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

VoiceEncapPeer101
        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
Target:
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

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Extended Pings can record routes that packets take

June 16, 2007 at 6:55 pm | Posted in Blogroll | 1 Comment

CME1#ping
Protocol [ip]:
Target IP address: 32.29.1.1
Repeat count [5]:
Datagram size [100]:
Timeout in seconds [2]:
Extended commands [n]: y
Source address or interface:
Type of service [0]:
Set DF bit in IP header? [no]:
Validate reply data? [no]:
Data pattern [0xABCD]:
Loose, Strict, Record, Timestamp, Verbose[none]: record
Number of hops [ 9 ]:
Loose, Strict, Record, Timestamp, Verbose[RV]:
Sweep range of sizes [n]:
Type escape sequence to abort.
Sending 5, 100-byte ICMP Echos to 32.29.1.1, timeout is 2 seconds:
Packet has IP options:  Total option bytes= 39, padded length=40
 Record route: <*>
   (0.0.0.0)
   (0.0.0.0)
   (0.0.0.0)
   (0.0.0.0)
   (0.0.0.0)
   (0.0.0.0)
   (0.0.0.0)
   (0.0.0.0)
   (0.0.0.0)

Reply to request 0 (1 ms).  Received packet has options
 Total option bytes= 40, padded length=40
 Record route:
   (32.29.1.2)
   (32.29.1.1)
   (32.29.1.1)
   (32.29.1.2) <*>
   (0.0.0.0)
   (0.0.0.0)
   (0.0.0.0)
   (0.0.0.0)
   (0.0.0.0)
 End of list

Reply to request 1 (4 ms).  Received packet has options
 Total option bytes= 40, padded length=40
 Record route:
   (32.29.1.2)
   (32.29.1.1)
   (32.29.1.1)
   (32.29.1.2) <*>
   (0.0.0.0)
   (0.0.0.0)
   (0.0.0.0)
   (0.0.0.0)
   (0.0.0.0)
 End of list

Reply to request 2 (4 ms).  Received packet has options
 Total option bytes= 40, padded length=40
 Record route:
   (32.29.1.2)
   (32.29.1.1)
   (32.29.1.1)
   (32.29.1.2) <*>
   (0.0.0.0)
   (0.0.0.0)
   (0.0.0.0)
   (0.0.0.0)
   (0.0.0.0)
 End of list

Reply to request 3 (1 ms).  Received packet has options
 Total option bytes= 40, padded length=40
 Record route:
   (32.29.1.2)
   (32.29.1.1)
   (32.29.1.1)
   (32.29.1.2) <*>
   (0.0.0.0)
   (0.0.0.0)
   (0.0.0.0)
   (0.0.0.0)
   (0.0.0.0)
 End of list

Reply to request 4 (4 ms).  Received packet has options
 Total option bytes= 40, padded length=40
 Record route:
   (32.29.1.2)
   (32.29.1.1)
   (32.29.1.1)
   (32.29.1.2) <*>
   (0.0.0.0)
   (0.0.0.0)
   (0.0.0.0)
   (0.0.0.0)
   (0.0.0.0)
 End of list

Success rate is 100 percent (5/5), round-trip min/avg/max = 1/2/4 ms

VoIP debug commands

June 3, 2007 at 1:09 am | Posted in VoIP | 1 Comment

We should timestamp the debug outputs with miliseconds, so that we can easily map the events shown on different Cisco devices.

CE1(config)#service timestamps debug datetime ?
  localtime      Use local time zone for timestamps
  msec           Include milliseconds in timestamp
  show-timezone  Add time zone information to timestamp
  year           Include year in timestamp
  <cr>

CE1(config)#service timestamps debug datetime msec

CE1#debug vtsp dsp
This CLI command is now ‘debug voip dsm dsp’
CE1#debug voip ?
  aaa           VoIP AAA
  ais           AIS (Application Information System)
  application   Application Framework library debugging
  avlist        avlist debugs
  call          Voice call debugging
  ccapi         Call Control API
  dcapi         Device Control API
  dialpeer      dialpeer debugging
  dsm           DSP Stream Manager Debugs
  dsmp          Distributed Streams Media Processor
  dspapi        Generic DSP API
  eddri         eddri debug
  enum          ENUM information
  event-log     Event logging
  hpi           HPI (54x) DSP messages
  profile       debug voip profile
  rawmsg        Raw Message
  register      voice-register debug information
  rtcp          Enable VOIP RTCP debugging trace
  rtp           Enable VOIP RTP debugging trace
  source-group  Source-group debugging
  srtcp         Enable VOIP SRTCP debugging trace
  srtp          Enable VOIP IOS SRTP debugging trace
  statistics    Voice Statistics
  translation   Debug translation rules
  tsp           Telephony Service information
  uri           Debug Voice URI
  vtsp          Voice Telephony Call Control information

CE1#debug voip dsm ?
  all      Enable All DSM debugging
  dsp      Enable dsp message trace
  error    Enable DSM error debugging
  session  Session Debugging
  stats    Stats Debugging
  tone     Tone Debugging

CE1#debug voip dsm dsp
DSP Stream Manager dsp debugging is on

Debug call legs, signalling information

CE1#debug voip ccapi ?
  all           Enable all debugs
  default       Enable default debugs
  detail        detail debug
  error         major call and software errors debug
  function      function debug
  individual    individual CCAPI debug
  inout         CCAPI Function in (enter) and out (exit)
  protoheaders  CCAPI protocol headers/bodies passing info
  service       service debug
  <cr>

CE1#debug voip ccapi inout
voip ccapi inout debugging is on
show dial-peer voice summary
CE1#sh dial-peer voice summary
dial-peer hunt 0
             AD                                    PRE PASS                OUT
TAG    TYPE  MIN  OPER PREFIX    DEST-PATTERN      FER THRU SESS-TARGET    STAT PORT
100    pots  up   up             200                0                      up   2/0
101    pots  up   up             201                0                      up   2/1
86868- voip  up   up             86868648           1  syst ipv4:202.86.50.100 
648  
Check what dial-peer a number will be matched to.

show dialplan number xxxxx

CE1#show dialplan number 201
Macro Exp.: 201

VoiceEncapPeer101
        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 = 4733, Charged Units = 0,
        Successful Calls = 2, Failed Calls = 1, Incomplete Calls = 0
        Accepted Calls = 9, Refused Calls = 0,
        Last Disconnect Cause is “10  “,
        Last Disconnect Text is “normal call clearing (16)”,
        Last Setup Time = 15828582.
Matched: 201   Digits: 3
Target:
H.323

H.225 call setup, TCP port 1720, similar to A.931
H.245 capability exchange (Codec, Media, Channel etc), TCP pot 11000+
Media (UDP), RTP port 16384-32767 (Odd+Even ports per session)
debug ras

CE1#debug ras
H.323 RAS Messages debugging is on

show interaction with gatekeeper

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