Simple VoIP trouble-shooting commands

June 17, 2007 at 11:06 pm | Posted in VoIP | 1 Comment

show voice port (Check for configuration information, including loopstart or groundstart.)
show call history voice
show call history voice last 1
show call active voice
show ip route
show logging
show debugging
show dial-peer voice
show dialplan number
show dialplan (on both the local and remote routers—verify that the data is configured correctly.)
show num-exp (if number expansion is configured)—check that the partial number on the local router maps to the correct full E.164 telephone number on the remote router.
debug vpm signal (Collect debug information about signaling events, such as detection of a ring, a call connection, or a disconnect.)
debug vpm spi—verify the output string the router dials is correct. This is not availble on 124 IOS ???
debug cch323 rtp—check RTP packet transport.
debug cch323 h225—check the call setup.

CME1#show dialplan number 201
Macro Exp.: 201

        peer type = voice, information type = voice,
        description = `’,
        tag = 101, destination-pattern = `201′,
        answer-address = `’, preference=0,
        CLID Restriction = None
        CLID Network Number = `’
        CLID Second Number sent
        CLID Override RDNIS = disabled,
        source carrier-id = `’, target carrier-id = `’,
        source trunk-group-label = `’,  target trunk-group-label = `’,
        numbering Type = `unknown’
        group = 101, Admin state is up, Operation state is up,
        Outbound state is up,
        incoming called-number = `’, connections/maximum = 0/unlimited,
        DTMF Relay = disabled,
        URI classes:
            Destination =
        huntstop = disabled,
        in bound application associated: ‘DEFAULT’
        out bound application associated: ”
        dnis-map =
        permission :both
        incoming COR list:maximum capability
        outgoing COR list:minimum requirement
        Translation profile (Incoming):
        Translation profile (Outgoing):
        incoming call blocking:
        translation-profile = `’
        disconnect-cause = `no-service’
        advertise 0x40 capacity_update_timer 25 addrFamily 0 oldAddrFamily 0
        type = pots, prefix = `’,
        forward-digits default
        session-target = `’, voice-port = `2/1′,
        direct-inward-dial = disabled,
        digit_strip = enabled,
        register E.164 number with H323 GK and/or SIP Registrar = TRUE
        fax rate = system,   payload size =  20 bytes
        supported-language = ”

        Time elapsed since last clearing of voice call statistics never
        Connect Time = 0, Charged Units = 0,
        Successful Calls = 0, Failed Calls = 0, Incomplete Calls = 0
        Accepted Calls = 0, Refused Calls = 0,
        Last Disconnect Cause is “”,
        Last Disconnect Text is “”,
        Last Setup Time = 0.
Matched: 201   Digits: 3
CME1#sh voice port 2/0

Foreign Exchange Station 2/0 Slot is 0, Sub-unit is 2, 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
 Non Linear Mute is disabled
 Non Linear Threshold is -21 dB
 Music On Hold Threshold is Set to -38 dBm
 In Gain is Set to 0 dB
 Out Attenuation is Set to 3 dB
 Echo Cancellation is enabled
 Echo Cancellation NLP mute is disabled
 Echo Cancellation NLP threshold is -21 dB
 Echo Cancel Coverage is set to 64 ms
 Echo Cancel worst case ERL is set to 6 dB
 Playout-delay Mode is set to adaptive
 Playout-delay Nominal is set to 60 ms
 Playout-delay Maximum is set to 250 ms
 Playout-delay Minimum mode is set to default, value 40 ms
 Playout-delay Fax is set to 300 ms
 Connection Mode is normal
 Connection Number is not set
 Initial Time Out is set to 10 s
 Interdigit Time Out is set to 10 s
 Call Disconnect Time Out is set to 60 s
 Supervisory Disconnect Time Out is set to 750 ms
 Ringing Time Out is set to 180 s
 Wait Release Time Out is set to 30 s
 Companding Type is u-law
 Region Tone is set for AU

 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
 Station name None, Station number None
 Translation profile (Incoming):
 Translation profile (Outgoing):

 Voice card specific Info Follows:
 Signal Type is loopStart
 Ring Frequency is 50 Hz
 Hook Status is On Hook
 Ring Active Status is inactive
 Ring Ground Status is inactive
 Tip Ground Status is active
 Digit Duration Timing is set to 100 ms
 InterDigit Duration Timing is set to 100 ms
 Hookflash-in Timing is set to max=1000 ms, min=150 ms
 Hookflash-out Timing is set to 400 ms
 No disconnect acknowledge
 Ring Cadence is defined by CPTone Selection
 Ring Cadence are [4 2] [4 20] * 100 msec
 Ringer Equivalence Number is set to 1


1 Comment »

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  1. good article on VoIP. would be adding this in my site Tech Was. Thanks looking forward for other good articles

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