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Cisco Module FXO Analog Personality f MC3810

1. e amp m immediate start e amp m melcas delay e amp m melcas immed e amp m melcas wink e amp m wink start ext sig fgd eana fxo ground start fxo loop start fxo melcas fxs ground start fxs loop start fxs melcas r2 analog r2 digital r2 pulse Cisco AS5300 Universal Access Servers T1 Router config controller dsO group ds0 group no timeslots timeslot list service data fax voice type e amp m fgb e amp m fgd e amp m immediate start fxs ground start fxs loop start fgd eana fgd os vl itu sas ground start sas loop start none Cisco AS5300 Universal Access Servers E1 Router config controller dsO group ds0 group no timeslots timeslot list type none p7 r2 analog r2 digital r2 lsv181 digital r2 pulse Cisco AS5800 Universal Access Servers T1 Router config controller ds0 group ds0 group no timeslots timeslot list type e amp m fgb e amp m fgd e amp m immediate start fxs ground start fxs loop start fgd eana rl itu rl modified rl turkey sas ground start sas loop start none Cisco AS5800 Universal Access Servers E1 Voice Ports Router config controller dsO group ds0 group no timeslots timeslot list type e amp m fgb e amp m fgd e amp m immediate start fxs ground start fxs loop start p7 r2 analog r2 digital r2 pulse sas ground start sas loop start none Cisco 7200 and 7500 Series Series Routers T1 and E1 Voi
2. Purpose Cisco 2600 and 3600 Series Routers Analog Voice Ports Router test voice port slot subunit port detector m lead battery reversal loop current ring tip ground ring ground ring trip disable Cisco 2600 and 3600 Series Routers Digital Voice Ports Router test voice port slot port ds0 group detector m lead battery reversal loop current ring tip ground ring ground ring trip disable Cisco MC3810 Multiservice Concentrators Analog Voice Ports Router test voice port slot port detector m lead battery reversal loop current ring tip ground ring ground ring trip disable Cisco MC3810 Multiservice Concentrators Digital Voice Ports Router test voice port slot ds0 group detector m lead battery reversal loop current ring tip ground ring ground ring trip disable Identifies the voice port on which you want to end the test Enter a keyword for the detector under test and the keyword disable to end the forced state Note For each signaling type E amp M FXO FXS only the applicable keywords are displayed The disable keyword is displayed only when a detector is in the forced state Cisco IOS Voice Video and Fax Configuration Guide Configuring Voice Ports E Troubleshooting Analog and Digital Voice Port Configurations Loopback Function Tests To establish loopbacks on a voice port use the following commands in privileged EXEC mode Command Purpo
3. in a cadence pattern with up to four on off time cycles In the following procedure the following commands were not supported until Cisco IOS Release 12 2 2 T freq max deviation freq max power freq min power freq power twist and freq max delay Cisco IOS Voice Video and Fax Configuration Guide E Configuring Voice Ports W Configuring Digital Voice Ports To create a voice class that defines the specific tone or tones to be detected and then apply the voice class to the voice port use the following commands beginning in global configuration mode Command Purpose Step1 Router config voice class dualtone tag Creates a voice class for defining one tone detection pattern The range for the tag number is from 1 to 10000 The tag number must be unique on the router For more information about configuring voice classes see the Configuring Dial Plans Dial Peers and Digit Manipulation chapter in this configuration guide Step2 Router config voice class freq pair tone id Specifies the two frequencies in Hz for a tone to oS ap a SLL be detected or one frequency if a nondual tone is to be detected If the tone to be detected contains only one frequency enter 0 for frequency 2 The arguments are as follows tone id Ranges from to 16 There is no default frequency 1 and frequency 2 Ranges from 300 to 3600 or you can enter 0 for frequency 2 There is no default Note Repeat this
4. MC3810 Multiservice Concentrators Analog Voice Ports Router test voice port slot port detector relay e lead loop ring ground battery reversal power denial ring tip ground on off Cisco MC3810 Multiservice Concentrators Digital Voice Ports Router test voice port slot ds0 group relay e lead loop ring ground battery reversal power denial ring tip ground on off Step2 Cisco 2600 and 3600 Series Routers Analog Voice Ports Identifies the voice port on which you want to end Router test voice port slot subunit port relay the test Enter a keyword for the relay under test e lead loop ring ground battery reversal and the keyword disable to end the forced state power denial ring tip ground disable Note For each signaling type E amp M FXO Cisco 2600 and 3600 Series Routers Digital Voice Ports FXS only the applicable keywords are displayed The disable keyword is Router test voice port slot port ds0 group relay displayed only when a relay is i th e lead loop ring ground battery reversal power denial ring tip ground disable forced state Cisco M C3810 M ultiservice Concentrators Analog Voice Ports Router test voice port slot port detector relay e lead loop ring ground battery reversal power denial ring tip ground disable Cisco M C3810 Multiservice Concentrators Digital Voice Ports Router test voice port slot ds0 group relay e lead loop ring g
5. a VCM in the same chassis For more information refer to the following publications Cisco MC3810 Multiservice Concentrator Hardware Installation Guide Overview of the Cisco MC3810 Series Configuring Cisco MC3810 Series Concentrators to Use High Performance Compression Modules Cisco AS5300 Universal Access Server The Cisco AS5300 Universal Access Server includes three expansion slots One slot is for either an Octal T1 E1 PRI feature card eight ports or a Quad T1 E1 PRI feature card four ports and the other two can be used for voice fax or modem feature cards Because a single voice fax feature card VFC can support up to 48 T1 or 60 E1 voice calls the Cisco AS5300 Voice Gateway system can support a total of 96 or 120 simultaneous voice calls The use of VFCs requires Cisco IOS release 12 0 2XH or later Cisco AS5300 VFCs are coprocessor cards each with a powerful reduced instruction set computing RISC engine and dedicated high performance DSPs to ensure predictable real time voice processing The design couples this coprocessor with direct access to the Cisco AS5300 routing engine for streamlined packet forwarding For more information refer to the following publications Cisco AS5300 Chassis Installation Guide Cisco AS5300 Module Installation Guide Cisco AS5800 Universal Access Server The Cisco AS5800 Universal Access Server consists of two primary system components the Cisco 5814 dial shelf DS which hold
6. and transMit Slot is 1 Sub unit is 0 Port is 1 lt lt voice port 1 0 1 Type of VoicePort is E amp M Operation State is DORMANT Administrative State is UP No Interface Down Failure Description is not set Noise Regeneration is enabled Non Linear Processing is enabled Music On Hold Threshold is Set to 38 dBm In Gain is Set to 0 dB Out Attenuation is Set to 0 dB Echo Cancellation is enabled Echo Cancel Coverage is set to 8 ms Connection Mode is normal Connection Number is not set Initial Time Out is set to 10 s Interdigit Time Out is set to 10 s Region Tone is set for US mi Cisco 10S Voice Video and Fax Configuration Guide __Configuring Voice Ports Verifying Analog and Digital Voice Port Configurations show controller Command Examples In the following sections output examples of the following types are shown Cisco 3600 Series Router T1 Controller page 103 Cisco MC3810 Multiservice Concentrator E1 Controller page 103 Cisco AS5800 Universal Access Server T1 Controller page 103 Cisco 3600 Series Router T1 Controller The following output is from a Cisco 3600 series router with a T1 controller Router show controller T1 1 1 0 T1 1 0 0 is up Applique type is Channelized T1 Cablelength is long gain36 Odb No alarms detected alarm trigger is not set Framing is ESF Line Code is B8ZS Clock Source is Line Data in current interval 180 seconds elapsed 0 Line Code Violations 0 Path Code V
7. excellent clock source and the PLL usually requires that the CO provide that source and a PBX usually must receive clocking from the other voice port Figure 19 Dual E1 Ports Receiving Clocking from the Line lt Clock ae x a lt Clock PBX 26921 Looped time The following configuration sets up this clocking method controller El 1 0 lt lt description connected to the CO framing crc4 linecoding hdb3 clock source line primary ds0 group timeslots 1 15 type e amp m wink start 1 E Cisco IOS Voice Video and Fax Configuration Guide __Configuring Voice Ports amp Configuring Digital Voice Ports W controller El 1 1 lt lt description connected to the PBX framing crc4 linecoding hdb3 clock source line ds0 group timeslots 1 15 type e amp m wink start The clock source line primary command tells the router to use this voice port to drive the PLL All other voice ports configured as clock source line are then put into an implicit loop timed mode If the primary voice port fails or goes down the other voice port instead receives the clock that drives the PLL In this configuration port 1 1 might see controlled slips but these should not force it down This method prevents the PBX from seeing slips Note When terminating two T1 E1 lines on a two port interface card such as the VWIC 2MFT if both controllers are set for line clocking but the lines are not within clocking t
8. faxes over an IP network To make a voice connection the router must have a supported VIC installed The Cisco 1750 router supports two slots for either WAN interface cards WICs or VICs and supports one VIC only slot For analog connections two port VICs are available to support FXO FXS and E amp M signaling VICs provide direct connections to telephone equipment analog phones analog fax machines key systems or PBXs or to a PSTN For more information refer to the Cisco 1750 Voice over IP Quick Start Guide Cisco IOS Voice Video and Fax Configuration Guide E Configuring Voice Ports HE Analog Voice Ports Configuration Task List Cisco 2600 Series and Cisco 3600 Series Routers The Cisco 2600 and 3600 series routers are modular multifunction platforms that combine dial access routing local area network to local area network LAN services and multiservice integration of voice video and data in the same device Voice network modules installed in Cisco 2600 series or Cisco 3600 series routers convert telephone voice signals into data packets that can be transmitted over an IP network The voice network modules have no connectors VICs installed in the network modules provide connections to the telephone equipment or network VICs work with existing telephone and fax equipment and are compatible with H 323 standards for audio and video conferencing The Cisco 2600 series router can house one network module In the Cisco
9. from 6 to 14 The default is 0 Router config voiceport output attenuation value Specifies the amount of attenuation in decibels at the transmit side of the interface decreasing the signal A system wide loss plan can be implemented using the input gain and output attenuation commands The default value for this command assumes that a standard transmission loss plan is in effect meaning that normally there must be 6 dB attenuation between phones The value argument is any integer from 6 to 14 The default is 0 Router config voiceport impedance 600c 600r complex1 Specifies the terminating impedance of a voice port interface which needs to match the specifications from the specific telephony system to which it is connected complex2 600c Specifies 600 ohms complex 600r Specifies 600 ohms real 900c Specifies 900 ohms complex complex1 Specifies Complex 1 complex2 Specifies Complex 2 The default is 600r E Cisco IOS Voice Video and Fax Configuration Guide __Configuring Voice Ports Command Verifying Analog and Digital Voice Port Configurations W Purpose Step4 Router config voiceport loss plan plan1 plan2 Cisco MC3810 multiservice concentrators FXO plan5 plan6 plan7 plans plan9 or FXS analog voice ports only Specifies the analog to digital gain offset loss plan For definitions of each plan see the Cisco IOS Voice V
10. ground start differs from digital E amp M because the A and B bits do not track each other as they do in digital E amp M signaling that is A is not necessarily equal to B When the CO delivers a call it seizes a channel goes off hook by setting the A bit to 0 The CO equipment also simulates ringing by toggling the B bit The terminating equipment goes off hook when it is ready to answer the call Digits are usually not delivered for incoming calls E amp M connections can use one of three different signaling types to acknowledge on hook and off hook states wink start immediate start and delay start E amp M wink start is usually preferred but not all COs and PBXs can handle wink start signaling The E amp M connection between the router and switch CO or PBX must match the CO or PBX E amp M signaling type or calls cannot be connected properly E amp M signaling is normally used for trunks It is normally the only way that a CO switch can provide two way dialing with Direct Inward Dialing DID In all the E amp M protocols off hook is indicated by A B 1 and on hook is indicated by A B 0 robbed bit signaling If dial pulse dialing is used the A and B bits are pulsed to indicate the addressing digits The are several further important subclasses of E amp M robbed bit signaling E amp M Wink Start Feature Group B In the original wink start handshaking protocol the terminating side responds to an off hook from the originating side w
11. is the initial timeout duration A valid entry is an integer from 0 to 120 The default is 10 Step3 Router config voiceport timeouts interdigit seconds Configures the number of seconds that the system waits after the caller has input the initial digit or a subsequent digit of the dialed string If the timeout ends before the destination is identified a tone sounds and the call ends This value is important when using variable length dial peer destination patterns dial plans The seconds argument is the interdigit timeout wait time in seconds A valid entry is an integer from 0 to 120 The default is 10 Step4 Router config voiceport timeouts ringing seconds Specifies the duration that the voice port allows infinity ringing to continue if a call is not answered The keyword and argument are as follows infinity Indicates ringing should continue until the caller goes on hook seconds Specifies the number of seconds to allow ringing without answer The range is from 5 to 60000 The default is 180 Step5 Router config voiceport timeouts wait release Specifies the duration that a voice port stays in the seconds infinity call failure state while the Cisco device sends a busy tone reorder tone or an out of service tone to the port The keyword and argument are as follows infinity Indicates the voice port should not be released as long as the call failure state remains seconds Spec
12. peer or voice port Limit is five 5 strings All strings must be valid E 164 numbers up to 32 digits in length Router config voiceport calling number outbound null Cisco AS5300 universal access server only Suppresses ANI No ANI is passed when this voice port is selected Disconnect Supervision Commands PBX and PSTN switches use several different methods to indicate that a call should be disconnected because one or both parties have hung up The commands in this section are used to configure the router to recognize the type of signaling in use by the PBX or PSTN switch connected to the voice port These methods include the following Battery reversal disconnect Battery denial disconnect e Supervisory tone disconnect STD Battery reversal occurs when the connected switch changes the polarity of the line in order to indicate changes in call state such as off hook or in this case call disconnect This is the signaling looked for when the battery reversal command is enabled on the voice port which is the default configuration Battery denial sometimes called power denial occurs when the connected switch provides a short approximately 600 ms interruption of line power to indicate a change in call state This is the signaling looked for when the supervisory disconnect command is enabled on the voice port which is the default configuration Supervisory tone disconnect occurs when the connected switch provides a
13. port is from 0 to 11 Step3 T1 Selects frame type for T1 or E1 line Ro ter config c ntroller a framingo s 85f The keywords and arguments are as follows El T1 lines Router config controller framing crc4 no crc4 v 268 super farie australia SUP esf extended super frame Ellines crce4 Provides 4 bits of error protection no cre4 Disables erc4 australia Optional Specifies the E1 frame type used in Australia The default for T1 is sf The default for E1 is erc4 Hl Cisco 10S Voice Video and Fax Configuration Guide __Configuring Voice Ports Configuring Digital Voice Ports W Command Purpose Step 4 Router config controller clock source line primary Configures the clock source secondary internal The keywords and arguments are as follows line Specifies that the PLL on this port derives clocking from the external source to which the port is connected generally the CO primary Optional Specifies that the PLL on this port derives clocking from the external source and puts the other port generally connected to the PBX into looped time mode Both ports are configured with line but only the port connected to the external source is configured with primary secondary Optional Indicates a backup external source for clocking if the primary clocking shuts down Configure the clock source line secondary command on the controller that ha
14. to Configure Digital Voice Ports Gather the following information about the telephony network connection that the voice port will be making Line interface T1 or El Signaling interface FXO FXS or E amp M If the interfaces are Primary Rate Interface PRI or BRI see the Configuring ISDN Interfaces for Voice chapter in this configuration guide and Cisco IOS Terminal Services Configuration Guide Line coding AMI or B8ZS for T1 and AMI or HDB3 for El Framing format SF D4 or ESF for T1 and CRC4 or no CRC4 for E1 Number of channels Table 8 describes voice port hardware configurations for various platforms After the controllers have been configured the show voice port summary command can also be used to determine available voice port numbers If the show voice port command and a specific port number is entered the default voice port configuration for that port displays Table 8 Router Platform Digital Voice Slot Port Designations Voice Hardware Slot Number Port Number Cisco 2600 series Digital T1 E1 Packet Voice Trunk Network Module NM HDV with VWIC 1MFT or VWIC 2MFT One network module can be installed in a Cisco 2600 series router slot is the router location of the voice module 1 port is the VWIC location in the network module Oto 1 Cisco 3600 series Digital T1 E1 Packet Voice Trunk Network Module NM HDV with VWIC 1MFT or VWIC 2MFT One network m
15. to change some of the voice port values to improve voice quality or to match parameters on proprietary PBXs to which you are connecting use the commands in the current section and also in the Fine Tuning Analog and Digital Voice Ports section on page 78 After the voice port has been configured make sure that the ports are operational by following the steps described in the following sections Verifying Analog and Digital Voice Port Configurations page 97 Troubleshooting Analog and Digital Voice Port Configurations page 108 For more information on these and other voice port commands see the Cisco IOS Voice Video and Fax Command Reference The commands keywords and arguments that you are able to use may differ slightly from those presented here based on your platform Cisco IOS release and configuration When in doubt use Cisco IOS command help command to determine the syntax choices that are available To configure basic analog voice port parameters on Cisco 1750 Cisco 2600 series Cisco 3600 series and Cisco MC3810 routers use the following commands beginning in global configuration mode Purpose Cisco 1750 and MC3810 Enters voice port configuration mode Router config voice port slot port The arguments are as follows Cisco 2600 and 3600 series slot Specifies the number of the router slot Router config voice port slot subunit port where the voice network module is installed Cis
16. types are shown Cisco 3600 Series Router Analog E amp M Voice Port page 99 Cisco 3600 Series Router Analog FXS Voice Port page 100 Cisco 3600 Series Router Digital E amp M Voice Port page 101 Cisco AS5300 Universal Access Server T1 CAS Voice Port page 101 Cisco 7200 Series Router Digital E amp M Voice Port page 102 Cisco 3600 Series Router Analog E amp M Voice Port The following output is from a Cisco 3600 series router analog E amp M voice port Router show voice port 1 0 0 E amp M Slot is 1 Sub unit is 0 Port is 0 Type of VoicePort is E amp M Operation State is unknown Administrative State is unknown The Interface Down Failure Cause is 0 Alias is NULL Cisco IOS Voice Video and Fax Configuration Guide m Configuring Voice Ports W Verifying Analog and Digital Voice Port Configurations Noise Regeneration is disabled Non Linear Processing is disabled Music On Hold Threshold is Set to 0 dBm In Gain is Set to 0 dB Out Attenuation is Set to 0 dB Echo Cancellation is disabled Echo Cancel Coverage is set to 16ms Connection Mode is Normal Connection Number is Initial Time Out is set to 0 s Interdigit Time Out is set to 0 s Analog Info Follows Region Tone is set for northamerica Currently processing none Maintenance Mode Set to None not in mtc mode Number of signaling protocol errors are 0 Voice card specific Info Follows Signal Type is wink start Operation Type is 2 wire mpeda
17. 0x5C6705AF 0x0 0x385722B0 RemoteIPAddress 171 68 235 18 RemoteUDPPort 16580 RoundTripDelay 29 ms SelectedQoS best effort tx_DtmfRelay inband voice SessionProtocol cisco SessionTarget ipv4 171 68 235 18 OnTimeRvPlayout 63690 GapFillWithSilence 0 ms GapFillWithPrediction 180 ms GapFillWithInterpolation 0 ms GapFillWithRedundancy 0 ms HiWaterPlayoutDelay 70 ms LoWaterPlayoutDelay 30 ms ReceiveDelay 40 ms LostPackets 0 ms EarlyPackets 1 ms LatePackets 18 ms VAD disabled CoderTypeRate g729r8 CodecBytes 20 cvVoIPCallHistoryIcpif 0 SignalingType cas show Call history voice Command Example The following output is from a Cisco 7200 series router Router show call history voice GENERIC SetupTime 94893250 ms Index 450 PeerAddress 52258 PeerSubAddress PeerId 50000 Cisco IOS Voice Video and Fax Configuration Guide __Configuring Voice Ports Verifying Analog and Digital Voice Port Configurations PeerlfIndex 35 LogicalIfIndex 0 DisconnectCause 10 DisconnectText normal call clearing Connect Time 948 93780 DisconectTime 95015500 CallOrigin 1 ChargedUnits 0 InfoType 2 TransmitPackets 32258 TransmitBytes 645160 ReceivePackets 20061 ReceiveBytes 401220 VOIP ConnectionId 0x142E62FB 0x5C6705B3 0x0 0x388F851C RemoteIPAddress 171 68 235 18 RemoteUDPPort 16552 RoundTripDelay 23 ms SelectedQoS best effort tx_DtmfRelay inband voice SessionProtocol cisco SessionT
18. 1 or El packet voice trunk network module can be installed in a Cisco 2600 series router or in a Cisco 3620 router A Cisco 3640 router can support three network modules and a Cisco 3660 router can support up to six network modules Cisco IOS Voice Video and Fax Configuration Guide __Configuring Voice Ports Configuring Digital Voice Ports W The MFT VWICs that are used in the packet voice trunk network modules are available in one and two port configurations for T1 and for E1 and in two port configurations with drop and insert capability for T1 and E1 MFTs support the following kinds of traffic Data As WICs for T1 or El applications including fractional data line use the T1 version includes a fully managed DSU CSU and the E1 version includes a fully managed DSU e Packet voice As VWICs included with the digital T1 or El packet voice trunk network module to provide connections to PBXs and COs the MFTs enable packet voice applications e Multiplexed voice and data Some two port T1 or E1 VWICs can provide drop and insert multiplexing services with integrated DSU CSUs For example when used with a digital T1 packet voice trunk network module drop and insert allows 64 kbps DSO channels to be taken from one T1 and digitally cross connected to 64 kbps DSO channels on another T1 Drop and insert sometimes called TDM cross connect uses circuit switching rather than the DSPs that VoIP technology employs Drop and insert is d
19. 12 Step 13 Step 14 Step 15 Step 16 Command Purpose Router config voiceport timing interdigit milliseconds Specifies the DTMF interdigit duration in milliseconds Valid entries are from 50 to 500 The default is 100 Router config voiceport timing percentbreak percent Cisco MC3810 multiservice concentrators FXO and E amp M ports only Specifies the percentage of the break period for the dialing pulses if different from the default The range is from 20 to 80 The default is 50 Router config voiceport timing pulse pulses per second FXO and E amp M only Specifies the pulse dialing rate in pulses per second Valid entries are from 10 to 20 The default is 20 Router config voiceport timing pulse digit milliseconds FXO only Configures the pulse digit signal duration The range of the pulse digit signal duration is from 10 to 20 The default is 20 Router config voiceport timing pulse interdigit FXO and E amp M only Specifies pulse dialing interdigit timing in milliseconds Valid entries are from 100 to 1000 The default is 500 Router config voiceport timing wink duration milliseconds E amp M only Specifies maximum wink signal duration in milliseconds for a wink start signal Valid entries are from 100 to 400 The default is 200 Router config voiceport timing wink wait milliseconds E amp M only Specifies maximum wink wait duration i
20. 24 full duplex channels or timeslots and an E1 line contains 30 The signal on each channel is transmitted at 64 kbps a standard known as digital signal 0 DSO the channels are known as DSO channels The ds0 group command creates a logical voice port a DSO group from some or all of the DSO channels which allows you to address those channels easily as a group in voice port configuration commands Digital voice ports are found at the intersection of a packet voice network and a digital circuit switched telephone network The digital voice port interfaces that connect the router or access server to T1 or E1 lines pass voice data and signaling between the packet network and the circuit switched network Signaling is the exchange of information about calls and connections between two ends of a communication path For instance signaling communicates to the call s end points whether a line is idle or busy whether a device is on hook or off hook and whether a connection is being attempted An end point can be a CO switch a PBX a telephony device such as a telephone or fax machine or a voice equipped router acting as a gateway There are two aspects to consider about signaling on digital lines one aspect is the actual information about line and device states that is transmitted and the second aspect is the method used to transmit the information on the digital lines The actual information about line and device states is communicated over digit
21. 3600 series the Cisco 3620 router has slots for up to two network modules the Cisco 3640 router has slots for up to four network modules and the Cisco 3660 router has slots for up to six network modules Typically one of the slots is used for LAN connectivity For analog telephone connections low density voice fax network modules that contain either one or two VIC slots are installed in the network module slots Each VIC is specific to a particular telephone signaling interface FXS FXO or E amp M therefore the VIC determines the type of signaling on that module For more information refer to the following Cisco 2600 Series Hardware Installation Guide Cisco 3600 Series Hardware Installation Guide Cisco Network Module Hardware Installation Guide Cisco M C3810 Multiservice Concentrator To support analog voice circuits a Cisco MC3810 multiservice concentrator must be equipped with an AVM which supports six analog voice ports By installing specific signaling modules known as analog personality modules APMs the analog voice ports may be equipped for the following signaling types in various combinations FXS FXO and E amp M For FXS the analog voice ports use an RJ 11 connector interface to connect to analog telephones or fax machines two wire or to a key system four wire For FXO the analog voice ports use an RJ 11 physical interface to connect to a CO trunk For E amp M connections the analog voice ports use an RJ 1C
22. 600 and 3600 Series Routers Digital Voice Ports test conaitionistalreadyc aeayated Router test voice port slot port ds0 group inject tone disable Cisco MC3810 Multiservice Concentrators Analog Voice Ports Router test voice port slot port detector inject tone disable Cisco MC3810 Multiservice Concentrators Digital Voice Ports Router test voice port slot ds0 group inject tone disable Cisco IOS Voice Video and Fax Configuration Guide B vc 113 Configuring Voice Ports E Troubleshooting Analog and Digital Voice Port Configurations Relay Related Function Tests To test relay related functions on a voice port use the following commands in privileged EXEC mode Command Purpose Step1 Cisco 2600 and 3600 Series Routers Analog Voice Ports Identifies the voice port you want to test Enter a Router test voice port slot subunit port relay keyword for the relay under test and specify e lead loop ring ground battery reversal whether to force it to the on or off state power denial ring tip ground on off Note For each signaling type E amp M FXO Cisco 2600 and 3600 Series Routers Digital Voice Ports FXS only the applicable keywords are displayed The disable keyword is Router test voice port slot port ds0 group relay displ d 1 h l aan A e lead loop ring ground battery reversal Ispiay CO OD Y Wena TE AY AS nL Uae power denial ring tip ground on off forced state Cisco
23. Access Servers Platform Analog Digital Cisco 803 and 804 Yes No Cisco 1750 Yes No Cisco 2600 series Yes Yes Cisco 3600 series Yes Yes Cisco MC3810 Yes Yes Cisco AS5300 No Yes Cisco AS5800 No Yes Cisco 7200 series No Yes Cisco 7500 series No Yes Telephony Signaling Interfaces Voice ports on routers and access servers physically connect the router or access server to telephony devices such as telephones fax machines PBXs and PSTN central office CO switches These devices may use any of several types of signaling interfaces to generate information about on hook status ringing and line seizure The router s voice port hardware and software need to be configured to transmit and receive the same type of signaling being used by the device with which they are interfacing so that calls can be exchanged smoothly between the packet network and the circuit switched network The signaling interfaces discussed in this chapter include foreign exchange office FXO foreign exchange station FXS and receive and transmit E amp M which are types of analog interfaces Some digital connections emulate FXO FXS and E amp M interfaces and they are discussed in the second half of this chapter It is important to know which signaling method the telephony side of the connection is using and to match the router configuration and voice interface hardware to that signaling method The next three illustrations show h
24. BX is detected by the digital signal processor DSP the analog FXO port goes on hook This feature prevents an analog FXO port from remaining in an off hook state after an incoming call is ended FXO supervisory disconnect tone enables interoperability with PSTN and PBX systems whether or not they transmit supervisory tones Note 2 This feature applies only to analog FXO ports with loop start signaling on the Cisco 2600 and 3600 series routers and on Cisco MC3810 multiservice concentrators with high performance compression modules HCMs To configure a voice port to detect incoming tones you need to know the parameters of the tones expected from the PBX or PSTN Then create a voice class that defines the tone detection parameters and finally apply the voice class to the applicable analog FXO voice ports This procedure configures the voice port to go on hook when it detects the specified tones The parameters of the tones need to be precisely specified to prevent unwanted disconnects due to detection of nonsupervisory tones or noise A supervisory disconnect tone is normally a dual tone consisting of two frequencies however tones of only one frequency can also be detected Use caution if you configure voice ports to detect nondual tones because unwanted disconnects can result from detection of random tone frequencies You can configure a voice port to detect a tone with one on off time cycle or you can configure it to detect tones
25. Configuring Voice Ports Voice ports are found at the intersections of packet based networks and traditional telephony networks and they facilitate the passing of voice and call signals between the two networks Physically voice ports connect a router or access server to a line from a circuit switched telephony device in a PBX or the public switched telephone network PSTN Basic software configuration for voice ports describes the type of connection being made and the type of signaling to take place over this connection Additional commands provide fine tuning for voice quality enable special features and specify parameters to match those of proprietary PBXs This chapter includes the following sections Voice Port Configuration Overview page 36 Analog Voice Ports Configuration Task List page 40 Configuring Digital Voice Ports page 54 Fine Tuning Analog and Digital Voice Ports page 78 Verifying Analog and Digital Voice Port Configurations page 97 Troubleshooting Analog and Digital Voice Port Configurations page 108 Not all voice port commands are covered in this chapter Some are described in the Configuring Trunk Connections and Conditioning Features chapter or the Configuring ISDN Interfaces for Voice chapter in this configuration guide The voice port configuration commands included in this chapter are fully documented in the Cisco IOS Voice Video and Fax Command Reference To identify the hardware platfor
26. Maximum Playout Delay is 160 milliseconds Analog Info Follows Region Tone is set for northamerica Currently processing Voice Maintenance Mode Set to None not in mtc mode Number of signaling protocol errors are 0 Impedance is set to 600r Ohm Analog interface A D gain offset Analog interface D A gain offset 3 dB 3 dB Voice card specific Info Follows Signal Type is loopStart Ring Frequency is 20 Hz Hook Status is On Hook Ring Active Status is inactive Ring Ground Status is inactive Tip Ground Status is active Digit Duration Timing is set to Ring Cadence are 20 40 100 ms InterDigit Duration Timing is set to 100 ms 100 msec InterDigit Pulse Duration Timing is set to 500 ms Cisco 3600 Series Router Digital E amp M Voice Port The following output is from a Cisco 3600 series router digital E amp M voice port Router show voice port 1 0 1 receEive and transMit Slot is 1 Type of VoicePort is E amp M Operation State is DORMANT Administrative State is UP No Interface Down Failure Description is not set Noise Regeneration is enabled Non Linear Processing is enabled Sub unit is 0 Port is 1 Music On Hold Threshold is Set to 38 dBm In Gain is Set to 0 dB Out Attenuation is Set to 0 dB Echo Cancellation is enabled Echo Cancel Coverage is set to 8 ms Connection Mode is normal Connection Number is not set Initial Time Out is set to 10 s Interdigit Time Out is set to 10 Region Tone i
27. SP voice channels are in the busy state codec complexity cannot be changed When all of the DSP channels are in the idle state changes can be made to the codec complexity selection To configure the DSP interface use the following commands beginning in privileged EXEC mode Command Purpose Stepl1 Router show interfaces dspfarm Displays the DSP voice channel activity If any DSP voice channels are in the busy state codec complexity cannot be changed When all of the DSP channels are in the idle state continue to Step 2 Step2 Router configure terminal Enters global configuration mode Step3 Cisco 7200 series Enters DSP interface configuration mode The Router config dspint dspfarm slot port arguments are as follows slot port Specifies the slot and port numbers Cisco 7500 series of the interface Router config dspint dspfarm slot port adapter port adapter port Specifies the adapter and port numbers of the interface Cisco IOS Voice Video and Fax Configuration Guide VC 64 __Configuring Voice Ports Configuring Digital Voice Ports W Command Purpose Step4 Router config dspfarm codec high med Specifies the codec complexity based on the codec standard being used The keyword specified for codec affects the choice of codecs available when the codec dial peer configuration command is used The keywords are as follows high Supports two voice channels encode
28. TATE 1 1 g729r8 y S_CONNECT FXSLS_CONNECT 1 2 FXSLS_ONHOOK 1 3 EM_ONHOO 1 4 EM_ONHOOK 1 5 a FXOLS_ONHOOK 1 6 FXOLS_ONHOOK Cisco 3600 Series Router Digital Voice Port The following output is from a Cisco 3600 series router Router show voice call summary PORT CODEC VAD VISP STATE VPM STATE 015 1 g729r8 y S_CONNECT S_TSP_CONNECT 015 2 g729r8 y S_CONNECT S_TSP_CONNECT 7015 3 g729r8 y S_CONNECT S_TSP_CONNECT 015 4 g729r8 y S_CONNECT S_TSP_CONNECT 015 5 g729r8 y S_CONNECT S_TSP_CONNECT 015 6 g729r8 y S_CONNECT S_TSP_CONNECT 015 7 g729r8 y S_CONNECT S_TSP_CONNECT 015 8 g729r8 y S_CONNECT S_TSP_CONNECT 015 9 g729r8 y S_CONNECT S_TSP_CONNECT 015 10 g729r8 y S_CONNECT S_TSP_CONNECT 015 11 g729r8 y S_CONNECT S_TSP_CONNECT 015 12 g729r8 y S_CONNECT S_TSP_CONNECT show Call active voice Command Example The following output is from a Cisco 7200 series router Router show call active voice GENERIC SetupTime 94523746 ms Index 448 PeerAddress 73072 PeerSubAddress PeerId 70000 PeerIfIndex 37 Cisco IOS Voice Video and Fax Configuration Guide Configuring Voice Ports W Verifying Analog and Digital Voice Port Configurations LogicallIfIndex 0 Connect Time 94524043 DisconectTime 94546241 CallOrigin 1 ChargedUnits 0 InfoType 2 TransmitPackets 6251 TransmitBytes 125020 ReceivePackets 3300 ReceiveBytes 66000 VOIP ConnectionId 0x142E62FB
29. Table 5 for an explanation of E amp M types The default is 1 Cisco IOS Voice Video and Fax Configuration Guide __Configuring Voice Ports Configuring Digital Voice Ports W Command Purpose Step3 Router config voiceport eptone locale Selects a two letter locale keyword for the voice call progress tones and other locale specific parameters to be used on this voice port Voice call progress tones include dial tone busy tone and ringback tone which vary with geographical region Other parameters include ring cadence and compand type Cisco routers comply with the ISO3166 locale name standards to see valid choices enter a question mark following the cptone command The default is us Step4 Router config voiceport compand type u law a law Cisco 2600 and 3600 series routers and Cisco MC3810 multiservice concentrators only Specifies the companding standard used This command is used in cases when the DSP is not used such as local cross connects and overwrites the compand type value set by the cptone command The keywords are as follows a law Specifies the ITU T PCM a law companding standard used primarily in Europe The default for E1 is a law u law Specifies the ITU T PCM mu law companding standard used in North America and Japan The default for T1 is u law Note If you have a Cisco MC3810 multiservice concentrator or Cisco 3660 router the compand type a law command m
30. Telecommunication Union Telecommunication Standardization Sector ITU T standard in which it is defined For example two common codecs are the G 711 and the G 729 codecs The various codecs use different algorithms to encode analog voice into digital bit streams and have different bit rates frame sizes and coding delays associated with them The codecs also differ in the amount of perceived voice quality they achieve Specialized hardware and software in the digital signal processors DSPs perform codec transformation and compression functions and different DSPs may offer different selections of codecs Select the same type of codec as the one that is used at the other end of the call For instance if a call was coded with a G 729 codec it must be decoded with a G 729 codec Codec choice is configured on dial peers For more information see the Configuring Dial Plans Dial Peers and Digit Manipulation chapter in this configuration guide Codec complexity refers to the amount of processing power that a codec compression technique requires some require more processing power than others Codec complexity affects call density which is the number of calls that can take place on the DSP interfaces which can be HCMs port adapter DSP farms or voice cards depending on the type of router in this case the Cisco MC3810 multiservice concentrator The greater the codec complexity the fewer the calls that can be handled Codec complexity is ei
31. To configure codec complexity on the Cisco MC3810 multiservice concentrator using HCMs use the following commands beginning in privileged EXEC mode Command Purpose Router show voice dsp Checks the DSP voice channel activity If any DSP voice channels are in the busy state the codec complexity cannot be changed When all the DSP channels are in the idle state continue to Step 2 Router configure terminal Enters global configuration mode Router config voice card 0 Enters voice card configuration mode and specifies voice card 0 Router config voicecard codec complexity high medium For analog voice ports Specifies codec complexity based on the codec standard being used This setting restricts the codecs available in dial peer configuration All voice cards in a router must use the same codec complexity setting The keywords are as follows high Specifies two voice channels encoded in any of the following formats G 71lulaw G 71 lalaw G 723 1 15 3 G 723 1 Annex A r5 3 G 723 1 r6 3 G 723 1 Annex A 16 3 G 726 116 G 726 124 G 726 132 G 728 G 729 G 729 Annex B and fax relay medium default Specifies four voice channels encoded in any of the following formats G 71lulaw G 71 lalaw G 726 r16 G 726 124 G 726 132 G 729 Annex A G 729 Annex B with Annex A and fax relay Note Iftwo HCMs are installed this command configures both HCMs at once Config
32. Today s circuit switched telephone networks dedicate a bidirectional 64 kbps channel for the duration of each conversation regardless of whether anyone is speaking at the moment This means that in a normal voice conversation at least 50 percent of the bandwidth is wasted when one or both parties are silent This figure can actually be much higher when normal pauses and breaks in conversation are taken into account Packet switched voice networks on the other hand can use this wasted bandwidth for other purposes when voice activity detection VAD is configured VAD works by detecting the magnitude of speech in decibels and deciding when to cut off the voice from being framed VAD has some technological problems however which include the following General difficulties determining when speech ends e Clipped speech when VAD is slow to detect that speech is beginning again Automatic disabling of VAD when conversations take place in noisy surroundings VAD is configured on dial peers by default it is enabled For more information see the Configuring Dial Plans Dial Peers and Digit Manipulation chapter in this configuration guide Two parameters associated with VAD music threshold and comfort noise are configured on voice ports If VAD is enabled use the following commands to adjust parameter values associated with VAD beginning in voice port configuration mode Purpose Step1 Router config voiceport music thres
33. Type 20Hz Suspend Poll 0 Call Type PCM Encoding Distinctive Ring Guard timer 1000msec Disconnect Silence timer 6dB Two Party Direction Rx only u law Disc Type OSI 0 mse 5 sec SPI Addr 3 DSLAC Revision 4 SLIC Cmd OD TX TS 00 RX TS 00 Op Fn 6F Op Fn2 00 Op Cond 00 AISN 6D ELT B5 EPG 32 52 00 00 SLIC Pin Direction 1F CODEC Coefficients GX AO 00 GR 3A Al Z EA 23 2A 35 A5 9F C2 AD 3A AE 22 46 C2 FO B 29 FA 8F 2A CB A9 23 92 2B 49 F5 37 1D 01 X AB 40 3B 9F A8 7E 22 97 36 A6 2A AE R 01 11 01 90 01 90 01 90 01 90 01 90 GZ 60 ADAPT B 91 B2 8F 62 31 CSM Finite State Machine Call 0 State idle Call Id 0x0 Active no Call 1 State idle Call Id 0x0 Active no Call 2 State idle Call Id 0x0 Active no Time Slot Control 0 Cisco IOS Voice Video and Fax Configuration Guide gy Configuring Voice Ports WE Configuring Digital Voice Ports Troubleshooting Tip for Cisco 803 and 804 Routers Check to ensure that all cables are securely connected Configuring Digital Voice Ports The digital voice port commands discussed in this section configure channelized T1 or E1 connections for information on ISDN connections see Configuring ISDN Interfaces for Voice in this configuration guide The T1 or El lines that connect a telephony network to the digital voice ports on a router or access server contain channels for voice calls a T1 line contains
34. Voice Video and Fax Configuration Guide __Configuring Voice Ports Analog Voice Ports Configuration Task List Hi Configuring Platform Specific Analog Voice Hardware This section describes the general types of analog voice port hardware available for the router platforms included in this chapter Cisco 800 Series Routers page 43 e Cisco 1750 Modular Router page 43 Cisco 2600 Series and Cisco 3600 Series Routers page 44 Cisco MC3810 Multiservice Concentrator page 44 Note For current information about supported hardware see the release notes for the platform and Cisco IOS release being used Cisco 800 Series Routers Cisco 803 and Cisco 804 routers support data and voice applications The data applications on these routers are implemented through the ISDN port and the voice applications are implemented with ISDN Basic Rate Interface BRI through the telephone ports If a Cisco 803 or 804 router is being used connect two devices such as an analog touch tone telephone fax machine or modem through two fixed telephone ports the gray PHONE 1 and PHONE 2 ports that have RJ 11 connectors Each device is connected to basic telephone services through the ISDN line For more information refer to the Cisco 800 Series Routers Hardware Installation Guide Cisco 1750 M odular Router The Cisco 1750 modular router provides Voice over IP VoIP functionality and can carry voice traffic for example telephone calls and
35. X physical interface to connect to an analog PBX two wire or four wire Optional high performance voice compression modules HCMs can replace standard voice compression modules VCMs to operate according to the voice compression coding algorithm codec specified when the Cisco MC3810 concentrator is configured The HCM2 provides four voice channels at high codec complexity and eight channels at medium complexity The HCM6 provides 12 voice channels at high complexity and 24 channels at medium complexity One or two HCMs can be installed in a Cisco MC3810 multiservice concentrator but an HCM may not be combined with a VCM in one chassis For more information refer to the Cisco MC3810 Multiservice Concentrator Hardware Installation Guide 2 Note For current information about supported hardware see the release notes for the platform and Cisco IOS release being used E Cisco IOS Voice Video and Fax Configuration Guide __Configuring Voice Ports Analog Voice Ports Configuration Task List Hi Configuring Codec Complexity for Analog Voice Ports on the Cisco MC3810 with High Performance Compression Modules The term codec stands for coder decoder A codec is a particular method of transforming analog voice into a digital bit stream and vice versa and also refers to the type of compression used Several different codecs have been developed to perform these functions and each one is known by the number of the International
36. XC ona VIP2 or VIP4 in Cisco 7500 series routers If the VIP is inserted in interface processor slot 3 and port adapter slot 0 then the addresses of the PA VXB or PA VXC are 3 0 0 or 3 0 1 interface processor slot 3 port adapter slot 0 and interfaces 0 and 1 Interface processor slot 0 to 12 depends on the number of slots in the router Port adapter slot always 0 or 1 Interface port 0 or 1 Cisco IOS Voice Video and Fax Configuration Guide Configuring Voice Ports WE Configuring Digital Voice Ports The following is show voice port summary sample output for a Cisco MC3810 multiservice concentrator Router show voice port summary IN OUT PORT CH SIG TYPE ADMIN OPER STATUS STATUS EC 0 17 18 fxo ls down down idle on hook y 0 18 19 fxo ls up dorm idle on hook y 0 19 20 fxo l1ls up dorm idle on hook y 0 20 21 fxo l1s up dorm idle on hook y 0321 22 fxo l1ls up dorm idle on hook y 0322 23 x0 1s up dorm idle on hook y 0 23 24 e amp m imd up dorm idle idle y Platform Specific Digital Voice Hardware This section briefly describes digital voice hardware on the following platforms Cisco 2600 series and Cisco 3600 series routers Cisco MC3810 multiservice concentrator Cisco AS5300 universal access server Cisco AS5800 universal access server Cisco 7200 series and Cisco 7500 series routers N Note For current information about supported hardware see the release notes for the pl
37. a calling party disconnects The seconds argument is the number of seconds of the interval and ranges from 0 to 10 The default depends on the country chosen in the pots country command Step10 Router config pots distinctive ring guard time Optional Specifies the delay after which a milirseconds telephone port can be rung after a previous call is disconnected The milliseconds argument is the number of milliseconds of the delay and ranges from 0 to 1000 The default depends on the country chosen in the pots country command Verifying Analog Telephone Connections on Cisco 803 and 804 Routers After configuring analog telephone connections perform the following steps to verify proper operation Step1 Pick up the handset of an attached telephony device and check for a dial tone Step2 Review the configuration using the show pots status command which displays settings of physical characteristics and other information on telephone interfaces Router show pots status POTS Global Configuration Country United States Dialing Method Overlap Tone Source Remote CallerId Support YES Line Type 600 ohm PCM Encoding u law Disc Type OSI Ringing Frequency 20Hz Distinctive Ring Guard timer 0 msec Disconnect timer 1000 msec Disconnect Silence timer 5 sec TX Gain 6dB RX Loss 6dB Filter Mask 6F Adaptive Cntrl Mask 0 POTS PORT 1 Hook Switch Finite State Machine State On Hook Event 0 Hook Switch Regi
38. al lines using signaling methods that emulate the methods used in analog circuit switched networks FXS FXO and E amp M The method used to transmit the information describes the way that the emulated analog signaling is transmitted over digital lines which may be common channel signaling CCS or channel associated signaling CAS CCS sends signaling information down a dedicated channel and CAS takes place within the voice channel itself This chapter describes CAS signaling which is sometimes called robbed bit signaling because user bandwidth is robbed by the network for signaling A bit is taken from every sixth frame of voice data to communicate on or off hook status wink ground start dialed digits and other information about the call In addition to setting up and tearing down calls CAS provides the receipt and capture of dialed number identification DNIS and automatic number identification ANI information which are used to support authentication and other functions The main disadvantage of CAS signaling is its use of user bandwidth to perform these signaling functions For signaling to pass between the packet network and the circuit switched network both networks must use the same type of signaling The voice ports on Cisco routers and access servers can be configured to match the signaling of most COs and PBXs as explained in this chapter This section discusses the following topics Prerequisites for Configuring Digital Vo
39. amples of the following types are shown Cisco 3640 Router Analog Voice Port Cisco MC3810 Multiservice Concentrator Digital Voice Port Cisco IOS Voice Video and Fax Configuration Guide __Configuring Voice Ports Verifying Analog and Digital Voice Port Configurations W Cisco 3640 Router Analog Voice Port The following output is from a Cisco 3640 router Router show voice port summary IN OUT PORT CH SIG TYPE ADMIN OPER STATUS STATUS EC 2 0 0 e amp m wnk up dorm idle idle y 2 0 1 e amp m wnk up dorm idle idle y 2 1 0 fxs ls up dorm on hook idle y 2 1 1 fxs ls up dorm on hook idle y Cisco M C3810 M ultiservice Concentrator Digital Voice Port The following output is from a Cisco MC3810 multiservice concentrator Router show voice port summary N OUT PORT CH SIG TYPE ADMIN OPER STATUS STATUS EC 0 17 18 fxo ls down down idle on hook y 0 18 19 xo ls up dorm idle on hook y 0219 20 fxo ls up dorm idle on hook y 0 20 21 fxo ls up dorm idle on hook y 0421 22 fxo ls up dorm idle on hook y 0 22 23 fxo ls up dorm idle on hook y 0 23 24 e amp m imd up dorm idle idle y 1 1 Exe lg up dorm on hook idle y 1 2 fxe le up dorm on hook idle y 1 3 e amp m imd up dorm idle idle y 1 4 e amp m imd up dorm idle idle y 1 5 gt fxo ls up dorm idle on hook y 1 6 fxo ls up dorm idle on hook y show voice port Command Examples In the following sections output examples of the following
40. and For output examples see the show controller Command Examples section on page 103 Router show controller t1 el controller number To display voice channel configuration information for all DSP channels use the show voice dsp command For output examples see the show voice dsp Command Examples section on page 104 Router show voice dsp To verify the call status for all voice ports use the show voice call summary command For output examples see the show voice call summary Command Examples section on page 105 Router show voice call summary To display the contents of the active call table which shows all of the calls currently connected through the router or concentrator use the show call active voice command For output examples see the show call active voice Command Example section on page 105 Router show call active voice To display the contents of the call history table use the show call history voice command To limit the display to the last calls connected through this router use the keyword last and define the number of calls to be displayed with the argument number To limit the display to a shortened version of the call history table use the brief keyword For output examples see the show call history voice Command Example section on page 106 Router show call history voice last number brief show voice port summary Command Examples In the following sections output ex
41. arameter values have been set for the controller framing clock source line code cable length If these values do not match those of the telephony connection you are making reconfigure the controller See the show controller Command Examples section on page 103 for output No connectivity Enter the show voice port command with the voice port number that you are troubleshooting which will tell you Ifthe voice port is up If it is not use the no shutdown command to make it active What parameter values have been set for the voice port including default values these do not appear in the output for the show running config command If these values do not match those of the telephony connection you are making reconfigure the voice port See the show voice port Command Examples section on page 99 for sample output Telephony device buzzes or does Use the show voice port command to confirm that ring frequency not ring is configured correctly It must match the connected telephony equipment and may be country dependent Cisco 10S Voice Video and Fax Configuration Guide VC 108 _Configuring Voice Ports Troubleshooting Analog and Digital Voice Port Configurations Table 10 Troubleshooting Voice Port Configurations continued Problem Suggested Action Distorted speech Use the show voice port command to confirm the cptone keyword setting also called region tone is US Set
42. arget ipv4 171 68 235 18 OnTimeRvPlayout 398000 GapFillWithSilence 0 ms GapFillWithPrediction 1440 ms GapFillWithInterpolation 0 ms GapFillWithRedundancy 0 ms HiWaterPlayoutDelay 97 ms LoWaterPlayoutDelay 30 ms ReceiveDelay 49 ms LostPackets 1 ms EarlyPackets 1 ms LatePackets 132 ms VAD disabled CoderTypeRate g729r8 CodecBytes 20 cvVoIPCallHistorylIcpif 0 Cisco IOS Voice Video and Fax Configuration Guide gy Configuring Voice Ports WE Troubleshooting Analog and Digital Voice Port Configurations Troubleshooting Analog and Digital Voice Port Configurations The following sections will assist in analyzing and troubleshooting voice port problems Troubleshooting Chart page 108 Voice Port Testing Commands page 110 Troubleshooting Chart Table 10 lists some problems you might encounter after configuring voice ports and has some suggested remedies Table 10 Troubleshooting Voice Port Configurations Problem Suggested Action No connectivity Ping the associated IP address to confirm connectivity If you cannot successfully ping your destination refer to the Cisco IOS IP Configuration Guide No connectivity Enter the show controller t1 or show controller e1 command with the controller number for the voice port you are troubleshooting This will tell you Ifthe controller is up If it is not use the no shutdown command to make it active Whether alarms have been reported What p
43. atform and Cisco IOS release you are using Cisco 2600 Series and Cisco 3600 Series Routers Digital voice hardware on Cisco 2600 series and Cisco 3600 series modular access routers includes the high density voice HDV network module and the multiflex trunk MFT voice WAN interface card VWIC When an HDV is used in conjunction with an MFT and packet voice DSP modules PVDMs the HDV module is also called a digital packet voice trunk network module The digital T1 or E1 packet voice trunk network module supports T1 or El applications including fractional use The T1 version integrates a fully managed data service unit channel service unit DSU CSU and the E1 version includes a fully managed DSU The digital T1 or El packet voice trunk network module provides per channel T1 or E1 data rates of 64 or 56 kbps for WAN services Frame Relay or leased line Digital T1 or El packet voice trunk network modules for Cisco 2600 and 3600 series routers allow enterprises or service providers using the voice equipped routers as customer premise equipment CPE to deploy digital voice and fax relay These network modules receive constant bit rate telephony information over T1 or E1 interfaces and convert that information to a compressed format so that it can be sent over a packet network The digital T1 or E1 packet voice trunk network modules can connect either to a PBX or similar telephony device or to a CO to provide PSTN connectivity One digital T
44. bit modification commands for digital voice ports modify sent or received bit patterns Different versions of E amp M use different ABCD signaling bits to represent idle and seize For example North American CAS E amp M represents idle as OXXX and seize as 1X XX where X indicates that the state of the BCD bits is ignored In MELCAS E amp M idle is 1101 and seize is 0101 The commands in this section are provided to modify bit patterns to match particular E amp M schemes To manipulate bit patterns for digital voice ports use the following commands as necessary in voice port configuration mode Cisco IOS Voice Video and Fax Configuration Guide E Configuring Voice Ports WE Configuring Digital Voice Ports Command Purpose Step 1 Router config voiceport condition tx a bit tx b bit tx c bit tx d bit rx a bit rx c bit rx d bit on off invert Manipulates sent or received bit patterns to match expected patterns on a connected device Repeat the command for each transmit and or receive bit to be modified but be careful not to destroy the information content of the bit pattern The default is that the signaling format is not manipulated for all transmit or receive A B C and D bits The keywords are as follows on Sets the bit to 1 permanently off Sets the bit to 0 permanently invert Changes the state to the opposite of the original transmit or receive state Note The show v
45. ce Ports Router config controller dsO group ds0 group no timeslots timeslot list type e amp m delay e amp m immediate e amp m wink fxs ground start fxs loop start fxo ground start fxo loop start Defines the T1 channels for use by compressed voice calls and the signaling method that the router uses to connect to the PBX or CO Note This step shows the basic syntax and signaling types available with the ds0 group command For the complete syntax see the Cisco IOS Voice Video and Fax Command Reference Release 12 2 The keywords and arguments are as follows ds0 group no ldentifies the DSO group number from 0 to 23 for T1 or from 0 to 30 for El timeslots timeslot list Specifies the single time slot number single range of numbers or multiple ranges of numbers separated by commas For T1 E1 allowable values are from 1 to 24 Examples are as follows 2 3 5 1 7 9 1 12 service Indicates the type of calls to be handled by this DSO group data fax or voice type Refers to the signaling type of the telephony connection being made Types include the following e amp m delay dial Specifies the originating endpoint that sends an off hook signal and waits for the off hook signal followed by an on hook signal from the destination e amp m fgb E amp M Type II Feature Group B e amp m fgd E amp M Type II Feature Group D E Cisco IOS Voice Video a
46. ck Tells the router to collect all digits dialed and to send the digits in one message The default depends on the country chosen in the pots country command E Cisco IOS Voice Video and Fax Configuration Guide __Configuring Voice Ports Analog Voice Ports Configuration Task List Hi Command Purpose Step4 Router config pots disconnect supervision osi Optional Specifies how the router notifies the reversal connected telephones fax machines or modems when the calling party has disconnect The keywords are as follows osi open switching interval Specifies the duration for which DC voltage applied between tip and ring conductors of a telephone port is removed reversal Specifies the polarity reversal of the tip and ring conductors of a telephone port The default depends on the country chosen in the pots country command Step5 Router config pots encoding alaw ulaw Optional Specifies the pulse code modulation PCM encoding scheme for telephones fax machines or modems connected to a Cisco 800 series router The keywords are as follows alaw Specifies the ITU T PCM encoding scheme used to represent analog voice samples as digital values ulaw Specifies the North American PCM encoding scheme used to represent analog voice samples as digital values The default depends on the country chosen in the pots country command Step6 Router config pots tone source local
47. co 2600 and Cisco 3600 series routers or the router slot number where the analog voice module is installed Cisco MC3810 multiservice concentrator port Indicates the voice port Valid entries are O or 1 subunit Specifies the location of the VIC Note The slash must be entered between slot and port Valid entries vary by router platform see Table 7 on page 42 or enter the show voice port summary command for available values FXO or FXS Selects the access signaling type to match that of Router config voiceport signal loop start the telephony connection you are making The ground start keywords are as follows loop start default Uses a closed circuit to indicate off hook status used for residential loops ground start Uses ground and current detectors preferred for PBXs and trunks Cisco 10S Voice Video and Fax Configuration Guide E Configuring Voice Ports HE Analog Voice Ports Configuration Task List Command Purpose E amp M The keywords are as follows Router config voiceport signal wink start iimediate start delay dialj wink start default Indicates that the calling side seizes the line then waits for a short off hook wink from the called side before proceeding immediate start Indicates that the calling side seizes the line and immediately proceeds used for E amp M tie trunk interfaces delay dial Indicates that the calling side se
48. command for each additional tone to be specified Step3 Router config voice class freq max deviation Specifies the maximum frequency deviation that frequency will be detected in Hz The frequency argument ranges from 10 to 125 The default is 10 Step4 Router config voice class freq max power dBmO Specifies the maximum tone power that will be detected in dBmO The dBmO argument ranges from 0 to 20 The default is 10 Step5 Router config voice class freq min power dBmO Specifies the minimum tone power that will be detected in dBmO The dBmO argument ranges from 10 to 35 The default is 30 Step6 Router config voice class freq power twist dBmO Specifies the power difference allowed between the two frequencies in dBmO The dBmO argument ranges from 0 to 15 The default is 6 Step7 Router config voice class freq max delay time Specifies the timing difference allowed between the two frequencies in 10 millisecond increments The time argument ranges from 10 to 100 100 ms to 1 s The default is 20 200 ms Step 8 Router config voice class cadence min on time time Specifies the minimum tone on time that will be detected in 10 millisecond increments The time argument ranges from 0 to 100 0 ms to 1 s Step9 Router config voice class cadence max off time time Specifies the maximum tone off time that will be detected in 10 millisecond increments The time argument ranges from 0 to 5000 0
49. company the Cisco IOS software you are using Cisco IOS Voice Video and Fax Configuration Guide E Configuring Voice Ports HE Analog Voice Ports Configuration Task List Table 6 Analog Voice Port Configurations Telephony Signaling Section Containing Voice Port Interface Router Platform Voice Hardware Required Configuration Instructions End user Cisco 803 Configuring Analog Telephone telephoneor Cisco 804 Connections on Cisco 803 and fax 804 Routers FXO Cisco 1750 VIC 2FXO VIC 2FXO EU Configuring Basic Parameters Cisco 2600 series on Analog FXO FXS or E amp M Cisco 3600 series Voice Ports Cisco MC3810 MC3810 AVM6 MC3810 APM FXO FXS Cisco 1750 VIC 2FXS Cisco 2600 series Cisco 3600 series Cisco MC3810 MC3810 AVM6 MC3810 APM FXS E amp M Cisco 1750 VIC 2E M Cisco 2600 series Cisco 3600 series Cisco MC3810 MC3810 AVM6 MC3810 APM EM Table 7 Analog Voice Slot Port Designations Chassis Slot Voice NM Slot Voice Port Router Platform Voice Hardware Numbers Numbers Numbers Cisco 803 804 Analog POTS Cisco 1750 Analog VIC 0 to 1 0 to 1 Cisco 2600 series Voice fax network module Varies based 1 Otol with two port VIC on router Cisco 3600 series Voice fax network module 1 3620 0 to 1 0 to 1 with two port voice over 3640 0 to 3 interface cards VICs a 3660 1 to 6 Cisco MC3810 Analog voice module AVM 1 lto6 E Cisco IOS
50. concentrators Router config voice port slot port e subunit Specifies the voice interface card VIC where the voice port is located port identifies the analog voice port number Step 14 Router config voiceport supervisory disconnect Assigns an FXO supervisory disconnect tone dualtone mid call pre connect voice class tag voice class to the voice port The keywords are as follows mid call Specifies tone detection during the entire call pre connect Specifies tone detection only during call set up Step15 Router config voiceport supervisory disconnect Configures the voice port to disconnect on receipt anytone of any tone Timeouts Commands To change timeouts parameters use the following commands as appropriate in voice port configuration mode Cisco IOS Voice Video and Fax Configuration Guide ce Configuring Voice Ports WE Configuring Digital Voice Ports Command Purpose Step1 Router config voiceport timeouts call disconnect Configures the call disconnect timeout value in seconds seconds Valid entries range from 0 to 120 The default is 60 Step2 Router config voiceport timeouts initial seconds Sets the number of seconds that the system waits between the caller input of the initial digit and the subsequent digit of the dialed string If the wait time expires before the destination is identified a tone sounds and the call ends The seconds argument
51. ctor to its original state To configure this feature use the following commands in privileged EXEC mode Command Purpose Cisco 2600 and 3600 Series Routers Analog Voice Ports Router test voice port slot subunit port detector m lead battery reversal loop current ring tip ground ring ground ring trip on off Cisco 2600 and 3600 Series Routers Digital Voice Ports Router test voice port slot port ds0 group detector m lead battery reversal loop current ring tip ground ring ground ring trip on off Cisco MC3810 Multiservice Concentrators Analog Voice Ports Router test voice port slot port detector m lead battery reversal loop current ring tip ground ring ground ring trip on off Identifies the voice port you want to test Enter a keyword for the detector under test and specify whether to force it to the on or off state Note For each signaling type E amp M FXO FXS only the applicable keywords are displayed The disable keyword is displayed only when a detector is in the forced state Cisco IOS Voice Video and Fax Configuration Guide __Configuring Voice Ports Step 2 Cisco MC3810 Multiservice Concentrators Digital Voice Ports Router test voice port slot ds0 group detector m lead battery reversal loop current ring tip ground ring ground ring trip on off Troubleshooting Analog and Digital Voice Port Configurations i Command
52. d in any of the following formats G 711 G 726 G 729 G 729 Annex B G 723 1 G 723 1 Annex A G 728 and fax relay med default Supports up to four calls using the following codecs G 711 G 726 G 729 Annex A G 729 Annex B with Annex A and fax relay Step5 Router config dspfarm description Enters a string to include descriptive text about this DSP interface connection This information is displayed in the output for show commands and does not affect the operation of the interface in any way Configuring Controller Settings for Digital T1 E1 Voice Ports The purpose of configuring controllers for digital T1 E1 voice ports is to match the configuration of the router to the line characteristics of the telephony network connection being made so that voice and signaling can be transferred between them and so that logical voice ports or DSO groups may be established Figure 16 shows how a ds0 group command gathers some of the DSO time slots from a T1 line into a group that becomes a single logical voice port which can later be addressed as a single entity in voice port configurations Other DSO groups for voice can be created from the remaining time slots shown in the figure or the time slots can be used for data or serial pass through Note that all the controller commands in Figure 16 other than ds0 group apply to all the time slots in the T1 Cisco IOS Voice Video and Fax Configuration Guide mcs Configuri
53. e FXS interface This number must match the connected telephony equipment and Cisco MC3810 Multiservice Concentrator may be country dependent If not set properly the Router config voiceport ring frequency 20 30 a device may not ring or it may UZZ The keyword default is 25 on the Cisco 1750 router 2600 and 3600 series routers and 20 on the Cisco MC3810 multiservice concentrator Hl Cisco IOS Voice Video and Fax Configuration Guide __Configuring Voice Ports Command Analog Voice Ports Configuration Task List Hi Purpose Step 8 Router config voiceport ring number number FXO only Specifies the maximum number of rings to be detected before an incoming call is answered by the router The default is 1 Step 9 pattern02 pattern06 pattern10 interval pattern03 pattern07 pattern11 pattern04 pattern05 pattern08 pattern09 pattern12 Router config voiceport ring cadence pattern01 define pulse FXS only Specifies an existing pattern for ring or it defines a new one Each pattern specifies a ring pulse time and a ring interval time The keywords and arguments are as follows pattern01 through pattern12 name pre set ring cadence patterns Enter ring cadence to see ring pattern explanations define pulse interval specifies a user defined pattern pulse is a number one or two digits from 1 to 50 specifying ring pulse on time in hundreds of milli
54. e implementations echo cancellers are enabled using the echo cancel enable command and echo trails are configured using the echo cancel coverage command To configure parameters related to the echo canceller use the following commands beginning in voice port configuration mode E Cisco IOS Voice Video and Fax Configuration Guide __Configuring Voice Ports Configuring Digital Voice Ports W Command Purpose Stepl Router config voiceport echo cancel enable Enables the cancellation of voice that is sent and received on the same interface Echo cancellation coverage must also be configured The default is that echo cancellation is enabled Note Not valid for four wire E amp M interfaces Use no echo cancel enable to disable the feature Step2 Router config voiceport echo cancel coverage 8 16 Adjusts the echo canceller by the specified number 2d 32 of milliseconds The default is 16 Step3 Router config voiceport non linear Enables nonlinear processing residual echo suppression in the echo canceler which shuts off any signal if no near end speech is detected Echo cancelling must be enabled for this feature The default is that nonlinear processing is enabled Cisco IOS Voice Video and Fax Configuration Guide Configuring Voice Ports WE Configuring Digital Voice Ports Voice Level Adjustment As much as possible it is desirable to achieve a uniform input decibel level to the pac
55. e keyword specifies that the clock source is derived from the active line rather than from the free running internal clock The following rules apply to clock sourcing on the controller ports When both ports are set to line clocking with no primary specification port 0 is the default primary clock source and port 1 is the default secondary clock source When both ports are set to line and one port is set as the primary clock source the other port is by default the backup or secondary source and is loop timed If one port is set to clock source line or clock source line primary and the other is set to clock source internal the internal port recovers clock from the clock source line port if the clock source line port is up If it is down then the internal port generates its own clock If both ports are set to clock source internal there is only one clock source internal This section describes the five basic timing scenarios that can occur when a digital voice port is connected to a PBX or CO In all the examples that follow the PSTN or CO and the PBX are interchangeable for purposes of providing or receiving clocking Single Voice Port Providing Clocking In this scenario the digital voice hardware is the clock source for the connected device as shown in Figure 17 The PLL generates the clock internally and drives the clocking on the line Generally this method is useful only when connecting to a PBX key system or channe
56. e ports are described in the next three sections They are grouped by the configuration mode from which they are executed as follows Configuring Codec Complexity for Digital T1 E1 Voice Ports page 62 Codec complexity refers to the amount of processing power assigned to codec processing on a voice port On most router platforms that support codec complexity codec complexity is selected in voice card configuration mode although it is selected in DSP interface mode on the Cisco 7200 and 7500 series The value configured for codec complexity establishes the choice of codecs that are available on the dial peers See the Configuring Dial Plans Dial Peers and Digit Manipulation chapter in this configuration guide for more information about configuring dial peers Configuring Controller Settings for Digital T1 E1 Voice Ports page 65 Specific line characteristics must be configured to match those of the PSTN line that is being connected to the voice port These are typically configured in controller configuration mode Configuring Basic Voice Port Parameters for Digital T1 E1 Voice Ports page 76 Voice port configuration mode allows many of the basic voice call attributes to be configured to match those of the PSTN or PBX connection being made on this voice port In addition to the basic voice port parameters there are additional commands that allow for the fine tuning of the voice port configurations or for configuration of optional featu
57. ed Support for digital T1 packet voice trunk network modules is included in Plus feature sets The IP Plus feature set requires 8 MB of Flash memory other Plus feature sets require 16 MB Cisco 2600 and 3600 series routers For digital E1 packet voice trunk network modules install Cisco IOS Release 12 1 2 T 12 2 1 or a later release The minimum DRAM memory requirements are 48 MB with one or two Els 64 MB with three to eight Els 128 MB with 9 to 12 Els For high volume applications the memory required may be greater than these minimum values Support for digital E1 packet voice trunk network modules is included in Plus feature sets The IP Plus feature set requires 16 MB of Flash memory Cisco MC3810 concentrators HCMs require Cisco IOS Release 12 0 7 XK or 12 1 2 T 12 2 1 or a later release Cisco 7200 and 7500 series routers For digital T1 E1 voice port adapters install Cisco IOS Release 12 0 5 XE 12 0 7 T 12 2 1 or a later release The minimum DRAM memory requirement to support T1 E1 high capacity digital voice port adapters is 64 MB The memory required for high volume applications may be greater than that listed Support for T1 E1 high capacity digital voice port adapters is included in Plus feature sets The IP Plus feature set requires 16 MB of Flash memory Cisco IOS Voice Video and Fax Configuration Guide E Configuring Voice Ports WE Configuring Digital Voice Ports Preparing Information
58. ed on Cisco MC3810 analog voice ports on Cisco 1750 Cisco 2600 series and Cisco 3600 series routers only analog voice ports on VIC 2FXO cards are able to detect battery reversal Also use the no battery reversal command when a connected FXO port does not support battery reversal detection For FXS ports Use the no battery reversal command to configure the voice port not to reverse battery when it connects calls The default is to reverse battery when a call is connected then return to normal when the call is over providing positive disconnect See also the disconnect ack command Step 7 Step2 Router config voiceport no supervisory disconnect FXO only Enables the PBX or PSTN switch to provide STD By default the supervisory disconnect command is enabled Step3 Router config voiceport disconnect ack FXS only Configures the voice port to return an acknowledgment upon receipt of a disconnect signal The FXS port removes line power if the equipment on the FXS loop start trunk disconnects first This is the default The no disconnect ack command prevents the FXS port from responding to the on hook disconnect with a removal of line power E Cisco IOS Voice Video and Fax Configuration Guide __Configuring Voice Ports Configuring Digital Voice Ports W FXO Supervisory Disconnect Tone Commands If the FXO supervisory disconnect tone is configured and a detectable tone from the PSTN or P
59. efault is 0 Step5 Router config voiceport timing dial pulse min delay Specifies time in milliseconds between the nee ee egies generation of wink like pulses when the type is pulse Valid entries are from 0 to 5000 The default is 300 for the Cisco 3600 series routers and 140 for the Cisco MC3810 multiservice concentrators Step6 Router config voiceport timing dialout delay Cisco MC3810 multiservice concentrators only milliseconds Specifies dialout delay in milliseconds for the sending digit or cut through on an FXO trunk or an E amp M immediate trunk Valid entries are from 100 to 5000 The default is 300 Step 7 Router config voiceport timing digit milliseconds Specifies the DTMF digit signal duration in milliseconds Valid entries are from 50 to 100 The default is 100 Step8 Router config voiceport timing guard out FXO ports only Specifies the duration in pi seconds milliseconds of the guard out period that prevents this port from seizing a remote FXS port before the remote port detects a disconnect signal The range is from 300 to 3000 The default is 2000 Step9 Router config voiceport timing hookflash out Specifies the duration in milliseconds of the apa Se Coucs hookflash Valid entries are from 50 to 500 The default is 300 Cisco IOS Voice Video and Fax Configuration Guide m Configuring Voice Ports WE Configuring Digital Voice Ports Step 10 Step 11 Step
60. eize lIndicates that the pattern represents line seizure idle lIndicates that the pattern represents an idle condition 0000 1111 Represents the bit pattern to use Router config voiceport ignore rx a bit rx b bit Digital E1 E amp M voice ports on Cisco 2600 and rx c bit rx d bit 3600 series routers and Cisco MC3810 multiservice concentrators only Configures the voice port to ignore the specified receive bit for North American E amp M or E amp M MELCAS if patterns different from the defaults are required See the command reference for the default signaling patterns as defined in ANSI and CEPT standards Calling Number Outbound Commands On the Cisco AS5300 universal access server platform if T1 CAS is configured with the Feature Group D FGD Exchange Access North American FGD EANA signaling the automatic number identification ANTI can be sent for outgoing calls by using the calling number outbound command FGD EANA is a FGD signaling protocol of type EANA which provides certain call services such as emergency USA 911 calls ANI is an SS7 Signaling System 7 feature in which a series of digits analog or digital are included in the call to identify the telephone number of the calling device In other words ANI identifies the number of the calling party ANI digits are used for billing purposes by Internet service providers ISPs among other things The commands in this section can be issued
61. er call volume experienced on business telephones Loop start signaling has no means of preventing two sides from seizing the same line simultaneously a condition known as glare Also loop start signaling does not provide switch side disconnect supervision for FXO calls The telephony switch the connection in the PSTN another PBX or key system expects the router s FXO interface which looks like a telephone to the switch to hang up the calls it receives through its FXO port However this function is not built into the router for received calls it only operates for calls originating from the FXO port Another access signaling method used by FXO and FXS interfaces to indicate on hook or off hook status to the CO is ground start signaling It works by using ground and current detectors that allow the network to indicate off hook or seizure of an incoming call independent of the ringing signal and allow for positive recognition of connects and disconnects For this reason ground start signaling is typically used on trunk lines between PBXs and in businesses where call volume on loop start lines can result in glare See the Disconnect Supervision Commands section on page 82 and FXO Supervisory Disconnect Tone Commands section on page 85 for voice port commands that configure additional recognition of disconnect signaling In most cases the default voice port command values are sufficient to configure FXO and FXS voice ports Trunk ci
62. es Analog show voice port s ot port summary Cisco 3600 series Digital show voice port slot port ds0 group no summary Cisco MC3810 Analog show voice port s ot port summary Digital show voice port s ot ds0 group no summary Cisco AS5300 Digital show voice port controller ds0 group no Cisco AS5800 Digital show voice port shelf slot port dsO0 group no Cisco 7200 series Digital show voice port slot port adapter dsO0 group no Cisco 7500 series Digital show voice port slot port adapter slot ds0 group no Cisco IOS Voice Video and Fax Configuration Guide E Configuring Voice Ports W Verifying Analog and Digital Voice Port Configurations Step 5 Step 6 Step 7 Step 8 Step 9 Step 10 For digital T1 E1 connections to verify the codec complexity configuration use the show running config command to display the current voice card setting If medium complexity is specified the codec complexity setting is not displayed If high complexity is specified the setting codec complexity high is displayed The following example shows an excerpt from the command output when high complexity has been specified Router show running config hostname router alpha voice card 0 codec complexity high For digital T1 E1 connections to verify that the controller is up and that no alarms have been reported and to display information about clock sources and other controller settings use the show controller comm
63. escribed in the Configuring Trunk Connections and Trunk Conditioning Features chapter in this configuration guide The digital T1 or El packet voice trunk network module contains five 72 pin Single In line Memory Module SIMM sockets or banks numbered 0 through 4 for PVDMs Each socket can be filled with a single 72 pin PVDM and there must be at least one packet voice data module PVDM 12 in the network module to process voice calls Each PVDM holds three digital signal processors DSPs so with five PVDM slots populated a total of 15 DSPs are provided High complexity codecs support two simultaneous calls on each DSP and medium complexity codecs support four calls on each DSP A digital T1 or El packet voice trunk network module can support the following numbers of channels When the digital T1 or El packet voice trunk network module is configured for high complexity codec mode up to six voice or fax calls can be completed per PVDM 12 using the following codecs G 711 G 726 G 729 G729 Annex A E1 G 729 Annex B G 723 1 G723 1 Annex A T1 G 728 and fax relay When the digital T1 or El packet voice trunk network module is configured for medium complexity codec mode up to 12 voice or fax calls can be completed per PVDM 12 using the following codecs G 711 G 726 G 729 Annex A G 729 Annex B with Annex A and fax relay For more information refer to the following publications Cisco 2600 Series Hardware Installatio
64. ests To inject a test tone into a voice port use the following commands in privileged EXEC mode Command Purpose Step1 Cisco 2600 and 3600 Series Routers Analog Voice Ports Identifies the voice port you want to test and enter Router test voice port slot subunit port inject tone keywords for the direction to send the test tone and local network 1000hz 2000hz 200hz 3000hz for the frequency of the test tone 300hz 3200hz 3400hz 500hz quiet Note A call must be established on the voice Cisco 2600 and 3600 Series Routers Digital Voice Ports port under test Router test voice port slot port ds0 group inject tone local network 1000hz 2000hz 200hz 3000hz 300hz 3200hz 3400hz 500hz quiet Cisco M C3810 Multiservice Concentrators Analog Voice Ports Router test voice port slot port detector inject tone local network 1000hz 2000hz 200hz 3000hz 300hz 3200hz 3400hz 500hz quiet Cisco MC3810 Multiservice Concentrators Digital Voice Ports Router test voice port slot ds0 group inject tone local network 1000hz 2000hz 200hz 3000hz 300hz 3200hz 3400hz 500hz quiet Step2 Cisco 2600 and 3600 Series Routers Analog Voice Ports Identifies the voice port on which you want to end Router test voice port slot subunit port inject tone the test and enter the keyword disable to end the disable test tone Note The disable keyword is available only if a Cisco 2
65. eywords and arguments are as follows pattern01 through patternl2 Specifies preset ring cadence patterns Enter ring cadence to see ring pattern explanations define pulse interval Specifies a user defined pattern as follows pulse is a number 1 or 2 digits from 1 to 50 specifying ring pulse on time in hundreds of milliseconds interval is a number 1 or 2 digits from 1 to 50 specifying ring interval off time in hundreds of milliseconds The default is the pattern specified by the configured cptone locale command Step8 Router config voiceport description string Attaches a text string to the configuration that describes the connection for this voice port This description appears in various displays and is useful for tracking the purpose or use of the voice port The string argument is a character string from 1 to 255 characters in length The default is that no description is attached to the configuration Step 9 Router config voiceport no shutdown Activates the voice port Fine Tuning Analog and Digital Voice Ports Normally default parameter values for voice ports are sufficient for most networks Depending on the specifics of your particular network however you may need to adjust certain parameters that are configured on voice ports Collectively these commands are referred to as voice port tuning commands Note The commands keywords and arguments that you are able to use may d
66. g as they pertain to the clocking method Looped time clocking The voice port takes the clock received on its Rx receive pair and regenerates it on its Tx transmit pair While the port receives clocking the port is not driving the PLL on the card but is spoofing that is fooling the port so that the connected device has a viable clock and does not see slips that is loss of data bits PBXs are not designed to accept slips on a T1 or El line and such slips cause a PBX to drop the link into failure mode While in looped time mode the router often sees slips but because these are controlled slips they usually do not force failures of the router s voice port Slips These messages indicate that the voice port is receiving clock information that is out of phase out of synchronization Because the router has only a single PLL it can experience controlled slips while it receives clocking from two different time sources The router can usually handle controlled slips because its single PLL architecture anticipates them Note Physical layer issues such as bad cabling or faulty clocking references can cause slips Eliminate these slips by addressing the physical layer or clock reference problems In the dual voice ports receiving clocking from the line scenario the PLL derives clocking from the CO and puts the voice port connected to the PBX into looped time mode This is usually the best method because the CO provides an
67. gured in dial peer configuration mode When there are numerous dial peers to configure it might be simpler to configure playout delay on a voice port If there are conflicting playout delay configurations on a voice port and also on a dial peer the dial peer configuration takes precedence To configure the playout delay jitter buffer use the following commands beginning in dial peer or voice port configuration mode Command Purpose Stepl Router config voiceport playout delay mode adaptive Determines the mode in which the jitter buffer will fixed operate for calls on this voice port The keywords are as follows adaptive Adjusts the jitter buffer size and amount of playout delay during a call based on current network conditions fixed Defines the jitter buffer size as fixed so that the playout delay does not adjust during a call A constant playout delay is added The default is adaptive Cisco IOS Voice Video and Fax Configuration Guide E Configuring Voice Ports WE Configuring Digital Voice Ports Command Purpose Step2 Router config voiceport playout delay nominal value Tunes the playout buffer to accommodate packet maximum value minimum default low high jitter caused by switches in the WAN The keywords and arguments are as follows nominal Defines the amount of playout delay applied at the beginning of a call by the jitter buffer in the gateway I
68. hold number Specifies the minimal decibel level of music played when calls are put on hold The decibel level affects how voice activity detection VAD treats the music data Valid entries range from 70 to 30 When used with VAD if the level is set too high the remote end hears no music if it is set too low there is unnecessary voice traffic The default is 38 Step2 Router config voiceport comfort noise This parameter creates subtle background noise to fill silent gaps during calls when VAD is enabled on voice dial peers If comfort noise is not generated the resulting silence can fool the caller into thinking the call is disconnected instead of being merely idle The default is that comfort noise is enabled Cisco IOS Voice Video and Fax Configuration Guide Configuring Voice Ports WE Configuring Digital Voice Ports Voice Quality Tuning Commands The commands in this section configure parameters to improve voice quality Common voice quality issues include the following e Delay in Voice Networks Jitter Adjustment e Echo Adjustment Voice Level Adjustment Delay in Voice Netw orks Delay is the time it takes for voice packets to travel between two endpoints Excessive delay can cause quality problems with real time traffic such as voice However because of the speed of network links and the processing power of intermediate devices some delay is expected When listening to speech
69. ice Ports page 55 Preparing Information to Configure Digital Voice Ports page 56 Platform Specific Digital Voice Hardware page 58 Configuring Basic Parameters on Digital T1 E1 Voice Ports page 61 Cisco IOS Voice Video and Fax Configuration Guide __Configuring Voice Ports Configuring Digital Voice Ports W Prerequisites for Configuring Digital Voice Ports Digital T1 or El packet voice capability requires specific service software and hardware Obtain T1 or E1 service from the service provider or from your PBX Create your company s dial plan Establish a working telephony network based on your company s dial plan Establish a connection to the network LAN or WAN Set up a working IP and Frame Relay or ATM network For more information about configuring IP refer to the Cisco IOS IP Configuration Guide Release 12 2 Install appropriate voice processing and voice interface hardware on the router See the Platform Specific Digital Voice Hardware section on page 58 Cisco 2600 and 3600 series routers For digital T1 packet voice trunk network modules install Cisco IOS Release 12 0 5 XK 12 0 7 T 12 2 1 or a later release The minimum DRAM memory requirements are as follows 32 MB with one or two T1 lines 48 MB with three or four T1 lines 64 MB with five to ten T1 lines 128 MB with more than ten T1 lines The memory required for high volume applications may be greater than that list
70. ideo and Fax Command Reference The default is the plan1 keyword Step5 Router config voiceport idle voltage high low Cisco MC3810 multiservice concentrators analog FXS ports only Specifies the talk battery tip to ring voltage condition when the port is idle The keywords are as follows high Specifies that the voltage is high 48V low Specifies that the voltage is low 24V and is the default Verifying Analog and Digital Voice Port Configurations Step 1 Step 2 Step 3 Step 4 After configuring the voice ports on your router perform the following steps to verify proper operation Pick up the handset of an attached telephony device and check for a dial tone If you have dial tone check for DTMF detection If the dial tone stops when you dial a digit then the voice port is most likely configured properly To identify port numbers of voice interfaces installed in your router use the show voice port summary command For examples of the output see the show voice port summary Command Examples section on page 98 To verify voice port parameter settings use the show voice port command with the appropriate syntax from Table 9 For sample output see the show voice port Command Examples section on page 99 Table 9 Show Voice Port Command Syntax Platform Voice Port Type Command Syntax Cisco 1750 Analog show voice port s ot port summary Cisco 2600 seri
71. idle 01 015 24 0 65589 83889 C549 015 00 g729r8 3 3 busy idle 0 0 1 015 6 0 66889 83331 01 g729r8 8 busy idle 0 0 1 015 2 0 65690 81700 02 g729r8 busy idle 0 0 1 015 9 0 66422 82099 03 g729r8 busy idle 0 0 1 015 25 0 65566 83852 Router show voice dsp TYPE DSP CH CODEC VERS STATE STATE RST AI PORT TS ABORT TX RX PAK CNT C549 007 00 medium 3 3 IDLE idle 0 0 1 0 4 0 0 0 13 C549 008 00 medium 3 3 IDLE idle 0 0 1 0 5 0 0 0 3 C549 009 00 medium 3 3 IDLE idle 0 0 1 0 6 0 0 0 a13 C549 010 00 medium 3 3 IDLE idle 0 0 1 0 7 0 0 0 e138 C549 011 00 medium 3 3 IDLE idle 0 0 1 0 8 0 0 0 23 C549 012 00 medium 3 3 IDLE idle 0 0 1 0 9 0 0 0 3 C542 001 01 g7llulaw 3 3 IDLE idle 0 0 2 0 0 0 512 519 213 C542 002 01 g7llulaw 3 3 IDLE idle 0 0 2 0 1 0 505 502 3 C542 003 01 g7llalaw 3 3 IDLE idle 0 0 2 1 0 0 28756 28966 pals C542 004 01 g7llulaw 3 3 IDLE idle 0 0 2 1 1 0 834 838 3 mi Cisco 10S Voice Video and Fax Configuration Guide __Configuring Voice Ports Verifying Analog and Digital Voice Port Configurations show voice call summary Command Examples In the following sections output examples of the following types are shown Cisco MC3810 Multiservice Concentrator Analog Voice Port Cisco 3600 Series Router Digital Voice Port Cisco MC3810 M ultiservice Concentrator Analog Voice Port The following output is from a Cisco MC3810 multiservice concentrator Router show voice call summary PORT CODEC VAD VTSP STATE VPM S
72. iffer slightly from those presented here based on your platform Cisco IOS release and configuration When in doubt use Cisco IOS command help command to determine the syntax choices that are available Hl Cisco IOS Voice Video and Fax Configuration Guide __Configuring Voice Ports Configuring Digital Voice Ports W The voice port tuning commands are grouped into these categories and explained in the following sections e Auto Cut Through Command page 79 Bit Modification Commands for Digital Voice Ports page 79 Calling Number Outbound Commands page 81 Disconnect Supervision Commands page 82 FXO Supervisory Disconnect Tone Commands page 85 Timeouts Commands page 87 Timing Commands page 89 DTMF Timer Inter Digit Command for Cisco AS5300 Access Servers page 90 Voice Quality Tuning Commands page 92 Full descriptions of the commands in this section can be found in the Cisco IOS Voice Video and Fax Command Reference Release 12 2 Auto Cut Through Command The auto cut through command allows you to connect to PBXs that do not provide an M lead response To configure auto cut through use the following command in voice port configuration mode Command Purpose Router config voiceport auto cut through E amp M only Enables call completion on a router when a PBX does not provide an M lead response Bit Modification Commands for Digital Voice Ports The
73. ifies the number of seconds to allow before the call is released The range is from 3 to 3600 The default is 30 Cisco IOS Voice Video and Fax Configuration Guide VC 88 __Configuring Voice Ports Configuring Digital Voice Ports W Timing Commands To change timing parameters use the following commands as appropriate in voice port configuration mode Command Purpose Step1 Router config voiceport timing clear wait E amp M only Specifies the minimum amount of milliseconds time between the inactive seizure signal and clearing of the call Valid entries for the milliseconds argument are from 200 to 2000 milliseconds The default is 400 Step2 Router config voiceport timing delay duration E amp M only Specifies the delay signal duration for milliseconds delay dial signaling in milliseconds Valid entries are from 100 to 5000 The default is 2000 Step3 Router config voiceport timing delay start E amp M only Specifies minimum delay time in mantaseconds milliseconds from outgoing seizure to outdial address Valid entries are from 20 to 2000 The default is 300 for the Cisco 3600 series routers and 150 for the Cisco MC3810 multiservice concentrators Step4 Router config voiceport timing delay with integrity Cisco MC3810 multiservice concentrators E amp M HEE SOROS ports only Specifies duration of the wink pulse for the delay dial in milliseconds Valid entries are from 0 to 5000 The d
74. in voice port or dial peer mode because the syntax is the same To configure your digital T1 E1 packet voice trunk network module to generate outbound ANI digits on a Cisco AS5300 universal access server use the following commands in voice port configuration mode Cisco IOS Voice Video and Fax Configuration Guide E Configuring Voice Ports W Configuring Digital Voice Ports Step 1 Step 2 Step 3 Purpose Router config voiceport calling number outbound Cisco AS5300 universal access server only range stringl string Specifies ANI to be sent out when the T1 CAS fgd eana command is configured as signaling type The string and string2 arguments are valid E 164 telephone number strings Both strings must be of the same length and cannot be more than 32 digits long Only the last four digits are used for specifying the range string to string2 and for generating the sequence of ANI by rotating through the range until string2 is reached and then starting from string again If strings are less than four digits in length then entire strings are used Router config voiceport calling number outbound Cisco AS5300 universal access server only stringl string2 string3 string4 Specifies ANI to be sent out when the T1 CAS fgd eana command is configured as signaling type This option configures a sequence of discrete strings string string5 to be passed out as ANI for successive calls using the dial
75. iolations 0 Slip Secs 0 Fr Loss Secs 0 Line Err Secs 0 Degraded Mins 0 Errored Secs 0 Bursty Err Secs 0 Severely Err Secs 0 Unavail Secs Cisco MC3810 Multiservice Concentrator E1 Controller The following output is from a Cisco MC3810 multiservice concentrator with an E1 controller Router show controller E1 1 0 El 1 0 is up Applique type is Channelized El Cablelength is short 133 Description El WIC card Alpha No alarms detected Framing is CRC4 Line Code is HDB3 Clock Source is Line Primary Data in current interval 1 seconds elapsed 0 Line Code Violations 0 Path Code Violations 0 Slip Secs 0 Fr Loss Secs 0 Line Err Secs 0 Degraded Mins 0 Errored Secs 0 Bursty Err Secs 0 Severely Err Secs 0 Unavail Secs Cisco AS5800 Universal Access Server T1 Controller The following output is from a Cisco AS5800 universal access server with a T1 controller Router show controller t1 2 Tl 2 48 up No alarms detected Version info of slot 0 HW 2 Firmware 16 PLD Rev 0 Manufacture Cookie Info EEPROM Type 0x0001 EEPROM Version 0x01 Board ID 0x42 Board Hardware Version 1 0 Item Number 73 2217 4 Board Revision AO Serial Number 06467665 PLD ISP Version 0 0 Manufacture Date 14 Nov 1997 Framing is ESF Line Code is B8ZS Clock Source is Internal Cisco IOS Voice Video and Fax Configuration Guide Bm vc 103 Configuring Voice Ports W Verifying Analog and Digital Voice Port Configurations Data i
76. ith a short wink transition from on hook to off hook and back again This wink tells the originating side that the terminating side is ready to receive addressing digits After receiving addressing digits the terminating side then goes off hook for the duration of the call The originating endpoint maintains off hook for the duration of the call E amp M Wink Start Feature Group D In Feature Group D wink start with wink acknowledge handshaking protocol the terminating side responds to an off hook from the originating side with a short wink transition from on hook to off hook and back again just as in the original wink start This wink tells the originating side that the terminating side is ready to receive addressing digits After receiving addressing digits the terminating side provides another wink called an acknowledgment wink that tells the originating side that the terminating side has received the dialed digits The terminating side then goes off hook to indicate connection This last indication can be due to the ultimate called endpoint s having answered The originating endpoint maintains an off hook condition for the duration of the call E amp M Immediate Start In the immediate start protocol the originating side does not wait for a wink before sending addressing information After receiving addressing digits the terminating side then goes off hook for the duration of the call The originating endpoint maintains off hook f
77. izes the line and waits then checks to determine whether the called side is on hook before proceeding if not it waits until the called side is on hook before sending digits Used for E amp M tie trunk interfaces Note Configuring the signal keyword for one voice port on a Cisco 2600 or 3600 series router VIC changes the signal value for both ports on the VIC Step3 Router config voiceport eptone locale Selects the two letter locale for the voice call progress tones and other locale specific parameters to be used on this voice port Cisco routers comply with the ISO 3166 locale name standards To see valid choices enter a question mark following the eptone command The default is us Step4 Router config voiceport dial type dtmf pulse FXO only Specifies the dialing method for outgoing calls Step5 Router config voiceport operation 2 wire 4 wire E amp M only Specifies the number of wires used for voice transmission at this interface the audio path only not the signaling path The default is 2 wire Step6 Router config voiceport type 1 2 3 5 E amp M only Specifies the type of E amp M interface to which this voice port is connecting See Table 5 on page 40 for an explanation of E amp M types The default is 1 Step7 Cisco 1750 Router and 2600 and 3600 Series Routers FXS only Selects the ring frequency in hertz Router config voiceport ring frequency 25 50 used on th
78. justment section on page 96 Volume too low or missed DTMF Increase speaker s output level or listener s input level See the Voice Level Adjustment section on page 96 Echo interval is greater than 25 ms sounds like a separate voice Configure the echo cancel enable command and increase the value for the echo cancel coverage keyword See the Echo Adjustment section on page 94 Too much echo Reduce the output level at the speaker s voice port See the Voice Level Adjustment section on page 96 Cisco IOS Voice Video and Fax Configuration Guide E Configuring Voice Ports WE Troubleshooting Analog and Digital Voice Port Configurations Voice Port Testing Commands These commands allow you to force voice ports into specific states for testing They require the use of Cisco IOS Release 12 0 7 XK or 12 1 2 T or a later release and they apply only to Cisco 2600 and 3600 series routers and to Cisco MC3810 multiservice concentrators The following types of voice port tests are covered Detector Related Function Tests page 110 Loopback Function Tests page 112 Tone Injection Tests page 113 Relay Related Function Tests page 114 Fax Voice Mode Tests page 114 Detector Related Function Tests Step 1 Using the test voice port detector command you are able to force a particular detector into an on or off state perform tests on the detector and then return the dete
79. ket voice network in order to limit or eliminate any voice distortion due to incorrect input and output decibel levels Adjustments to levels may be required by the type of equipment connected to the network or by local country specific conditions Incorrect input or output levels can cause echo as can an impedance mismatch Too much input gain can cause clipped or fuzzy voice quality If the output level is too high at the remote router s voice port the local caller will hear echo If the local router s voice port input decibel level is too high the remote side will hear clipping If the local router s voice port input decibel level is too low or the remote router s output level is too low the remote side voice can be distorted at a very low volume and DTMF may be missed Use the input gain and output attenuation commands to adjust voice levels and the impedance command to set the impedance value to match that of the voice circuit to which the voice port connects To change parameters related to voice levels use the following commands as appropriate in voice port configuration mode Purpose Router config voiceport input gain value Specifies in decibels the amount of gain to be inserted at the receiver side of the interface increasing or decreasing the signal After an input gain setting is changed the voice call must be disconnected and reestablished before the changes take effect The value argument is any integer
80. l bank A Cisco VoIP gateway rarely provides clocking to the CO because CO clocking is much more reliable The following configuration sets up this clocking method for a digital E1 voice port controller El 1 0 framing crc4 linecoding hdb3 clock source internal dsO group timeslots 1 15 type e amp m wink start Figure 17 Single Voice Port Providing Clocking E10 Va Clock gt 26919 Single Voice Port Receiving Internal Clocking In this scenario the digital voice hardware receives clocking from the connected device CO telephony switch or PBX see Figure 18 The PLL clocking is driven by the clock reference on the receive Rx side of the digital line connection Figure 18 Single E1 Port Receiving Clocking from the Line iG lt Clock Cosme 26920 Cisco IOS Voice Video and Fax Configuration Guide E Configuring Voice Ports WE Configuring Digital Voice Ports The following configuration sets up this clocking method controller T1 1 0 framing esf linecoding ami clock source line ds0 group timeslots 1 12 type e amp m wink start Dual Voice Ports Receiving Clocking from the Line In this scenario the digital voice port has two reference clocks one from the PBX and another from the CO as shown in Figure 19 Because the PLL can derive clocking from only one source this case is more complex than the two preceding examples Before looking at the details consider the followin
81. ler On a Cisco 3640 router for example ds0 group 1 timeslots 1 24 type e amp m wink automatically creates the voice port 1 0 1 when issued in the configuration mode for controller 1 0 On a Cisco MC3810 universal concentrator when you are in the configuration mode for controller 0 the command ds0 group 1 timeslots 1 24 type e amp m wink creates logical voice port 0 1 To map individual DSOs define additional DSO groups under the T1 E1 controller specifying different time slots Defining additional DSO groups also creates individual DSO voice ports Defines the emulated analog signaling method that the router uses to connect to the PBX or PSTN Cisco 10S Voice Video and Fax Configuration Guide __Configuring Voice Ports S Note Configuring Digital Voice Ports W Most digital T1 E1 connections used for switch to switch or switch to router trunks are E amp M connections but FXS and FXO connections are also supported These are normally used to provide emulated OPX Off Premises eXtension from a PBX to remote stations FXO ports connect to FXS ports The FXO or FXS connection between the router and switch CO or PBX must use matching signaling or calls cannot connect properly Either ground start or loop start signaling is appropriate for these connections Ground start provides better disconnect supervision to detect when a remote user has hung up the telephone but ground start is not available on all PBXs Digital
82. llowing commands beginning in global configuration mode Command Purpose Step1 Cisco 2600 and 3600 Series Routers Enters voice port configuration mode The Router config voice port slot port ds0 group no arguments are defined as the following O slot Specifies the router location where the Cisco MC3810 Multiseries Concentrators network module Cisco 2600 3600 and Router config voice port slot ds0 group no MC3810 or voice port adapter Cisco AS5300 AS5800 7200 and 7500 is Cisco AS5300 Universal Access Server installed This is the same number as the Router config voice port controller ds0 group no controller for the T1 E1 voice port A p port Indicates the voice interface card Cisco AS5800 Universal Access Server ae location Router config voice port shelf slot port ds0 group no ds0 group no Specifies the logical voice port that was created with the ds0 group Cisco 7200 Series Routers controller command Router config voice port controller Indicates the controller for the slot port adapter ds0 group no T1 E1 voice port Cisco 7500 Series Routers shelf Specifies the dial shelf which is Router config voice port always 0 slot port adapter slot ds0 group no port adapter Indicates the port adapter for the voice port Step2 Router config voiceport type 1 2 3 5 E amp M only Specifies the type of E amp M interface to which this voice port is connected See
83. m or software image information associated with a feature in this chapter use the Feature Navigator on Cisco com to search for information about the feature or refer to the software release notes for a specific release For more information see the Identifying Supported Platforms section in the Using Cisco IOS Software chapter Cisco IOS Voice Video and Fax Configuration Guide E Configuring Voice Ports WE Voice Port Configuration Overview Voice Port Configuration Overview Voice ports on routers and access servers emulate physical telephony switch connections so that voice calls and their associated signaling can be transferred intact between a packet network and a circuit switched network or device For a voice call to occur certain information must be passed between the telephony devices at either end of the call such as the devices on hook status the line s availability and whether an incoming call is trying to reach a device This information is referred to as signaling and to process it properly the devices at both ends of the call segment that is those directly connected to each other must use the same type of signaling The devices in the packet network must be configured to convey signaling information in a way that the circuit switched network can understand They must also be able to understand signaling information received from the circuit switched network This is accomplished by installing appr
84. ms to 50 s Cisco 10S Voice Video and Fax Configuration Guide __Configuring Voice Ports Configuring Digital Voice Ports W Command Purpose Step10 Router config voice class cadence list cadence id Optional Specifies a tone cadence pattern to be Cer ee ee er eg eee eye ee me detected Specify an on time and off time for each cycle 2 off time cycle 3 on time cycle 3 off time 2 a a cycle of the cadence pattern cycle 4 on time cycle 4 off time The arguments are as follows cadence id Ranges from to 10 There is no default cycle N on time and cycle N off time Range from 0 to 1000 0 ms to 10 s The default is 0 Step11 Router config voice class cadence variation time Optional Specifies the maximum time that the tone onset can vary from the specified onset time and still be detected in 10 millisecond increments The time argument ranges from 0 to 200 0 ms to 2 s The default is 0 Step 12 Router config voice class exit Exits voice class configuration mode Step 13 Cisco 2600 and 3600 Series Routers Enters voice port configuration mode Router config voice port slot subunit port The arguments are as follows Cisco MC3810 Multiservice Concentrators slot Specifies the slot number where the voice network module is installed Cisco 2600 and Cisco 3600 series routers or the router slot number where the analog voice module is installed Cisco MC3810 multiservice
85. n Guide Cisco 3600 Series Hardware Installation Guide Cisco Network Module Hardware Installation Guide Cisco IOS Release 12 0 7 T online document Configuring 1 and 2 Port T1 E1 Multiflex Voice WAN Interface Cards on Cisco 2600 and 3600 Series Routers Cisco MC3810 Multiservice Concentrator To support a T1 or El digital voice interface the Cisco MC3810 multiservice concentrator must be equipped with a digital voice interface card DVM The DVM interfaces with a digital PBX channel bank or video codec It supports up to 24 channels of compressed digital voice at 8 kbps or it can cross connect channelized data from user equipment directly onto the router s trunk port for connection to a carrier network The DVM is available with a balanced interface using an RJ 48 connector or with an unbalanced interface using Bayonet Neill Concelman BNC connectors Optional HCMs can replace standard VCMs to operate according to the voice compression coding algorithm codec specified when the Cisco MC3810 multiservice concentrator is configured The HCM2 provides 4 voice channels at high codec complexity and 8 channels at medium complexity The Cisco IOS Voice Video and Fax Configuration Guide c5 Configuring Voice Ports WE Configuring Digital Voice Ports HCM6 provides 12 voice channels at high complexity and 24 channels at medium complexity You can install one or two HCMs in a Cisco MC3810 but an HCM can not be combined with
86. n current interval 269 seconds elapsed 0 Line Code Violations 0 Path Code Violations 0 Slip Secs 0 Fr Loss Secs 0 Line Err Secs 0 Degraded Mins 0 Errored Secs 0 Bursty Err Secs 0 Severely Err Secs 0 Unavail Secs show voice dsp Command Examples The following output is from a Cisco 3640 router when a digital voice port is configured Router show voice dsp TYPE DSP CH CODEC VERS STATE STATE RST AI PORT TS ABORT TX RX PAK CNT C549 010 00 g729r8 3 3 busy idle 0 0 1 015 1 0 67400 85384 01 g729r8 8 busy idle 0 0 1 015 7 0 67566 83623 02 g729r8 busy idle 0 01 015 13 0 65675 81851 03 g729r8 busy idle 0 01 015 20 0 65530 83610 C549 011 00 g729r8 3 3 busy idle 0 0 1 015 2 0 66820 84799 01 g729r8 8 busy idle 0 0 1 015 8 0 59028 66946 02 g729r8 busy idle 0 01 015 14 0 65591 81084 03 g729r8 busy idle 0 0 T 015 21 0 66336 82739 C549 012 00 g729r8 3 3 busy idle 0 0 1 015 3 0 59036 65245 01 g729r8 8 busy idle 0 0 1 015 9 0 65826 81950 02 g729r8 busy idle 0 0 1 015 5 0 65606 80733 03 g729r8 busy idle 0 01 015 22 0 65577 83532 C549 013 00 g729r8 3 3 busy idle 0 On 1 085 4 0 67655 82974 01 g729r8 8 busy idle 0 0 1 015 0 0 65647 82088 02 g729r8 busy idle 0 0 1 015 7 0 66366 80894 03 g729r8 busy idle 0 01 015 23 0 66339 82628 C549 014 00 g729r8 3 3 busy idle 0 0 1 015 5 0 68439 84677 01 g729r8 8 busy idle 0 0 1 015 il 0 65664 81737 02 g729r8 busy idle 0 0 1 015 8 0 65607 81820 03 g729r8 busy
87. n fixed mode this is also the maximum size of the jitter buffer throughout the call value Specifies the range that depends on type of DSP and configured codec complexity For medium codec complexity the range is from 0 to 150 ms For high codec complexity and DSPs that do not support codec complexity the range is from 0 to 250 ms maximum adaptive mode only Specifies the jitter buffer s upper limit 80ms or the highest value to which the adaptive delay is set minimum adaptive mode only Specifies the jitter buffer s lower limit 10 ms or the lowest value to which the adaptive delay is set default Specifies 40 ms Echo Adjustment Echo is the sound of your own voice reverberating in the telephone receiver while you are talking When timed properly echo is not a problem in the conversation however if the echo interval exceeds approximately 25 milliseconds it is distracting Echo is controlled by echo cancellers In the traditional telephony network echo is generally caused by an impedance mismatch when the four wire network is converted to the two wire local loop In voice packet based networks echo cancellers are built into the low bit rate codecs and are operated on each DSP By design echo cancellers are limited by the total amount of time they wait for the reflected speech to be received which is known as an echo trail The echo trail is normally 32 milliseconds In Cisco System s voic
88. n milliseconds for a wink start signal Valid entries are from 100 to 5000 The default is 200 DTMF Timer Inter Digit Command for Cisco AS5300 Access Servers Step 1 Step 2 Step 3 Step 4 To configure the DTMF timer for Cisco AS5300 access servers use the following commands beginning in global configuration mode Command Purpose Router config controller T1 number Configures a T1 controller and enters controller configuration mode Router config dsO group channel number timeslots range type signaling type dtmf dnis Configures channelized T1 timeslots which enables a Cisco AS5300 modem to answer and send an analog call Router config cas custom channel Customizes E1 R2 signaling parameters for a particular E1 channel group on a channelized E1 line Router conf ctrl cas dtmf timer inter digit milliseconds Configures the DTMF inter digit timer for a DSO group Cisco IOS Voice Video and Fax Configuration Guide __Configuring Voice Ports Configuring Digital Voice Ports W Verifying DTMF Timer Inter Digit Command To verify the DTMF timer use the following command in EXEC mode Command Purpose Router show running config Displays the configuration information currently running on the terminal Voice Activity Detection Commands Related to Voice Port Configuration Mode Command In normal voice conversations only one person speaks at a time
89. nce is set to 600r Ohm E amp M Type is unknown Dial Type is dtmf n Seizure is inactive Out Seizure is inactive Digit Duration Timing is set to 0 ms nterDigit Duration Timing is set to 0 ms Pulse Rate Timing is set to 0 pulses second nterDigit Pulse Duration Timing is set to 0 ms Clear Wait Duration Timing is set to 0 ms Wink Wait Duration Timing is set to 0 ms Wink Duration Timing is set to 0 ms Delay Start Timing is set to 0 ms Delay Duration Timing is set to 0 ms Cisco 3600 Series Router Analog FXS Voice Port The following output is from a Cisco 3600 series router analog FXS voice port Router show voice port 1 2 Voice port 1 2 Slot is 1 Port is 2 Type of VoicePort is FXS Operation State is UP Administrative State is UP No Interface Down Failure Description is not set Noise Regeneration is enabled Non Linear Processing is enabled In Gain is Set to 0 dB Out Attenuation is Set to 0 dB Echo Cancellation is enabled Echo Cancel Coverage is set to 8 ms Connection Mode is normal Connection Number is not set Initial Time Out is set to 10 s Interdigit Time Out is set to 10 s Coder Type is g729ar8 Companding Type is u law Voice Activity Detection is disabled Ringing Time Out is 180 s Wait Release Time Out is 30 s Nominal Playout Delay is 80 milliseconds mi Cisco 10S Voice Video and Fax Configuration Guide __Configuring Voice Ports Verifying Analog and Digital Voice Port Configurations i
90. nd Fax Configuration Guide __Configuring Voice Ports Configuring Digital Voice Ports W Command Purpose e amp m immediate start E amp M Immediate Start e amp m melcas delay E amp M Mercury Exchange Limited Channel Associated Signaling MELCAS delay start signaling support e amp m melcas immed E amp M MELCAS immediate start signaling support e amp m melcas wink E amp M MELCAS wink start signaling support e amp m wink start The originating endpoint sends an off hook signal and waits for a ext sig For the specified channel automatically generates the off hook signal and stays in the off hook state fgd eana Feature Group D Exchange Access North American fgd os Feature Group D Operator Services fxo melcas MELCAS Foreign Exchange Office signaling support fxs melcas MELCAS Foreign Exchange Station signaling support fxs ground start FXS Ground Start fxs loop start FXS Loop Start none Null Signaling for External Call Control p7 Specifies the p7 switch type rl itu R1 ITU sas ground start SAS Ground Start sas loop start SAS Loop Start The r1 and r2 keywords refer to line signaling based on international signaling standards The r1 itu keywords are based on signaling standards in countries besides the United States An ITU variant means that there are multiple R1 standards in a particular country but that Cisc
91. network module or WAN interface card to provide the connection to the network LAN or WAN Establish a working IP and Frame Relay or ATM network For more information about configuring IP refer to the Cisco IOS IP Configuration Guide Release 12 2 Install appropriate voice processing and voice interface hardware on the router See the Configuring Platform Specific Analog Voice Hardware section on page 43 Preparing to Configure Analog Voice Ports Before configuring an analog voice port assemble the following information about the telephony connection that the voice port will be making If connecting to a PBX it is important to understand the PBX s wiring scheme and timing parameters This information should be available from your PBX vendor or the reference manuals that accompany your PBX Telephony signaling interface FXO FXS or E amp M Locale code usually the country for call progress tones If FXO type of dialing DTMF touch tone or pulse If FXO type of start signal loop start or ground start If E amp M type I H IL or V If E amp M type of line two wire or four wire If E amp M type of start signal wink immediate delay dial Table 6 should help you determine which hardware and configuration instructions are appropriate for your situation Table 7 on page 42 shows slot and port numbering which differs for each of the voice enabled routers More current information may be available in the release notes that ac
92. ng Voice Ports WE Configuring Digital Voice Ports Figure 16 T1 Controller Configuration on Cisco 2600 or 3600 Series Routers Network module slot 1 Ti VWIC slot 0 2 az Ce E ae Creates DSO group or logical voice port 1 0 1 by grouping 12 time slots together Configures T1 controller 1 0 controller t1 1 07 framing esf clock source line linecode b8zs 37760 dsO group 1 timeslots 1 12 type e amp m wink start Voice port controller configuration includes setting the parameters described in the following sections Framing Formats on Digital T1 E1 Voice Ports Clock Sources on Digital T1 E1 Voice Ports Line Coding on Digital T1 E1 Voice Ports DSO Groups on Digital T1 E1 Voice Ports Another controller command that might be needed cablelength is discussed in the Cisco IOS Interface Command Reference Release 12 2 Framing Formats on Digital T1 E1 Voice Ports The framing format parameter describes the way that bits are robbed from specific frames to be used for signaling purposes The controller must be configured to use the same framing format as the line from the PBX or CO that connects to the voice port you are configuring Digital T1 lines use super frame SF or extended super frame ESF framing formats SF provides two state continuous supervision signaling in which bit values of 0 are used to represent on hook and bit values of 1 are
93. o supports the ITU variant Step 7 Router config controller no shutdown Activates the controller Cisco IOS Voice Video and Fax Configuration Guide EE Configuring Voice Ports WE Configuring Digital Voice Ports Configuring Basic Voice Port Parameters for Digital T1 E1 Voice Ports For FXO and FXS connections the default voice port parameter values are often adequate However for E amp M connections it is important to match the characteristics of your PBX so voice port parameters may need to be reconfigured from their defaults Each voice port that you address in digital voice port configuration is one of the logical voice ports that you created with the ds0 group command Companding from compression and expansion used in Step 4 of the following table is the part of the PCM process in which analog signal values are logically rounded to discrete scale step values on a nonlinear scale The decimal step number is then coded in its binary equivalent prior to transmission The process is reversed at the receiving terminal using the same nonlinear scale 3 0 The commands keywords and arguments that you are able to use may differ slightly from those presented here based on your platform Cisco IOS release and configuration When in doubt use Cisco IOS command help command to determine the syntax choices that are available To configure basic parameters for digital T1 E1 voice ports use the fo
94. odule can be installed in a Cisco 3620 router A Cisco 3640 router can support three modules and as many as six can be installed in a Cisco 3660 router slot is the router location of the voice module 3620 0 to 1 3640 0 to 3 3660 0 to 5 port is the VWIC location in the network module Oto 1 E Cisco IOS Voice Video and Fax Configuration Guide __Configuring Voice Ports Table 8 Router Platform Voice Hardware Configuring Digital Voice Ports W Digital Voice Slot Port Designations continued Slot Number Port Number Cisco MC3810 Digital voice module DVM Voice compression module VCM3 or VCM6 or High compression module HCM2 or HCM6 VCM3 and VCM6 do not support codec complexity options 1 Cisco AS5300 One Octal T1 E1 feature card eight ports or one Quad T1 E1 feature card four ports and one or two VFCs for voice and fax features controller is Octal 0 to 7 Quad 0 to 3 Cisco AS5800 Up to four 12 port T1 E1 trunk cards and up to eight VFCs Shelf is 1 slot is 0 to 5 Oto 11 Cisco 7200 series e Two port T1 E1 enhanced digital voice port adapters PA VXC high capacity PA VXB moderate capacity Port adapter slot 0 is reserved for the Fast Ethernet port on the I O controller if present Port adapter slot from 1 to 4 or from 1 to 6 Interface port 0 to 1 Cisco 7500 series PA VXB and PA V
95. oice port adapters are supported on Cisco 7200 and Cisco 7500 series routers two port high capacity up to 48 or 120 channels of compressed voice depending on codec choice and two port moderate capacity up to 24 or 48 channels of compressed voice These single width port adapters incorporate two universal ports configurable for either T1 or E1 connection for use with high performance digital signal processors DSPs Integrated CSU DSUs echo cancellation and DSO drop and insert functionality eliminate the need for external line termination devices and multiplexers For more information refer to the following publications Cisco 7200 VXR Installation and Configuration Guide Cisco 7500 Series Installation and Configuration Guide Two Port T1 El Moderate Capacity and High Capacity Digital Voice Port Adapter Installation and Configuration Note For current information about supported hardware see the release notes for the platform and Cisco IOS release being used Configuring Basic Parameters on Digital T1 E1 Voice Ports This section describes commands for basic digital voice port configuration Make sure you have all the data recommended in the Preparing Information to Configure Digital Voice Ports section on page 56 before starting this procedure Cisco IOS Voice Video and Fax Configuration Guide E Configuring Voice Ports WE Configuring Digital Voice Ports The basic steps for configuring digital voic
96. oice port command reports at the protocol level and the show controller command reports at the driver level The driver is not notified of any bit manipulation using the condition command As a result the show controller command output does not account for the bit conditioning Step 2 Router config voiceport define tx bits seize idle 0000 0001 0010 0011 0101 0110 0111 1000 1001 1010 1101 1110 1111 Digital E1 E amp M voice ports on Cisco 2600 and 3600 series routers and Cisco MC3810 multiservice concentrators only Defines specific transmit or receive signaling bits to match the bit patterns required by a connected device for North American E amp M and E amp M MELCAS voice signaling if patterns different from the preset defaults are required Also specifies which bits a voice port monitors and which bits it ignores if patterns that are different from the defaults are required See the define command for the default signaling patterns as defined in American National Standards Institute ANSI and code excited linear prediction compression CEPT standards The keywords are as follows tx bits Indicates the pattern applies to transmit signaling bits Cisco IOS Voice Video and Fax Configuration Guide __Configuring Voice Ports Step 3 Command Configuring Digital Voice Ports W Purpose rx bits Indicates the pattern applies to receive signaling bits s
97. olerance of one another one of the controllers is likely to experience slips To prevent slips ensure that the two T1 E1 lines are within clocking tolerance of one another even if the lines are from different providers Dual Voice Ports One Receives Clocking and One Provides Clocking In this scenario the digital voice hardware receives clocking for the PLL from E1 0 and uses this clock as a reference to clock E1 1 see Figure 20 If controller E1 0 fails the PLL internally generates the clock reference to drive El 1 Figure 20 Dual E1 ports One Receiving and One Providing Clocking lt Clock y y PB E11 Clock gt 26922 x lt The following configuration sets up this clocking method controller El 1 0 framing crc4 linecoding hdb3 clock source line dsO group timeslots 1 15 type e amp m wink start controller El 1 1 framing crc4 linecoding hdb3 clock source internal dsO group timeslots 1 15 type e amp m wink start Cisco IOS Voice Video and Fax Configuration Guide Configuring Voice Ports WE Configuring Digital Voice Ports Dual Voice Ports Router Provides Both Clocks In this scenario the router generates the clock for the PLL and therefore for both voice ports see Figure 21 Figure 21 Dual E1 Ports both Clocks from the Router lt gt E1 1 1 Clock _ gt PBX 26923 The following configuration sets u
98. on To configure codec complexity use the following commands beginning in privileged EXEC mode Command Purpose Step1 Router show voice dsp Checks the DSP voice channel activity If any DSP voice channels are in the busy state codec complexity cannot be changed When all of the DSP channels are in the idle state continue to Step 2 Step2 Router configure terminal Enters global configuration mode Step3 Router config voice card slot Enters voice card configuration mode for the card or cards in the slot specified For the Cisco 2600 and 3600 series routers the slot argument ranges from 0 to 5 For the Cisco MC3810 multiservice concentrator slot must be 0 Step4 Router config voicecard codec complexity Specifies codec complexity based on the codec standard being high med used This setting restricts the codecs available in dial peer configuration All voice cards in a router must use the same codec complexity setting The keywords are as follows high Optional Specifies up to six voice or fax calls completed per PVDM 12 using the following codecs G 711 G 726 G 729 G 729 Annex B G 723 1 G 723 1 Annex A G 728 and fax relay e med Optional Supports up to 12 voice or fax calls completed per PVDM 12 using the following codecs G 711 G 726 G 729 Annex A G 729 Annex B with Annex A and fax relay The default is med Note On the Cisco MC3810 multiservice concentrator this command is
99. one equipment keysets and PBXs An FXO interface is used for trunk or tie line connections to a PSTN CO or to a PBX that does not support E amp M signaling when local telecommunications authority permits This interface is of value for off premise station applications A standard RJ 11 modular telephone cable connects the FXO voice interface card to the PSTN or PBX through a telephone wall outlet FXO and FXS interfaces indicate on hook or off hook status and the seizure of telephone lines by one of two access signaling methods loop start or ground start The type of access signaling is determined by the type of service from the CO standard home telephone lines use loop start but business telephones can order ground start lines instead E Cisco IOS Voice Video and Fax Configuration Guide __Configuring Voice Ports E amp M Interfaces Voice Port Configuration Overview il Loop start is the more common of the access signaling techniques When a handset is picked up the telephone goes off hook this action closes the circuit that draws current from the telephone company CO and indicates a change in status which signals the CO to provide dial tone An incoming call is signaled from the CO to the handset by sending a signal in a standard on off pattern which causes the telephone to ring Loop start has two disadvantages however that usually are not a problem on residential telephones but that become significant with the high
100. opriate voice hardware in the router or access server and by configuring the voice ports that connect to telephony devices or the circuit switched network The illustrations below show examples of voice port usage In Figure 10 one voice port connects a telephone to the wide area network WAN through the router In Figure 11 one voice port connects to the PSTN and another to a telephone the router acts like a small PBX Figure 12 shows how two PBXs can be connected over a WAN to provide toll bypass Figure 10 Telephone to WAN Voice port Serial or 1 0 0 Ethernet port EN a G 37754 Figure 11 Telephone to PSTN Voice port Voice port 1 0 0 0 0 1 o B Figure 12 PBX to PBX over a WAN Voice port Serial or Serial or Voice port 1 0 0 Ethernet port Ethernet port 1 0 0 s oe gt eS g ry A Vv EEE a gt 37756 E Cisco IOS Voice Video and Fax Configuration Guide __Configuring Voice Ports Voice Port Configuration Overview W Cisco provides a variety of Cisco IOS commands for flexibility in programming voice ports to match the physical attributes of the voice connections that are being made Some of these connections are made using analog means of transmission while others use digital transmission Table 4 shows the analog and digital voice port connection support of the router platforms discussed in this chapter Table 4 Analog and Digital Voice port Support on Cisco Routers and
101. or E amp M access signaling and is the default for E amp M voice ports Wink start was developed to minimize glare a condition found in immediate start E amp M in which both ends attempt to seize a trunk at the same time In wink start the calling side seizes the line by going off hook on its E lead then waits for a short temporary off hook pulse or wink from the other end on its M lead before sending address information The switch interprets the pulse as an indication to proceed and then sends the dialed digits as DTMF or dialed pulses In delay dial signaling the calling station seizes the line by going off hook on its E lead After a timed interval the calling side looks at the status of the called side If the called side is on hook the calling side starts sending information as DTMF digits otherwise the calling side waits until the called side goes on hook and then starts sending address information Table 5 E amp M Wiring and Signaling Methods M Lead Signal Battery Lead Signal Ground Lead E amp M Type E Lead Configuration Configuration Configuration Configuration I Output relay to Input referenced to ground ground II Output relay to SG Input referenced to Feed for M Return for E ground connected to 48V __ galvanically isolated from ground Ill Output relay to Input referenced to Connected to 48V Connected to ground ground ground V Output relay to Input referenced to gro
102. or the duration of the call Feature Group D is supported on Cisco AS5300 platforms and on Cisco 2600 3600 and 7200 series with digital T1 packet voice trunk network modules Feature Group D is not supported on E1 or analog voice ports Cisco IOS Voice Video and Fax Configuration Guide E Configuring Voice Ports WE Configuring Digital Voice Ports To configure controller settings for digital T1 E1 voice ports use the following commands beginning in global configuration mode Command Purpose Step1 Cisco 7200 and 7500 series Defines the card as T1 or El and stipulates the Router config card type t1 el slot location The keywords and arguments are as follows t1 el Defines the type of card slot A value from 0 to 5 Step2 Cisco 2600 and 3600 series Cisco M C3810 and Cisco 7200 series Enters controller configuration mode Router config controller t1 el slot port The keywords and arguments are as follows Cisco AS5300 t1 el1 tThe type of controller Router config controller t1 el number slot port The backplane slot number and port number for the interface being Cisco AS5800 configured Router config controller t1 el shelf slot port number The network processor module 5 number the range is from 0 to 2 Cisco 7500 series Router config controller t1 el slot port adapter slot shelf slot port Indicates the controller ports the range for
103. ow the different signaling interfaces are associated with different uses of voice ports In Figure 13 FXS signaling is used for end user telephony equipment such as a telephone or fax machine Figure 14 shows an FXS connection to a telephone and an FXO connection to the PSTN at the far side of a WAN this might be a telephone at a local office going over a WAN to a router at headquarters that connects to the PSTN In Figure 15 two PBXs are connected across a WAN by E amp M interfaces This illustrates the path over a WAN between two geographically separated offices in the same company Cisco IOS Voice Video and Fax Configuration Guide E Configuring Voice Ports WE Voice Port Configuration Overview Figure 13 FXS Signaling Interfaces Voice port Serial or Serial or Voice port 1 0 0 Ethernet port Ethernet port 1 0 0 a EA lea FXS Q xs 5 Figure 14 FXS and FXO Signaling Interfaces Voice port Serial or Serial or Voice port 1 0 0 Ethernet port Ethernet port 1 0 0 A Soo 37758 Figure 15 E amp M Signaling Interfaces Voice port Serial or Serial or Voice port 1 0 0 Ethernet port Ethernet port 1 0 0 PBX PBX A ea ea AN en E amp M 37759 L FXS and FXO Interfaces An FXS interface connects the router or access server to end user equipment such as telephones fax machines or modems The FXS interface supplies ring voltage and dial tone to the station and includes an RJ 11 connector for basic teleph
104. p this clocking method controller El 1 0 framing crc4 linecoding hdb3 clock source internal dsO group timeslots 1 15 type e amp m wink start 1 controller El 1 1 framing esf linecoding b8zs clock source internal ds0 group timeslots 1 15 type e amp m wink start Line Coding on Digital T1 E1 Voice Ports Digital T1 E1 interfaces require that line encoding be configured to match that of the PBX or CO that is being connected to the voice port Line encoding defines the type of framing used on the line T1 line encoding methods include alternate mark inversion AMI and binary 8 zero substitution B8ZS AMI is used on older T1 circuits and references signal transitions with a binary 1 or mark B8ZS a more reliable method is more popular and is recommended for PRI configurations as well B8ZS encodes a sequence of eight zeros in a unique binary sequence to detect line coding violations Supported E1 line encoding methods are AMI and high density bipolar 3 HDB3 which is a form of zero suppression line coding DSO Groups on Digital T1 E1 Voice Ports For digital voice ports a single command ds0 group performs the following functions Defines the T1 E1 channels for compressed voice calls Automatically creates a logical voice port The numbering for the logical voice port created as a result of this command is controller ds0 group no where controller is defined as the platform specific address for a particular control
105. rature For more information on codec complexity see the Configuring Codec Complexity for Analog Voice Ports on the Cisco MC3810 with High Performance Compression Modules section on page 45 Two configuration task tables are shown below one for the Cisco 2600 and 3600 series routers and the Cisco MC3810 concentrator which use voice card configuration mode and the second for the Cisco 7200 and 7500 series routers which use DSP interface configuration mode Cisco IOS Voice Video and Fax Configuration Guide __Configuring Voice Ports Configuring Digital Voice Ports W Cisco 2600 and 3600 Series and Cisco MC3810 This procedure applies to voice ports on digital packet voice trunk network modules on Cisco 2600 series and Cisco 3600 series routers and to voice ports on HCMs on Cisco MC3810 multiservice concentrators 2 Note On Cisco 2600 and 3600 series routers with digital T1 E1 packet voice trunk network modules codec complexity cannot be configured if DSO groups are configured Use the no ds0 group command to remove DSO groups before configuring codec complexity L Note On the Cisco MC3810 multiservice concentrator with high compression modules check the DSP voice channel activity with the show voice dsp command If any DSP voice channels are in the busy state you cannot change the codec complexity When all of the DSP channels are in the idle state you can make changes to the codec complexity selecti
106. rcuits connect telephone switches to one another they do not connect end user equipment to the network The most common form of analog trunk circuit is the E amp M interface which uses special signaling paths that are separate from the trunk s audio path to convey information about the calls The signaling paths are known as the E lead and the M lead The name E amp M is thought to derive from the phrase Ear and Mouth or rEceive and transMit although it could also come from Earth and Magnet The history of these names dates back to the days of telegraphy when the CO side had a key that grounded the E circuit and the other side had a sounder with an electromagnet attached to a battery Descriptions such as Ear and Mouth were adopted to help field personnel determine the direction of a signal in a wire E amp M connections from routers to telephone switches or to PBXs are preferable to FXS FXO connections because E amp M provides better answer and disconnect supervision Like a serial port an E amp M interface has a data terminal equipment data communications equipment DTE DCE type of reference In the telecommunications world the trunking side is similar to the DCE and is usually associated with CO functionality The router acts as this side of the interface The other side is referred to as the signaling side like a DTE and is usually a device such as a PBX Five distinct physical configurations for the signaling part of the interface Types I V
107. remote Optional Specifies the source of dial ringback and busy tones for telephones fax machines or modems connected to a Cisco 800 series router The keywords are as follows local default Specifies that the router supplies the tones remote Specifies that the telephone switch supplies the tones Step7 Router config pots ringing freq 20Hz 25Hz Optional Specifies the frequency at which 50Hz telephones fax machines or modems connected to a Cisco 800 series router ring The keywords are as follows 20Hz Indicates that connected devices ring at 20 Hz 25Hz lIndicates that connected devices ring at 25 Hz 50Hz lIndicates that connected devices ring at 50 Hz The default depends on the country chosen in the pots country command Cisco IOS Voice Video and Fax Configuration Guide E Configuring Voice Ports W Analog Voice Ports Configuration Task List Command Purpose Step 8 Router config pots disconnect time interval Optional Specifies the interval at which the disconnect method is applied if connected telephones fax machines or modems fail to detect that a calling party has disconnected The interval argument is the number of milliseconds of the interval and ranges from 50 to 2000 The default depends on the country chosen in the pots country command Step9 Router config pots silence time seconds Optional Specifies the interval of silence after
108. res In most cases the default values for these commands are sufficient for establishing voice port configurations If it is necessary to change some of these parameters to improve voice quality or to match parameters in proprietary PBXs to which you are connecting use the commands in the Fine Tuning Analog and Digital Voice Ports section on page 78 After voice port configuration make sure the ports are operational by following the steps described in these sections Verifying Analog and Digital Voice Port Configurations page 97 Troubleshooting Analog and Digital Voice Port Configurations page 108 For more information on voice port commands refer to the Cisco IOS Voice Video and Fax Command Reference Configuring Codec Complexity for Digital T1 E1 Voice Ports On the Cisco 2600 3600 7200 and 7500 routers codec complexity can be configured separately for each T1 E1 digital packet voice trunk network module or port adapter On a Cisco MC3810 multiservice concentrator only a single codec complexity setting is used even when two HCMs are installed The value specified in this task affects the choice of codecs available when the codec dial peer configuration command is configured For details on the number of calls that can be handled simultaneously using each of the codec standards refer to the entries for codec and codec complexity in the Cisco IOS Voice Video and Fax Command Reference and to platform specific product lite
109. round battery reversal power denial ring tip ground disable Fax Voice Mode Tests The test voice port switch fax command forces a voice port into fax mode for testing After you enter this command you can use the show voice call or show voice call summary command to check whether the voice port is able to operate in fax mode If no fax data is detected by the voice port the voice port remains in fax mode for 30 seconds and then reverts automatically to voice mode The disable keyword ends the forced mode switch however the fax mode ends automatically after 30 seconds The disable keyword is available only while the voice port is in fax mode E Cisco IOS Voice Video and Fax Configuration Guide __Configuring Voice Ports Troubleshooting Analog and Digital Voice Port Configurations i To force a voice port into fax mode and return it to voice mode use the following commands in privileged EXEC mode Command Purpose Step1 Cisco 2600 and 3600 Series Routers Analog Voice Ports Identifies the voice port you want to test Enter the Router test voice port slot subunit port switch fax keyword fax to force the voice port into fax mode Cisco 2600 and 3600 Series Routers Digital Voice Ports Router test voice port slot port ds0 group switch fax Cisco MC3810 Multiservice Concentrators Analog Voice Ports Router test voice port slot port detector switch fax Cisco MC3810 Multiservice Concentrators Digital Voice Por
110. s channelized trunk cards and connects to the PSTN and the Cisco 7206 router shelf RS which holds port adapters and connects to the IP backbone The dial shelf acts as the access concentrator by accepting and consolidating all types of remote traffic including voice dial in analog and digital ISDN data and industry standard WAN and remote connection types The dial shelf also contains controller cards voice feature cards modem feature cards trunk cards and dial shelf interconnect cards One or two dial shelf controllers DSCs provide clock and power control to the dial shelf cards Each DSC contains a block of logic that is referred to as the common logic and system clocks This block of logic can use a variety of sources to generate the system timing including an E1 or T1 T3 input signal from the BNC connector on the DSC s front panel The configuration commands for the master clock specify the various clock sources and a priority for each source see the Clock Sources on Digital T1 E1 Voice Ports section on page 66 The Cisco AS5800 voice feature card is a multi DSP coprocessing board and software package that adds VoIP capabilities to the Cisco AS5800 platform The Cisco AS5800 voice feature card when used with other cards such as LAN WAN and modem cards provides a gateway for up to 192 packetized voice fax calls and 360 data calls per card A Cisco AS5800 can support up to 1 344 voice calls in split dial shelf configuration with
111. s set for US S Cisco AS5300 Universal Access Server T1 CAS Voice Port The following output is from a Cisco AS5300 universal access server T1 CAS voice port Router show voice port DSO Group 1 0 1 0 Type of VoicePort is CAS Operation State is DORMANT Administrative State is UP No Interface Down Failure Description is not set Noise Regeneration is enabled Non Linear Processing is enabled Music On Hold Threshold is Set to 38 dBm Cisco IOS Voice Video and Fax Configuration Guide E Configuring Voice Ports W Verifying Analog and Digital Voice Port Configurations In Gain is Set to 0 dB Out Attenuation is Set to 0 dB Echo Cancellation is enabled Echo Cancel Coverage is set to 8 ms Playout delay Mode is set to default Playout delay Nominal is set to 60 ms Playout delay Maximum is set to 200 ms Connection Mode is normal Connection Number is not set Initial Time Out is set to 10 s Interdigit Time Out is set to 10 s Call Disconnect Time Out is set to 60 s Ringing Time Out is set to 180 s Companding Type is u law Region Tone is set for US Wait Release Time Out is 30 s Station name None Station number None Voice card specific Info Follows DSO channel specific status info IN OUT PORT CH SIG TYPE OPER STATUS STATUS TIP RING Cisco 7200 Series Router Digital E amp M Voice Port The following output is from a Cisco 7200 series router digital E amp M voice port Router show voice port 1 0 1 receEive
112. s the next best known clocking internal Optional Specifies that the clock is generated from the voice port s internal PLL For more information about clock sources see the Clock Sources on Digital T1 E1 Voice Ports section on page 66 The default is line Step5 Tllines Specifies the line encoding to use Router config controller linecode ami b8zs The keywords are as follows Ellines ami Specifies the alternate mark inversion Router config controller linecode ami hdb3 AMI line code type T1 and E1 b8zs Specifies the binary 8 zero substitution B8ZS line code type T1 only hdb3 Specifies the high density bipolar 3 HDB3 line code type E1 only The default for T1 is ami The default for E1 is hdb3 Cisco IOS Voice Video and Fax Configuration Guide i EEE Configuring Voice Ports WE Configuring Digital Voice Ports Step 6 Command Purpose Cisco 2600 and 3600 Series Routers and Cisco MC3810 Multiservice Concentrators T1 Router config controller dsO group ds0 group no timeslots timeslot list type e amp m delay dial e amp m fgd e amp m immediate start e amp m wink start ext sig fgd eana fxo ground start fxo loop start fxs ground start fxs loop start Cisco 2600 and 3600 Series Routers and Cisco MC3810 Multservice Concentrators E1 Router config controller dsO group ds0 group no timeslots timeslot list type e amp m delay dial
113. se Step1 Cisco 2600 and 3600 Series Routers Analog Voice Ports Identifies the voice port you want to test and enters Router test voice port slot subunit port loopback a keyword for the loopback direction local twork Reece R EKOSE Note A call must be established on the voice Cisco 2600 and 3600 Series Routers Digital Voice Ports pounder tesk Router test voice port slot port ds0 group loopback local network Cisco M C3810 Multiservice Concentrators Analog Voice Ports Router test voice port slot port detector loopback local network Cisco M C3810 Multiservice Concentrators Digital Voice Ports Router test voice port slot ds0 group loopback local network Step2 Cisco 2600 and 3600 Series Routers Analog Voice Ports Identifies the voice port on which you want to end Router test voice port slot subunit port loopback the test and enters the keyword disable to end the disable loopback Cisco 2600 and 3600 Series Routers Digital Voice Ports Router test voice port slot port ds0 group loopback disable Cisco MC3810 Multiservice Concentrators Analog Voice Ports Router test voice port slot port detector loopback disable Cisco MC3810 Multiservice Concentrators Digital Voice Ports Router test voice port slot ds0 group loopback disable E Cisco IOS Voice Video and Fax Configuration Guide __Configuring Voice Ports Troubleshooting Analog and Digital Voice Port Configurations i Tone Injection T
114. seconds and interval is a number one or two digits from 1 to 50 specifying ring interval off time in hundreds of milliseconds The default is the pattern specified by the cptone locale that has been configured Step 10 Router config voiceport description string Attaches a text string to the configuration that describes the connection for this voice port This description appears in various displays and is useful for tracking the purpose or use of the voice port The string argument is a character string from 1 to 255 characters in length The default is that there is no text string describing the voice port attached to the configuration Step 11 Router config voiceport no shutdown Activates the voice port If a voice port is not being used shut the voice port down with the shutdown command Cisco IOS Voice Video and Fax Configuration Guide E Configuring Voice Ports HE Analog Voice Ports Configuration Task List Configuring Analog Telephone Connections on Cisco 803 and 804 Routers Step 1 Step 2 Step 3 Multiple devices analog telephone fax machine or modem can be connected to a Cisco 803 or 804 telephone port The number of devices that can be connected depends on the ringer equivalent number REN of each device that is to be connected The REN can usually be found on the bottom of a device The REN of the router telephone port is 5 so if the REN of each device to be connec
115. special tone to indicate a change in call state Some PBXs and PSTN CO switches provide a 600 millisecond interruption of line power as a supervisory disconnect and others provide supervisory tone disconnect STD This is the signal that the router is looking for when the no supervisory disconnect command is configured on the voice port Cisco IOS Voice Video and Fax Configuration Guide __Configuring Voice Ports Configuring Digital Voice Ports W Note In some circumstances you can use the FXO Disconnect Supervision feature to enable analog FXO ports to monitor call progress tones for disconnect supervision that are returned from a PBX or from the PSTN For more information see the FXO Supervisory Disconnect Tone Commands section on page 85 Cisco IOS Voice Video and Fax Configuration Guide E Configuring Voice Ports WE Configuring Digital Voice Ports To change parameters related to disconnect supervision use the following commands as appropriate in voice port configuration mode Command Purpose Stepl Router config voiceport no battery reversal Analog only Enables battery reversal The default is that battery reversal is enabled For FXO ports Use the no battery reversal command to configure a loop start voice port not to disconnect when it detects a second battery reversal The default is to disconnect when a second battery reversal is detected This functionality is support
116. ster 10 Suspend Poll 0 CODEC Finite State Machine State Idle Event 0 Connection None Call Type Two Party Direction Rx only Line Type 600 ohm PCM Encoding u law Disc Type OSI Ringing Frequency 20Hz Distinctive Ring Guard timer 0 msec Disconnect timer 1000 msec Disconnect Silence timer 5 sec TX Gain 6dB RX Loss 6dB E Cisco IOS Voice Video and Fax Configuration Guide __Configuring Voice Ports Filter Mask 6F Adaptive Cntrl Mask 0 Analog Voice Ports Configuration Task List Hi CODEC Registers SPI Addr 2 DSLAC Revision 4 SLIC Cmd OD TX TS 00 RX TS 00 Op Fn 6F Op Fn2 00 Op Cond 00 AISN 6D ELT B5 EPG 32 52 00 00 SLIC Pin Direction 1F CODEC Coefficients GX AO 00 GR 3A Al Z EA 23 2A 35 A5 9F C2 AD 3A AE 22 46 C2 FO B 29 FA 8F 2A CB A9 23 92 2B 49 F5 37 1D 01 X AB 40 3B 9F A8 7E 22 97 36 A6 2A AE R 01 11 01 90 01 90 01 90 01 90 01 90 GZ 60 ADAPT B 91 B2 8F 62 31 CSM Finite State Machine Call 0 State idle Call Id 0x0 Active no Call 1 State idle Call Id 0x0 Active no Call 2 State idle Call Id 0x0 Active no POTS PORTS 2 Hook Switch Finite State Machine State On Hook Hook Switch Register Event 0 20 CODEC Finite State Machine Event 0 None 600 ohm Ringing Frequency Disconnect timer TX Gain 6dB RX Loss Filter Mask 6F Adaptive Cntrl Mask 0 CODEC Registers State Idle Connection Line
117. ted is 1 a maximum of five devices can be connected to that particular telephone port These routers support touch tone analog telephones only they do not support rotary telephones To configure standard features for analog telephone connections on Cisco 803 and 804 routers use the following commands in global configuration mode Command Purpose Router config pots country country Specifies the country to use for country specific default settings for physical characteristics Enter pots country for a list of supported countries and the codes to enter A default country is not defined Router config pots line type typel type2 type3 Optional Specifies the impedance of telephones fax machines or modems connected to a Cisco 800 series router The keywords are as follows typel Specifies the resistance used for the POTS connection typically 600 ohms type2 Specifies the resistance used for the POTS connection typically 900 ohms type3 Specifies the resistance used for the POTS connection typically 300 400 ohms The default depends on the country chosen in the pots country command Router config pots dialing method overlap enblock Optional Specifies how the router collects and sends digits dialed on connected telephones fax machines or modems The keywords are as follows overlap Tells the router to send each digit dialed in a separate message e enblo
118. the human ear normally accepts up to about 150 ms of delay without noticing delays The ITU G 114 standard recommends no more than 150 ms of one way delay for a normal voice conversation Once the delay exceeds 150 ms a conversation is more like a walkie talkie conversation in which one person must wait for the other to stop speaking before beginning to talk You can measure delay fairly easily by using ping tests at various times of the day with different network traffic loads If network delay is excessive it must be reduced for adequate voice quality Several different types of delay combine to make up the total end to end delay associated with voice calls Propagation delay Amount of time it takes the data to physically travel over the media Handling delay Amount of time it takes to process data by adding headers taking samples forming packets etc Queuing delay Amount of time lost due to congestion Variable delay or jitter Amount of time that causes the conversation to break and become unintelligible Jitter is described in detail below Propagation handling and queuing delay are not addressed by voice port commands and fall outside the scope of this chapter J itter Adjustment Delay can cause unnatural starting and stopping of conversations but variable length delays also known as jitter can cause a conversation to break and become unintelligible Jitter is not usually a problem with PSTN calls beca
119. ther medium or high The difference between medium and high complexity codecs is the amount of CPU power necessary to process the algorithm and therefore the number of voice channels that can be supported by a single DSP All medium complexity codecs can also be run in high complexity mode but fewer usually half as many channels will be available per DSP For details on the number of calls that can be handled simultaneously using each of the codec standards refer to the entries for the codec and codec complexity commands in the Cisco IOS Voice Video and Fax Command Reference On a Cisco MC3810 concentrator only a single codec complexity setting is used even when two HCMs are installed The value that is specified in this task affects the choice of codecs available when the codec dial peer configuration command is configured See the Configuring Dial Plans Dial Peers and Digit Manipulation chapter in this configuration guide Note On the Cisco MC3810 with high performance compression modules check the DSP voice channel activity with the show voice dsp command If any DSP voice channels are in the busy state the codec complexity cannot be changed When all the DSP channels are in the idle state changes can be made to the codec complexity selection Cisco IOS Voice Video and Fax Configuration Guide E Configuring Voice Ports HE Analog Voice Ports Configuration Task List Step 1 Step 2 Step 3 Step 4
120. ting a wrong cptone could result in faulty voice reproduction during analog to digital or digital to analog conversions Music on hold is not heard Reduce the music threshold level Background noise is not heard Enable the comfort noise command Long pauses occur in conversation like speaking on a walkie talkie Overall delay is probably excessive the standard for adequate voice quality is 150 ms one way transit delay Measure delay by using ping tests at various times of the day with different network traffic loads If delay must be reduced areas to examine include propagation delay of signals between the sending and receiving endpoints voice encoding delay and the voice packetization time for various VoIP codecs Jerky or choppy speech Variable delay or jitter is being introduced by congestion in the packet network Two possible remedies are to Reduce the amount of congestion in your packet network Pings between VoIP endpoints will give an idea of the round trip delay of a link which should never exceed 300 ms Network queuing and dropped packets should also be examined Increase the size of the jitter buffer with the playout delay command See the Jitter Adjustment section on page 92 Clipped or fuzzy speech Reduce input gain See the Voice Level Adjustment section on page 96 Clipped speech Reduce the input level at the listener s router See the Voice Level Ad
121. ts Router test voice port slot ds0 group switch fax Step2 Cisco 2600 and 3600 Series Routers Analog Voice Ports Identifies the voice port on which you want to end Router test voice port slot subunit port switch the test Enter the keyword disable to return the disable voice port to voice mode Cisco 2600 and 3600 Series Routers Digital Voice Ports Router test voice port slot port ds0 group switch disable Cisco MC3810 Multiservice Concentrators Analog Voice Ports Router test voice port slot port detector switch disable Cisco MC3810 Multiservice Concentrators Digital Voice Ports Router test voice port slot ds0 group switch disable Cisco IOS Voice Video and Fax Configuration Guide gL vc 115 Configuring Voice Ports E Troubleshooting Analog and Digital Voice Port Configurations Cisco IOS Voice Video and Fax Configuration Guide
122. two 7206VXR router shelves Cisco 10S Voice Video and Fax Configuration Guide VC 60 __Configuring Voice Ports Configuring Digital Voice Ports W For more information refer to the following publications Cisco AS5800 Universal Access Server Operation Administration Maintenance and Provisioning Guide Cisco AS5800 Access Server Hardware Installation Guide Cisco 7200 and Cisco 7500 Series Routers Cisco 7200 and Cisco 7500 series routers support multimedia routing and bridging with a wide variety of protocols and media types The Cisco 7000 family versatile interface processor VIP is based on a RISC engine optimized for I O functions To this engine are attached one or two port adapters or daughter boards which provide the media specific interfaces to the network The network interfaces provide connections between the routers peripheral component interconnect PCI buses and external networks Port adapters can be placed in any available port adapter slot in any desired combination T1 E1 high capacity digital voice port adapters for Cisco 7200 and Cisco 7500 series routers allow enterprises or service providers using the equipped routers as customer premise equipment to deploy digital voice and fax relay These port adapters receive constant bit rate telephony information over T1 E1 interfaces and can convert that information to a compressed format for transmission as voice over IP VoIP Two types of digital v
123. und 48V Analog Voice Ports Configuration Task List Analog voice port interfaces connect routers in packet based networks to analog two wire or four wire analog circuits in telephony networks Two wire circuits connect to analog telephone or fax devices and four wire circuits connect to PBXs Typically connections to the PSTN CO are made with digital interfaces This section describes how to configure analog voice ports and covers the following topics Configuring Codec Complexity for Analog Voice Ports on the Cisco MC3810 with High Performance Compression Modules page 45 Configuring Basic Parameters on Analog FXO FXS or E amp M Voice Ports page 46 Configuring Analog Telephone Connections on Cisco 803 and 804 Routers page 50 E Cisco IOS Voice Video and Fax Configuration Guide __Configuring Voice Ports Analog Voice Ports Configuration Task List Hi Three other sections later in the chapter provide help with fine tuning and troubleshooting Fine Tuning Analog and Digital Voice Ports page 78 Verifying Analog and Digital Voice Port Configurations page 97 Troubleshooting Analog and Digital Voice Port Configurations page 108 Prerequisites for Configuring Analog Voice Ports Obtain two or four wire line service from your service provider or from a PBX Complete your company s dial plan Establish a working telephony network based on your company s dial plan Install at least one other
124. uring Basic Parameters on Analog FXO FXS or E amp M Voice Ports This section describes commands for basic analog voice port configuration All the data recommended in the Preparing to Configure Analog Voice Ports section on page 41 should be gathered before starting this procedure If configuring a Cisco MC3810 multiservice concentrator that has HCMs codec complexity should also be configured following the steps in the Configuring Codec Complexity for Analog Voice Ports on the Cisco MC3810 with High Performance Compression Modules section on page 45 Note If you have a Cisco MC3810 multiservice concentrator or Cisco 3660 router the compand type a law command must be configured on the analog ports only The Cisco 2660 3620 and 3640 routers do not require the configuration of th compand type a law command however if you request a list of commands the compand type a law command will display E Cisco IOS Voice Video and Fax Configuration Guide __Configuring Voice Ports Step 1 Step 2 3 0 Command Analog Voice Ports Configuration Task List Hi In addition to the basic voice port parameters described in this section there are commands that allow voice port configurations to be fine tuned In most cases the default values for fine tuning commands are sufficient for establishing FXO and FXS voice port configurations E amp M voice ports are more likely to require some configuration If it is necessary
125. use different methods to signal on hook off hook status as shown in Table 5 Cisco voice implementation supports E amp M Types I II III and V The physical E amp M interface is an RJ 48 connector that connects to PBX trunk lines which are classified as either two wire or four wire This refers to whether the audio path is full duplex on one pair of wires two wire or on two pair of wires four wire A connection may be called a four wire E amp M circuit although it actually has six to eight physical wires It is an analog connection although an analog E amp M circuit may be emulated on a digital line For more information on digital voice port configuration of E amp M signaling see the DSO Groups on Digital T1 E1 Voice Ports section on page 70 Cisco IOS Voice Video and Fax Configuration Guide E Configuring Voice Ports HE Analog Voice Ports Configuration Task List PBXs built by different manufacturers can indicate on hook off hook status and telephone line seizure on the E amp M interface by using any of three types of access signaling that are as follows Immediate start is the simplest method of E amp M access signaling The calling side seizes the line by going off hook on its E lead and sends address information as dual tone multifrequency DTMF digits or as dialed pulses on Cisco 2600 series routers and Cisco 3600 series routers following a short fixed length pause Wink start is the most commonly used method f
126. use the bandwidth of calls is fixed and each call has a dedicated circuit for the duration of the call However in VoIP networks data traffic might be bursty and jitter from the packet network can become an issue Especially during times of network congestion packets from the same conversation can arrive at different interpacket intervals disrupting the steady even delivery needed for voice calls Cisco voice gateways have built in jitter buffering to compensate for a certain amount of jitter the playout delay command can be used to adjust the jitter buffer Normally the defaults in effect are sufficient for most networks However a small playout delay from the jitter buffer can cause lost packets and choppy audio and a large playout delay can cause unacceptably high overall end to end delay Cisco 10S Voice Video and Fax Configuration Guide __Configuring Voice Ports Configuring Digital Voice Ports W Note Prior to Cisco IOS Release 12 1 5 T playout delay was configured in voice port configuration mode For Cisco IOS Release 12 1 5 T and later releases in most cases playout delay should be configured in dial peer configuration mode on the VoIP dial peer that is on the receiving end of the voice traffic that is to be buffered This dial peer senses network conditions and relays them to the DSPs which adjust the jitter buffer as necessary When multiple applications are configured on the gateway playout delay should be confi
127. used to represent off hook ESF robs four bits instead of two yet has little impact on voice quality ESF is required for 64 kbps operation on DSO and is recommended for Primary Rate Interface PRI configurations E1 lines can be configured for cyclic redundancy check CRC4 or no cyclic redundancy check with an optional argument for E1 lines in Australia Clock Sources on Digital T1 E1 Voice Ports Digital T1 E1 interfaces use timers called clocks to ensure that voice packets are delivered and assembled properly All interfaces handling the same packets must be configured to use the same source of timing so that packets are not lost or delivered late The timing source that is configured can be external from the line or internal to the router s digital interface Hl Cisco IOS Voice Video and Fax Configuration Guide __Configuring Voice Ports Configuring Digital Voice Ports W If the timing source is internal timing derives from the onboard phase lock loop PLL chip in the digital voice interface If the timing source is line external then timing derives from the PBX or PSTN CO to which the voice port is connected It is generally preferable to derive timing from the PSTN because their clocks are maintained at an extremely accurate level This is the default setting for the clocks When two or more controllers are configured one should be designated as the primary clock source it will drive the other controllers The lin
128. ust be configured on the analog ports only The Cisco 2660 3620 and 3640 routers do not require the compand type a law command configured however if you request a list of commands the compand type a law command will display Step5 Cisco 2600 series and 3600 series FXS only Selects the ring frequency in hertz Router config voiceport ring frequency 25 50 used on the FXS interface This number must match the connected telephony equipment and Cisco MC3810 can be country dependent If not set properly the Router config voiceport ring frequency 20 30 attached telephony device may not ring or it may buzz The default is 25 on the Cisco 2600 and 3600 series routers and 20 on the Cisco MC3810 multiservice concentrators Cisco IOS Voice Video and Fax Configuration Guide 5m Configuring Voice Ports WE Configuring Digital Voice Ports Command Purpose Step6 Router config voiceport ring number number FXO only Specifies the maximum number of rings to be detected before an incoming call is answered by the router The default is 1 Step7 Router config voiceport ring cadence pattern01 FXS only Specifies an existing pattern for ring pattern02 pattern03 pattern04 pattern05 or defines a new one Each pattern specifies a Bere rege beg eer T pataros paEkarnoa ring pulse time and a ring interval time The pattern10 patternll pattern12 define pulse 8 P 8 intervaly k
129. valid only with one or more HCMs installed and voice card 0 must be specified If two HCMs are installed this command configures both HCMs at once Cisco IOS Voice Video and Fax Configuration Guide EE Configuring Voice Ports WE Configuring Digital Voice Ports Cisco AS5300 Universal Access Server Codec support on the Cisco AS5300 universal access server is determined by the capability list on the voice feature card which defines the set of codecs that can be negotiated for a voice call The capability list is created and populated when VCWare is unbundled and DSPWare is added to VFC Flash memory The capability list does not indicate codec preference it simply reports the codecs that are available The session application decides which codec to use Codec support is configured on dial peers rather than on voice ports see the Configuring Dial Plans Dial Peers and Digit Manipulation chapter in this configuration guide Cisco AS5800 Universal Access Server Selection of codec support on Cisco AS5800 access servers is made during dial peer configuration See the Configuring Dial Plans Dial Peers and Digit Manipulation chapter in this configuration guide Cisco 7200 Series and Cisco 7500 Series Routers On Cisco 7200 series and Cisco 7500 series routers codec complexity is configured on the DSP interface 2 Note Check the DSP voice channel activity using the show interfaces dspfarm command If any D

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