Wednesday, July 15, 2015

Cisco Voice Troubleshooting Tools

 TranslatorX:
 TranslatorX allows you to quickly parse through Cisco CallManager trace files and search for Q.931, H.225, SCCP (Skinny), MGCP, or SIP messages.

Triple Combo:
This tool helps engineers understand Callmanager trace files and helps in understanding the call legs for troubleshooting.
Just drag and drop the SDL SDI trace files on Triple Combo program.

Cisco Voice Log Translator:
Cisco VLT is a troubleshooting tool that reads complex System Diagnostic Interface (SDI) trace-log message files from a Cisco Unified Communications Manager and translates them into a user-friendly, English-based format. You can sort, organize, analyze, and interpret messages and display raw or translated message text using offline message files on your system.

CME Quick Configuration Tool (QCT):
If you don’t like CLI to configure Callmanager Express, here’s a GUI tool which basically dumps a config file on CME based on your input.
http://www.cisco.com/asiapac/campaigns/businessdialogue/files/IPC_Exp_QCT_DS.pdf
Downloads of the QCT tool match the version of CME you are running. You can download the tool here.
http://www.cisco.com/cgi-bin/tablebuild.pl/cme-qct

Cisco Callmanager/Presence/CVP Real Time Monitoring Tool:
Built in basic monitoring & trace collection tool.

Cisco Unified Analysis Manager:
This tool is something new Cisco added as part of RTMT.
Basically it’s an advance version of Dialed Number Analyzer.

Xlog:
This is another tool to convert Callmanager trace files raw data to a readable call flow type.

Voice Codec Bandwidth Calculator:
Use the Voice Codec Bandwidth Calculator to determine the bandwidth used by different codecs with various voice protocols over different media.

Convert UNIX time:
As you know Callmanager trace files are in unix time format.
This is a handy tool when troubleshooting Callmanager.

Remote control an IP Phone:
Take Remote Control of Cisco IP Phones from anywhere in the world with network connectivity.
Works good for offshore support.
Simply associate the phone with a user, type in phone ip address, user name password to get control.
You can make calls; see what’s displayed on the LCD.

Soft Phone:
Don’t want to buy Cisco license for IP Communicator.
Try this less expensive Soft phone, works better than Cisco’s IP Communicator and good part is you can run multiple instance on the same machine, this feature is kind of a blessing of guys who are preparing for CCIE voice.

Documentation for Cisco CallManager and Cisco Unity systems:
CMReports is quick and easy way to create Cisco Callmanager and Unity documentation.
Simply export the database, upload it to cmreports.com and within minutes you will get your documentation in your Inbox.

Tuesday, January 3, 2012

All about MGCP show Commands

MGCP show Commands
The show commands are useful for displaying the current status of the configuration as well as verifying that the changes that you made took effect. The following commands are described:
Details about these commands can be found in the Cisco IOS Voice Command Reference.

show ccm-manager

If your MGCP network includes Cisco CallManager, use this command to verify the active and redundant configured Cisco CallManager servers. This command also indicates if the gateway is currently registered with Cisco CallManager.
Note Note: The following show ccm-manager command output was captured in a separated environment.
Router# show ccm-manager
MGCP Domain Name: Router
Total number of host: 2
Priority Status Host
============================================================
Primary Registered 10.89.129.210
First backup Backup ready 10.89.129.211
Second backup Undefined 
Current active Call Manager: 10.89.129.210
Current backup Call Manager: 10.89.129.211
Redundant link port: 2428
Failover Interval: 30 seconds
Keepalive Interval: 15 seconds
Last keepalive sent: 1d00h (elapsed time: 00:00:03)
Last MGCP traffic time: 1d00h (elapsed time: 00:00:03)
Last switchover time: 04:49:39 from (10.89.129.211)
Switchback mode: Graceful

show mgcp

Use this command to verify the status of the router's MGCP parameters. You should see the IP address of the server that you are using (172.16.1.252 in this case.) All of the other parameters were left at their default behavior in this configuration.
VG200A# show mgcp
MGCP Admin State ACTIVE, Oper State ACTIVE - Cause Code NONE
MGCP call-agent: 172.16.1.252  Initial protocol service is MGCP
MGCP block-newcalls DISABLED
MGCP dtmf-relay codec all mode out-of-band
MGCP modem passthrough:  CA
MGCP request timeout 500, MGCP request retries 3
MGCP gateway port: 2427, MGCP maximum waiting delay 3000
MGCP restart delay 0, MGCP vad DISABLED
MGCP simple-sdp ENABLED
MGCP codec type g711ulaw, MGCP packetization period 20
MGCP JB threshold lwm 30, MGCP JB threshold hwm 150
MGCP LAT threshold lmw 150, MGCP LAT threshold hwm 300
MGCP PL threshold lwm 1000, MGCP PL threshold hwm 10000
MGCP playout mode is adaptive 60, 4, 200 in msec
MGCP IP ToS low delay disabled, MGCP IP ToS high throughput disabled
MGCP IP ToS high reliability disabled, MGCP IP ToS low cost disabled
MGCP IP precedence 5, MGCP default package: line-package
MGCP supported packages: gm-package dtmf-package trunk-package line-package
                         hs-package
VG200A#
Table: Explanation of Fields in the show mgcp Command
Field OutputDescription
MGCP Admin State ...The administrative and operational state of the MGCP daemon. The administrative state controls starting and stopping the application using the mgcp and mgcp block-newcalls commands. The operational state controls normal MGCP operations.
MGCP call-agentThe address of the call agent specified in the mgcp command.
MGCP block-newcalls enabledThe state of the mgcp block-newcalls command.
MGCP request timeoutThe setting for the mgcp request timeout command.
MGCP request retriesThe setting for the mgcp request retries command.
MGCP gateway portThe UDP port specification.
MGCP maximum waiting delayThe setting for the mgcp max-waiting-delay command.
MGCP restart delayThe setting for the mgcp restart-delay command.
MGCP VADThe setting for the mgcp vad command.
MGCP codec typeThe setting for the mgcp codec command.
MGCP packetization periodThe packetization period parameter setting for the mgcp codec command.
MGCP JB threshold low water markThe jitter buffer minimum threshold parameter setting for the mgcp quality-threshold command.
JB threshold high water markThe jitter buffer maximum threshold parameter setting for the mgcp quality-threshold command.
MGCP LAT threshold low water markThe latency minimum threshold parameter setting for the mgcp quality-threshold command.
LAT threshold high water markThe latency maximum threshold parameter setting for the mgcp quality-threshold command.
MGCP PL threshold low water markThe packet loss minimum threshold parameter setting for the mgcp quality-threshold command.
PL threshold high water markThe packet loss minimum threshold parameter setting for the mgcp quality-threshold command.
MGCP IP ToS low delayThe low-delay parameter setting for the mgcp ip-tos command.
MGCP IP ToS high throughputThe high-throughput parameter setting for the mgcp ip-tos command.
MGCP IP ToS high reliability The high-reliability parameter setting for the mgcp ip-tos command.
MGCP IP ToS low costThe low-cost parameter setting for the mgcp ip-tos command.
MGCP IP precedenceThe precedence parameter setting for the mgcp ip-tos command.
MGCP default package typeThe default-package parameter setting for the mgcp default-package command.
Supported MGCP packagesThe packages supported in this session.

show mgcp endpoint

Use this command to show the voice ports (endpoints) that are under MGCP control in the router. This command verifies which voice ports have been bound to the MGCP application. This is related to the application mgcp command and the port commands that were entered when configuring the POTS dial peer.
VG200A#show mgcp endpoint
voice-port 1/0/0
voice-port 1/0/1
voice-port 1/1/0
voice-port 1/1/1
VG200A#

show mgcp connection

Use this command to display any active MGCP connections. The endpoint in this example is Slot1/Module 1/Port 0. This corresponds to the MGCP Member Configuration identifier in Cisco CallManager. This tells you which port on the router is the endpoint in the call.
In the screen output below there is one active call.
VG200A#show mgcp connection
Endpoint         Call_ID(C)               Conn_ID(I) (P)ort     (M)ode (S)tate (C)odec EC   
(R)esult[EA]
1. aaln/S1/SU1/0 C=A000000001000008,23,24 I=0xD       P=16390,0  M=4    S=4,4   CO=1   
EC=1  R=0,0
Total number of active calls 1
VG200A#
Table: Explanation of Fields in the show mgcp connection Command
Field OutputDescription
EndpointThe endpoint for each call shown in the digital endpoint naming convention of slot number (S0) and digital line (DS1-0) number (1).
Call_ID(C)The MGCP call ID send by the call agent, the internal Call Control Application Programming Interface (CCAPI) call ID for this endpoint, and the peer call legs CCAPI call ID.

(CCAPI is an API to provide call control facilities to applications.)
Conn_ID(I)The connection ID generated by the gateway and sent in the ACK message.
(P)ortThe ports used for this connection. The first port is the local UDP port. The second port is the remote UDP port.
(M)odeThe call mode, where:

0-Indicates an invalid value for mode.

1-Indicates the gateway should only send packets.

2-Indicates the gateway should only receive packets.

3-Indicates the gateway can send and receive packets.

4-Indicates the gateway should neither send nor receive packets.

5-Indicates the gateway should place the circuit in loopback mode.

6-Indicates the gateway should place the circuit in test mode.

7-Indicates the gateway should use the circuit for network access for data.

8-Indicates the gateway should place the connection in network loopback mode.

9-Indicates the gateway should place the connection in network continuity test mode.

10-Indicates the gateway should place the connection in conference mode.

All other values are used for internal debugging.
(S)tateThe call state. The values are used for internal debugging purposes.
(C)odecThe codec identifier. The values are used for internal debugging purposes.
(E)vent [SIFL]Used for internal debugging.
(R)esult [EA]Used for internal debugging.

show voice port mod_num/slot_num/port_num

Use this command to verify the current status and configuration of the voice ports on the router.
The following is sample output from the show voice port command for a foreign exchange office (FXO) voice port:
VG200A#show voice port 1/0/0
Foreign Exchange Office 1/0/0 Slot is 1, Sub-unit is 0, Port is 0
 Type of VoicePort is FXO
 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
 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
 Ringing Time Out is set to 180 s
 Companding Type is u-law
 Region Tone is set for US
 Analog Info Follows:
 Currently processing none
 Maintenance Mode Set to None (not in mtc mode)
 Number of signaling protocol errors are 0
 Impedance is set to 600r Ohm
 Wait Release Time Out is 30 s
 Station name None, Station number None
 Voice card specific Info Follows:
 Signal Type is loopStart
 Number Of Rings is set to 1
 Supervisory Disconnect active
 Hook Status is On Hook
 Ring Detect Status is inactive
 Ring Ground Status is inactive
 Tip Ground Status is inactive
 Dial Type is dtmf
 Digit Duration Timing is set to 100 ms
 InterDigit Duration Timing is set to 100 ms
 Pulse Rate Timing is set to 10 pulses/second
 InterDigit Pulse Duration Timing is set to 750 ms
 Percent Break of Pulse is 60 percent
 GuardOut timer is 2000 ms
VG200A#
The following is sample output from the show voice port command for a foreign exchange station (FXS) voice port:
VG200A#show voice port 1/1/0
Foreign Exchange Station 1/1/0 Slot is 1, Sub-unit is 1, Port is 0
 Type of VoicePort is FXS
 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
 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
 Ringing Time Out is set to 180 s
 Companding Type is u-law
 Region Tone is set for US
 Analog Info Follows:
 Currently processing none
 Maintenance Mode Set to None (not in mtc mode)
 Number of signaling protocol errors are 0
 Impedance is set to 600r Ohm
 Wait Release Time Out is 30 s
 Station name None, Station number None
 Voice card specific Info Follows:
 Signal Type is loopStart
 Ring Frequency is 25 Hz
 Hook Status is On Hook
 Ring Active Status is inactive
 Ring Ground Status is inactive
 Tip Ground Status is inactive
 Digit Duration Timing is set to 100 ms
 InterDigit Duration Timing is set to 100 ms
 Ring Cadence is defined by CPTone Selection
 Ring Cadence are [20 40] * 100 msec
VG200A#
Table: Explanation of Fields in the show voice port Command
Field OutputDescription
Administrative StateAdministrative state of the voice port.
AliasUser-supplied alias for this voice port.
Clear Wait Duration TimingTime of inactive seizure signal to declare call cleared.
Connection ModeConnection mode of the interface
Connection NumberFull E.164 telephone number used to establish a connection with the trunk or PLAR mode.
Currently ProcessingType of call currently being processed: none, voice, or fax.
Delay Duration TimingMaximum delay signal duration for delay dial signaling.
Delay Start TimingTiming of generation of delayed start signal from detection of incoming seizure.
Dial TypeOut-dialing type of the voice port.
Digit Duration TimingDTMF Digit duration in milliseconds.
E&M TypeType of E&M interface.
Echo Cancel CoverageEcho Cancel Coverage for this port.
Echo CancellationWhether or not echo cancellation is enabled for this port.
Hook Flash Duration TimingMaximum length of hook flash signal.
Hook StatusHook status of the FXO/FXS interface.
ImpedanceConfigured terminating impedance for the E&M interface.
In GainAmount of gain inserted at the receiver side of the interface.
In SeizureIncoming seizure state of the E&M interface.
Initial Time OutAmount of time the system waits for an initial input digit from the caller.
InterDigit Duration TimingDTMF interdigit duration in milliseconds.
InterDigit Pulse Duration TimingPulse dialing interdigit timing in milliseconds.
Interdigit Time OutAmount of time the system waits for a subsequent input digit from the caller.
Maintenance Mode Maintenance mode of the voice-port.
Music On Hold ThresholdConfigured Music-On-Hold Threshold value for this interface.
Noise RegenerationWhether or not background noise should be played to fill silent gaps if VAD is activated.
Number of signaling protocol errorsNumber of signaling protocol errors.
Non-Linear ProcessingWhether or not Non-Linear Processing is enabled for this port.
Operations StateOperation state of the port.
Operation TypeOperation of the E&M signal: 2-wire or 4-wire.
Out AttenuationAmount of attenuation inserted at the transmit side of the interface.
Out SeizureOutgoing seizure state of the E&M interface.
PortPort number for this interface associated with the voice interface card.
Pulse Rate TimingPulse dialing rate in pulses per second (pps).
Regional ToneConfigured regional tone for this interface.
Ring Active StatusRing active indication.
Ring FrequencyConfigured ring frequency for this interface.
Ring Ground StatusRing ground indication
Signal TypeType of signaling for a voice port: loop-start, ground-start, wink-start, immediate, and delay-dial.
SlotSlot used in the voice interface card for this port.
Sub-unitSub-unit used in the voice interface card for this port.
Tip Ground StatusTip ground indication.
Type of VoicePortType of voice port: FXO, FXS, and E&M.
The Interface Down Failure CauseText string describing why the interface is down.
Wink Duration TimingMaximum wink duration for wink start signaling.
Wink Wait Duration TimingMaximum wink wait duration for wink start signaling.

show mgcp statistics

Use this command to show statistical information related to MGCP activity on the router.
The following is sample output from a 28xx router with IOS 12.4(22)T
2851#sh mgcp statistics 
 UDP pkts rx 28349, tx 28576
 Unrecognized rx pkts 0, MGCP message parsing errors 0
 Duplicate MGCP ack tx 1, Invalid versions count 0
 CreateConn rx 353, successful 353, failed 0
 DeleteConn rx 353, successful 353, failed 0
 ModifyConn rx 629, successful 629, failed 0
 DeleteConn tx 0, successful 0, failed 0
 NotifyRequest rx 217, successful 217, failed 0
 AuditConnection rx 0, successful 0, failed 0
 AuditEndpoint rx 263, successful 233, failed 30
 RestartInProgress tx 27, successful 27, failed 0
 Notify tx 26530, successful 26530, failed 0
 ACK tx 1785, NACK tx 30
 ACK rx 26533, NACK rx 0
 Collisions: Passive 0, Active 0

 IP address based Call Agents statistics:
 IP address 10.10.14.36, Total msg rx 28349,
                  successful 28318, failed 30
 System resource check is DISABLED. No available statistic

 DS0 Resource Statistics
 -----------------------
 Utilization: 3.33 percent           <-- E1 utilization
 Total channels: 60                  <-- two E1 whereas one port shutdown
 Addressable channels: 30            <-- just on E1 operational
 Inuse channels: 1                   <-- currently used channels
 Disabled channels: 30               <-- second E1 not operational
 Free channels: 29                   <-- available channels 

The following is sample output from Voice Gateway VG200:
VG200A#show mgcp statistics
 UDP pkts rx 3791, tx 3830
 Unrecognized rx pkts 0, MGCP message parsing errors 0
 Duplicate MGCP ack tx 0, Invalid versions count 0
 CreateConn rx 12, successful 12, failed 0
 DeleteConn rx 12, successful 12, failed 0
 ModifyConn rx 42, successful 42, failed 0
 DeleteConn tx 0, successful 0, failed 0
 NotifyRequest rx 8, successful 8, failed 0
 AuditConnection rx 0, successful 0, failed 0
 AuditEndpoint rx 20, successful 20, failed 0
 RestartInProgress tx 6, successful 6, failed 0
 Notify tx 3704, successful 3704, failed 0
 ACK tx 68, NACK tx 0
 ACK rx 3703, NACK rx 0
 IP address based Call Agents statistics:
 IP address 172.16.1.252, Total msg rx 3791,
                  successful 3791, failed 0
VG200A# 
Table: Explanation of Fields in the show mgcp statistics Command
Field OutputDescription
UDP pktsThe number of UDP packets received (rx) and transmitted (tx).
Unrecognized rx pktsThe number of packets received that are of unknown type.
MGCP message parsing errorsThe number of MGCP message parsing errors.
Duplicate MGCP ack txThe number of duplicate MGCP ACK transmission messages.
Invalid versions countThe number of invalid versions.
CreateConn rx ...The number of Create Connection messages received from the call agent by the media gateway. Messages received are classified as being successful or failed.
DeleteConn rx ...The number of Delete Connection messages received from the call agent by the media gateway. Messages received are classified as being successful or failed.
ModifyConn rx ...The number of Modify Connection messages received from the call agent by the media gateway. Messages received are classified as being successful or failed.
DeleteConn tx ...The number of Delete Connection messages sent by the call agent. Messages received are classified as being successful or failed.
NotifyRequest rx ...The number of Notify messages received by the call agent from the media gateway. Messages received are classified as being successful or failed.
AuditConnection rx ...The number of Audit Connection messages received from the call agent by the media gateway. Messages received are classified as being successful or failed.
AuditEndpoint rx ...The number of Audit Endpoint messages received from the call agent by the media gateway. Messages received are classified as being successful or failed.
RestartInProgress tx ...The number of Restart In Progress (RSIP) messages transmitted by the call agent. Messages received are classified as being successful or failed.
Notify tx ...The number of Notify messages transmitted by the call agent. Messages received are classified as being successful or failed.
ACK tx ...The number of acknowledgement messages transmitted by the call agent.
NACK tx ...The number of negative acknowledgement messages transmitted by the call agent.
ACK rx ...The number of acknowledgement messages received by the gateway.
NACK rx ...The number of negative acknowledgement messages received by the gateway.
IP addressThe IP address of the call agent.
Total msg rx ...The total number of messages received by the gateway. Messages received are classified as being successful or failed.

Other debug mgcp Commands

Use debug mgcp {all | error | events | packets | parser} when you are experiencing problems that you believe are not related to configuration errors or hardware problems. It is recommended that you keep an example of each debug command from a working configuration to use as a baseline for comparison when you are experiencing problems.

MGCP Connections and Endpoints Verification

controller T3 9/0
 framing m23
 cablelength 224
 t1 1-8 controller
!
controller T1 9/0:1
 framing esf
 ds0-group 0 timeslots 1-24 type e&m-fgb mf dnis
!
controller T1 9/0:2
 framing esf
 ds0-group 0 timeslots 1-24 type e&m-fgb dtmf dnis
!
controller T1 9/0:3
 framing esf
 ds0-group 0 timeslots 1-24 type e&m-immediate-start
!
controller T1 9/0:4
 framing esf
 ds0-group 0 timeslots 1-24 type e&m-fgb
!
controller T1 9/0:5
 framing esf
 pri-group timeslots 1-24 service mgcp
!
controller T1 9/0:6
 framing esf
 ds0-group 0 timeslots 1-24 type none service mgcp
!
controller T1 9/0:7
 framing esf
 extsig mgcp
 guard-timer 10 on-expiry  reject 
 pri-group timeslots 1-24 service mgcp
!
controller T1 9/0:8
 framing esf
 extsig mgcp
 guard-timer 10 on-expiry  reject 
 ds0-group 0 timeslots 1-24 type none service mgcp
!

Troubleshooting MGCP SRC CAC

Troubleshooting MGCP SRC CAC
To help identify SRC CAC problems, use the following commands in privileged EXEC mode:

CommandPurpose
Router# show call threshold {status [unavailable] | stats} Displays status of configured triggers or statistics for application programming interface (API) calls that were made to global and interface resources
Router# show mgcp statistics Displays MGCP statistics, including those for MGCP SRC VoIP CAC
Router# clear call threshold statsClears call threshold statistics
Router# clear mgcp src-stats Clears statistics gathered for MGCP SRC CAC
Router# debug call thresholdDisplays details of trigger actions
Router# debug mgcp src Provides debug information for MGCP SRC CAC calls

Troubleshooting MGCP RSVP CAC

To identify and trace RSVP CAC problems, use the following commands in privileged EXEC mode:
CommandPurpose
Router# show call fallback cacheDisplays a network congestion level check result if one has been cached
Router# show call rsvp-sync stats Displays statistics for calls that attempted RSVP reservation
Router# show call rsvp-sync confDisplays the configuration settings for RSVP synchronization
Router# show ip rsvp reservation Displays the RSVP-related receiver information currently in the database
Router# debug call rsvp-sync func-trace Displays messages about software functions called by RSVP
Router# debug call rsvp-sync events Displays events that occur during RSVP setup
Router# debug ip rsvp detail Displays detailed information about RSVP-enabled and Subnetwork Bandwidth Manager (SBM) message processing

Troubleshooting MGCP SA Agent CAC

To help identify Service Assurance (SA) Agent CAC problems, use the following commands in privileged EXEC mode:
CommandPurpose
Router# show call fallback cache Displays a network congestion level check result if one has been cached
Router# debug call fallback probes Verifies that probes are being sent correctly
Router# debug call fallback detail Displays details of the VoIP call fallback
Router# show rtr application {tabular | full} Displays global information about the SA agent feature. There are a number of other options for the show rtr command; use CLI help to browse a list of choices
Router# debug rtr error Enables logging of SA agent run-time errors
Router# debug rtr traceTraces the execution of an SA agent operation

MGCP Call Routing and Dial Peers

Verifying Digits Received and Sent on the POTS Call Leg
Once the on-hook and off-hook signaling are verified to be working correctly, the next step in troubleshooting and debugging a VoIP call is to verify that the correct digits are being received or sent on the voice port (digital or analog). A dial peer is not matched or the switch (CO or PBX) cannot ring the correct station if incomplete or incorrect digits are being sent or received. Some commands that can be used to verify the digits received/sent are:
  • show dialplan number-This command is used to show which dial peer is reached when a particular telephone number is dialed.
  • debug vtsp session-This command displays information on how each network indication and application request is processed, signaling indications, and DSP control messages.
  • debug vtsp dsp -This command displays the digits as they are received by the voice port.
  • debug vtsp all-This command enables the following debug voice telephony service provider (VTSP) commands: debug vtsp session, debug vtsp error, and debug vtsp dsp.

show dialplan number

The show dialplan number digit_string command displays the dial peer that is matched by a string of digits. If multiple dial peers can be matched, they are all shown in the order in which they are matched. The output of this command looks like this:
Router# show dialplan number 5000
Macro Exp.: 5000
VoiceOverIpPeer2
        information type = voice,
        tag = 2, destination-pattern = `5000',
        answer-address = `', preference=0,
        group = 2, Admin state is up, Operation
        state is up,
        incoming called-number = `', 
        connections/maximum = 0/unlimited,
        application associated: 
        type = voip, session-target = 
        `ipv4:192.168.10.2',
        technology prefix: 
        ip precedence = 5, UDP checksum = 
        disabled, session-protocol = cisco, 
        req-qos = best-effort, 
        acc-qos = best-effort, 
dtmf-relay = cisco-rtp, 
        fax-rate = voice,   
        payload size =  20 bytes
        codec = g729r8,   
        payload size =  20 bytes,
        Expect factor = 10, Icpif = 30,
        signaling-type = cas,
        VAD = enabled, Poor QOV Trap = disabled, 
        Connect Time = 25630, Charged Units = 0,
        Successful Calls = 25, Failed Calls = 0,
        Accepted Calls = 25, Refused Calls = 0,
        Last Disconnect Cause is "10  ",
        Last Disconnect Text is "normal call 
        clearing.",
        Last Setup Time = 84427934.
        Matched: 5000   Digits: 4
        Target: ipv4:192.168.10.2

debug vtsp dsp

debug vtsp dsp shows the digits as they are received by the voice port. The following output shows the collection of DTMF digits from the DSP:
Router# debug vtsp dsp 
Voice telephony call control dsp debugging is on
!-- ACTION: Caller picked up handset and dialed 
!-- digits 5000.
!-- The DSP detects DTMF digits. Digit 5 was 
!-- detected with ON time of 130msec.
*Mar 10 17:57:08.505: vtsp_process_dsp_message: 
MSG_TX_DTMF_DIGIT_BEGIN: digit=5, 
*Mar 10 17:57:08.585: vtsp_process_dsp_message: 
MSG_TX_DTMF_DIGIT_OFF: digit=5, 
duration=130
*Mar 10 17:57:09.385: vtsp_process_dsp_message: 
MSG_TX_DTMF_DIGIT_BEGIN: digit=0
*Mar 10 17:57:09.485: vtsp_process_dsp_message: 
MSG_TX_DTMF_DIGIT_OFF: digit=0, 
duration=150
*Mar 10 17:57:10.697: vtsp_process_dsp_message: 
MSG_TX_DTMF_DIGIT_BEGIN: digit=0
*Mar 10 17:57:10.825: vtsp_process_dsp_message: 
MSG_TX_DTMF_DIGIT_OFF: digit=0, 
duration=180
*Mar 10 17:57:12.865: vtsp_process_dsp_message: 
MSG_TX_DTMF_DIGIT_BEGIN: digit=0
*Mar 10 17:57:12.917: vtsp_process_dsp_message: 
MSG_TX_DTMF_DIGIT_OFF: digit=0, 
duration=100
Router# debug vtsp session 
Voice telephony call control session debugging is on
!--- <some output have been omitted>
!-- ACTION: Caller picked up handset. 
!-- The DSP is allocated, jitter buffers, VAD 
!-- thresholds, and signal levels are set.
*Mar 10 18:14:22.865: dsp_set_playout: [1/0/0 (69)]
packet_len=18 channel_id=1 packet_id=76 mode=1 
initial=60 min=4 max=200 fax_nom=300
*Mar 10 18:14:22.865: dsp_echo_canceller_control: 
[1/0/0 (69)] packet_len=10 channel_id=1 packet_id=66
flags=0x0
*Mar 10 18:14:22.865: dsp_set_gains: [1/0/0 (69)] 
packet_len=12 channel_id=1 packet_id=91 
in_gain=0 out_gain=65506
*Mar 10 18:14:22.865: dsp_vad_enable: [1/0/0 (69)] 
packet_len=10 channel_id=1 packet_id=78 
thresh=-38act_setup_ind_ack 
*Mar 10 18:14:22.869: dsp_voice_mode: [1/0/0 (69)] 
packet_len=24 channel_id=1 packet_id=73 coding_type=1
voice_field_size=80 
VAD_flag=0 echo_length=64 comfort_noise=1 
inband_detect=1 digit_relay=2 
AGC_flag=0act_setup_ind_ack(): dsp_dtmf_mod
e()act_setup_ind_ack: passthru_mode = 0, 
no_auto_switchover = 0dsp_dtmf_mode
(VTSP_TONE_DTMF_MODE)
!-- The DSP is put into "voice mode" and dial-tone is 
!-- generated.
*Mar 10 18:14:22.873: dsp_cp_tone_on: [1/0/0 (69)] 
packet_len=30 channel_id=1 packet_id=72 tone_id=4 
n_freq=2 freq_of_first=350 freq_of_second=440 amp_of_first=
4000 amp_of_second=4000 direction=1 on_time_first=65535 
off_time_first=0 on_time
_second=65535 off_time_second=0

If you determine that the digits are not being sent or received properly, then you might need to use either a digit-grabber (test tool) or T1 tester to verify that the digits are being sent at the correct frequency and timing interval. If they are being sent "incorrectly" for the switch (CO or PBX), some values on the router or switch (CO or PBX) might need to be adjusted so that they match and the devices can interoperate. These are usually digit duration and interdigit duration values. If the digits appear to be sent correctly, you can also check any number translation tables in the switch (CO or PBX) that might be adding or removing digits.

Verifying End-to-End VoIP Signaling on the VoIP Call Leg

After verifying that voice-port signaling is working properly and that the correct digits have been received, move to the VoIP call control troubleshooting and debugging. The following factors explain why call control debugging can be a complex job:
  • H.323 is made up of three layers of call-negotiation and call-establishment: H.225, H.245, and H.323. These protocols use a combination of TCP and UDP to set up and establish a call.
  • End-to-End VoIP debugging shows a number of Cisco IOS state-machines, and problems with any state-machine can cause a call to fail.
  • End-to-End VoIP debugging can be very verbose and create a lot of debug output.