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Cisco 2851 IOS 12.4(3h) using VIC2-2BRI-NT/TE and Call Manager 5.0.4.2000-1

I have the BRI conneted to a BT ISDN service and am unable to make or receive calls to the PSTN. The ISDN connection shows as level 3 up and the gateway is talking to the Call Manager fine.

I have put in a dial-peer that allows a particular PSTN number to be router out of the BRI, when we attempt to make the call the call manager has a clear code of 34 (no circuit/channel available)

I have attached the config of the 2851 gateway for reference, I have configured a plan to dial out to try and get that working, a plan exists for dialling in putting call to IP Phone 1006 but calls dont work in either direction.
Config.txt
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Scatterson

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Hi,

I have changed the config of dial-peer voice 50 pots to use pattern 0.T and still have the same issue.

Following is the output of show isdn status.
voice_gateway#show isdn status
Global ISDN Switchtype = basic-net3

%Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 2. Layer 3 output may not apply

ISDN BRI0/1/0 interface
        dsl 2, interface ISDN Switchtype = basic-net3
        L2 Protocol = Q.921 0x0000  L3 Protocol(s) = CCM MANAGER 0x0003
    Layer 1 Status:
        ACTIVE
    Layer 2 Status:
        TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED
    Layer 3 Status:
        0 Active Layer 3 Call(s)
    Active dsl 2 CCBs = 0
    The Free Channel Mask:  0x80000003

%Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 3. Layer 3 output may not apply

ISDN BRI0/1/1 interface
        dsl 3, interface ISDN Switchtype = basic-net3
        L2 Protocol = Q.921 0x0000  L3 Protocol(s) = CCM MANAGER 0x0003
    Layer 1 Status:
        ACTIVE
    Layer 2 Status:
        TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED
    Layer 3 Status:
        0 Active Layer 3 Call(s)
    Active dsl 3 CCBs = 0
    The Free Channel Mask:  0x80000003
    Total Allocated ISDN CCBs = 0

Thanks for your help.
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Hi,

voice_gateway#show 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
200    pots  up   up                                1                      down 0/1/0
100    voip  up   up             *                  0  syst ipv4:192.168.43.240
50     pots  up   up             0.T                0                      up   0/1/0

voice_gateway#csim start 01473261830
csim: called number = 01473261830, loop count = 1 ping count = 0

csim: loop = 1, failed = 1
csim: call attempted = 1, setup failed = 1, tone failed = 0


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I have setup dial peer 100 and 200 to try and make incomming calls from the PSTN to a single IP Phone. As nothing is working I am not sure of the use of wildcards and have therefore focussed on outbound calls using dial-peer 50. Should I remove dial-peer 100 and 200 for now?

The clear code I get form call manager when trying to make an outbound call is 34 (no circuit/channel available), is it possible that I dont have the correct version of IOS to support the voice card? I have found something that indicates I may need a dspfarm configuring, my configuration is set to no dspfarm on voice-card 0, could this be the issue?

Thanks,
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I have turned on q.931 debug but I have no messages output to my telnet session, do I need to be on the console terminal to see these messages?

I am using MGCP as we have a call manager.

The number from the provider for the BRI interface is 01473250178, we have no ddi range on the number.
Hi,

how do I view the ISDN diagnostic outputs? I have now a terminal connected to the serial port of the router and srill I do not see any diagnostic information when I make a call

Thanks,
Hi,

I have managed to turn on debugging of Q931 but I do not see any messages when I try and make a call

I have enabled debugging of cdp packets and I see these messages on my screen.

So it seems like the call is not being setup over the BRI interface by the Gateway or the gateway is not seeing the cards as available voice ports.
Hi,

I have managed to enable logging and made a csim start which resulted in the following:-

voice_gateway#csim start 01473261830
voice_gateway#
*Aug 26 15:40:20.047: csim: called number = 01473261830, loop count = 1 ping count = 0

*Aug 26 15:40:20.047: csimSetupPeer peer type(1), destPat(01473261830), matched(1), target()
*Aug 26 15:40:20.047: //-1/xxxxxxxxxxxx/CCAPI/ccCallSetupRequest:
   Destination=, Calling IE Present=FALSE, Mode=0,
   Outgoing Dial-peer=50, Params=0x44BBEFF0, Progress Indication=NULL(0)
*Aug 26 15:40:20.047: //-1/xxxxxxxxxxxx/CCAPI/ccCheckClipClir:
   In: Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
*Aug 26 15:40:20.047: //-1/xxxxxxxxxxxx/CCAPI/ccCheckClipClir:
   Out: Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
*Aug 26 15:40:20.047: //-1/xxxxxxxxxxxx/CCAPI/ccCallSetupRequest:
   Destination Pattern=0.T, Called Number=01473261830, Digit Strip=TRUE
*Aug 26 15:40:20.051: //-1/xxxxxxxxxxxx/CCAPI/ccCallSetupRequest:
   Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
   Called Number=01473261830(TON=Unknown, NPI=Unknown),
   Redirect Number=, Display Info=
   Account Number=, Final Destination Flag=FALSE,
   Guid=985F1AAE-918D-11DE-8020-C6EB743CC32B, Outgoing Dial-peer=50
*Aug 26 15:40:20.051: //-1/xxxxxxxxxxxx/CCAPI/cc_api_display_ie_subfields:
   ccCallSetupRequest:
   cisco-username=
   ----- ccCallInfo IE subfields -----
   cisco-ani=
   cisco-anitype=0
   cisco-aniplan=0
   cisco-anipi=0
   cisco-anisi=0
   dest=01473261830
   cisco-desttype=0
   cisco-destplan=0
   cisco-rdie=FFFFFFFF
   cisco-rdn=
   cisco-rdntype=-1
   cisco-rdnplan=-1
   cisco-rdnpi=-1
   cisco-rdnsi=-1
   cisco-redirectreason=-1
*Aug 26 15:40:20.051: //-1/xxxxxxxxxxxx/CCAPI/ccIFCallSetupRequestPrivate:
   Interface=0x45E5EC34, Interface Type=6, Destination=, Mode=0x0,
   Call Params(Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
   Called Number=01473261830(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
   Subsriber Type Str=, FinalDestinationFlag=FALSE, Outgoing Dial-peer=50, Call Count On=FALSE,
   Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)
*Aug 26 15:40:20.051: csimSetupPeer call setup err rc(-5)
*Aug 26 15:40:20.051: csim: loop = 1, failed = 1
*Aug 26 15:40:20.051: csim: call attempted = 1, setup failed = 1, tone failed = 0

*Aug 26 15:40:20.051: //-1/xxxxxxxxxxxx/CCAPI/ccAppShutdownMode:
   ccAppShutdownMode: remove it from the queue


I am completely lost so hopefully this helps?
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Hi,

Slight change to the issue.

I tried moving the cards around in the router and in the process went back to a configuration without the call manager. In this setup I was able to use csim start followed by the number and was able to call out of the router over the PSTN and ring a POTS phone. When I re-connected the call manager i could not get calls to work using csim stat or calling through the call manager. When the call manager is in the loop I loose all q931 signalling, i make a call with debug turned on and i see no activity on the signalling channel.

When the call manager is handling the number translation, prefixing etc do I need to remove the dial-peers from the gateway? is there a particular configuration for the dial peers when using the call manager?

The point at which the calls break is when the isdn bind-13 ccm-manager service mgcp is added to the interface BRI0/1/0 configuration. This seems to be added automatically by the call manager when connected to the gateway.

Thanks for all your help, I think I am getting very close to resolving this, unfortunately I need to get this working with the call manager as we are doing some integration work with our products and the cisco call manager CDR records, if we bypass the call manager we will not generate CDRs.

Many thanks
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Ok sounds good but how do I configure this?

Thanks,
On the call manager, when I add a gateway I get the coice of MGCP or SCCP as the protocol
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Ah, great thanks, will give it a go
Fantastic, I now have outgoing calls working but am struggling with the dial-peer tables and translation tables for an incomming call from the PSTN.

The incomming called number will be presented as 250178 and I need this to be routed to IP Phone 1006, any thouights on how these tales should look?

Thanks
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