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Cisco IOS Voice Troubleshooting and Monitoring Guide, Release 12.4T
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Guide to Cisco IOS Voice Troubleshooting and Monitoring Features
- Part 1: Troubleshooting Voice Overview
- Part 2 : Troubleshooting Voice Telephony
- Part 3 : Troubleshooting Voice Technologies
- Part 4 : Troubleshooting Telephony Applications
- Part 5 : Troubleshooting Fax
- Part 6 : Monitoring the Voice Network
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Cause Codes and Debug Values
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Feedback
Table Of Contents
Troubleshooting Analog Voice Interfaces to the IP Network
Disabling a Port on a Multiple Port Card
No LED When Phone Off the Hook
Disabling a Port on a Multiple Port Card
Troubleshooting FXO Answer and Disconnect Supervision
Monitoring and Maintaining FXO Answer and Disconnect Supervision
Troubleshooting Caller ID Problems
Calling Number Lost, Name Delivered
Calling Number Delivered, Name Lost
E&M Signaling Unit Side and Trunk Circuit Side Compatibility Issues
Troubleshooting E&M Interfaces at the Physical Level
Preparing to Troubleshoot E&M Physical Problems
Troubleshooting Type I Interfaces
Troubleshooting Type II Interfaces
Troubleshooting Type III Interfaces
Troubleshooting Type V Interfaces
Confirming the PBX E&M Configuration Parameters
Confirming the Cisco IOS Gateway Configuration
Verifying the Wiring Arrangement Between the PBX and the Cisco Gateway
Verifying Supervision Signaling
Verifying That the Cisco Equipment and PBX Are Sending and Receiving Digits
Verifyinng That the Gateway Sends the Expected Digits to the PBX
Verify That the Gateway Receives the Expected Digits from the PBX
Verifying Direct Inward Dialing Voice-Port Configuration
Detector-Related Function Tests
Troubleshooting Analog Voice Interfaces to the IP Network
If you are troubleshooting an analog connection, you must understand what type of circuit and interface your voice port is using. 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. Analog voice telephony interfaces include foreign exchange office (FXO), foreign exchange station (FXS), and receive and transmit (E&M). Direct Inward Dialing (DID) is a service offered by telephone companies that enables callers to dial directly to an extension on a PBX without the assistance of an operator or automated call attendant.
To troubleshoot analog voice interfaces, see the following sections:
If you are troubleshooting a connection to a PBX, you might find the PBX interoperability notes useful. These notes contain configuration information for Cisco gateways and several types of PBXs. To access these notes, use the following website:
http://www.cisco.com/warp/public/779/largeent/avvid/inter_operability/
FXS Interfaces
An FXS interface connects the router or access server to end-user equipment such as telephones, fax machines, and modems. The FXS interface supplies ring, voltage, and dial tone to the station and includes an RJ-11 connector for basic telephone equipment, keysets, and PBXs. In Figure 10, FXS signaling is used for end-user telephony equipment, such as a telephone or fax machine.
Figure 10 FXS Signaling Interfaces
If you are having trouble with an FXS port, check the following sections:
FXS Hardware Troubleshooting
An FXS interface connects directly to a standard telephone, fax machine, or similar device and supplies ring, voltage, and dial tone.
Troubleshoot FXS hardware by checking the following sections:
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Disabling a Port on a Multiple Port Card
Software Compatibility
To ensure that your FXS card is compatible with your software, check the following:
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Cabling
Two types of cabling are supported for Cisco FXS interfaces. They are described in the following sections:
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Note
RJ-11 Connectors
The two-port and four-port FXS interface cards support the RJ-11 connector. Illustrations of the connector ports are shown in Figure 11 and Figure 12. Information about LEDs can be found in the "Connecting Voice Interface Cards to a Network" chapter of the Cisco Interface Card Hardware Installation Guide.
Figure 11 Two-Port FXS Card Front Panel
Figure 12 Four-Port FXS/DID Card Front Panel
For information about the VIC-2FXS interface card, refer to Understanding Foreign Exchange Station (FXS) Voice Interface Cards, document ID 7938.
RJ-21 Connectors on the High-Density Analog Telephony Network Module
The High-Density Analog Telephony network module supports an RJ-21 connector. This network module supports both FXS and FXO traffic. An illustration of the connector port is shown in Figure 13. Information about LEDs and pinouts can be found in the "Connecting High-Density Analog Telephony Network Modules to a Network" chapter of the Cisco Network Modules Hardware Installation Guide.
Figure 13 High-Density Analog Telephony Network Module
Shutdown Port
If the port is not working, be sure the port is not shut down. Enter the show voice port command with the voice port number that you are troubleshooting. The output will tell you:
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Disabling a Port on a Multiple Port Card
If you shut down a port on a multiple-port card, you can disable all of the ports on that card. If only one port is bad and the others are working, in many cases you can disable the bad port and then use the working ports until a replacement arrives. To disable a bad port, use one of the following methods:
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Ring Voltage Problems
Telephone exchanges and FXSs need to supply DC battery and AC ringing to enable the connected telephone equipment to transmit speech energy and to power the telephone equipment's ringing device. This section discusses what voltages are supplied by various Cisco FXS interfaces and how to overcome some known issues regarding voltage levels.
Ringing Voltages
The industry standard for PBX and key systems requires that the ring detection circuit be able to detect a ringing signal as low as 40 Vrms. This voltage takes into account the effects of load and cabling voltage drop on a ringing signal generated from a central office (CO). Conversely, the CO (exchange) must supply ringing with enough power to drive the maximum load over the maximum cable length. In order to meet this requirement, a CO-based unit must present a ringing signal with an amplitude of approximately 85 to 100 Vrms. Cisco voice gateways are intended for use as on premise services (ONS) equipment that is colocated or fairly close to equipment that detects ringing, so it can therefore use a lower ringing voltage and still meet the 40 Vrms 5 Ringer Equivalence Number (REN) requirement.
Idle Battery Voltage
Cisco voice gateways were designed for ONS connections and by default the FXS interface supplies either -24 Vdc or -36 Vdc idle battery, whereas off premise services, such as a CO, would require voltages of -48 V because it might have to interconnect over much greater cable lengths. Certain Cisco FXS interfaces can be configured to supply higher voltages.
Idle Line Voltages
Table 24 shows idle line voltages supplied by various Cisco gateway FXS interfaces.
Ring Voltage Problems
Voltage problems can cause three types of problems:
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Certain automated devices, such as fax machines, answer machines, multiline phones and voice mail systems, look at the line voltage in order to deduce if the line is busy or idle. If another device is off hook, then the line voltage drops, and the automated system does not answer or initiate a call. If the threshold being used is close to -24 V or higher, this can cause the device not to work as expected.
Certain phones might not ring when the default ring voltage and ring frequency are applied from the Cisco FXS interface.
Answering and Call Initiation Problems with Automated Telephony Devices
In voice port configuration mode, configure the idle-voltage command on the voice port of the FXS to increase idle battery voltage from -24 V to -48 V. The idle-voltage low setting designates -24 V and the idle-voltage high setting designates -48 V.
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Ringing Problems
Phone manufacturers sometimes use frequency filters known as antitinkle circuits to prevent ringer devices from sounding while the user is dialing. Sometimes it is necessary to adjust the frequency of the ring to suit the connected device.
Configure the ring frequency for Cisco modular access routers by issuing the following command:
Router(config-voiceport)# ring frequency ?25 ring frequency 25 Hertz50 ring frequency 50 HertzConfigure the ring frequency for the Cisco IAD2400 platform by issuing the following command:
Router(config-voiceport)# ring frequency ?20 ring frequency 20 Hertz30 ring frequency 30 HertzTo prevent ringer devices from sounding, you can also provide a voltage threshold so that the lower voltages, which can be produced during dialing, are ignored. Increasing the voltage can overcome this.
Configure the DC offset voltage on Cisco IAD2400 series routers by issuing the following command:
Router(config-voiceport)# ring dc-offset ?10-volts Ring DC offset 10 volts20-volts Ring DC offset 20 volts24-volts Ring DC offset 24 volts
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FXS Ring Failure in the United Kingdom
A telephone approved for the United Kingdom might fail to ring when connected to a Cisco FXS port. The failure results from a physical interoperability issue and is independent of Cisco hardware or software. British Telecom did not implement RJ-11 type connectors when it adopted plug-and-socket connection methodology. RJ-11 connectors allow parallel connectivity for the transmission path and the ringer circuit. They were not used because older telephones needed to have their ringer circuits connected in series due to a requirement for high current.
Outside the United Kingdom, ringer circuitry is self-contained in each phone. The U.K. implementation puts the capacitor, which provides the AC ring path, and the antitinkle feature (prevents the bell or ringer from sounding when pulse dialing is used) externally in the first socket, connected to the local loop.
In the United Kingdom, certain British Approval Board for Telecommunications (BABT) telephones fail to ring when they are connected to FXS ports on Cisco voice-enabled routers and switches. Outgoing calls can be made and voice communication in both directions can be established. However, incoming calls do not ring the telephone. These telephones functioned correctly before they were connected to the FXS ports.
Because a proprietary connection system is implemented, you must use an adapter to connect the telephone to an FXS port. The adapter must be a master that contains the capacitor, or the telephone fails to ring.
For a schematic and more information, refer to Understanding Why Telephones in the United Kingdom Connected to Cisco FXS Interfaces May Fail to Ring, document ID 25800.
Unbreakable Dial Tone
A common problem encountered in a VoIP network is being unable to break dial tone. The router seizes a line on the local PBX but when digits are dialed, the dial tone stays. The calling party is unable to pass the dual-tone multifrequency (DTMF) tones or digits to the terminating device, resulting in callers being unable to dial the desired extension or interact with a device that needs DTMF tones such as a voice mail or an interactive voice response (IVR) application. This problem can result from a number of sources, for example:
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Make sure the dial type is set as DTMF on both the router and the PBX. The FXS port does not pass on the digits; therefore, this setting is not available on an FXS port. However, this setting can be changed on FXO and E&M ports:
Router(config-voiceport)# dial-type ?dtmf touch-tone dialermf mf-tone dialerpulse pulse dialerFor more information, refer to Inability To Break Dialtone in a Voice over IP Network, document ID 22376.
No LED When Phone Off the Hook
Verify if you have an analog or digital card. If you have an analog card like the VIC-2FXS or the VIC2-4FXS, you might have one of the following problems:
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If you have a digital card like the NM-2V, you might have bad DSPs.
Use the following procedure if there is no LED when your phone is off hook:
Step 1
Step 2
Step 3
Step 4
FXO Interfaces
An FXO interface is used for trunk, or tie line, connections to a PSTN CO or to a PBX that does not support E&M signaling (when local telecommunications authority permits). This interface is of value for off-premises station applications. Figure 14 shows an FXS connection to a telephone and an FXO connection to the PSTN at the far side of a WAN.
Figure 14 FXS and FXO Signaling Interfaces
If you are having trouble with an FXO port, check the following sections:
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FXO Hardware Troubleshooting
An FXO interface is used for trunk connections. Troubleshoot FXO hardware by checking the following sections:
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Software Compatibility
To ensure that your card is compatible with your software, check the following:
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Cabling
Two types of cabling are supported for Cisco FXO interfaces. They are described in the following sections:
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Note
RJ-11 Connectors
The two-port and four-port FXO interface cards support the RJ-11 connector. Illustrations of the connector ports are shown in Figure 15 and Figure 16. Information about LEDs can be found in the "Connecting Voice Interface Cards to a Network" chapter of the Cisco Interface Card Hardware Installation Guide.
Figure 15 Two-Port FXO Card Front Panel
Figure 16 Four-Port FXO Card Front Panel
RJ-21 Connectors on the High-Density Analog Telephony Network Module
The High-Density Analog Telephony network module supports an RJ-21 connector. This network module supports both FXS and FXO traffic. An illustration of the connector port is shown in Figure 17. Information about LEDs and pinouts can be found in the "Connecting High-Density Analog Telephony Network Modules to a Network" chapter of the Cisco Network Modules Hardware Installation Guide.
Figure 17 High-Density Analog Telephony Network Module
Shutdown Port
If the port is not working, be sure the port is not shut down. Enter the show voice port command with the voice port number that you are troubleshooting. The output will tell you:
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Disabling a Port on a Multiple Port Card
If you shut down a port on a multiple-port card, you can disable all of the ports on that card. If only one port is bad and the others are working, in many cases you can disable the bad port and use the working ports until a replacement arrives. To disable a bad port, use one of the following methods:
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FXO Disconnect Failure
When loop-starting signaling is used, an FXO interface looks like a phone to the switch that it is connecting to. The FXO interface closes the loop to indicate off hook. The switch always provides a battery so there is no disconnect supervision from the switch side. Because a switch expects a phone user or modem to hang up the phone when the call is terminated on either side, it also expects the FXO port on the router to hang up. However, the FXO port expects the switch to tell it when to hang up. Because the port relies on the switch, there is no guarantee that a near- or far-end FXO port will disconnect the call once either end of the call has hung up.
The most common symptoms of this problem are phones that continue to ring when the caller has cleared, or FXO ports that remain busy after the previous call should have been cleared.
To troubleshoot this problem, refer to Understanding FXO Disconnect Problem, document ID 8120.
Troubleshooting FXO Answer and Disconnect Supervision
This section describes troubleshooting the FXO Answer and Disconnect Supervision feature for analog FXO voice ports. This feature applies to analog FXO voice ports with loop-start signaling connected to PSTNs, PBXs, or key systems.
The FXO Answer and Disconnect Supervision feature enables analog FXO ports to monitor call-progress tones and to monitor voice and fax transmissions returned from a PBX or from the PSTN.
Answer supervision can be accomplished in two ways: by detecting battery reversal, or by detecting voice, fax, or modem tones. If an FXO voice port is connected to the PSTN and battery reversal is supported, use the battery reversal method. Voice ports that do not support battery reversal must use the answer supervision method, in which answer supervision is triggered when the DSP detects voice, modem, or fax transmissions. Configuring answer supervision automatically enables disconnect supervision; however, you can configure disconnect supervision separately if answer supervision is not configured.
Disconnect supervision can be configured to detect call-progress tones sent by the PBX or PSTN (for example, busy, reorder, out-of-service, number-unavailable), or to detect any tone received (for example, busy tone or dial tone). When an incoming call ends, the DSP detects the associated call-progress tone, causing the analog FXO voice port to go on-hook.
This section provides solutions to problems that you might encounter when implementing the
FXO Answer and Disconnect Supervision feature.Typical problems with the answer supervision feature are as follows:
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If any call legs have IVR configured, ensure that the IVR version is 2.0.
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Answer supervision—either by battery-reversal detection or by call-progress tone detection—is not configured on the voice-port in use.
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The level on the sensitivity parameter in the supervisory answer dualtone command is set too low. Configure the sensitivity for high.
If incorrect disconnect cause codes are reported, check the following:
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Note
If calls are not billed correctly, it might be that answer supervision is not being triggered. For answer supervision to be triggered, voice, fax, or data tones originating at the called-party end must be detected.
To configure the FXO Supervisory Disconnect Tone, refer to the Voice Port Configuration document.
Monitoring and Maintaining FXO Answer and Disconnect Supervision
To monitor the status of the FXO Answer and Disconnect Supervision feature, use the show voice port command, which causes the FXO voice port to be monitored. The following table illustrates the use of the show voice port command for monitoring voice port 1/1/0.
Unbreakable Dial Tone
A common problem encountered in a VoIP network is being unable to break dial tone. The router puts a seizure on the local PBX but when digits are dialed, the dial tone stays. The calling party is unable to pass the DTMF tones or digits to the terminating device, resulting in callers being unable to dial the desired extension or interact with the device that needs DTMF tones, such as a voice mail or IVR application. Here are some possible causes of the problem:
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Make sure the dial type is set as DTMF on both the router and the PBX. The FXS port does not pass on the digits, therefore this setting is not available on an FXS port. However, this setting can be changed on FXO and E&M ports:
Router(config-voiceport)# dial-type ?dtmf touch-tone dialermf mf-tone dialerpulse pulse dialerFor more information, refer to Inability To Break Dialtone in a Voice over IP Network, document ID 22376.
Troubleshooting Caller ID Problems
Several debugs can be used to troubleshoot the Caller ID feature on the routers. The voice port module (VPM) signaling debugs, such as the debug vpm signal command, track the standard debugs with Caller ID feature turned on. These debugs are analyzed from the perspective of the terminating router and its FXO port; the caller ID is sent from this end. The following example shows an FXO port receiving caller ID. In this example, the phone sends the caller ID to the FXO port.
Nov 20 10:40:15.861 EST: [1/0/0] htsp_start_caller_id_rx Nov 20 10:40:15.861 EST: [1/0/0] htsp_set_caller_id_rx:BELLCORE Nov 20 10:40:15.861 EST: htsp_timer - 10000 msec Nov 20 10:40:17.757 EST: [1/0/0, FXOLS_RINGING, E_DSP_SIG_0100] Nov 20 10:40:17.757 EST: fxols_ringing_not Nov 20 10:40:17.761 EST: htsp_timer_stop Nov 20 10:40:17.761 EST: htsp_timer - 10000 msec Nov 20 10:40:18.925 EST: [1/0/0] htsp_stop_caller_id_rx Nov 20 10:40:21.857 EST: [1/0/0, FXOLS_RINGING, E_DSP_SIG_0000] Nov 20 10:40:23.857 EST: [1/0/0, FXOLS_RINGING, E_DSP_SIG_0100] Nov 20 10:40:23.857 EST: fxols_ringing_not Nov 20 10:40:23.861 EST: htsp_timer_stop htsp_setup_ind Nov 20 10:40:23.861 EST: [1/0/0] get_fxo_caller_id:Caller ID received. Message type=128 length=31 checksum=74 Nov 20 10:40:23.861 EST: [1/0/0] Caller ID String 80 1C 01 08 31 31 32 30 31 35 34 30 02 07 35 35 35 31 32 31 32 07 07 4F 75 74 73 69 64 65 74 Nov 20 10:40:23.865 EST: [1/0/0] get_fxo_caller_id calling num=5551212 calling name=Outside calling time=11/20 15:40 Nov 20 10:40:23.869 EST: [1/0/0, FXOLS_WAIT_SETUP_ACK, E_HTSP_SETUP_ACK] Nov 20 10:40:23.873 EST: fxols_wait_setup_ack: Nov 20 10:40:23.873 EST: [1/0/0] set signal state = 0xCtimestamp = 0 Nov 20 10:40:23.985 EST: [1/0/0, FXOLS_PROCEEDING, E_DSP_SIG_0100] fxols_proceed_clear Nov 20 10:40:23.985 EST: htsp_timer_stop2 Nov 20 10:40:24.097 EST: [1/0/0, FXOLS_PROCEEDING, E_DSP_SIG_0110] fxols_rvs_battery Nov 20 10:40:24.097 EST: htsp_timer_stop2 Nov 20 10:40:24.733 EST: [1/0/0, FXOLS_PROCEED_RVS_BT, E_HTSP_PROCEEDING] fxols_offhook_proc Nov 20 10:40:24.733 EST: htsp_timer - 120000 msec Nov 20 10:40:24.745 EST: [1/0/0, FXOLS_PROCEED_RVS_BT, E_HTSP_VOICE_CUT_THROUGH] fxols_proc_voiceIn this example, everything was working fine and both Name and Number Display were properly delivered to the phone. In the two scenarios below, the calling number is missing in one case, and the name display is missing in the other.
Calling Number Lost, Name Delivered
In the following example, the calling number is lost, but the name is delivered:
Nov 17 17:39:34.164 EST: [1/1/0] htsp_set_caller_id_txcalling num=display_info=Outside called num=9913050 Nov 17 17:39:34.164 EST: [1/1/0] Caller ID String 80 1601 08 31 31 31 37 32 32 33 39 04 01 4F 07 07 4F 75 7473 69 64 65 88In the Caller ID String, looking at "04 01 4F" translates to:
04 : Reason for Absence of DN
01 : Length of message
4F : "Out of Area"Calling Number Delivered, Name Lost
In the following example, the calling number is delivered, but the name is lost:
Nov 17 17:53:24.034 EST: [1/1/0] htsp_set_caller_id_txcalling num=5550109display_info= called num=5550011Nov 17 17:53:24.034 EST: [1/1/0] Caller ID String 8016 01 08 31 31 31 37 32 32 35 33 02 07 35 35 35 31 3231 32 08 01 4F 05In the Caller ID String, looking at "08 01 4F" translates to:
08 : Reason for Absence of Display
01 : Length
4F : Out of AreaFor more information, refer to Caller ID Name Delivery Issues on Cisco IOS Gateways, document ID 23444.
E&M Interfaces
The difference between a conventional two-wire telephone interface such as FXS or FXO and an E&M interface is that the E&M interface has wires that pass the audio signals plus wires to act as an input (to sense an incoming call) or an output (to indicate an outgoing call). These control leads are normally called the E lead (input) and the M lead (output). Depending on the type of E&M interface, the signaling leads could be controlled by connecting them to the ground, switching a -48-Vdc source, or completing a current loop between the two devices.
E&M interfaces can normally be two- or four-wire operation, which does not refer to the total number of physical connections on the port but rather to the way that audio is passed between the devices. Two-wire operation means the transmitting and receiving audio signals are passed through a single pair of wires (one pair equals two wires). Four-wire operation uses one pair for transmitting and another pair for receiving audio.
In Figure 18, two PBXs are connected across a WAN by E&M interfaces. This topology illustrates the path over a WAN between two geographically separated offices in the same company.
Figure 18 E&M Signaling Interfaces
This section contains the following topics:
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E&M Hardware Troubleshooting
The E&M interface typically connects remote calls from an IP network to a PBX. Troubleshoot Cisco E&M hardware by checking the following sections:
Software Compatibility
For interface cards inserted into Cisco modular access routers, check the compatibility tables in the "Overview of Cisco Interface Cards" chapter in the Cisco Interface Cards Installation Guide.
Cabling
E&M is a signaling technique for two-wire and four-wire telephone and trunk interfaces. The E&M interface typically connects remote calls from an IP network to a PBX. The card is connected to the PSTN or PBX through a telephone wall outlet by a straight-through RJ-48C cable.
Note
The connector port for the E&M voice interface card is shown in Figure 19. Information about LEDs can be found in the "Connecting Voice Interface Cards to a Network" chapter of the Cisco Interface Cards Installation Guide.
Note
Figure 19 Two-Port E&M Card Front Panel
To verify that the analog E&M hardware is being recognized by the Cisco IOS platform, use the following commands:
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show version Command on a Cisco 3640 Platform
Router# show versionCisco Internetwork Operating System SoftwareIOS (tm) 3600 Software (C3640-IS-M), Version 12.1(2), RELEASE SOFTWARE (fc1)Copyright (c) 1986-2000 by cisco Systems, Inc.Compiled Wed 10-May-00 07:20 by lindaImage text-base: 0x600088F0, data-base: 0x60E38000ROM: System Bootstrap, Version 11.1(20)AA2, EARLY DEPLOYMENT RELEASE SOFTWARE(fc1)Router uptime is 0 minutesSystem returned to ROM by power-on at 11:16:21 cst Mon Mar 12 2001System image file is "flash:c3640-is-mz.121-2.bin"cisco 3640 (R4700) processor (revision 0x00) with 126976K/4096K bytes of memory.Processor board ID 16187704R4700 CPU at 100Mhz, Implementation 33, Rev 1.0Bridging software.X.25 software, Version 3.0.0.SuperLAT software (copyright 1990 by Meridian Technology Corp).2 Ethernet/IEEE 802.3 interface(s)2 Voice FXS interface(s)2 Voice E&M interface(s)DRAM configuration is 64 bits wide with parity disabled.125K bytes of non-volatile configuration memory.32768K bytes of processor board System flash (Read/Write)20480K bytes of processor board PCMCIA Slot0 flash (Read/Write)Configuration register is 0x2102show running-config Command on a Cisco 3640 Platform
Router# show running-configBuilding configuration...Current configuration:!!--- Some output omitted.version 12.1service timestamps debug uptimeservice timestamps log uptime!hostname Router!voice-port 3/0/0!voice-port 3/0/1!voice-port 3/1/0!voice-port 3/1/1!endShutdown Port
Check to make sure the port is not shut down. Enter the show voice port command with the voice port number that you are troubleshooting. The output will tell you:
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E&M Interface Types
This section describes the standard analog E&M interface types I, II, III, and V (IV is not supported by Cisco platforms). The following topics are covered:
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E&M Signaling Unit Side and Trunk Circuit Side Compatibility Issues
E&M signaling defines a trunk circuit side and a signaling unit side for each connection. Cisco's analog E&M interface functions as the signaling unit side, so it expects the other side to be a trunk circuit. When you use E&M interface model Type II or Type V, you can connect two signaling unit sides back to back by appropriate crossing of the signaling leads. When using the E&M Type I or Type III interface, you cannot connect two signaling unit sides back to back.
Many PBX brands have E&M analog trunk cards that can operate as either the trunk circuit side or the signaling unit side. Because the Cisco E&M interfaces are fixed as the signaling unit side of the interface, it may be necessary to change the E&M trunk settings on the PBX to operate as the trunk circuit side. If Type I or III E&M is being used, this is the only way the PBX can work with the Cisco E&M interface.
Some PBX products (and many key systems) can operate only as the signaling unit side of the E&M interface. They cannot interoperate with the Cisco E&M interface if Type I or Type III is chosen. If Type II or Type V E&M is being used, PBX products fixed as "signaling unit" side can still be used with the Cisco E&M interface via Type II or Type V.
Each E&M signaling type has a unique circuit model and connection diagram. The following sections describe the different types. Table 25 shows the E&M supervisory signal description.
E&M Type I Interface Model
E&M Type I is the original E&M lead signaling arrangement, and it is the most common interface type in North America. Table 26 shows the sent signal states for on- and off-hook signaling.
Table 26 E&M Type I Signal States
M
Ground
Battery
E
Open
Ground
The gateway grounds its E-lead to signal a trunk seizure. The PBX applies battery to its M-lead to signal a seizure. Cisco gateways expect to see off-hook conditions on the M-lead, and they signal off-hook to a remote device on the E-lead. E&M Type I 2-wire operation is shown in Figure 20. E&M Type I 4-wire operation is shown in Figure 21.
Figure 20 E&M Type I 2-Wire Audio Operation
Figure 21 E&M Type I 4-Wire Audio Operation
Note
Figure 22 E&M Cabling Pins
Considerations for Type I interfaces include:
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E&M Type II Interface Model
E&M Type II provides a 4-wire fully looped arrangement that provides full isolation between the trunks and signaling units. Type II is usually used on Centrex lines and Nortel PBX systems. Table 27 shows the sent signal states for on- and off-hook signaling.
Table 27 E&M Type II Signal States
M
Open
Battery
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Open
Ground
The gateway grounds its E-lead to signal a trunk seizure. The PBX applies battery to its M-lead to signal a seizure. Cisco gateways expect to see off-hook conditions on the M-lead, and they signal off-hook to a remote device on the E-lead. E&M Type II 2-wire operation is shown in Figure 23. E&M Type II 4-wire operation is shown in Figure 24.
Figure 23 E&M Type II 2-Wire Audio Operation
Figure 24 E&M Type II 4-Wire Audio Operation
Note
Considerations for Type II interfaces include:
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E&M Type III Interface Model
E&M Type III is a partially looped four-wire E&M arrangement with ground isolation. The signaling unit provides both the battery and the ground. Table 28 shows the sent signal states for on- and off-hook signaling.
Table 28 E&M Type III Signal States
M
Ground
Battery
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Open
Ground
The router senses loop current on the M-lead for an inbound seizure and grounds its E-lead for an outbound seizure. Cisco routers/gateways expect to see off-hook conditions on the M-lead, and they signal off-hook to a remote device on the E-lead. E&M Type III 2-wire operation is shown in Figure 25. E&M Type III 4-wire operation is shown in Figure 26.
Figure 25 E&M Type III 2-Wire Audio Operation
Figure 26 E&M Type III 4-Wire Audio Operation
Note
Considerations for Type III interfaces include:
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E&M Type V Interface Model
E&M Type V is widely used outside North America (nearly a worldwide standard.) Type V is a symmetrical two-wire lead arrangement that signals in both directions (open for on-hook and ground for off-hook.) Table 29 shows the sent signal states for on- and off-hook signaling.
Table 29 E&M Type V Signal States
M
Open
Ground
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Open
Ground
The gateway grounds its E-lead to signal a trunk seizure. The PBX grounds its M-lead to signal a seizure. Cisco gateways expect to see off-hook conditions on the M-lead, and they signal off-hook to remote device on the E-lead. E&M Type V 2-wire operation is shown in Figure 27. E&M Type V 4-wire operation is shown in Figure 28.
Figure 27 E&M Type V 2-Wire Audio Operation
Figure 28 E&M Type V 4-Wire Audio Operation
Note
Considerations for Type V interfaces include:
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Troubleshooting E&M Interfaces at the Physical Level
E&M provides the highest quality analog interface available, but it also is the most difficult to administer due to the number of leads, configurations, and protocol issues. Usually it is helpful to have the appropriate reference diagram available when verifying the connections.
Preparing to Troubleshoot E&M Physical Problems
Use the information in the following sections to prepare to troubleshoot E&M physical problems:
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Hardware Troubleshooting Tools
Test equipment is not required for every installation, but sometimes you need to use it to isolate problems with analog E&M ports. The most useful equipment is a digital multimeter and a technician's line test set. These tools allow measurement of signaling states and voltages, and monitoring of audio signals. A digital multimeter is used to measure the DC loop voltage and AC ringing voltage on FXS ports, E- or M-lead signaling transitions, voltages on E or M leads, and DC resistance of E&M signaling leads.
In the terminating mode of operation, the technician's line test set acts like a normal telephone handset when connected to a loopstart trunk, allowing telephone numbers to be dialed on the built-in keypad. When switched to the monitoring mode (bridging mode), the unit presents a high impedance to the TX or RX audio pairs of the E&M port, allowing the audio signals and tones to be heard on the built-in loudspeaker. This mode helps you find problems with one-way audio, incorrect digits being sent or received, distortion and level problems, and possible sources of noise and echo.
For an effective troubleshooting kit, have the following items available:
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Precautions
Warning
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PBX Interconnection
The majority of PBXs interface with peripheral equipment using cable distribution frames (DFs). Multipair cables are run from the PBX equipment cabinet to the distribution frame, where they are jumpered (cross-connected) to the external devices. These DFs have various names, but the most common terms for them are 110 block, 66 block, and Krone frame. The DF is generally the place where all connections are made between the router voice port and the PBX, so it is where most wiring errors are made and would obviously be the best place to perform testing and troubleshooting.
Use Rollover Cable for E&M Port-to-Port Testing
Past experience of Cisco Technical Assistance Center (TAC) engineers has shown that most E&M-related faults are due to incorrect wiring or PBX port programming. To assist in determining if the fault is external to the router, you can use the standard rollover console cable that is supplied with every Cisco router as an E&M cross-over cable. This cross-over cable connects the signaling output of one port to the input of the other port and maintains an audio path between the two ports. You can configure a dial peer so that a test call is sent out one port and looped back into the second port, proving the operation of the router.
The rollover console cable has the following RJ-45 connector wiring:
1-------8
2-------7
3-------6
4-------5
5-------4
6-------3
7-------2
8-------1
The signaling cross-over occurs as pins 2 (M-lead) and 7 (E-lead) on one port are connected to pins 7 (E-lead) and 2 (M-lead) on the other port. The two ports share a common internal ground. The cross-over on pins 4 and 5 (audio pair) has no effect on the audio signal. By setting both voice ports to 2-wire, type 5 operation, the E&M ports become symmetrical and an outward seizure on one port is seen as an incoming seizure on the second port. Any DTMF digits sent out immediately come back in and are then matched on another dial peer. If the test calls are successful, there is little doubt about the operation of the router voice ports. In the following example, the assumption is made that there are working devices on the IP network that can originate and accept VoIP calls.
The voice ports and dial peers are configured like this:
voice-port 1/0/0!--- First port under test.operation 2-wiresignal-type winktype 5!voice-port 1/0/1!--- Second port under test.operation 2-wiresignal-type winktype 5Cisco - Analog E&M Troubleshooting Guidelines (Cisco IOS Platforms)!dial-peer voice 100 pots!--- Send call out to port 1/0/0, strip the 100 and prefix with a called!--- number 200.destination-pattern 100port 1/0/0prefix 200!dial-peer voice 200 voip!--- Incoming test call for 200 comes!--- in on port 1/0/1 and is sent to 10.1.1.1 as VoIP call.destination-pattern 200session-target ipv4:10.1.1.1!When a VoIP call comes in to the router with a called number of 100, it is sent out port 1/0/0. By default, any explicitly matched digits on a POTS dial peer are assumed to be an access code and stripped off before the call is made. To route the call correctly, these digits need to be replaced. In this case the prefix command prepends the digits 200 as the called number. This call is immediately looped back in on port 1/0/1. The digits match on dial-peer 200 and make the new call to the designated IP address. The devices originating and accepting the VoIP calls should then have an audio connection that is across the IP network and goes out and back through the E&M ports. This connection proves the router is working properly and indicates that the fault is external to the router. The majority of faults are due to incorrect cabling or PBX port programming issues.
Troubleshooting Type I Interfaces
The four-wire Type I interface from the PBX (set up for the trunk circuit side) has the following characteristics:
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Confirm the Cable Interface from the PBX
If you think the cable is bad, pull the suspect voice cable from the router and leave the other side connected to the PBX. Then do the following:
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Additional Troubleshooting Tips
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Troubleshooting Type II Interfaces
The four-wire Type-II interface from the PBX (setup for trunk circuit side) has the following characteristics:
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Confirm the Cable Interface from the PBX
Pull the suspect voice cable from the router and leave the other side connected to the PBX. Then do the following:
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Additional Troubleshooting Tips
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Troubleshooting Type III Interfaces
The four-wire Type-III interface from the PBX has the following characteristics:
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Confirm the Cable Interface from the PBX
Pull the suspect voice cable from the router and leave the other side connected to the PBX. Then do the following:
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Additional Troubleshooting Tips
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Troubleshooting Type V Interfaces
The four-wire Type-V interface from the PBX has the following characteristics:
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Confirm the Cable Interface from the PBX
Pull the suspect voice cable from the router and leave the other side connected to the PBX. Then do the following:
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Additional Troubleshooting Tips
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Confirming E&M Configuration
The following items should be checked to confirm the E&M configuration:
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Confirming the PBX E&M Configuration Parameters
The Cisco gateway needs to match the PBX configuration. One of the challenges of configuring and troubleshooting analog E&M circuits are the amount of configuration variables.
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For information about how specific PBX types interoperate with your gateway, go to the PBX interoperability portal, which is located here:
http://www.cisco.com/warp/public/779/largeent/avvid/inter_operability/
Note
Confirming the Cisco IOS Gateway Configuration
The Cisco gateway configuration should match the connected PBX configuration. Use the following commands to verify the Cisco IOS platform configuration:
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Sample Output of show voice port Command
Router# show voice port 1/0/0recEive And transMit 1/0/0 Slot is 1, Sub-unit is 0, Port is 0Type of VoicePort is E&MOperation State is DORMANTAdministrative State is UPThe Last Interface Down Failure Cause is Administrative ShutdownDescription is not setNoise Regeneration is enabledNon Linear Processing is enabledMusic On Hold Threshold is Set to -38 dBmIn Gain is Set to 0 dBOut Attenuation is Set to 0 dBEcho Cancellation is enabledEcho Cancel Coverage is set to 8 msConnection Mode is normalConnection Number is not setInitial Time Out is set to 10 sInterdigit Time Out is set to 10 sCall-Disconnect Time Out is set to 60 sRegion Tone is set for USAnalog Info Follows:Currently processing noneMaintenance Mode Set to None (not in mtc mode)Number of signaling protocol errors are 0Impedance is set to 600r OhmVoice card specific Info Follows:Signal Type is immediateOperation Type is 2-wireE&M Type is 5Dial Type is dtmfIn Seizure is inactiveOut Seizure is inactiveDigit Duration Timing is set to 100 msInterDigit Duration Timing is set to 100 msPulse Rate Timing is set to 10 pulses/secondInterDigit Pulse Duration Timing is set to 500 msClear Wait Duration Timing is set to 400 msWink Wait Duration Timing is set to 200 msWink Duration Timing is set to 200 msDelay Start Timing is set to 300 msDelay Duration Timing is set to 2000 msDial Pulse Min. Delay is set to 140 msVerifying the Wiring Arrangement Between the PBX and the Cisco Gateway
Physical wiring is often the primary source for analog E&M problems. It is imperative that you verify that the cable/wiring you are using is appropriate for the E&M setup in place. A few things to consider:
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See the "Troubleshooting E&M Interfaces at the Physical Level" section for more information on the wiring arrangement.
Verifying Supervision Signaling
In this step, verify that on-hook/off-hook signals are being transmitted between the PBX and the gateway. If you are accessing the router through the console port, enter the command terminal monitor, otherwise no debug output is displayed.
Follow these steps to verify supervision signaling:
SUMMARY STEPS
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DETAILED STEPS
Step 1
Step 2
Step 3
In this example, the PBX is seizing the router trunk. The router E&M voice port transitions from on-hook to off-hook. This shows that on-hook, off-hook signaling is being received from the PBX.
Router# debug vpm signalVoice Port Module signaling debugging is enabled*Mar 2 05:54:43.996: htsp_process_event: [1/0/0, 1.4 , 34]em_onhook_offhookhtsp_setup_ind*Mar 2 05:54:44.000: htsp_process_event: [1/0/0, 1.7 , 8]*Mar 2 05:54:44.784: htsp_process_event: [1/0/0, 1.7 , 10]*Mar 2 05:54:44.784: htsp_process_event: [1/1/0, 1.2 , 5]fxsls_onhook_setuphtsp_alerthtsp_alert_notify*Mar 2 05:54:44.788: htsp_process_event: [1/0/0, 1.7 , 11]*Mar 2 05:54:44.788: htsp_process_event: [1/1/0, 1.5 , 11]fxsls_waitoff_voiceIf no output is displayed, there is probably a problem with the E&M supervision signaling.
Table 30 describes some possible problems and the corresponding solutions.
Table 30 E&M Supervisory Signaling Troubleshooting Table
No dial tone from the Cisco port.
No port seizure activity seen on the Cisco gateway.
The PBX is not configured to seize the E&M port connected to the Cisco equipment.
Configure the PBX to seize the trunk.
The port is seized but the call does not go through.
There is an E&M Type (I, II, III or V) mismatch between the PBX and the gateway.
Verify (and change if necessary) the E&M type configured on the Cisco equipment. See the "Confirming the Cisco IOS Gateway Configuration" section.
The port has unbreakable dial tone.
The Cisco gateway is unable to send digits when a port is seized.
Incorrect wiring arrangement (cabling) for the supervisory signaling leads (E and M leads for Type I and V; E, M, SB, and SG leads for Types II and III).
Wiring issues are usually the primary source of analog E&M problems. Make sure the cable used corresponds to the required PBX and Cisco gateway pinout, interface type, and audio operation setup. For more information see the "Troubleshooting E&M Interfaces at the Physical Level" section.
The port on the Cisco gateway cannot be seized.
The Cisco gateway is unable to send digits.
The Cisco gateway configuration changes are not enabled.
Issue the shutdown/no shutdown command sequence on the E&M voice port after the configuration changes.
Calls cannot be made in two directions.
On-hook or off-hook signals have been sent one way only.
This is probably an indication of a defective cable, where one path of the signaling leads is wired correctly and the other side is not.
Verifying That the Cisco Equipment and PBX Are Sending and Receiving Digits
After confirming successful supervisory (on-hook/off-hook) signaling between the PBX and the gateway, you need to verify that address information (DTMF digits or pulse dial) is being passed between both ends.
Note
There are three start dial supervision line protocols that analog E&M uses to define how the equipment passes address information:
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Make sure both the Cisco gateway and the PBX are configured with the same start dial supervision protocol. Verify that information is being passed by performing the following steps:
SUMMARY STEPS
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DETAILED STEPS
Step 1
Step 2
Step 3
Router# show debuggingVoice Port Module signaling debugging is onVoice Telephony dsp debugging is onRouter#*Mar 1 03:16:19.207: htsp_process_event: [1/0/0, 1.4 , 34] em_onhook_offhookhtsp_setup_*Mar 1 03:16:19.207: htsp_process_event: [1/0/0, 1.7 , 8]*Mar 1 03:16:19.339: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT_BEGIN: digit=2,rtp_=0x9961CF03*Mar 1 03:16:19.399: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT_OFF: digit=2,duration=*Mar 1 03:16:19.539: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT_BEGIN: digit=0,rtp_=0x9961CF032.*Mar 1 03:16:19.599: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT_OFF: digit=0,duration=*Mar 1 03:16:19.739: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT_BEGIN: digit=0,rtp_=0x9961CF03*Mar 1 03:16:19.799: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT_OFF: digit=0,duration=*Mar 1 03:16:19.939: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT_BEGIN: digit=0,=rtp_=0x9961CF03*Mar 1 03:16:19.999: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT_OFF: digit=0,duration=*Mar 1 03:16:19.999: htsp_process_event: [1/0/0, 1.7 , 10]*Mar 1 03:16:19.999: htsp_process_event: [1/1/0, 1.2 , 5] fxsls_onhook_setuphtsp_alerthtsp_*Mar 1 03:16:20.003: htsp_process_event: [1/0/0, 1.7 , 11]*Mar 1 03:16:20.003: htsp_process_event: [1/1/0, 1.5 , 11] fxsls_waitoff_voice*Mar 1 03:16:27.527: htsp_process_event: [1/1/0, 1.5 , 34] fxsls_waitoff_offhook*Mar 1 03:16:27.531: htsp_process_event: [1/0/0, 1.7 , 6] em_offhook_connectem_stop_Step 4
Log Buffer (1000000 bytes):*Mar 1 03:45:31.287: htsp_process_event: [1/1/1, 1.2 , 34] fxsls_onhook_offhookhtsp_setup_ind*Mar 1 03:45:31.291: htsp_process_event: [1/1/1, 1.3 , 8]*Mar 1 03:45:33.123: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT_BEGIN: digit=1, rtp_timestamp=0xCD4365D8*Mar 1 03:45:33.283: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT_OFF: digit=1,duration=205*Mar 1 03:45:33.463: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT_BEGIN: digit=0, rtp_timestamp=0xCD4365D8*Mar 1 03:45:33.643: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT_OFF: digit=0,duration=225*Mar 1 03:45:33.823: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT_BEGIN: digit=0, rtp_timestamp=0xCD4365F0*Mar 1 03:45:34.003: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT_OFF: digit=0,duration=222*Mar 1 03:45:34.203: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT_BEGIN: digit=0, rtp_timestamp=0xCD4365F0*Mar 1 03:45:34.411: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT_OFF: digit=0,duration=252*Mar 1 03:45:34.415: htsp_process_event: [1/1/1, 1.3 , 10]*Mar 1 03:45:34.415: htsp_process_event: [1/0/0, 1.4 , 5] em_onhook_setup em_offhook*Mar 1 03:45:34.415: htsp_process_event: [1/0/0, 1.13 , 43] em_start_timer: 1200 ms*Mar 1 03:45:34.715: htsp_process_event: [1/0/0, 1.10 , 34] em_wink_offhookem_stop_timers em_start_timer: 1200 ms*Mar 1 03:45:34.923: htsp_process_event: [1/0/0, 1.11 , 22] em_wink_onhook em_stop_timers em_send_digit htsp_dial*Mar 1 03:45:34.923: digit=1, components=2, freq_of_first=697, freq_of_second=1209, amp_of_first=16384, amp_of_second=16384*Mar 1 03:45:34.923: digit=0, components=2, freq_of_first=941, freq_of_second=1336, amp_of_first=16384, amp_of_second=16384*Mar 1 03:45:34.923: digit=0, components=2, freq_of_first=941, freq_of_second=1336, amp_of_first=16384, amp_of_second=16384*Mar 1 03:45:34.923: digit=0, components=2, freq_of_first=941, freq_of_second3.=1336, amp_of_first=16384, amp_of_second=16384*Mar 1 03:45:35.727: vtsp_process_dsp_message: MSG_TX_DIALING_DONE*Mar 1 03:45:35.727: htsp_process_event: [1/0/0, 1.7 , 19] em_offhook_digit_donehtsp_alerthtsp_alert_notifyTable 31 shows digit send and receive problems and the corresponding solutions. These problems can be diagnosed if you notice that the wink timers are being triggered.
Table 31 Digit Send and Receive Troubleshooting Table
Start dial supervision mismatch or timing issues between the PBX and gateway.
Make sure both end systems are configured with the same start dial protocol.
Audio operation mismatch (for example, one side configured for 2-wire, the other for 4-wire) or wiring problems on the audio path.
Verify the gateway configuration and PBX configuration and the wiring arrangement. For more information see the "Troubleshooting E&M Interfaces at the Physical Level" section.
Note
Wiring problems in the audio path.
Verify the wiring arrangement. See the "Troubleshooting E&M Interfaces at the Physical Level" section.
In the 4-wire audio mode, some PBX and key system products reverse the normal usage of the tip and ring and tip-1 and ring-1 pairs. In that case, to match up the audio pairs with the Cisco E&M audio pairs, you might need to connect tip and ring on the PBX side to tip-1 and ring-1 on the Cisco side, and tip-1 and ring-1 on the PBX side to tip and ring on the Cisco side. If the audio pairs are not correctly matched up in 4-wire mode, there is no end-to-end audio path in either direction. If the E&M interface is configured to send dial strings as dial pulse (which works by pulsing on the E or M lead), it is possible to establish a call even with the 4-wire audio pairs reversed, but there will be little or no audio path in either direction after the call is established (there might be low-level transmission of audio, but the sound levels will be far too low for comfort). If you are using DTMF to send dial strings, the E&M interface goes off hook at the start of the call, but the call does not complete, because one end sends the DTMF tones on the wrong audio pair, and the other end does not receive these DTMF tones.
Verifyinng That the Gateway Sends the Expected Digits to the PBX
Once the two end devices are able to successfully send supervision and address signaling (on-hook, off-hook, digits), we can assume that the troubleshooting process is complete for analog E&M signaling, and it is now in the dial plan domain. For more information about dial plan design, refer to the Voice Design and Implementation Guide, document ID 5756.
If incomplete or incorrect digits are sent by the Cisco equipment, then the Telco switch (CO or PBX), cannot ring the correct station.
On POTS dial peers, the only digits that are sent to the other end are the ones specified with the command destination-pattern and the wild card character ("."). The POTS dial peer command prefix can be used to include a dial-out prefix that the system enters automatically instead of people dialing it. Refer to the following output example for a sample configuration.
!!--- Some output ommited.!!--- E&M Voice Port!voice-port 1/0/0type 2signal immediate!!--- FXS Voice Portvoice-port 1/1/0!dial-peer voice 1 potsdestination-pattern 2000port 1/1/0!!--- Dial peer 2 is in charge of forwarding calls to the E&M voiceport 1/0/0.!--- In this case the digit "1" in the destination pattern will be dropped and the syste!--- will transmit the 3 digits matched by the "." wildcard.!--- Notice that since the PBX is expecting the "1000" string, the prefix command is used.!dial-peer voice 2 potsdestination-pattern 1...port 1/0/0prefix 1!Verify That the Gateway Receives the Expected Digits from the PBX
Verify that the digits received from the PBX match a dial peer in the gateway. If incomplete or incorrect digits are sent by the PBX, a dial peer cannot be matched. Use the command debug vtsp dsp to view the digits received by the analog E&M voice port.
To verify which dial peers match a specific string use the command show dialplan number. Refer to the following sample output example.
Router# show dialplan number 1000Macro Exp.: 1000VoiceEncapPeer2information type = voice,tag = 2, destination-pattern = `1...',answer-address = `', preference=0,group = 2, Admin state is up, Operation state is up,incoming called-number = `', connections/maximum = 0/unlimited,application associated:type = pots, prefix = `1',session-target = `', voice-port = `1/0/0',direct-inward-dial = disabled,register E.164 number with GK = TRUEConnect Time = 19644, Charged Units = 0,Successful Calls = 63, Failed Calls = 2,Accepted Calls = 65, Refused Calls = 0,Last Disconnect Cause is "10 ",Last Disconnect Text is "normal call clearing.",Last Setup Time = 28424467.Matched: 1000 Digits: 1Target:Router# show dialplan number 2000Macro Exp.: 2000VoiceEncapPeer1information type = voice,tag = 1, destination-pattern = `2000',answer-address = `', preference=0,group = 1, Admin state is up, Operation state is up,incoming called-number = `', connections/maximum = 0/unlimited,application associated:type = pots, prefix = `',session-target = `', voice-port = `1/1/1',direct-inward-dial = disabled,register E.164 number with GK = TRUEConnect Time = 19357, Charged Units = 0,Successful Calls = 68, Failed Calls = 8,Accepted Calls = 76, Refused Calls = 0,Last Disconnect Cause is "10 ",Last Disconnect Text is "normal call clearing.",Last Setup Time = 28424186.Matched: 2000 Digits: 4Target:Unbreakable Dial Tone
A common problem occurs when the router seizes the local PBX but as digits are dialed, the dial tone stays. The calling party is unable to pass the DTMF tones or digits to the terminating device, resulting in callers being unable to dial the desired extension or interact with the device that needs DTMF tones, such as a voice mail or interactive voice response (IVR) application. This problem can result from a number of reasons such as:
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For more information, refer to Inability To Break Dialtone in a Voice over IP Network , document ID 22376.
Analog DID Interfaces
Direct inward dialing (DID) is a service offered by telephone companies that enables callers to dial directly to an extension on a PBX without the assistance of an operator or automated call attendant. This service makes use of DID trunks, which forward only the last three to five digits of a phone number to the PBX. The DID state machine is identical to the E&M state machine.
Figure 29 shows a hypothetical topology in which a user connected to the PSTN (User A) dials various numbers and is connected to the appropriate extensions on a PBX.
Figure 29 DID Support for Cisco 2600 and Cisco 3600 Series Routers
555-1234
234
User C
555-1345
345
User D
555-1456
456
User B
555-1678
678
No dial-peer match found; fast busy tone is played
DID Hardware Troubleshooting
A DID voice interface connects directly to a standard telephone, fax machine, or similar device and supplies ring, voltage, and dial tone.
Troubleshoot DID hardware by checking the following sections:
Software Compatibility
For interface cards inserted into Cisco 1600 series, Cisco 1700 series, Cisco 2600 series, Cisco 3600 series, Cisco 3700 series, and Cisco ICS 7750 platforms, refer to the compatibility tables in the "Overview of Cisco Interface Cards" chapter in the Cisco Interface Cards Installation Guide.
Cabling
The two-port and four-port DID interface cards support the RJ-11 connector. Illustrations of the connector ports are shown in Figure 30 and Figure 31. Information about LEDs can be found in the "Connecting Voice Interface Cards to a Network" chapter of the Cisco Interface Card Hardware Installation Guide.
Figure 30 Two-Port Analog DID Voice Interface Card
Figure 31 Four-Port Analog FXS/DID Voice Interface Card
For more information about the VIC-2DID interface card, refer to Understanding 2 Port Direct Inward Dial (2 DID) Voice Interface Cards, document ID 15268.
Shutdown Port
Check to make sure that the port is not shut down. Enter the show voice port command with the voice port number that you are troubleshooting, which will tell you:
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Verifying Direct Inward Dialing Voice-Port Configuration
To verify voice-port configuration, enter the show voice port command. You can specify a voice port or view the status of all configured voice ports. In the following example, the specified port is configured for DID.
Router# show voice port 1/1/0Foreign Exchange Station with Direct Inward Dialing (FXS-DID) 1/1/0 Slot is1, Sub-unit is 1, Port is 0Type of VoicePort is DID-INOperation State is DORMANTAdministrative State is UPNo Interface Down FailureDescription is not setNoise Regeneration is enabledNon Linear Processing is enabledMusic On Hold Threshold is Set to -38 dBmIn Gain is Set to 0 dBOut Attenuation is Set to 0 dBEcho Cancellation is enabledEcho Cancel Coverage is set to 8 msPlayout-delay Mode is set to defaultPlayout-delay Nominal is set to 60 msPlayout-delay Maximum is set to 200 msPlayout-delay Minimum mode is set to default, value 4 msPlayout-delay Fax is set to 300 msConnection Mode is normalConnection Number is not setInitial Time Out is set to 10 sInterdigit Time Out is set to 10 sCall Disconnect Time Out is set to 3 sRinging Time Out is set to 180 sWait Release Time Out is set to 3 sCompanding Type is u-lawRegion Tone is set for USAnalog Info Follows:Currently processing noneMaintenance Mode Set to None (not in mtc mode)Number of signaling protocol errors are 0Impedance is set to 600r OhmStation name Chalil Mohanan, Station number 1234567Voice card specific Info Follows:Signal Type is wink-startDial Type is dtmfIn Seizure is inactiveOut Seizure is inactiveDigit Duration Timing is set to 100 msInterDigit Duration Timing is set to 100 msPulse Rate Timing is set to 10 pulses/secondInterDigit Pulse Duration Timing is set to 750 msClear Wait Duration Timing is set to 400 msWink Wait Duration Timing is set to 200 msWait Wink Duration Timing is set to 550 msWink Duration Timing is set to 200 msDelay Start Timing is set to 300 msDelay Duration Timing is set to 2000 msDial Pulse Min. Delay is set to 140 msPercent Break of Pulse is 60 percentAuto Cut-through is disabledDialout Delay for immediate start is 300 msVoice Port Testing Commands
Voice port testing commands allow you to force voice ports into specific states for testing. The following types of voice-port tests are covered:
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Detector-Related Function Tests
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 detector to its original state.
To configure this feature, enter these commands beginning in privileged EXEC mode:
Loopback Function Tests
To establish loopbacks on a voice port, enter the following commands beginning in privileged EXEC mode:
Tone Injection Tests
To inject a test tone into a voice port, enter the following commands beginning in privileged EXEC mode:
Relay-Related Function Tests
To test relay-related functions on a voice port, enter the following commands beginning in privileged EXEC mode:
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.
To force a voice port into fax mode and return it to voice mode, enter the following commands, beginning in privileged EXEC mode:
Any Internet Protocol (IP) addresses used in this document are not intended to be actual addresses. Any examples, command display output, and figures included in the document are shown for illustrative purposes only. Any use of actual IP addresses in illustrative content is unintentional and coincidental.
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