number substitution Network requirements As shown in Figure 32, a PBX to forms a local telephony network at place A and place B respectively. The following requirements must be met: • These two local telephony networks communicate through two voice gateways. Subscribers in one PBX network can make ordinary calls to remote subscribers in the other PBX network over a VoIP network. • Configure two FXO trunk lines between each router and its PBX and enable hunt group to realize trunk line backup.
[RouterB-voice-dial-substitute21101] rule 1 0101688 0001 [RouterB-voice-dial-substitute21101] rule 2 0103366 0002 [RouterB-voice-dial-substitute21101] rule 3 0102323 0003 # Configure a number substitution rule list for calling numbers of outgoing calls.
# Configure a number substitution rule list for calling numbers of incoming calls. [RouterA-voice-dial-substitute101] quit [RouterA-voice-dial] number-substitute 102 [RouterA-voice-dial-substitute102] dot-match left-right [RouterA-voice-dial-substitute102] rule 1 ^...0001$ ...1234 [RouterA-voice-dial-substitute102] rule 2 ^...0002$ ...6788 [RouterA-voice-dial-substitute102] rule 3 ^...0003$ ...6565 [RouterA-voice-dial-substitute102] quit # Configure number substitution rules.
Configuration consideration The select-rule rule-order 1 4 command can implement load sharing. Because the first rule "exact match" cannot distinguish the priority between Router B and Router C, Router A will use the fourth rule "longest idle time" to make sure that the resources of the two gateways are fully, equally utilized. Configuration procedure 1. Configure Router A: # Configure an Ethernet address. system-view [RouterA] interface ethernet 2/1 [RouterA-Ethernet2/1] ip address 1.1.1.
[RouterB-voice-dial] entity1001 pots [RouterB-voice-dial-entity1001] match-template 010.... [RouterB-voice-dial-entity1001] line 1/1 [RouterB-voice-dial-entity1001] send-number all [RouterB-voice-dial-entity1001] quit # Configure rules in the match order for voice entity selection. [RouterB-voice-dial] select-rule rule-order 1 4 3. Configure Router C: # Configure an Ethernet address. system-view [RouterC] interface ethernet 2/1 [RouterC-Ethernet2/1] ip address 1.1.1.
Figure 34 Network diagram Configuration procedure 1. Configure Router A: # Configure an Ethernet address. system-view [RouterA] interface ethernet 2/1 [RouterA-Ethernet2/1] ip address 1.1.1.1 24 [RouterA-Ethernet2/1] quit # Configure a VoIP entity. [RouterA] voice-setup [RouterA-voice] dial-program [RouterA-voice-dial] entity 2000 voip [RouterA-voice-dial-entity2000] match-template 010.... [RouterA-voice-dial-entity2000] address sip ip 1.1.1.
# Configure a VoIP entity. [RouterB] voice-setup [RouterB-voice] dial-program [RouterB-voice-dial] entity 1000 voip [RouterB-voice-dial-entity1000] match-template 010.... [RouterB-voice-dial-entity1000] address sip ip 1.1.1.3 [RouterB-voice-dial-entity1000] quit # Configure POTS entities.
[RouterC-voice-dial-entity1003] return # Enable hunt group for voice subscriber lines.
[RouterA] voice-setup [RouterA-voice] dial-program [RouterA-voice-dial] subscriber-group 1 [RouterA-voice-dial-group1] match-template 1100.. [RouterA-voice-dial-group1] quit [RouterA-voice-dial] subscriber-group 2 [RouterA-voice-dial-group2] match-template 1200.. [RouterA-voice-dial-group2] quit # Configure VoIP entities for place B and place C. [RouterA-voice-dial] entity 2000 voip [RouterA-voice-dial-entity2000] address sip proxy [RouterA-voice-dial-entity2000] match-template 2...
4. Display information of the configured subscriber groups: display voice subscriber-group all Current configuration of subscriber group 1 # Description : Referenced by entities: Type: POTS Tag: 2100 Include match templates: Match-template: 1100.. # END Current configuration of subscriber group 2 # Description : Referenced by entities: Type: POTS Tag: 2100 Type: POTS Tag: 3100 Include match templates: Match-template: 1200..
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). SIP is responsible for signaling control in IP networks and communication with soft switch platforms.
information of the called UA is available and the calling UA is allowed to make the call, the proxy server then forwards the request to the called UA. Redirect server A redirect server sends a new connection address to a requesting client. 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.
• Scalable system—Allows expansion as enterprises grow. • Support to remote users—With SIP, an enterprise network can extend to all its users, wherever they are. • Competitive advantage potentials—More SIP-based services are emerging. • Consistent communication method—Management becomes easier as the result of consistency in dialup mode and system access method used by branches, SOHOs, and traveling personnel.
As shown in Figure 36, a SIP UA sends its registrar a REGISTER request at startup or in response to an administrative registration operation, carrying all the information that must be recorded. Upon receipt of the request, the registrar sends back a response notifying receipt of the request, and a 200 OK (SUCCESS) message if the registration is accepted.
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 As shown in Figure 39, when a SIP redirect server receives a session request, it sends back a response indicating the address of the called SIP endpoint instead of forwarding the request.
Figure 39 Call redirection procedure for UAs Internet User agent User agent Redirect Server INVITE 100 Trying 302 Moved Temporarily ACK INVITE 100 Trying 200 OK ACK This is a common application. Fundamentally, a redirect server can respond with the address of a proxy server as well. The subsequent call procedures are the same as the call procedures involving proxy servers.
using their own digital certificates and can communicate with each other only after 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. For more information about signaling encryption, see "Configuring TLS for SIP sessions." SIP over TLS requires the configuration of TLS security policies.
TLS-SRTP combinations TLS protects control signaling, and SRTP encrypts and authenticates voice media flows. You can use them separately or together. As shown in Table 15, there are four combinations of TLS and SRTP. Table 15 TLS-SRTP combinations TLS SRTP Description Signaling packets are secured. Personal information is protected. On On Media packets are secured. Call conversations are protected. Recommended. Off On On Off Off Off Signaling packets are not secured.
The device only supports encapsulating QSIG messages within SIP messages. 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. As shown in Figure 41, two VPNs Voice and Data exist on the network. The two VPNs cannot access each other. After you specify the Voice VPN instance on PE 1, Telephone 3 can make SIP calls with telephones 1 and 2 in the VPN Voice.
Task Remarks Configuring outbound SIP proxy server information for a SIP UA Optional. Configuring transport layer protocol for SIP calls Required when TCP or TLS is specified as the transport layer protocol. Configuring SIP extensions Strict SIP routing The feature is enabled by default, and requires no configuration. Configuring out-of-band SIP DTMF transmission mode Optional. Configuring source IP address binding for SIP messages Optional. Configuring a domain name for the SIP UA Optional.
Authentication information selection rule If the authentication information is configured with the user command, POTS entity view is preferred. Otherwise, the authentication information configured in SIP client view is used. If realm is configured on a SIP UA, make sure the value is the same as that configured on the server. Otherwise, the SIP UA will fail the authentication due to mismatch.
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. Otherwise, no register request can be initiated.
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. • Use the timer registration divider command to set the expiration percentage.
failure-triggered re-registration is enabled and the error code is 5xx (except for 502, 504, 505, and 513), 403, or 408, the device will re-register the number on the registrar. To configure call failure-triggered re-registration: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter SIP client view. sip N/A 4. Enable call failure-triggered re-registration. call-fallback register Optional. Not enabled by default.
Step Command Remarks Optional. Enable checking the status of voice subscriber lines associated with POTS voice entities. 5. line-check enable By default, before registering numbers for a POTS voice entity, the device checks the status of the voice subscriber line associated to the POTS voice entity. The device can send REGISTER requests for numbers only when the status of the line is up.
• Use the proxy server. If a SIP server is present, use the SIP proxy server for SIP message interaction. • Use the domain name as the destination address. Without the need of obtaining the IP address of the destination, the client only needs to know the unique domain name of the destination on the network, and communicates with the destination through DNS. • Use the E.164 Number to URI Mapping (ENUM). ENUM translates telephone numbers into Internet addresses.
Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter SIP client view. Sip N/A Optional. The default DNS lookup method is a-record. If you configure the destination port by using the address sip { dns domain-name [ port port-number ] | enum-group group-number }, proxy dns domain-name [ port port-number ], or mwi-server dns domain-name [ port port-number ] command, the DNS lookup mode can only be Type-A. 4. Set the DNS lookup mode.
Configuring user information When accessing the service provider network, configure the user information provided by the service provider to confirm user identity. To configure user information: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter voice dial program view. dial-program N/A 4. Enter POTS voice entity view. entity entity-number pots N/A Optional. 5. Configure user information.
Configuring UDP or TCP for outgoing SIP calls The execution of the transport command in SIP client view specifies the global transport layer protocol for outgoing SIP calls. If you want to configure a different transport layer protocol for individual calls, you can specify the transport layer protocol in corresponding VoIP voice entity view. If the transport layer protocol configured in VoIP voice entity view and that configured in SIP client view are different, the former is adopted.
Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter SIP client view. sip N/A 4. Set the aging time for TCP connections. timer connection age tcp tcp-age-time Optional. By default, the aging time for TCP connections is 5 minutes. 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.
Step Command Remarks Not configured by default. 4. Reference the SSL policy for the client. crypto ssl-client-policy client-policy-name 5. Reference the SSL policy for the server. crypto ssl-server-policy server-policy-name Not configured by default. For more information about SSL policies, see Security Configuration Guide. To specify TLS as the global transport layer protocol for outgoing SIP calls: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view.
To specify TLS as the transport layer protocol for incoming SIP calls: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter SIP client view. sip N/A 4. Specify TLS as the transport layer protocol for incoming SIP calls. listen transport tls By default, the TLS listening port is disabled. To set the aging time for TLS connection: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3.
Specifying the URL scheme for outgoing SIP calls The device provides two URL schemes: SIP and SIP secure (SIPS). You can choose either as needed. To ensure transmission security, specify the SIPS scheme. The execution of the url command in SIP client view specifies the global URL scheme. If you want to configure a different URL scheme for individual outgoing SIP calls, specify the URL scheme in corresponding VoIP voice entity view.
Step Command Remarks Optional. By default, no URL scheme is configured. 5. Specify the URL scheme for outgoing SIP calls. url { sip | sips } This command is effective only when the type of the VoIP voice entity is SIP. You can specify the SIPS scheme only when the transport layer protocol is TLS. Configuring SIP extensions Strict SIP routing Strict SIP routing is supported.
Configuring source IP address binding for SIP messages Perform this task to specify a source IP address for SIP signaling or media messages by using one of the following methods. • Static IP address binding—Uses a static address as the source IP address. • Interface binding—Uses the IP address of a source interface as the source IP address. In a large-scale network, an interface obtains its IP address from a DHCP or PPPoE server.
Condition Result Configure a source address binding when the physical layer or link layer state of the corresponding interface is down. The source address binding does not take effect and the gateway automatically gets a source IP address for packets. The DHCP lease duration expires and the bound interface dynamically obtains a new IP address from the DHCP server The new IP address will be used as the source IP address. The SIP registrar is enabled.
Step Configure the device to obtain the destination number from the To header field for sending a SIP request. 4. Configure the device to use the address in the To header field as the address in the From header field. 5. Command Remarks sip-comp callee By default, the destination number is obtained from the request-line, which is the start line in an SIP request message.
To configure SIP compatibility for fax pass-through and modem pass-through: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter SIP client view. sip N/A 4. Configure SIP compatibility for fax pass-through and modem pass-through. sip-comp { t38 | x-parameter } * By default, the compatibility options are not carried in re-INVITE requests.
Step Command Remarks 4. Configure the User-Agent header field in a SIP request. sip-comp agent product-name product-version By default, the User-Agent header field in a SIP request is not configured. 5. Configure the Server header field in a SIP response. sip-comp server product-name product-version By default, the Server header field in a SIP response is not configured.
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. The default SIP status code to PSTN release cause code mappings and PSTN release cause code to SIP status mappings are used for communication between a SIP network and a PSTN.
Configuration procedure To configure periodic refresh of SIP sessions: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter SIP client view. sip N/A Optional. By default,. • The periodic refresh of SIP 4. Enable periodic refresh of SIP sessions and set the maximum and minimum session expiration time. timer session-expires seconds [ minimum min-seconds ] sessions is not enabled automatically.
Configuring VRF-aware SIP Before you perform this task, you must have completed L3VPN configurations. 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.
Task Command Remarks Display the temporarily saved SIP user identifications and the mapping information of the users’ contact addresses. display voice sip dynamic-contact-address [ | { begin | exclude | include } regular-expression ] Available in any view. Clear all call statistics of the SIP UA reset voice sip statistics Available in user view. Clear a specified SIP connection over a specific transport layer protocol. reset voice sip connection { tcp | tls } id conn-id Available in user view.
[RouterA-voice-dial-entity2222] quit [Router1-voice-dial] entity 1111 pots [RouterA-voice-dial-entity1111] line 1/0 [RouterA-voice-dial-entity1111] match-template 1111 2. Configure Router B: # Configure the Ethernet interface. system-view [RouterB] interface ethernet 2/1 [RouterB-Ethernet2/1] ip address 192.168.2.2 255.255.255.0 # Configure voice entities.
The configuration of SIP server is not dealt with in this example because it varies with devices. 1. Configure Router A: # Configure the Ethernet interface. system-view [RouterA] interface ethernet 2/1 [RouterA-Ethernet2/1] ip address 192.168.2.1 255.255.255.0 [RouterA-Ethernet2/1] quit # Configure SIP. [RouterA] voice-setup [RouterA-voice] sip [RouterA-voice-sip] registrar ipv4 192.168.2.3 [RouterA-voice-sip] proxy ipv4 192.168.2.
[RouterB-voice-dial-entity2222] quit [RouterB-voice-dial] entity 1111 voip [RouterB-voice-dial-entity1111] address sip proxy [RouterB-voice-dial-entity1111] match-template 1111 [RouterB-voice-dial-entity1111] quit Configuration verification After the local numbers of the two sides are registered on the registrar successfully, you can make calls between telephone 1111 and telephone 2222 through the proxy server.
system-view [RouterB] interface ethernet 2/1 [RouterB-Ethernet2/1] ip address 192.168.2.2 255.255.255.0 # Configure voice entities. 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.
[RouterA-voice-dial-entity2222] match-template 2222 [RouterA-voice-dial-entity2222] outband sip [RouterA-voice-dial-entity2222] quit [RouterA-voice-dial] entity 1111 pots [RouterA-voice-dial-entity1111] line 1/0 [RouterA-voice-dial-entity1111] match-template 1111 [RouterA-voice-dial-entity1111] outband sip 2. Configure Router B: # Configure the IP address of the Ethernet interface. system-view [RouterB] interface ethernet 2/1 [RouterB-Ethernet2/1] ip address 192.168.2.2 255.255.255.
Configuration procedure 1. Configure Router A: # Configure the IP address of the Ethernet interface. system-view [RouterA] interface ethernet 2/1 [RouterA-Ethernet2/1] ip address 192.168.2.1 255.255.255.0 [RouterA-Ethernet2/1] quit # Configure the voice entities. [RouterA] voice-setup [RouterA-voice] dial-program [RouterA-voice-dial] entity 2222 voip [RouterA-voice-dial-entity2222] address sip ip 192.168.2.
Configuring TCP 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 from Telephone 1111 to telephone 2222 be carried over TCP. Figure 47 Network diagram Confguration procedure 1. Configure Router A: # Configure the IP address of the Ethernet interface. system-view [RouterA] interface ethernet 2/1 [RouterA-Ethernet2/1] ip address 192.168.2.1 255.255.255.
[RouterB] voice-setup [RouterB-voice] sip [RouterB-voice-sip] listen transport tcp [RouterB-voice-sip] quit # Configure the voice entities. [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] entity 1111 voip [RouterB-voice-dial-entity1111] address sip ip 192.168.2.
# Create a PKI entity aaa, enter its view, and then configure the common name of the entity as RouterA. [RouterA] pki entity aaa [RouterA-pki-entity-aaa] common-name RouterA [RouterA-pki-entity-aaa] quit # Create a PKI domain voice, enter its view, and then specify the trusted CA as voice.
[RouterA-voice-dial] quit 2. Configure Router B: # Configure the IP address of the Ethernet interface. system-view [RouterB] interface ethernet 2/1 [RouterB-Ethernet2/1] ip address 192.168.2.2 255.255.255.0 [RouterB-Ethernet2/1] quit # Create a PKI entity aaa, enter its view, and then configure the common name of the entity as RouterB.
[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] entity 1111 voip [RouterB-voice-dial-entity1111] address sip ip 192.168.2.1 port 5061 [RouterB-voice-dial-entity1111] match-template 1111 Configuration verification SIP calls from Telephone 1111 to telephone 2222 are carried over TLS.
[RouterA-voice-sip] listen transport tls [RouterA-voice-sip] url sips [RouterA-voice-sip] transport tls [RouterA-voice-sip] quit # Configure voice entities. [RouterA-voice] dial-program [RouterA-voice-dial] entity 2222 voip [RouterA-voice-dial-entity2222] address sip ip 192.168.2.
Configuring SRTP for SIP calls Network requriements Two routers Router A and Router B work as SIP UAs. It is required that SIP calls use the SRTP protocol to protect call conversations. Figure 50 Network diagram Configuration procedure 1. Configure Router A: # Configure the IP address of the Ethernet interface. system-view [RouterA] interface ethernet 2/1 [RouterA-Ethernet2/1] ip address 192.168.2.1 255.255.255.
[RouterB-voice] sip [RouterB-voice-sip] media-protocol srtp [RouterB-voice-sip] quit # Specify 2222 as a local number of POTS voice entity 2222. [RouterB-voice] dial-program [RouterB-voice-dial] entity 2222 pots [RouterB-voice-dial-entity2222] line 1/0 [RouterB-voice-dial-entity2222] match-template 2222 # Configure VoIP voice entity 1111, and configure the IP address of the peer VoIP gateway as 192.168.2.1, and the called number as 1111.
the registrar. Otherwise, the voice gateway will not initiate an authentication request. If the realm argument is not locally configured, the voice gateway will not judge the configuration of realm on the server and consider the server is trusted. Failed to set up point-to-point calls Symptom The UA could not set up point-to-point calls. Solution Check that the IP address and the port number of the remote voice gateway are correctly configured.
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.
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. Optional. Configuring a call route Applicable to calls initiated from internal users to external users.
Step Command Remarks By default, no IP address is configured, that is, there is no local SIP server. 5. Configure the IP address of the local SIP server as the IP address of a local interface. server-bind ipv4 ipv4-address [ port port-number ] [ expires time-interval ] Note that the ipv4-address argument can be the IP address of an interface on the local router, or a loopback address such as 127.0.0.1.
Step Command Remarks By default, the local SIP server is disabled. 7. Enable the local SIP server. The functions of the local SIP server can take effect after you execute this command. server enable Configuring user information Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter SIP server view. sip-server N/A 4. Create a user to be registered with and enter register user view.
Step 4. Specify a trusted node. Command Remarks trusted-point ipv4 ipv4-address [ port port-number ] By default, no trusted node is 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.
Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter SIP server view. sip-server N/A 4. Create a user to be registered with and enter register user view. register-user tag By default, no user is created to be registered. 5. Apply a call rule set in register user view. By default, no call rule set is applied register user view. srs tag Only one call rule set can be applied in register user view.
Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter SIP server view. sip-server N/A 4. Enter call route view. call-route N/A 5. Configure a call route. trunk tag called-number called-pattern ipv4 dest-ip-addr [ port port-number ] [ area-prefix prefix ] By default, no call route is configured.
[RouterC-voice-server] server-bind ipv4 2.1.1.2 [RouterC-voice-server] server enable # Configure authentication information for Phone 1000 and Phone 5000.
[RouterB-voice-sip] register-enable on 4. Verify the configurations: [RouterC-voice-server-user5000] display voice sip-server register-user all 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.
[RouterC-voice-server-user5000] authentication username 5000 password simple 5000 2. Configure Router A: # Configure voice entities.
Phone 1000 and Phone 5000 have already registered with the local SIP server, Router C, which periodically sends Options messages to VCX to detect the link reachability. If the link is unreachable, the local SIP server accepts registrations and calls originated by Phone 1000 and Phone 5000. If the link is reachable, the local SIP server rejects registrations and calls, and Phone 1000 and Phone 5000 then register with VCX, which accepts their registrations and calls.
[RouterA-voice-server-user1000] number 1000 [RouterA-voice-server-user1000] quit [RouterA-voice-server] register-user 5000 [RouterA-voice-server-user5000] number 5000 [RouterA-voice-server-user5000] return # Configure voice entities.
----------------------------------------------------------------------1000 1000 online 1.1.1.2:5060 5000 { 5000 online 2.1.1.1:5060 When the IP link recovers, the local SIP server on Router A is disabled, and phones re-register with VCX. Configuring the call authority control Network requirements As shown in Figure 55, the DNs for Department A in a company are 1000 through 1999, while those for Department B are 5000 through 5999.
[RouterC-voice-server-user5555] authentication username 5555 password simple 5555 [RouterC-voice-server-user5555] quit # Configure a call rule set. [RouterC-voice-server] call-rule-set [RouterC-voice-server-set] service 0 [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...
[RouterB-voice-dial-entity5000] match-template 5000 [RouterB-voice-dial-entity5000] line 2/0 [RouterB-voice-dial-entity5000] user 5000 password simple 5000 [RouterB-voice-dial-entity5000] quit [RouterB-voice-dial] entity 5555 pots [RouterB-voice-dial-entity5555] match-template 5555 [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
# Configure the router to operate in the alone mode. system-view [RouterC] voice-setup [RouterC-voice] sip-server [RouterC-voice-server] server-bind ipv4 2.1.1.2 [RouterC-voice-server] server enable # 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.
5000 5000 online 1.1.1.1:5060 { Make a call from the external number 55661000 to the internal number 88995000. The local SIP server Router C removes the area prefix 8899 from 88995000 to convert it into the internal short number 5000. Then, the local SIP server alerts Phone 5000, and both parties can communicate after Phone 5000 is picked up. Configuring a call route Network requirements As shown in Figure 57: • The internal numbers of a company are four-digit long and the area prefix is 8899.
[RouterA-voice-dial-entity1000] user 1000 password simple 1000 [RouterA-voice-dial-entity1000] quit [RouterA-voice-dial]entity 55665000 voip [RouterA-voice-dial-entity55665000] address sip proxy [RouterA-voice-dial-entity55665000] match-template 55665000 [RouterA-voice-dial-entity55665000] quit [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 3. Configure Router B: # Configure voice entities.
Configuring SIP trunk This chapter describes how to configure SIP trunk. Background As shown in Figure 58, on a typical telephone network, internal calls of the enterprise are made 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 shown in Figure 59.
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. • The internal calls of the enterprise are placed by the enterprise IP-PBX.
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 SIP trunk configuration task list Task Remarks Enabling the SIP trunk function Required.
Task Remarks Enabling delayed offer to early offer conversion Optional. Enabling codec transcoding Optional. Enabling address hiding Optional. Enabling call forwarding Optional. Enabling call transfer Optional. Enabling midcall signaling pass-through Optional. Enabling the SIP trunk function Before using various SIP trunk functions, first enable the SIP trunk function on the SIP trunk device. Do not use a device enabled with the SIP trunk function as a SIP UA.
Step 4. Command Add a member server to the SIP server group and configure the server information. Remarks By default, a SIP server group has no member server. address index-number { ipv4 ip-address | dns dns-name } [ port port-number ] [ transport { udp | tcp | tls } ] [ url { sip | sips } ] You can add at most five member servers to a SIP server group. An index represents the priority of a member server in the SIP server group. The smaller the index value, the higher the priority. Optional.
Step Enable the real-time switching function in the SIP server group. 4. Command Remarks hot-swap enable Disabled by default. 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 Voice Command Reference. Configuring the keepalive and redundancy functions Use the keepalive function to detect whether the SIP servers in a SIP server group are reachable.
Step Command Remarks Disabled by default. Enable the keepalive function and set the keepalive mode for the SIP server group. keepalive { options [ interval seconds ] | register } If the keepalive function is disabled, the current server is always the one with the highest priority in the SIP server group. 5. Return to system view. quit N/A 6. Enter SIP client view. sip N/A 7. Configure the redundancy mode for the SIP server group. redundancy mode { homing | parking } Optional. 4.
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 Configure a source address binding when ongoing calls exist. for ongoing SIP media sessions but takes effect for subsequent SIP media sessions. • A new source address binding for signal takes effect immediately for all SIP signaling sessions.
Step 5. 6. 7. 8. 9. Command Remarks Assign the host username allocated by the ITSP to the SIP trunk account. assign contact-user user-name Not assigned by default. Assign the host name allocated by the ITSP to the SIP trunk account. assign host-name host-name Associate the SIP trunk account with a SIP server group for registration. registrar server-group group-number [ expires time ] Configure the authentication username and password for the SIP trunk account.
Step 4. 5. 6. Enter VoIP voice entity view. Bind a SIP trunk account to the VoIP voice entity. Bind a SIP server group to the VoIP voice entity. Command Remarks entity entity-number voip N/A bind sip-trunk account account-index address sip server-group group-number By default, no SIP trunk account is bound to the VoIP voice entity. Only an existing SIP trunk account can be bound to a VoIP voice entity. By default, no SIP server group is bound to the VoIP voice entity.
Step Command Remarks 2. Enter voice view. voice-setup N/A 3. Enter voice dial program view. dial-program N/A 4. Enter VoIP voice entity view. entity entity-number voip N/A 5. Bind a SIP trunk account to the VoIP voice entity. bind sip-trunk account account-index 6. Enable the SIP proxy server. address sip proxy Not configured by default. 7. Return to system view. quit N/A 8. Enter SIP client view. sip N/A 9. Specify a SIP server group to be used as the proxy server.
Step Command Remarks 3. Enter voice dial program view. dial-program N/A 4. Enter VoIP voice entity view. entity entity-number voip N/A 5. Match a source host name prefix for the VoIP voice entity. 6. Match a destination host name prefix for the VoIP voice entity. Optional. match source host-prefix host-prefix Not specified by default. In other words, all source host names can be matched. Optional. match destination host-prefix host-prefix Not specified by default.
Step 4. Enter VoIP voice entity view. Command Remarks entity entity-number voip N/A Optional. 5. Enable codec transparent transfer. codec transparent By default, codec transparent transfer is disabled and the SIP trunk device is involved in the media negotiation between the calling and called parties. Note that to enable codec transparent transfer on the SIP trunk device, execute this command on all VoIP voice entities connected to the public and private networks.
Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter voice dial program view. dial-program N/A 4. Enter VoIP voice entity view. entity entity-number voip N/A 5. Enable DO-EO conversion. early-offer forced Optional. 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.
Address hiding does not take effect for a voice entity enabled with media flow-around. To enable address hiding: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter SIP client view. sip N/A Optional. 4. Enable address hiding. address-hiding enable By default, address hiding for SIP-to-SIP calls is disabled.
Step Command Remarks 2. Enter voice view. voice-setup N/A 3. Enter voice dial program view. dial-program N/A 4. Enter VoIP voice entity view. entity entity-number voip N/A 5. Enable call transfer. Optional. supplementary−service sip call-transfer By default, call forwarding is disabled.
SIP trunk configuration examples This section provides configuration examples of SIP trunk. Configuring a SIP server group with only one member server Network requirements As shown in Figure 62, the enterprise private network has a SIP trunk device deployed. Router A is a private network device, and Router B is a public network device. All calls between the private network and public network are made through the SIP trunk device. Figure 62 Network diagram Configuration procedure 1.
[TG-voice] sip-trunk enable # Create SIP server group 1. Add a SIP server into the server group: the index and the IPv4 address of the server are 1 and 10.1.1.2 respectively. [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.
Cofngiuration verification 1. On the SIP trunk device, display SIP trunk account information. [TG-voice-dial-entity2] display voice sip-trunk account ID User Group Server Exp Status 1 1 1802 Online 2000 10.1.1.2:5060 The output shows that the private network account 2000 has registered with the server at 10.1.1.2. 2. All calls between the private network and public network are made through the SIP trunk device.
# Create SIP server group 1. Add two SIP servers into the server group: the IP addresses are 10.1.1.2 and 10.1.1.3, and the server with the address 10.1.1.2 has a higher priority value. [TG-voice] server-group 1 [TG-voice-group-1] address 1 ipv4 10.1.1.2 [TG-voice-group-1] address 3 ipv4 10.1.1.3 # Enable the real-time switching function of SIP server group 1. Set the keepalive mode and redundancy mode for SIP server group 1 to OPTIONS and parking respectively.
Configuration procedure # Configure the call route for the outbound calls from private network user 2000 to public network user 1000 by binding SIP server group 1 to VoIP voice entity 1. system-view [TG] voice-setup [TG-voice] sip-trunk enable [TG-voice] dial-program [TG-voice-dial] entity 1 voip [TG-voice-dial-entity1] address sip server-group 1 [TG-voice-dial-entity1] bind sip-trunk account 1 [TG-voice-dial-entity1] match-template 1000 # Specify that calls with the source IP addresses 1.1.1.
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. H.323 uses the Client/Server model to establish calls through communication between the gateway and gatekeeper. H.
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_Reque
• Zone administration and security control • Call control signaling and call management • Routing control and accounting 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. They interact with the gatekeeper by sending H.225.0 RAS messages.
Admission control If the address of the called endpoint is available, the calling endpoint sends an Admission_Request (ARQ) message, based on which the gatekeeper decides whether to allow this endpoint to join a call process. This is how the gatekeeper controls admission. In the ARQ message sent to the gatekeeper, the calling endpoint might ask for direct call signaling (see Figure 67) or gatekeeper-routed call signaling (see Figure 68).
Figure 68 Gatekeeper-routed call signaling 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. However, it might not send this message.
Opening/closing logical channels The two endpoints open one or more logical channels between them for transporting media streams. (The logical channels are specified by IP address plus port number.) These channels are closed when the communication is over. Complete release Finally, either endpoint in communication can release resources by sending a Release Complete message. Disconnection The endpoints each send a Disconnect_Request (DRQ) to their own gatekeepers, which will confirm or reject the request.
Configuring basic H.323 gateway functions Configuration prerequisites Complete the required configuration of POTS and VoIP voice entities. 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.
For more information about POTS and VoIP entities, see "Configuring voice entities." Configuring registration password You can configure the gateway to send a password with the RPQ message sent for registration with the gatekeeper. When the gatekeeper receives this request, it compares the password with the one configured on it for the gateway. The gatekeeper will accept the request and return an RCF only when they are the same.
Step Enable security calling. 4. Command Remarks gk-security call enable Enabled by default. NOTE: Security calling must be disabled in a voice network where the terminating gatekeeper has no ability to process a call token. Displaying and maintaining the H.323 gateway Task Command Remarks Display the registration state information of a gateway. display voice gateway [ | { begin | exclude | include } regular-expression ] Available in any view. H.
Configuration procedure 1. Configure Router A: # Create a VoIP entity. system-view [RouterA] voice-setup [RouterA-voice] dial-program [RouterA-voice-dial] entity 0755 voip [RouterA-voice-dial-entity755] match-template 0755.... [RouterA-voice-dial-entity755] address ras [RouterA-voice-dial-entity755] quit # Create a POTS entity.
[RouterB-voice-dial-entity2001] return system-view [RouterB] interface loopback 0 [RouterB-Loopback0] ip address 2.2.2.2 255.255.255.255 # Enter gatekeeper client view. [RouterB-Loopback0] quit [RouterB] voice-setup [RouterB-voice] gk-client # Configure the gateway alias, and the name and IP address of the gatekeeper. [RouterB-voice-gk] gw-address 2.2.2.2 [RouterB-voice-gk] gw-id cityb-gw [RouterB-voice-gk] gk-id gk-center gk-addr 3.3.3.3 1719 # Configure the area ID.
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.
Call transfer Subscriber A (originator) and subscriber B (recipient) are in a conversation. Subscriber A presses the flash hook and the call is put on hold. Subscriber A dials another number to originate a call to 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.
Silent monitor and barge in services Silent monitor service—Allows a supervisor to monitor active calls without being heard. Barge in service—Allows a supervisor to participate in a monitored call, thus implementing three-party conference. For example, subscribers A and B are in a conversation, and subscriber C is the supervisor. If C wants to join the conversation, it sends a request to A. If A permits, the three-party conference can be held.
Task Remarks Configuring call waiting Optional. Configuring call hold Optional. Configuring call forwarding Optional. Configuring call transfer Optional. Configuring call backup Optional. Configuring hunt group Optional. Configuring incoming call barring Optional. Configuring outgoing call barring Optional. Configuring MWI Optional. Configuring three-party conference Optional. Configuring silent monitor and barge in Optional. Configuring calling party control Optional.
Configuring call waiting by using command lines Enabling call waiting In addition to being enable or disable the call waiting feature, you can set parameters related to this feature. These parameters include the number of times a call waiting tone pattern is played, the number of tones in a call waiting tone pattern, and the interval for playing a call waiting tone pattern. To enable call waiting: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice subscriber line view.
Configuring call hold This section describes how to enable call hold and configure the tone playing mode. Configuration prerequisites The router is equipped with an FXS voice interface card. Enabling call hold using command lines Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice subscriber line view. subscriber-line line-number N/A 3. Enable call hold. call-hold enable Disabled by default. NOTE: This command is applicable only for the FXS voice subscriber line.
Step 3. Configure the tone playing mode for call hold. Command Remarks call-hold-format { inactive | sendonly [ media-play media-id ] | moh-number string ] } The default is inactive. Configuration example Enable call hold for the voice subscriber line of Telephone A. Telephone A and Telephone B are in a conversation. The subscriber at Telephone A can interrupt the conversation with Telephone B by pressing the hook flash, and place a call to Telephone C after hearing a dial tone.
Enable keys Disable keys Remarks *60*number# #60# Enable and disable call forwarding unavailable, where "number" represents a forwarded-to number. Configuring call forwarding by using command lines In practice, you should set a reasonable, valid forwarded-to number and avoid setting the forwarded-to number to a wrong number or the original called number. Enabling call forwarding unconditional Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice subscriber line view.
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. To configure a call forwarding priority level: Step Command Remarks 1.
Call forwarding unavailable Place a call from Telephone A to Telephone B. The system forwards the call to Telephone C when the subscriber line of Telephone B is unavailable. Finally, Telephone A and Telephone C start a conversation. # Enable call forwarding unavailable for the voice subscriber line of Telephone B and forward the call from Telephone A to Telephone C (3000).
# Enable call transfer for the voice subscriber line of Telephone A. system-view [Sysname] subscriber-line 2/0 [Sysname-subscriber-line2/0] call-transfer enable 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.
Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice subscriber line view. subscriber-line line-number N/A 3. Enable hunt group for the voice subscriber line. hunt-group enable Disabled by default. 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.
[Sysname-subscriber-line2/1] hunt-group enable 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. • The system administrator performs configurations by using command lines on the device.
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: • Subscribers perform configurations by using keys on a telephone terminal. • The system administrator performs configurations by using command lines on the device.
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. To configure MWI: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice subscriber line view. subscriber-line line-number N/A 3.
Displaying and maintaining MWI Task Command Remarks Display the information of MWI. display voice ss mwi { all | number number } [ | { begin | exclude | include } regular-expression ] Available in any view. Display subscription information. display voice sip subscribe-state [ | { begin | exclude | include } regular-expression ] Available in any view.
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 Voice Command Reference. To configure three-party conference by using command lines: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice subscriber line view. subscriber-line line-number N/A 3.
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 1. Enter system view. system-view N/A 2. Enter voice subscriber line view. subscriber-line line-number N/A 3. Enable three-party conference in active participation mode. joined-conference enable Disabled by default.
Configuration prerequisites The router is equipped with an FXS voice interface card. Enabling and disabling Feature service setting by using keys The device supports the Feature service configuration for the FXS voice subscriber line on telephone terminals. To enable and disable the setting of the Feature service: Feature name Silent Monitor Enable keys *425*destination# Disable keys None Remarks On the VCX, Telephone B has the right to monitor Telephone A.
Feature name Directed Pickup Enable keys *455*pwd*pickup_ number# Disable keys None Remarks This feature allows a subscriber to answer a call ringing on the phone of a specific subscriber. To answer the call, the subscriber enables the Directed Pickup feature (feature code *455), enters a security code, and then enters the extension of the ringing phone. This transfers the call to the subscriber.
Feature name Block CallId for Current Call Enable keys *890*destination number# Disable keys None Remarks A subscriber can dial the feature code to hide the calling number for the current call and the called subscriber cannot see the calling number. Applied only once. Subscriber Speed Dial (range) *601*code# None After a speed dial number is configured for a subscriber on the VCX, the subscriber can dial the feature code to originate a call to the corresponding telephone. Applied only once.
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.
Call services configuration examples This section provides call services configuration examples. Call waiting Network requirements As shown in Figure 71, place a call from Telephone C to Telephone A which is already engaged in a 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.
[RouterA-subscriber-line1/0] call-waiting enable 2. Configure Router B: 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 3.
Figure 72 Network diagram Router A Router B Eth1/1 10.1.1.1/24 1000 Telephone A Eth1/2 10.1.1.2/24 Eth1/1 20.1.1.2/24 Router C Eth1/1 20.1.1.1/24 3000 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. 1.
[RouterC-voice-dial] entity 3000 pots [RouterC-voice-dial-entity3000] line 1/0 [RouterC-voice-dial-entity3000] match-template 3000 Call transfer Network requirements As shown in Figure 73, call transfer enables Telephone A to transfer Telephone B to Telephone C. After the call transfer is completed, Telephone B and Telephone C are in a conversation. The whole process is as follows: 1. Call Telephone B from Telephone A, so that Telephone B and Telephone A are in a conversation. 2.
# Enable call hold and call transfer. system-view [RouterA] subscriber-line 1/0 [RouterA-subscriber-line1/0] call-hold enable [RouterA-subscriber-line1/0] call-transfer enable 2. Configure voice entities on Router B. 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 3. Configure Router C.
Configuration procedure Before performing the following configuration, make sure that Router A, Router B and Router C are routable to each other. 1. Configure Router A: # Configure the voice entity with a higher priority.
MWI Network requirements As shown in Figure 75, Telephone A and Telephone B registered with the VCX through Router A and 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. Figure 75 Network diagram Configuration procedure 1. Configure VCX: 2.
Figure 77 Configuration page of call processing server (2) 3. 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 Configuration Option box, as shown in Figure 78. Figure 78 Configuration page of unified messaging server Select 9000 from the Main Voicemail Access Number List, as shown in Figure 79.
# 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. 4. Configure Router A: # Configure voice entities.
[RouterB-voice-dial-entity2000] quit [RouterB-voice-dial] quit [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.
# Enable the call hold function. system-view [RouterA] subscriber-line 1/0 [RouterA-subscriber-line1/0] call-hold 2. enable Configure Router B: # Configure the voice entity. 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.
Silent monitor and barge in Network requirements • Configure silent monitor for Telephone C to monitor the conversation between Telephone A and Telephone B. As shown in Figure 81, Telephone A and Telephone B are in a conversation. Dial the feature code *425*Number of Telephone A# at Telephone C to monitor the conversation between Telephone A and Telephone B. • Configure barge in for Telephone C to participate the conversation between Telephone A and Telephone B.
Figure 82 Telephone configuration page # Configure the monitoring authority Click Features of number 1000 to enter the feature configuration page, and then click Edit Feature of the Silent Monitor and Barge In feature to enter the page as shown in Figure 83. 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 number 1000. After this configuration, the page as shown in Figure 84 appears.
Figure 84 Silent monitor and barge in feature configuration page (2) After the above configuration, Telephone C with the number 3000 can monitor and barge in the conversations of Telephone A with the number 1000. 2. Configure Router A: # Configure VoIP voice entities to Router B and Router C.
# Configure VoIP voice entities to Router A and Router C.
[RouterC-voice-sip] proxy ipv4 100.1.1.101 [RouterC-voice-sip] register-enable on [RouterC-voice-sip] quit [RouterC-voice] quit # Enable the setting of feature service. [RouterC] subscriber-line 1/0 [RouterC-subscriber-line1/0] feature permit After the above configuration, dial feature code *425*1000# at Telephone C, and you can monitor the conversation between Telephone A and Telephone C. If you want to participate in the conversation, dial *428# at Telephone C.
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.
A call watch group can monitor either local interfaces or remote IP addresses (that is, IP connectivity to remote interfaces), but not both. The state of the E1/T1 interface associated with a call watch group is set as follows: • If local interfaces are monitored, the E1/T1 interface is set to watch-out state when all the monitored local interfaces are down.
Associating the E1/T1 interface with the call watch group To adapt the state of an E1/T1 interface to the state of the monitored local links in a call watch group, you must associate the E1/T1 interface with the call watch group. Note that, each E1/T1 interface can be associated with only one monitor group. To associate the E1/T1 interface with the call watch group: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter E1/T1 interface view. controller controller-type number N/A 3.
Figure 86 Network diagram Configuration procedure 1. Configure device Voice A: # Configure an IP address for each interface. (Details not shown.) # Configure call watch group 1 to monitor local interfaces Ethernet 1/1 and Ethernet 1/2. system-view [VoiceA] call-watch rule 1 local interface ethernet 1/1 [VoiceA] call-watch rule 1 local interface ethernet 1/2 # Associate E1 1/0 with call watch group 1 in hard mode. [VoiceA] controller e1 1/0 [VoiceA-E1 1/0] call-watch group 1 hard 2.
Figure 87 Network diagram Configuration procedure 1. 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. system-view [VoiceA] nqa server enable [VoiceA] nqa entry admin test1 [VoiceA-nqa-admin-test1] type icmp-echo [VoiceA-nqa-admin-test1-icmp-echo] destination ip 10.1.1.
2. Configure an IP address for each interface on Router A, Router B and Voice B. (Details not shown.
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.
Signals that a G3 fax machine receives and sends are modulated analog signals. Therefore the router processes fax signals in a different way than it processes telephone signals. The router needs to perform A/D or D/A conversion for fax signals (that is, the router demodulates analog signals from PSTN into digital signals, or modulates digital signals from the IP network into analog signals), but does not need to compress fax signals. A real-time fax process consists of five phases: 1. Fax call setup phase.
Configuration prerequisites VoIP configuration is completed, IP calls can be made successfully, and fax machines are connected correctly. Configuration procedure To configure a fax interworking protocol: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter voice dial program view. dial-program N/A 4. Enter voice entity view. entity entity-number { pots | voip } N/A Optional. By default, the number of two kinds of redundant packets is 0.
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. This method is used for the voice gateway to interwork with other devices in the pass-through mode. • To configure the pass-through mode: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter voice dial program view. dial-program N/A 4. Enter voice entity view.
Step Command Configure the NTE payload type for the NTE compatible-switching mode. 6. Remarks Optional. modem compatible-param payload-type By default, the value of the NTE payload type is 100. This command is valid only for the NTE-compatible switching mode. Enabling CNG fax switchover Configuration prerequisites • VoIP configuration is completed, IP calls can be made successfully, and fax machines are connected correctly.
Configuration prerequisites VoIP configuration is completed, IP calls can be made successfully, and fax machines are connected correctly. Configuration procedure To configure ECM for fax: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter voice dial program view. dial-program N/A 4. Enter voice entity view. entity entity-number { pots | vofr | voip } N/A By default, ECM is disabled on the gateway. Enable ECM for fax. 5.
Step Configure the signal transmission mode of fax faculty. 5. Command Remarks fax nsf-on By default, a standard faculty mode is adopted for fax faculty transmission. 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.
gateway. The transmitting gateway finalizes the packet transmission rate by comparing the received training result with its own training result. Point-to-point training—The gateways do not participate in the rate training between two fax terminals. In this mode, rate training is performed between two fax terminals and is transparent to the gateways. • Perform the following configuration in voice entity view.
Step Command Remarks 3. Enter voice dial program view. dial-program N/A 4. Enter voice entity view. entity entity-number { pots | vofr | voip } N/A 5. Configure the fax training mode. fax train-mode local By default, the PPP training is adopted. 6. Configure the threshold of local training. fax local-train threshold threshold By default, the threshold is 10. NOTE: When the local training mode is adopted, use the fax local-train threshold command to configure the threshold in percentage.
Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Configure the faculty set of the voice gateway in H.323 slow connection to contain T.38 faculty description.. voip h323-conf tcs-t38 By default, the faculty set contains T.38 faculty description. 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.
Step Command Remarks Optional. By default, T.38 (namely, T.38) protocol is used for fax. The number of low speed and high speed redundant packets is 0. 5. Configure the protocol for interworking with other devices globally. default entity fax protocol { t38 | standard-t38 } [ lb-redundancy number | hb-redundancy number ] default entity fax protocol pcm { g711alaw | g711ulaw } If the call control protocol is SIP, this command can be used only for the originator of the fax request (using private T.
Step Command Remarks 12. Configure the codec type and switching mode for SIP Modem pass-through function globally. default entity modem protocol pcm { standard | nte-compatible } { g711alaw | g711ulaw } Optional. By default, the SIP Modem pass-through function is not configured. For information about how to use the global default parameters for fax, see "Configuring voice entities." Displaying and maintaining FoIP configuration Task Command Remarks Display the statistics of the FoIP module.
[RouterA] voice-setup [RouterA-voice] dial-program [RouterA-voice-dial] default entity fax protocol standard-t38 # Configure VoIP voice entity 0755, and configure the IP address and the fax number of the peer VoIP gateway as 2.2.2.2 and 0755.… respectively. [RouterA-voice-dial] entity 0755 voip [RouterA-voice-dial-entity755] match-template 0755.... [RouterA-voice-dial-entity755] address sip ip 2.2.2.2 [RouterA-voice-dial-entity755] quit # Specify 0101001 as the local fax number of POTS voice entity 1001.
Configuration procedure 1. Configure Router A: # Set the switching mode to Re-Invite switching and the codec type to g711alaw for SIP modem pass-through. system-view [RouterA] voice-setup [RouterA-voice] dial-program [RouterA-voice-dial] default entity modem protocol pcm standard g711ulaw # Configure VoIP voice entity 2000, and configure the IP address and the fax number of the peer VoIP gateway as 2.2.2.2 and 2000 respectively.
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. Advantages A conventional interactive voice system uses fixed audio files and operations.
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.
Creating an IVR voice entity Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter voice dial program view. dial-program N/A 4. Create an IVR voice entity and enter IVR voice entity view. entity entity-number ivr By default, no IVR voice entity is created. Configuring an IVR voice entity To configure the root node: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3.
Step Command Remarks Optional. 7. Bind the subscriber group to the IVR voice entity. caller-group { deny | permit } subscriber-group-list-number 8. Configure the calling numbers permitted to originate calls to the IVR voice entity. caller-permit calling-string By default, no subscriber group is bound to an IVR voice entity, that is, any calling number is allowed. Optional. By default, incoming calls are not restricted. Optional. 9.
Step Command Remarks Optional. 13. Set the maximum-call-connection number to the IVR voice entity. max-call set-number By default, no maximum-call-connection set is bound to an IVR voice entity (that is, the IVR voice entity does not belong to any maximum-call-connection set and there is no limitation on the number of call connections). Optional. 14. Configure the priority for the IVR voice entity. priority priority-order 15. Disable the voice entity search function. select-stop 16.
Configuring IVR processing methods globally If you do not configure any error processing method or timeout processing method for a node, it uses the global methods to handle errors and timeout issues. To configure IVR processing methods globally: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter IVR management view. ivr-system N/A Optional. Configure IVR global processing method for handling subscriber input errors. 4.
Step 4. Create an IVR voice entity node and enter node view. 5. Configure description string for the node. Command Remarks node node-id { call | jump | service } N/A Optional. description string By default, no description is configured for an IVR voice entity 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.
Step 7. 8. 9. Command Remarks Specify the audio file that will be played to the subscriber when the node is waiting for the subscriber to press keys. media-play media-id [ play-times ] [ force ] Optional. Configure the input error processing method for the node. input-error { end-call | goto-pre-node | goto-node node-id } [ media-play media-id [ play-times ] | repeat repeat-times ] * Configure the input timeout processing method for the node.
Step 8. Command Configure the input timeout processing method for the node. timeout { end-call | goto-pre-node | goto-node node-id } [ expires seconds | media-play media-id [ play-times ] | repeat repeat-times ] * Remarks Optional. Not configured by default. NOTE: You can configure input timeout processing and input timeout processing methods as needed. If you do not configure those methods for a node, the system applies global processing methods.
Displaying and maintaining customizable IVR Task Command Remarks Display the IVR playing information. display voice ivr media-play [ | { begin | exclude | include } regular-expression ] Available in any view. Display IVR call information. display voice ivr call-info [ | { begin | exclude | include } regular-expression ] Available in any view. Display IVR media resource information. display voice ivr media-source [ | { begin | exclude | include } regular-expression ] Available in any view.
Configuration procedure 1. Configure Router A: # Configure POTS voice entity 100. 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.
# Configure global error processing and timeout processing methods: If the timeout value expires before the subscriber dials at Telephone A, Router B plays audio file timeout.wav. If the timeout value expires for four times, Router B terminates the call. { If the subscriber dials a wrong number at Telephone A, Router B plays the audio file input_error.wav. If the subscriber dials wrong numbers for three times, Router B terminates the call.
Configuration procedure 1. Configure Router A: For more information, see "Configure Router A:." 2. Configure Router B: # Configure Call node 10: { After the IVR access number 300 is matched, Router B plays the audio file welcome.wav { After the subscriber dials 500 at Telephone A, Router B executes the secondary call to Telephone B2.
Configuration procedure 1. Configure Router A: For more information, see "Configure Router A:." 2. Configure Router B: # Configure Call node 10: { After the IVR access number 300 is matched, the audio file welcome.wav is played. { After the subscriber dials 50 at Telephone A, the secondary call to Telephone B1 is executed. [RouterB-voice-ivr] node 10 call [RouterB-voice-ivr-node10] media-play 10001 [RouterB-voice-ivr-node10] call-normal matching For more information, see "Configure Router B:." 3.
# 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 2. Configure Router B: # Configure POTS voice entity 500.
After dialing the number 300 at Telephone A, the subscriber hears the voice prompts of the audio file welcome.wav. After that, if the subscriber dials 0 at Telephone A, and Telephone B1 will ring. Jump node configuration example Network requirements As shown in Figure 95, configure the IVR access number and customize Jump node functions on Router B.
# 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 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.
• If the timeout value expires before the subscriber dials at Telephone A, Router B plays the audio file timeout.wav. Figure 96 Network diagram Configuration procedure 1. Configure Router A: # Configure POTS voice entity 100.
[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 # 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.
[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 [RouterA-voice-dial-entity300] outband sip 2. Configure Router B: # Configure POTS voice entity 500.
[RouterB-voice-ivr] node 10 service [RouterB-voice-ivr-node10] operation 2 media-play 10004 1 [RouterB-voice-ivr-node10] operation 3 end-call [RouterB-voice-ivr-node10] select-rule operation-order 2 3 1 3. Verify configurations: After dialing the number 300 at Telephone A, the subscriber hears the voice prompts of the audio file bye.wav. After that, the call will be terminated.
system-view [RouterB] voice-setup [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] quit # Configure IVR voice entity 300 and specify node 1 as the root node.
# Configure Call node 10 to first play the audio file call.wav, and then if the subscriber dials 1 at Telephone A, originate a call to number 500. [RouterB-voice-ivr] node 10 call [RouterB-voice-ivr-node10] media-play 10005 2 force [RouterB-voice-ivr-node10] extension 1 call 500 [RouterB-voice-ivr-node10] quit # Configure Service node 20 to first play the audio file bye.wav, and then terminate the call.
Analysis After the subscriber presses the # key, the call jumps to node 10 that is not configured with any operation, and thus the call is terminated. Solution Configure functions for node 10: playing voice files and executing normal secondary call. Loopback node Symptom The subscriber dials the IVR access number 300 and presses the # key to jump to node 10, but the call is terminated.
[Sysname-voice-dial-entity300] quit [Sysname-voice-dial] quit [Sysname-voice] ivr-system [Sysname-voice-ivr] node 1 jump [Sysname-voice-ivr-node1] user-input # goto-node 2 [Sysname-voice-ivr-node1] quit [Sysname-voice-ivr] node 2 jump [Sysname-voice-ivr-node2] user-input # goto-node 3 [Sysname-voice-ivr-node2] quit [Sysname-voice-ivr] node 3 jump [Sysname-voice-ivr-node3] user-input # goto-node 4 [Sysname-voice-ivr-node3] quit [Sysname-voice-ivr] node 4 jump [Sysname-voice-ivr-node4] user-input # goto-node
[Sysname-voice-ivr-node1] extension 1201 call 7745231 Analysis When the subscriber dials 1201, the node executes an extension secondary call as soon as it receives the number 120. Solution There are some short numbers for special usage, such as 120, 110, and 114. Do not configure them as the prefixes of extension numbers. Otherwise, when subscribers dial extension numbers with these prefixes, the node executes the normal secondary call immediately.
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.
Figure 100 Protocols and standards that VoFR complies with Call flow in dynamic mode The following shows the call flow in the dynamic mode: 1. 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. 2. The calling party dials and the voice interface card collects and stores the dialed digits, that is, the called number. 3.
2. The calling party dials and the voice interface card collects and stores the dialed digits, that is, the called number. 3. Upon the completion of dialing, the voice gateway matches the called number against voice entities. 4. VoFR processes the call if a VoFR entity is matched. The processing method depends on the call mode configured for the VoFR entity. 5. In the FRF.
Configuring basic functions Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter voice dial program view. dial-program N/A 4. Create a VoFR entity and enter VoFR entity view. entity entity-number vofr N/A 5. Configure a match template for the VoFR entity. match-template match-string By default, no match template is configured for the VoFR entity. Configure the codec on basis of the priority levels.
NOTE: • In the dynamic mode, the DTMF transmission mode is determined by the configuration of the VoFR entity on the originating side. • In the FRF.11 trunk mode, the DTMF transmission mode is determined by the configurations of the VoFR entities on the originating and terminating sides. Enabling VAD Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter voice dial program view. dial-program N/A 4.
Step 5. Enter frame relay interface view Command Remarks interface serial interface-number N/A • Associate the frame relay class with a frame relay interface so that the settings of voice bandwidth can take effect on the frame relay interface: fr-class class-name 6. Associate the frame relay class with a frame relay interface or virtual circuit. • Associate the frame relay class with a frame relay virtual circuit: a. Enter interface fr dlci dlci-number DLCI view b.
• Configure POTS entities. • Configure VoFR entities. Configuring a call mode Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter voice dial program view. dial-program N/A 4. Enter VoFR entity view. entity entity-number vofr N/A 5. Configure a call mode. call-mode dynamic By default, the dynamic mode is adopted. 6. Configure a channel to the peer voice gateway.
Configuring VoFR packets to carry a timestamp Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter voice dial program view. dial-program N/A 4. Enter VoFR entity view. entity entity-number vofr N/A 5. Configure a match template for the VoFR entity. match-template match-string By default, no match template is configured for the VoFR entity. Optional. Configure VoFR packets to carry a timestamp. 6.
Configuring VoFR packets to carry a sequence number The terminating voice gateway determines whether there is any voice packet loss, duplicate voice packets, or out-of-sequence occurs according to sequence numbers. This helps for voice quality. 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 1. Enter system view.
Configuring PSTN-dialed number Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter voice dial program view. dial-program N/A 4. Enter VoFR entity view. entity entity-number vofr N/A 5. Configure a PSTN-dialed number in the FRF.11 trunk mode. trunk-id string By default, no PSTN-dialed number is configured in the FRF.11 trunk mode. Configuring call control protocol Step Command Remarks 1. Enter system view. system-view N/A 2.
To configure the trunk wait timer length in the FRF.11 trunk mode: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Configure the trunk wait timer length in the FRF.11 trunk mode. vofr frf11-timer time Optional. The default is 30 seconds. NOTE: All FRF.11 trunks use the same trunk wait timer length. Configuring VoFR packets to carry sequence number For details, see "Configuring VoFR packets to carry a sequence number.
system-view [RouterA] fr class vofr [RouterA-fr-class-vofr] voice bandwidth 32000 reserved [RouterA-fr-class-vofr] quit # 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.
[RouterB-voice-dial-entity10] match-template 0101001 [RouterB-voice-dial-entity10] address vofr-dynamic serial 1/0 100 [RouterB-voice-dial-entity10] quit # Configure the local POTS entity (07552001).
[RouterA-voice] dial-program [RouterA-voice-dial] entity 0755 vofr [RouterA-voice-dial-entity755] match-template 07552001 [RouterA-voice-dial-entity755] address vofr-dynamic serial 2/0 100 [RouterA-voice-dial-entity755] quit # Configure the POTS entity (0101001). [RouterA-voice-dial-entity755] entity 1001 pots [RouterA-voice-dial-entity1001] match-template 0101001 [RouterA-voice-dial-entity1001] line 3/0 2. Configure Router B according to the network requirements and network diagram in Figure 102. FRF.
[RouterA-fr-dlci-Serial2/0-100] vofr huawei-compatible dce [RouterA-fr-dlci-Serial2/0-100] quit [RouterA-Seria12/0] quit # Configure the VoFR entity (9) for the FRF.11 trunk.
Concurrent transmission of voice and data Network requirements As shown in Figure 104, Telephone A (010-1001) attached to voice Router A in City A directly communicates with Telephone B (0755-2001), which is attached to voice Router B in City B over a frame relay network. The PC in City A and the server in City B transmit data through these two routers. 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.
# Configure the VoFR entity (07552001). [RouterA] voice-setup [RouterA-voice] dial-program [RouterA-voice-dial] entity 0755 vofr [RouterA-voice-dial-entity755] match-template 07552001 [RouterA-voice-dial-entity755] address vofr-dynamic serial 1/0 100 [RouterA-voice-dial-entity755] 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 2/0 2.
# Configure the local POTS entity (07552001). [RouterB-voice-dial] entity 2001 pots [RouterB-voice-dial-entity2001] match-template 07552001 [RouterB-voice-dial-entity2001] line 2/0 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.
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.
When receiving an Authorization acknowledgment, the originating gateway sends a VoIP_Accounting_Start request (call segment 2) to the RADIUS server. 3. After receiving a VoIP_Accounting_Start acknowledgment (call segment 2) from the RADIUS server, the originating gateway originates a call to the terminated gateway over the IP network so as to set up a voice channel on the IP network side.
Accounting_Stop request to the RADIUS server, and releases the call upon receiving an Accounting_Stop acknowledgment. • start-no-ack—When the call setup begins, the voice gateway sends an Accounting_Start request to the RADIUS server, and directly connects the call without waiting for an Accounting_Start acknowledgment. If the voice gateway receives an Accounting_Start unacknowledgment from the RADIUS server after the call is connected, it immediately releases the call.
Voice prompt Voice prompts in Chinese and English are available in the card number/password process and caller number process with IVR. 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. • Voice port number. • IP address of the peer voice gateway.
• 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. • Configure the number of redial attempts. • Enable the language selection function. The configuration tasks for the one-stage dialing process differ from those for the two-stage dialing process.
Task Remarks 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. Configuring the number of redial attempts Optional. Enabling the language selection function Optional. Configuring voice RADIUS Configuring accounting method The voice gateway processes RADIUS accounting requests and responses in multiple ways.
authorization, and accounting functions on the voice gateway. This rule applies to one-stage dialing users and two-stage dialing users. Calls will be released in the case of an accounting failure. • Configuration prerequisites A voice interface card (for example, an FXS interface card) is inserted in the router. Configuration procedure To enable the accounting function for one-stage dialing users: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view.
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. Configuration prerequisites A voice interface card (for example, an FXS interface card) is inserted in the router. The authentication function is enabled for one-stage dialing users. Authentication is a prerequisite for authorization. The authentication function must be enabled before the authorization function.
Configuration procedure To configure a rule for saving CDRs: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter voice AAA client view. aaa-client N/A 4. Configure a rule for saving CDRs. cdr { buffer size-number | duration timer-length | threshold percentage } The default is 50 for the size-number argument, 86,400 seconds (namely, 24 hours) for the timer-length argument, and 80 for the percentage argument.
In the caller number process with IVR, a user can select a language in which prompt tones are played. After the user selects a language, the voice gateway plays tones in the selected language to prompt for a called number. • Configuration prerequisites You have configured an access number. Configuration procedure To configure a two-stage dialing process: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter dial program view.
client can communicate with each other at the network layer, and that a list of corresponding two-stage dialing users as well as authentication policies has been configured on the RADIUS server. The authentication function for two-stage dialing users is enabled for a specific access number, while the authentication function for one-stage dialing users is enabled in voice AAA client view. Configuration prerequisites You have configured an access number.
Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter dial program view. dial-program N/A 4. Enter access number view. gw-access-number access-number N/A 5. Enable the authorization function for two-stage dialing users. authorization Disabled by default.
Configuration procedure To configure the timeout interval for a user to dial the next digit in a two-stage dialing process: Step Command Remarks 1. Enter system view. system-view N/A 2. Enter voice view. voice-setup N/A 3. Enter dial program view. dial-program N/A 4. Enter access number view. gw-access-number access-number N/A 5. Configure the timeout interval for a user to dial the next digit in a two-stage dialing process.
NOTE: If a user is required to press the dial terminator # after dialing a card number but fails to do so, the voice gateway will prompt timeout and require the user to redial the card number. This rule also applies to a password. 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.
Enabling the language selection function The language selection function applies to only the caller number process with IVR. With the language selection function enabled, the voice gateway will play tones to prompt for a language first and then a called number after a user dials an access number. Configuration prerequisites You have configured an access number and entered access number view. Configuration procedure To configure the language options: Step Command Remarks 1. Enter system view.
Voice RADIUS configuration example Card number/password process configuration Network requirements Local telephone users are connected to voice subscriber lines of routers directly or through PBXs. The routers are connected to the IP network through WAN ports. The RADIUS server is deployed on the IP network. The number of digits in a card number is 10 and that in a password is 4. The access number is 12345. Authentication, authorization, and accounting are required for users who dial this access number.
[RouterA-radius-sch1] primary accounting 1.1.1.3 1813 # Configure RADIUS packets to carry unqualified usernames. [RouterA-radius-sch1] user-name-format without-domain # Configure the server type to a RADIUS server based on an extended protocol. [RouterA-radius-sch1] server-type extended # Configure the RADIUS scheme in the default domain.
[RouterB] domain system [RouterB-isp-system] authentication voip radius-scheme sch1 [RouterB-isp-system] authorization voip radius-scheme sch1 [RouterB-isp-system] accounting voip radius-scheme sch1 [RouterB-isp-system] quit # Configure the access number and set the dialing process to the card number/password process.
6. If the authorization fails, check that call or access restriction is not set for the IP phone service on the RADIUS SERVER. 7. Check the log generated on the RADIUS SERVER and remove the fault according to the errors. Symptom 2 RADIUS authentication/authorization requests of voice users are always rejected. Follow the steps below to remove the fault: 1. Follow the steps introduced in symptom 1 to see if the fault can be removed. 2.
Support and other resources Contacting HP For worldwide technical support information, see the HP support website: http://www.hp.
Conventions This section describes the conventions used in this documentation set. Command conventions Convention Description Boldface Bold text represents commands and keywords that you enter literally as shown. Italic Italic text represents arguments that you replace with actual values. [] Square brackets enclose syntax choices (keywords or arguments) that are optional. { x | y | ... } Braces enclose a set of required syntax choices separated by vertical bars, from which you select one.
Network topology icons Represents a generic network device, such as a router, switch, or firewall. Represents a routing-capable device, such as a router or Layer 3 switch. Represents a generic switch, such as a Layer 2 or Layer 3 switch, or a router that supports Layer 2 forwarding and other Layer 2 features. Represents an access controller, a unified wired-WLAN module, or the switching engine on a unified wired-WLAN switch. Represents an access point.
Index ABCDEFHIMOPRSTV Configuring basic parameters for a T1 voice interface,65 A Advantages,286 Configuring basic parameters for an E1 voice interface,63 Analog voice subscriber line configuration examples,55 Configuring call authority control,103 B Configuring call authority control,182 Background,196 Configuring call backup,238 Configuring call forwarding,234 Binding an FXS voice subscriber line to an FXO voice subscriber line,43 Configuring call hold,233 Binding logical voice subscriber line t
Configuring outbound SIP proxy server information for a SIP UA,147 Displaying and maintaining the H.
Mirroring PCM, RTP, or voice command data on an analog voice subscriber line,54 Specifying the URL scheme for outgoing SIP calls,152 O Support for basic QSIG call,137 Specifying the ID for a media resource,290 Support for transport layer protocols,135 Overview,266 T Overview,8 Overview,286 Troubleshooting,176 Overview,312 Troubleshooting analog voice subscriber line configuration,59 Overview,99 Troubleshooting digital voice subscriber line configuration,98 Overview,130 P Troubleshooting H.
