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Beginner

MGCP Controlled Gateway Call Drops

I am trying to configure MGCP controlled gateway but experiencing a call drop issue. A gateway has BRI card installed, when I try to make a call it rings but as soon as I receive call it just drop. I gathered some debug logs (Debug ISDNQ931). I have replaced DSP but still no luck.

When I ring first time after restart Gateway

Oct 31 16:04:23.730: ISDN BR0/1/0 Q931: RX <- RELEASE_COMP pd = 8  callref = 0x01

2951-rtr1#

Oct 31 16:04:32.423: ISDN BR0/1/0 Q931: RX <- SETUP pd = 8  callref = 0x01

        Sending Complete

        Bearer Capability i = 0x9090A3

                Standard = CCITT

                Transfer Capability = 3.1kHz Audio

                Transfer Mode = Circuit

                Transfer Rate = 64 kbit/s

        Channel ID i = 0x89

                Exclusive, B1

        Calling Party Number i = 0x2183, '7776137568'

                Plan:ISDN, Type:National

        Called Party Number i = 0x81, '306892'

                Plan:ISDN, Type:Unknown

Oct 31 16:04:32.435: ISDN BR0/1/0 Q931: TX -> CALL_PROC pd = 8  callref = 0x81

        Channel ID i = 0x89

                Exclusive, B1

Willenhall-2951-rtr1#

Oct 31 16:04:32.463: ISDN BR0/1/0 Q931: TX -> ALERTING pd = 8  callref = 0x81

        Progress Ind i = 0x8088 - In-band info or appropriate now available

Willenhall-2951-rtr1#

Oct 31 16:04:34.487: ISDN BR0/1/0 Q931: TX -> CONNECT pd = 8  callref = 0x81

Oct 31 16:04:34.511: ISDN BR0/1/0 Q931: TX -> DISCONNECT pd = 8  callref = 0x81

       Cause i = 0x80AF - Resource unavailable, unspecified

Oct 31 16:04:34.539: ISDN BR0/1/0 Q931: RX <- CONNECT_ACK pd = 8  callref = 0x01

Oct 31 16:04:34.551: ISDN BR0/1/0 Q931: TX -> STATUS pd = 8  callref = 0x81

       Cause i = 0x80E2 - Message not compatible with call state or not implemented

        Call State i = 0x0B

Oct 31 16:04:34.703: ISDN BR0/1/0 Q931: RX <- RELEASE pd = 8  callref = 0x01

Oct 31 16:04:34.707: ISDN BR0/1/0 Q931: TX -> RELEASE_COMP pd = 8  callref = 0x81

When I ring second time, it just beep and give this error message

Oct 31 16:05:40.233: ISDN BR0/1/0 Q931: RX <- SETUP pd = 8  callref = 0x01

        Sending Complete

        Bearer Capability i = 0x9090A3

                Standard = CCITT

                Transfer Capability = 3.1kHz Audio

                Transfer Mode = Circuit

                Transfer Rate = 64 kbit/s

        Channel ID i = 0x89

                Exclusive, B1

        Calling Party Number i = 0x2183, '7776137568'

                Plan:ISDN, Type:National

        Called Party Number i = 0x81, '306892'

                Plan:ISDN, Type:Unknown

Willenhall-2951-rtr1#

Oct 31 16:05:44.225: ISDN BR0/1/0 Q931: RX <- SETUP pd = 8  callref = 0x01

        Sending Complete

        Bearer Capability i = 0x9090A3

                Standard = CCITT

                Transfer Capability = 3.1kHz Audio

                Transfer Mode = Circuit

                Transfer Rate = 64 kbit/s

        Channel ID i = 0x89

                Exclusive, B1

        Calling Party Number i = 0x2183, '7776137568'

                Plan:ISDN, Type:National

        Called Party Number i = 0x81, '306892'

                Plan:ISDN, Type:Unknown

Willenhall-2951-rtr1#

Oct 31 16:05:48.194: ISDN BR0/1/0 Q931: RX <- DISCONNECT pd = 8  callref = 0x01

        Cause i = 0x82E6 - Recovery on timer expiry

Oct 31 16:05:48.198: ISDN BR0/1/0 Q931: TX -> RELEASE_COMP pd = 8  callref = 0x81

       Cause i = 0x80D1 - Invalid call reference value

17 REPLIES 17
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Beginner

Cause i = 0x80AF - Resource unavailable, unspecified

This above line Indicates that gateway have no idea what to do with the call.


Have you setup gateway  in CUCM ??

Do you configure dial peer point to the CUCM..??

Highlighted

It's a MGCP contolled VGW, so no dial peer would be needed.

Can you post the output from sh inventory, sh isdn status and sh ccm-manager. Also post a print screen on how the gateway is setup in CUCM.

If you also post your running config it will help us to help you.

Please remember to rate helpful responses and identify helpful or correct answers.

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Thanks for your reply, I have attached config files and screenshots. Please have a look

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Thanks for your reply, I have attached config files and screenshots. Please have a look

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Muhammad,

In my experience resource unavailable usually point to a dsp failure...Your gateway is registered so your mgcp gateway config is fine..

Please send the ff:

show voice dsp voice

show voice dsp group all

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"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

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Aokanlawon@ I have attached sh dsp.txt. Just wanted to update you I have already replaced DSP but no luck

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OK..

The dsp looks fine but I dont see any voice channels allocated...

Signaling channels allocated: 12

    Voice channels allocated: 0

Can you send cucm traces?

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"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

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Aokanlawon@ from cucm what traces you want I mean cisco tomcat cisco trace collection etc

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Aokanlawon@ When I ring on DDI number I can see voice channel been allocated, please see below. However as soon as I answer the call it gives a busy tone and disconnect.

Weird thing is when I retry number it doesnt allocate voice channel and kept showing 0 value unless reset BRI interface, ANY IDEA?

Willenhall-2951-rtr1#sh voice call status

CallID     CID  ccVdb      Port        Slot/DSP:Ch  Called #   Codec    MLPP Dial-peers

0x1C       121A 0x12AA3CA4 0/1/0.0          0/1:1  *           g711ulaw  0/0

1 active call found

Willenhall-2951-rtr1#sh voice dsp group all

DSP groups on slot 0:

dsp 1:

  State: UP, firmware: 26.3.8

  Max signal/voice channel: 32/32

  Max credits: 480

  num_of_sig_chnls_allocated: 12

  Transcoding channels allocated: 0

  Group: FLEX_GROUP_VOICE, complexity: FLEX

    Shared credits: 465, reserved credits: 0

    Signaling channels allocated: 12

    Voice channels allocated: 1

    Credits used (rounded-up): 15

    Voice channels:

      Ch01: voice port: 0/1/0.0, codec: g711ulaw, credits allocated: 15

  Slot: 0

  Device idx: 0

  PVDM Slot: 0

  Dsp Type: SP2600

  

After answer the call output of the sh voice dsp group all is


Willenhall-2951-rtr1#sh voice dsp group all
DSP groups on slot 0:
dsp 1:
  State: UP, firmware: 26.3.8
  Max signal/voice channel: 32/32
  Max credits: 480
  num_of_sig_chnls_allocated: 12
  Transcoding channels allocated: 0
  Group: FLEX_GROUP_VOICE, complexity: FLEX
    Shared credits: 480, reserved credits: 0
    Signaling channels allocated: 12
    Voice channels allocated: 0
    Credits used (rounded-up): 0
  Slot: 0
  Device idx: 0
  PVDM Slot: 0
  Dsp Type: SP2600

dsp 7:
  State: UP, firmware: 26.3.8
  Max signal/voice channel: 16/16
  Max credits: 240
  num_of_sig_chnls_allocated: 0
  Transcoding channels allocated: 0
  Group: FLEX_GROUP_VOICE, complexity: FLEX
    Shared credits: 240, reserved credits: 0
    Signaling channels allocated: 0
    Voice channels allocated: 0
    Credits used (rounded-up): 0
  Slot: 0
  Device idx: 0
  PVDM Slot: 1
  Dsp Type: SP2600


DSP groups on slot 1:
This command is not applicable to slot 1

DSP groups on slot 2:
This command is not applicable to slot 2

DSP groups on slot 3:
This command is not applicable to slot 3

DSP groups on slot 4:
This command is not applicable to slot 4

  0 DSP resource allocation failure

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What IOS are you running?

I was asking for CUCM SDI traces not tomcat traces...

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"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

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Cisco IOS Software, C2951 Software (C2951-UNIVERSALK9-M), Version 15.0(1)M4, RELEASE SOFTWARE (fc1)
Technical Support: http://www.cisco.com/techsupport
Copyright (c) 1986-2010 by Cisco Systems, Inc.
Compiled Thu 28-Oct-10 18:58 by prod_rel_team

ROM: System Bootstrap, Version 15.0(1r)M6, RELEASE SOFTWARE (fc1)

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Can you please tell me what specific logs shall I download instead of full SDI, then it may reduce the size. I am downloading logs through RTMT and we have several nodes

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First of all, you need to know the server the IP phone you are using for test is regiustered to.

Then you need to ensure that the trace settings on the server is set to detailed

Next do a test call, then use RTMT to collect the trace on the specific server for the time range that you made the call..Make sure its the SDI trace you collect..send it over.

Please check the trace before sending it that the calle and calling number is in the trace file and also send the called and calling number

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"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

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Aokanlawon: I have managed to get SDI traces, please see attached file SDI.zip. Called party number is 638094, sorry cannot provide you calling number as telco blocked CLI. Looking forward to your reply

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