Wednesday, 31 January 2018

Cisco Voice Troubleshooting commands

Credit goes to www.uccollabing.com

Show cdp – It will show CDP Timer and Holdtime Frequency

Show cdp neighbors detail – It will show details of neighbor with an IP Address and IOS version

Show cdp neighbors – It will show details of Device ID, Local Interface, Holdtime, Capability, Platform and Port ID

Show cdp interface – It will show details of the interface if it is Up physically, Line protocol is up, Encapsulation and Holdtime.

Show cdp traffic – It will show details of the CDP counters (CDP packets sent and received)

Show voice call summary – It will show all the active calls on the Gateway, Ports, Codec, VAD (enabled or not), VTSP state and VPM state

Show voice call status – It will show only the active calls, not all the ports. It includes Port, Called Number, and Dial Peer

Show call history voice record – It will show information about calls made to and from the voice router

Show voice port summary – It will show a detailed information on Ports, Channel, Signalling Type, Port Status, In Operation Status, Out Status and EC. It basically shows FXO/FXS/PRI ports in use.

Show gateway – It will show the state and version is H.323

Show dialplan number 1000 – It will show you what happens when the specified number is dialed

Show dial-peer voice summary – It will show you what all dial-peers that are currently working. Summary of dial-peers/destination.

Show voip rtp connections  – It will show all the current RTP connections which will have Local and Remote IP Address, Port Numbers, Call IDs.

show controllers T1    or   show controllers E1 – It will show the status of a controller if it is up or down

show call active voice brief – It will show the active call information which includes Call ID, Peer IP Address and Codecs

Show mgcp  – It will show mgcp settings on the gateway

Show dial-peer voice – It will show how voice dial peers are configured.

Show mgcp statistics  – It will show mgcp statistics relationship between the devices which will have stats for CRCX, DLCX, MDCX and RSIP,  IP addresses of Call Agents etc

Show mgcp endpoint   It will show information related to MGCP endpoints

Show mgcp connection  – It will show information about the current mgcp connections

show sip-ua register status – It will show SIP Registration information

show voice dsp – It will show the status of all the DSPs on the Gateway

show ccm-manager  – It will show information about the active and redundant configured Cisco Unified Communications Manager. This command also indicates if the gateway is currently registered with Cisco Unified Communications Manager.

Show isdn active – It will show if a call is in progress and which number is being dialed.

Show isdn status – It will show statistics of an ISDN connection and show if your PRI is up/established correctly

show ccm-manager fallback – It will show whether MGCP fallback is enabled or disabled, if enabled, whether it is currently active or not.

Show sip service – It will help to display the status of SIP call service in a SIP gateway

show sip-ua status – It will help to display status for the SIP user agent (UA), including whether call redirection is enabled or disabled

cdp enable – It will enabled CDP on an interface

no cdp enable – It will disable CDP on an interface

Test voice translation-rule – It will allow you to test a translation rule configured on the gateway.

Csim start XXXXX – It is a hidden command which helps to generate calls

Debug voice ccapi inout – It will show calls in and out of ports on the gateway

Debug mgcp packet  – It will monitor the packets exchanged between CUCM and router.

Debug isdn q921 – It will help to verify if you have a connection to the ISDN switch.

Debug isdn q931  – It will monitor information about call setup and tear down of ISDN network connections (layer 3) between local router (user side) and the network

Debug h225  – It will display additional information about the contents of H.225 Registration, Admission, and Status Protocol (RAS) messages.

Debug vtsp tone – It will help to view tone generated by the router which could be fast busy or dial tone or busy signal etc.

Debug voip ccapi inout – It will trace the execution path through the call control API, which serves as the interface between the call session application and the underlying network-specific software.

Debug vpm signal – It will help to view to view the on−hook and off−hook signaling for the voice ports.

Debug vtsp dsp  – It will help to view digits collection performed on the router

Debug ccsip calls – It will help to show all SIP call details as they are updated in the SIP call control block. This debug command can be used to monitor call records for suspicious clearing causes.

Debug ccsip messages – It will help you to enable all SIP message tracing, such as those that are exchanged between the SIP user-agent client (UAC) and the access server

Debug isdn event – It will show events occurring on the user side (on the router) of the ISDN interface. The ISDN events that can be displayed are Q.931 events (call setup and teardown of ISDN network connections)

Q.931 Call Disconnection Causes

These are cause codes from the debug voip ccapi inout command.
Call Disconnection Cause Value (in Hex) Meaning and Number (in Decimal)
CC_CAUSE_UANUM = 0x1 Unassigned number (1)
CC_CAUSE_NO_ROUTE = 0x3 No route to destination (3)
CC_CAUSE_NORM = 0x10 Normal call clearing (16)
CC_CAUSE_BUSY = 0x11 User busy (17)
CC_CAUSE_NORS = 0x12 No user response (18)
CC_CAUSE_NOAN = 0x13 No user answer (19)
CC_CAUSE_REJECT = 0x15 Call rejected (21)
CC_CAUSE_INVALID_NUMBER = 0x1C Invalid number (28)
CC_CAUSE_UNSP = 0x1F Normal, unspecified (31)
CC_CAUSE_NO_CIRCUIT = 0x22 No circuit (34)
CC_CAUSE_NO_REQ_CIRCUIT = 0x2C No requested circuit (44)
CC_CAUSE_NO_RESOURCE = 0x2F No resource (47)
CC_CAUSE_NOSV = 0x3F Service or option not available, or unspecified (63)


Codec Negotiation Values

These codec negotiation values are from the debug voip ccapi inout command.
Negotiation Value Meaning
codec=0x00000001 G711 ULAW 64K PCM
codec=0x00000002 G711 ALAW 64K PCM
codec=0x00000004 G729
codec=0x00000004 G729IETF
codec=0x00000008 G729a
codec=0x00000010 G726r16
codec=0x00000020 G726r24
codec=0x00000040 G726r32
codec=0x00000080 G728
codec=0x00000100 G723r63
codec=0x00000200 G723r53
codec=0x00000400 GSMFR
codec=0x00000800 G729b
codec=0x00001000 G729ab
codec=0x00002000 G723ar63
codec=0x00004000 G723ar53
codec=0x00008000 CLEAR_CHANNEL


Tone Types

Tone Types Meaning
CC_TONE_RINGBACK 0x1 Ring tone
CC_TONE_FAX 0x2 Fax tone
CC_TONE_BUSY 0x4 Busy tone
CC_TONE_DIALTONE 0x8 Dial tone
CC_TONE_OOS 0x10 Out of service tone
CC_TONE_ADDR_ACK 0x20 Address acknowledgement tone
CC_TONE_DISCONNECT 0x40 Disconnect tone
CC_TONE_OFF_HOOK_NOTICE 0x80 Tone indicating that the phone is off-hook
CC_TONE_OFF_HOOK_ALERT 0x100 A more urgent version of CC_TONE_OFF_HOOK_NOTICE
CC_TONE_CUSTOM 0x200 Custom tone-used when you are specifying a custom tone
CC_TONE_NULL 0x0 Null tone


FAX-Rate and VAD Capabilities Values

Values Meaning
CC_CAP_FAX_NONE 0x1 Fax disabled or not available
CC_CAP_FAX_VOICE 0x2 Voice call
CC_CAP_FAX_144 0x4 14,400 baud
CC_CAP_FAX_96 0x8 9,600 baud
CC_CAP_FAX_72 0x10 7,200 baud
CC_CAP_FAX_48 0x20 4,800 baud
CC_CAP_FAX_24 0x40 2,400 baud
CC_CAP_VAD_OFF 0x1 VAD disabled
CC_CAP_VAD_ON 0x2 VAD enabled

Monday, 29 January 2018

CUCM Remove prefix from call presentation

Hi,

If you have a translate rule in your gateway that adds 9 to the caller number for incoming calls, for call history, redial etc. Then you are facing an issue where your Jabber will not do Caller ID based on your contacts.
To override this behavior you can do the following:

  1. Create a new partition not used anywhere else. e.g. Transform_PT
  2. Create a new Calling Party Transformation Pattern 
    1. Pattern 9.!
    2. Partition  Transform_PT
    3. Discard Digits PreDot
  3.  Create a CSS e.g. Transform_CSS
    1. Assign the Transform_PT partition to the CSS
  4. Go to a test device and assign the Transform_CSS  

    1. under "Number Presentation Transformation"
    2. un-check "Use Device Pool Calling Party Transformation CSS"
    3. Assign the created CSS Transform_CSS
    4. Apply & Reset
  5. Test and if happy apply that to the Trunk for all users using step 4 guide.

Thursday, 11 January 2018

Cisco Show dialplan command

Hi,

If you ever wonder which dial-peer matches on specific destinations (called numbers), use the
Show dialplan number 1234 timeout 

Where 1234 is the called number and timeout is needed for matching on variable-length destination patterns.