Table Of Contents

Troubleshooting MGCP and Related Protocol Interfaces to the IP Network

MGCP Overview

MGCP Gateway Call Flow

User Access Verification

Troubleshooting Guidelines

Call Routing and Dial Peers

Verifying Digits Received and Sent on the POTS Call Leg

show dialplan number

debug vtsp dsp

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

Call Admission Control

Troubleshooting MGCP

Troubleshooting MGCP SRC CAC

Troubleshooting MGCP RSVP CAC

Troubleshooting MGCP SA Agent CAC

Verifying Connections and Endpoints

MGCP Testing Commands

show ccm-manager

show mgcp

show mgcp endpoint

show mgcp connection

show voice port mod_num/slot_num/port_num

show mgcp statistics

Other debug mgcp Commands

Troubleshooting MGCP and Related Protocol Interfaces to the IP Network

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Media gateway control protocol (MGCP) bases its call control and intelligence in centralized call agents, also called media gateway controllers. The call agents issue commands to simple, low-cost endpoints, which are housed in media gateways (MGs), and they also receive event reports from the gateways. MGCP messages between call agents and media gateways are sent over IP/UDP.

For information about configuring MGCP, refer to the MGCP and Related Protocols Configuration Guide.

This chapter contains the following sections:

•MGCP Overview

•Call Routing and Dial Peers

•Call Admission Control

•Verifying Connections and Endpoints

•MGCP Testing Commands

MGCP Overview

Two basic MGCP constructs are endpoints and connections. An endpoint is a source or sink for call data (RTP/IP) that is flowing through the gateway. A common type of endpoint is found at the physical interface between the POTS or PSTN service and the gateway; this type of endpoint might be an analog voice port or a digital DS0 group. There are other types of endpoints as well, and some are logical rather than physical. An endpoint is identified by a two-part endpoint name that contains the name of the entity on which it exists (for example, an access server or router) and the local name by which it is known (for example, a port identifier).

Call agents manage call flow using standard MGCP commands that are sent to the endpoints under their control. The commands are delivered in standard ASCII text, and can contain session descriptions transmitted in Session Description Protocol (SDP), a text-based protocol. These messages are sent over IP/UDP.

Call agents keep track of endpoint and connection status through the gateway's reporting of standard events that are detected from endpoints and connections. Call agents also direct gateways to apply certain standard signals when a POTS/PSTN connection expects them. For example, when someone picks up a telephone handset, an off-hook event is detected on an endpoint on the residential gateway to which the telephone is connected. The gateway reports the event to a call agent, which orders the gateway to apply the dial-tone signal to the endpoint reporting the off-hook event. The person picking up the handset hears dial tone.

Figure 41 shows a hypothetical MGCP network with both residential and trunking gateways. The residential gateway has telephone sets connected to the gateway's FXS voice ports. MGCP or NCS over IP/UDP is used for call control and reporting to the call agent, and Real-time Transmission Protocol (RTP) is used to transmit the actual voice data.

Figure 41 also shows two trunking gateways with T1 (or E1) connections to the PSTN. Incoming time-division multiplexing (TDM) data is sent through the gateway into the packet network through the use of RTP. MGCP or TGCP over IP/UDP is used for call control and reporting to the call agent. Signaling System 7 (SS7) data travels a different route, however, bypassing the trunking gateway entirely in favor of a specialized signaling gateway, where the signaling data is transformed to ISUP/IP format and relayed to the call agent. Communication between two signaling gateways in the same packet network can be done with ISUP/IP, H.323, or Session Initiation Protocol (SIP).

Figure 41 MGCP Network Model

MGCP Gateway Call Flow

The following example shows an MGCP call flow. The network topology is shown in Figure 42 . In this example, the FXS port is registered as an endpoint to Cisco CallManager.

Figure 42 MGCP Call Flow Topology

User Access Verification

In this example, the show log command is used to show the MGCP packets sent between the Cisco gateway and Cisco CallManager.

Router#

Router#show log

Syslog logging: enabled (0 messages dropped, 1 messages rate-limited, 0 flushes,

 0 overruns, xml disabled)

    Console logging: level debugging, 280 messages logged, xml disabled

    Monitor logging: level debugging, 109 messages logged, xml disabled

    Buffer logging: level debugging, 69 messages logged, xml disabled

    Logging Exception size (4096 bytes)

    Count and timestamp logging messages: disabled

    Trap logging: level informational, 39 message lines logged

Log Buffer (10000000 bytes):

The FXS port goes off hook and notifies the CallAgent of the status through observed event. Here, the MGCP packet is sent by the gateway.

*Mar  1 02:16:29.399: send_mgcp_msg, MGCP Packet sent to 172.6.104.54 

*Mar  1 02:16:29.399: NTFY 186 aaln/S1/SU0/0@Router MGCP 0.1

X: 1b

O: L/hd

Here, the gateway receives the response from the CallAgent.

*Mar  1 02:16:29.399: MGCP Packet received from 172.6.104.54-

200 186

The CallAgent directs the port to provide the dial tone and requests a notification if there is any change of events such as a port disconnect or digits received.

*Mar  1 02:16:29.411: MGCP Packet received from 172.6.104.54-

RQNT 40 AALN/S1/SU0/0@Router MGCP 0.1

X: 1c

R: L/hu, D/[0-9ABCD*#]

S: L/dl

Q: process,loop

*Mar  1 02:16:29.419: send_mgcp_msg, MGCP Packet sent to 172.6.104.54 

The FXS port (endpoint) notifies the Call Agent about the digits that it received.

*Mar  1 02:16:29.419: 200 40 OK

*Mar  1 02:16:33.595: send_mgcp_msg, MGCP Packet sent to 172.6.104.54 

*Mar  1 02:16:33.595: NTFY 187 AALN/S1/SU0/0@Router MGCP 0.1

X: 1c

O: D/1

*Mar  1 02:16:33.599: MGCP Packet received from 172.6.104.54-

200 187

*Mar  1 02:16:33.603: MGCP Packet received from 172.6.104.54-

RQNT 41 AALN/S1/SU0/0@Router MGCP 0.1

X: 1d

R: L/hu, D/[0-9ABCD*#], L/hf

S:

Q: process,loop

*Mar  1 02:16:33.607: send_mgcp_msg, MGCP Packet sent to 172.6.104.54

*Mar  1 02:16:33.607: 200 41 OK

*Mar  1 02:16:35.655: send_mgcp_msg, MGCP Packet sent to 172.6.104.54 

*Mar  1 02:16:35.655: NTFY 188 AALN/S1/SU0/0@Router MGCP 0.1

X: 1d

O: D/0

*Mar  1 02:16:35.659: MGCP Packet received from 172.6.104.54-

200 188

*Mar  1 02:16:37.275: send_mgcp_msg, MGCP Packet sent to 172.6.104.54 

*Mar  1 02:16:37.275: NTFY 189 AALN/S1/SU0/0@Router MGCP 0.1

X: 1d

O: D/0

*Mar  1 02:16:37.279: MGCP Packet received from 172.6.104.54-

200 189

*Mar  1 02:16:38.815: send_mgcp_msg, MGCP Packet sent to 172.6.104.54 

*Mar  1 02:16:38.815: NTFY 190 AALN/S1/SU0/0@Router MGCP 0.1

X: 1d

O: D/1

<---

*Mar  1 02:16:38.819: MGCP Packet received from 172.6.104.54-

200 190

*Mar  1 02:16:38.839: MGCP Packet received from 172.6.104.54-

CRCX 42 AALN/S1/SU0/0@Router MGCP 0.1

C: A00000000200001d

X: 1e

M: inactive

R: L/hu

Q: process,loop

*Mar  1 02:16:38.851: send_mgcp_msg, MGCP Packet sent to 172.6.104.54 

*Mar  1 02:16:38.851: 200 42 OK

I: 4

v=0

c=IN IP4 172.16.13.41

m=audio 19156 RTP/AVP 0 8 99 101 102 2 15 103 4 104 105 106 107 18 100

a=rtpmap:99 G.729a/8000

a=rtpmap:101 G.726-16/8000

a=rtpmap:102 G.726-24/8000

a=rtpmap:103 G.723.1-H/8000

a=rtpmap:104 G.723.1-L/8000

a=rtpmap:105 G.729b/8000

a=rtpmap:106 G.723.1a-H/8000

a=rtpmap:107 G.723.1a-L/8000

a=rtpmap:100 X-NSE/8000

a=fmtp:100 200-202

a=X-sqn:0

a=X-cap: 1 audio RTP/AVP 100

a=X-cpar: a=rtpmap:100 X-NSE/8000

a=X-cpar: a=fmtp:100 200-202

a=X-cap: 2 image udptl t38

<---

*Mar  1 02:16:38.855: MGCP Packet received from 172.6.104.54-

RQNT 43 AALN/S1/SU0/0@Router MGCP 0.1

X: 1f

R: L/hu

S: G/rt

Q: process,loop

At this time the call is sent to the IP Phone. The CA directs the IP Phone to start playing ringbacks.

*Mar  1 02:16:38.859: send_mgcp_msg, MGCP Packet sent to 172.6.104.54 

*Mar  1 02:16:38.859: 200 43 OK

<---

*Mar  1 02:16:46.159: MGCP Packet received from 172.6.104.54-

MDCX 44 AALN/S1/SU0/0@Router MGCP 0.1

C: A00000000200001d

I: 4

X: 20

L: p:20, a:PCMU, s:off

M: recvonly

R: L/hu

Q: process,loop

*Mar  1 02:16:46.167: send_mgcp_msg, MGCP Packet sent to 172.6.104.54 

*Mar  1 02:16:46.167: 200 44 OK

<---

*Mar  1 02:16:46.171: MGCP Packet received from 172.6.104.54-

RQNT 45 AALN/S1/SU0/0@Router MGCP 0.1

X: 21

R: L/hu, D/[0-9ABCD*#], L/hf

S:

Q: process,loop

*Mar  1 02:16:46.175: send_mgcp_msg, MGCP Packet sent to 172.6.104.54 --->

*Mar  1 02:16:46.175: 200 45 OK

*Mar  1 02:16:46.179: MGCP Packet received from 172.6.104.54-

MDCX 46 AALN/S1/SU0/0@Router MGCP 0.1

C: A00000000200001d

I: 4

X: 22

L: p:20, a:PCMU, s:off

M: sendrecv

R: L/hu, L/hf, D/[0-9ABCD*#]

S:

Q: process,loop

v=0

o=- 4 0 IN EPN AALN/S1/SU0/0@Router

s=Cisco SDP 0

t=0 0

c=IN IP4 10.17.179.33

m=audio 28390 RTP/AVP 96

a=rtpmap:96 PCMU

*Mar  1 02:16:46.187: send_mgcp_msg, MGCP Packet sent to 172.6.104.54 

*Mar  1 02:16:46.191: 200 46 OK

Now the call is answered. Note that media is cut in both directions.After about 12 seconds the called party on the IP Phone drops the call.

*Mar  1 02:16:58.355: MGCP Packet received from 172.6.104.54-

MDCX 47 AALN/S1/SU0/0@Router MGCP 0.1

C: A00000000200001d

I: 4

X: 23

M: recvonly

R: L/hu

Q: process,loop

*Mar  1 02:16:58.359: send_mgcp_msg, MGCP Packet sent to 172.6.104.54 

*Mar  1 02:16:58.359: 200 47 OK

*Mar  1 02:16:58.379: MGCP Packet received from 172.6.104.54-

DLCX 48 AALN/S1/SU0/0@Router MGCP 0.1

C: A00000000200001d

I: 4

X: 24

R: L/hu

S:

Q: process,loop

*Mar  1 02:16:58.383: send_mgcp_msg, MGCP Packet sent to 172.6.104.54 

*Mar  1 02:16:58.383: 250 48 OK

P: PS=608, OS=97280, PR=604, OR=96640, PL=0, JI=64, LA=0

*Mar  1 02:17:01.403: MGCP Packet received from 172.6.104.54-

RQNT 49 AALN/S1/SU0/0@Router MGCP 0.1

X: 25

R: L/hu, D/[0-9ABCD*#]

S: L/dl

Q: process,loop

Since this call was made from an FXS phone the calling party hears a dial tone

*Mar  1 02:17:01.411: send_mgcp_msg, MGCP Packet sent to 172.6.104.54 

*Mar  1 02:17:01.411: 200 49 OK

*Mar  1 02:17:03.135: send_mgcp_msg, MGCP Packet sent to 172.6.104.54 

*Mar  1 02:17:03.135: NTFY 191 AALN/S1/SU0/0@Router MGCP 0.1

X: 25

O: L/hu

*Mar  1 02:17:03.139: MGCP Packet received from 172.6.104.54-

200 191

*Mar  1 02:17:03.143: MGCP Packet received from 172.6.104.54-

RQNT 50 AALN/S1/SU0/0@Router MGCP 0.1

X: 26

R: L/hd

S:

Q: process,loop

*Mar  1 02:17:03.147: send_mgcp_msg, MGCP Packet sent to 172.6.104.54 

*Mar  1 02:17:03.147: 200 50 OK

Troubleshooting Guidelines

The following suggestions help with troubleshooting:

•Reset the MGCP statistical counters with the clear mgcp statistics command.

•If RTP traffic is not getting through, make sure IP routing is enabled. Use the show rtp statistics command, then turn on the debug ip udp command and track down the MGCP RTP packets.

Router# show rtp statistics

RTP Statistics info:

No. CallId Xmit-pkts Xmit-bytes Rcvd-pkts  Rcvd-bytes Lost pkts  Jitter Latenc

1   17492  0x8A      0x5640     0x8A       0x5640     0x0        0x0      0x0 

Router# show rtp statistics

RTP Statistics info:

No. CallId Xmit-pkts Xmit-bytes Rcvd-pkts  Rcvd-bytes Lost pkts  Jitter Latenc

1   17492  0xDA      0x8840     0xDB       0x88E0     0x0        0x160    0x0 

•If an RSIP message is not received by the call agent, make sure that the mgcp call-agent command or the MGCP profile call-agent command is configured with the correct call agent name or IP address and UDP port. Use the show mgcp command or the show mgcp profile command to display this information:

Router# show mgcp

MGCP Admin State ACTIVE, Oper State ACTIVE - Cause Code NONE

MGCP call-agent: 172.29.248.51  Initial protocol service is MGCP, v. 1.0

...

MGCP gateway port: 2727, MGCP maximum waiting delay 3000

...

Router# show mgcp profile

MGCP Profile nycprofile

Description: NY branch office configuration

Call-agent: 10.14.2.200 Initial protocol service is MGCP, v. 1.0

...

•To verify connections and endpoints, use the show mgcp command:

Router# show mgcp connection

Endpoint  Call_ID(C) Conn_ID(I) (P)ort (M)ode (S)tate (C)odec (E)vent[SIFL] 
(R)esult[EA]

1. S0/DS1-1/5      C=F123AB,5,6  I=0x3  P=16506,16602  M=3  S=4  C=1 E=2,0,0,2  R=0,0

2. S0/DS1-1/6      C=F123AB,7,8  I=0x4  P=16602,16506  M=3  S=4  C=1 E=0,0,0,0  R=0,0

Router# show mgcp endpoint 

T1/0 ds0-group 0 timeslots 1-24

T1/1 ds0-group 0 timeslots 1-24

T1/2 ds0-group 0 timeslots 1-24

T1/3 ds0-group 0 timeslots 1-24

•If an MGCP message is rejected, it might be because the remote media gateway does not support SDP mandatory parameters (the o=, s=, and t= lines). If this is the case, configure the mgcp sdp simple command to send SDP messages without those parameters.

•If you notice problems with voice quality, make sure that the cptone (voice-port configuration) command is set for the correct country code. Capturing RTP packets from the sniffer might help to debug the problem; you can decide such questions as whether the payload type or timestamps are set correctly.

•To check operation of interfaces, use the show interface command.

•To view information about activity on the T1 or E1 line, use the show controllers command. Alarms, line conditions, and other errors are displayed. The data is updated every 10 seconds; and every 15 minutes, the cumulative data is stored and retained for 24 hours.

•When necessary, you can enable debug traces for errors, events, media, packets, and parser. The command debug mgcp packets can be used to monitor message flow in general. Note that there is always a performance penalty when using debug commands. The sample output below shows the use of the optional input-hex keyword to enable display of hexadecimal values.

Router# debug mgcp {all | errors | events | packets {input-hex}| parser}

Router# debug mgcp packets input-hex 

Media Gateway Control Protocol input packets in hex value debugging is on

MGCP Packet received -

DLCX 49993 * MGCP 0.1

MGCP Packet received in hex -

44 4C 43 58 20 34 39 39 39 33 20 2A 20 4D 47 43 50 20 30 2E 31 A

send_mgcp_msg, MGCP Packet sent --->

250 49993 

Call Routing and Dial Peers

To troubleshoot xGCP call routing, see the following sections:

•Verifying Digits Received and Sent on the POTS Call Leg

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

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.

Call Admission Control

MGCP VoIP call admission control (CAC) has several commands available to analyze call statistics and operation of applications on the gateway. They are classified into these groups for clarity:

•Troubleshooting MGCP

•Troubleshooting MGCP SRC CAC

•Troubleshooting MGCP RSVP CAC

•Troubleshooting MGCP SA Agent CAC

Troubleshooting MGCP

To provide information about the operation of the MGCP application, use the following commands in privileged EXEC mode:

Command	Purpose
Router# debug mgcp all	Displays real-time information about MGCP errors, events, media, packets, parser, system resource check (SRC), and VoIP call admission control (CAC)
Router# debug mgcp errors {endpoint endpoint-name}	Displays MGCP errors
Router# debug mgcp events {endpoint endpoint-name}	Displays MGCP events
Router# debug mgcp media {endpoint endpoint-name}	Displays MGCP tone and signal information
Router# debug mgcp packets {endpoint endpoint-name | input-hex}	Displays MGCP packet information, with input packets optionally in hexadecimal format
Router# debug mgcp parser	Displays MGCP parser and builder information
Router# debug mgcp src	Displays MGCP SRC CAC information
Router# debug mgcp voipcac	Turns on debugging messages for the VoIP CAC process at the MGCP application layer

Troubleshooting MGCP SRC CAC

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

Command	Purpose
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 stats	Clears call threshold statistics
Router# clear mgcp src-stats	Clears statistics gathered for MGCP SRC CAC
Router# debug call threshold	Displays 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:

Command	Purpose
Router# show call fallback cache	Displays 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 conf	Displays 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:

Command	Purpose
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 trace	Traces the execution of an SA agent operation

Verifying Connections and Endpoints

To verify MGCP-controlled endpoints configured for SS7 and ISDN PRI, use the show mgcp endpoint command in privileged EXEC mode.

---

Note The show mgcp endpoint command does not show configured endpoints for CAS, including FGD-OS.

---

The following example shows MGCP endpoints for a NAS package:

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

!

The following output shows available varieties of CAS:

•9/0:5—ISDN backhauling

•9/0:6—SS7

•9/0:7—ISDN backhauling with NAS package

•9/0:8—SS7 with NAS package

slot7# show mgcp endpoint

T1 S9/0:5 pri-group timeslots 1-23 type backhaul

T1 S9/0:6 ds0-group 0 timeslots 1-24 type none

T1 S9/0:7 pri-group timeslots 1-23 type backhaul

T1 S9/0:8 ds0-group 0 timeslots 1-24 type none

NAS Endpts

T1 9/0:7 ds0-group 0 timeslots 1-24 type none

T1 9/0:8 ds0-group 0 timeslots 1-24 type none

slot7#

MGCP Testing 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:

•show ccm-manager

•show mgcp

•show mgcp endpoint

•show mgcp connection

•show voice port mod_num/slot_num/port_num

•show mgcp statistics

•Other debug mgcp Commands

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 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 50 Explanation of Fields in the show mgcp Command  

Field Output	Description
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-agent	The address of the call agent specified in the mgcp command.
MGCP block-newcalls enabled	The state of the mgcp block-newcalls command.
MGCP request timeout	The setting for the mgcp request timeout command.
MGCP request retries	The setting for the mgcp request retries command.
MGCP gateway port	The UDP port specification.
MGCP maximum waiting delay	The setting for the mgcp max-waiting-delay command.
MGCP restart delay	The setting for the mgcp restart-delay command.
MGCP VAD	The setting for the mgcp vad command.
MGCP codec type	The setting for the mgcp codec command.
MGCP packetization period	The packetization period parameter setting for the mgcp codec command.
MGCP JB threshold low water mark	The jitter buffer minimum threshold parameter setting for the mgcp quality-threshold command.
JB threshold high water mark	The jitter buffer maximum threshold parameter setting for the mgcp quality-threshold command.
MGCP LAT threshold low water mark	The latency minimum threshold parameter setting for the mgcp quality-threshold command.
LAT threshold high water mark	The latency maximum threshold parameter setting for the mgcp quality-threshold command.
MGCP PL threshold low water mark	The packet loss minimum threshold parameter setting for the mgcp quality-threshold command.
PL threshold high water mark	The packet loss minimum threshold parameter setting for the mgcp quality-threshold command.
MGCP IP ToS low delay	The low-delay parameter setting for the mgcp ip-tos command.
MGCP IP ToS high throughput	The 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 cost	The low-cost parameter setting for the mgcp ip-tos command.
MGCP IP precedence	The precedence parameter setting for the mgcp ip-tos command.
MGCP default package type	The default-package parameter setting for the mgcp default-package command.
Supported MGCP packages	The 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 51 Explanation of Fields in the show mgcp connection Command  

Field Output	Description
Endpoint	The 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)ort	The ports used for this connection. The first port is the local UDP port. The second port is the remote UDP port.
(M)ode	The 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)tate	The call state. The values are used for internal debugging purposes.
(C)odec	The 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 52 Explanation of Fields in the show voice port Command 

Field Output	Description
Administrative State	Administrative state of the voice port.
Alias	User-supplied alias for this voice port.
Clear Wait Duration Timing	Time of inactive seizure signal to declare call cleared.
Connection Mode	Connection mode of the interface
Connection Number	Full E.164 telephone number used to establish a connection with the trunk or PLAR mode.
Currently Processing	Type of call currently being processed: none, voice, or fax.
Delay Duration Timing	Maximum delay signal duration for delay dial signaling.
Delay Start Timing	Timing of generation of delayed start signal from detection of incoming seizure.
Dial Type	Out-dialing type of the voice port.
Digit Duration Timing	DTMF Digit duration in milliseconds.
E&M Type	Type of E&M interface.
Echo Cancel Coverage	Echo Cancel Coverage for this port.
Echo Cancellation	Whether or not echo cancellation is enabled for this port.
Hook Flash Duration Timing	Maximum length of hook flash signal.
Hook Status	Hook status of the FXO/FXS interface.
Impedance	Configured terminating impedance for the E&M interface.
In Gain	Amount of gain inserted at the receiver side of the interface.
In Seizure	Incoming seizure state of the E&M interface.
Initial Time Out	Amount of time the system waits for an initial input digit from the caller.
InterDigit Duration Timing	DTMF interdigit duration in milliseconds.
InterDigit Pulse Duration Timing	Pulse dialing interdigit timing in milliseconds.
Interdigit Time Out	Amount of time the system waits for a subsequent input digit from the caller.
Maintenance Mode	Maintenance mode of the voice-port.
Music On Hold Threshold	Configured Music-On-Hold Threshold value for this interface.
Noise Regeneration	Whether or not background noise should be played to fill silent gaps if VAD is activated.
Number of signaling protocol errors	Number of signaling protocol errors.
Non-Linear Processing	Whether or not Non-Linear Processing is enabled for this port.
Operations State	Operation state of the port.
Operation Type	Operation of the E&M signal: 2-wire or 4-wire.
Out Attenuation	Amount of attenuation inserted at the transmit side of the interface.
Out Seizure	Outgoing seizure state of the E&M interface.
Port	Port number for this interface associated with the voice interface card.
Pulse Rate Timing	Pulse dialing rate in pulses per second (pps).
Regional Tone	Configured regional tone for this interface.
Ring Active Status	Ring active indication.
Ring Frequency	Configured ring frequency for this interface.
Ring Ground Status	Ring ground indication
Signal Type	Type of signaling for a voice port: loop-start, ground-start, wink-start, immediate, and delay-dial.
Slot	Slot used in the voice interface card for this port.
Sub-unit	Sub-unit used in the voice interface card for this port.
Tip Ground Status	Tip ground indication.
Type of VoicePort	Type of voice port: FXO, FXS, and E&M.
The Interface Down Failure Cause	Text string describing why the interface is down.
Wink Duration Timing	Maximum wink duration for wink start signaling.
Wink Wait Duration Timing	Maximum wink wait duration for wink start signaling.

show mgcp statistics

Use this command to show statistical information related to MGCP activity on the router.

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 53 Explanation of Fields in the show mgcp statistics Command  

Field Output	Description
UDP pkts	The number of UDP packets received (rx) and transmitted (tx).
Unrecognized rx pkts	The number of packets received that are of unknown type.
MGCP message parsing errors	The number of MGCP message parsing errors.
Duplicate MGCP ack tx	The number of duplicate MGCP ACK transmission messages.
Invalid versions count	The 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 address	The 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.

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.

© 2007 Cisco Systems, Inc. All rights reserved.