H323 Connection to CUCM (Q931 error) and access to Call Logs in RTMT v7.0

[Tyler Durden]

Hi All

I have an issue with calls hitting my CUCM 7.0 from a Cisco 3825 Voice Gateway.

I have a PBX connected via E1 to the 3825 VG and at present I can call from the CallManager to the PBX through the 3825 Voice Gateway with no problems. However I am unable to call from the PBX to the CallManager through the 3825 Voice Gateway.

I've been using CallManager version 8.0 in my lab for some other testing and have been using the "Calllogs" via "View Real Time Data" within the RTMT. Now when I try to find the same tool in CallManager v 7.0 (my live network) I dont have the same options so cant find the sme tool??

When I use CallManager v8.0 I can find the Calllogs too via the following: Trace & Log Central / Real Time Trace / View Real Time Data

Products: - UCM

Services: - Cisco CallManager

Trace File Type: - calllogs

However, In version 7.0 I don't seem to have this option! Is this option avalabe in version 7.0 and if so where?

Here is a screen cap of calllogs I have been using in version 8.0

Here is some infomation on my setup and the errors I have been getting:

Calls from the CUCM to PBX via 3825 VG set as H323 Gateway: WORKING

Calls from the PBX to CUCM via 3825 VG set as H323 Gateway: NOT WORKING

A test call from 1022044 on the PBX to 77301 a DN on the CallManager with "DEBUG ISDN Q931" set on the Voice Gateway shows:

Cause i = 0x8081 - Unallocated/unassigned number

My Voice Gateway setup is as follows:

~~~~Sensative infomation removed~~~~

!

isdn switch-type primary-qsig

voice-card 0

no dspfarm

!

voice class codec 1

codec preference 1 g711alaw

codec preference 2 g711ulaw

!

voice translation-rule 1

rule 1 /.*\(7....\)$/ /\1/

!

voice translation-profile InboundE1

translate called 1

!

controller E1 0/1/0

framing NO-CRC4

pri-group timeslots 1-31

description QSIG connection to PBX 1

!

controller E1 0/1/1

framing NO-CRC4

pri-group timeslots 1-31

description QSIG connection to PBX 2

!

ip ssh version 1

!

interface Loopback0

ip address X.X.X.X 255.255.255.255

h323-gateway voip bind srcaddr X.X.X.X

!

interface Serial0/1/0:15

no ip address

encapsulation hdlc

isdn switch-type primary-qsig

isdn incoming-voice voice

no cdp enable

!

interface Serial0/1/1:15

no ip address

encapsulation hdlc

isdn switch-type primary-qsig

isdn incoming-voice voice

no cdp enable

!

voice-port 0/1/0:15

timeouts interdigit 2

description QSIG Port 1 to PBX1

!

voice-port 0/1/1:15

translation-profile incoming InboundE1

timeouts interdigit 2

description QSIG Port 2 to PBX2

!

dial-peer voice 700 voip

preference 5

destination-pattern 7....

progress_ind setup enable 3

modem passthrough nse codec g711ulaw

voice-class codec 1

session target ipv4:<CUCM1>

dtmf-relay h245-alphanumeric

fax rate disable

no vad

!

dial-peer voice 701 voip

preference 10

destination-pattern 7....

progress_ind setup enable 3

modem passthrough nse codec g711ulaw

voice-class codec 1

session target ipv4:<CUCM2>

dtmf-relay h245-alphanumeric

fax rate disable

no vad

!

dial-peer voice 999001 pots

description Calls to Tetra

destination-pattern 9102....

incoming called-number .

direct-inward-dial

port 0/1/0:15

forward-digits 7

!

dial-peer voice 99900 pots

description TETRA Group calls

destination-pattern 103....

incoming called-number .

direct-inward-dial

port 0/1/1:15

forward-digits all

!

dial-peer voice 999002 pots

description Calls to Tetra

destination-pattern 102....

incoming called-number .

direct-inward-dial

port 0/1/0:15

forward-digits all

!

gateway

timer receive-rtp 1200

!

call-manager-fallback

max-conferences 12 gain -6

transfer-system full-consult

ip source-address X.X.X.X port 2000

max-ephones 200

max-dn 200

!

end

[islam.kamal · Accepted solution]

Hello

Can you please please check the below points:-

1- Add the below config to your GW

voice class h323 1

  h225 timeout tcp establish 3

  h225 timeout setup 3

dial-peer voice 700 voip

voice class h323 1

dial-peer voice 701 voip

voice class h323 1

On CUCM  device -gateway

2-on CUCM :Check that your GW  have the CSS which can be reacheble by your phones.

3-on CUCM :gateway make significant digits "5" , as the below

Thank you

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[Ayodeji Okanlawon]

Lee,

Un allocated/un assigned number is certainly a CSS issue. Check the CSS on your gateway under inbound calls. Ensure that CSS has access to the partition of the phones

[Dennis Mink]

I would have also tested the translation rule and debug dialpeer .

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[Nishant Savalia]

Hi Lee,

You need to check few things for this:-

1). Collect "debug isdn q931" and share it.

2). Check whether call setup message is being sent to CUCM or not? (you can use debug voip dialpeer)

3). Check the dialed number is in any partition? If yes, then check appropriate CSS is applied at gateway level.

Regards,

Nishant Savalia

[Tyler Durden]

Hi Guy's

Many thanks for the helpful infomation, I have access to the system latter today and we post back in this thread latter!

I suspect it is a CSS Issue but will also add the config to the gateway, there has been some changes to the design so the GW needs to be cleaned up and some ISDN config removed.

One thing I would still like to find is the RTMT CallLogs! As mentioned before when I use CallManager v8.0 I can find the Calllogs via the following: Trace & Log Central / Real Time Trace / View Real Time Data

In CUCM v 7.0 I dont have this option or I cant find it?

Many thanks

Lee

[Ayodeji Okanlawon]

Lee,

That field is not available in CUCM7. I am not sure why you want it though. What you need is SDI trace log files. This is where you start your troubleshooting..

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

[Tyler Durden]

Hi Guy's

Many thanks for all the help with this, I'm working on a clossed network so hard to accress the internet so sorry for late reply!

I was able to get this working in about 30min's this morning, I changed the "CSS" on the "Inbound Routing Infomation - Inbound Calls".

Again, many thanks!

Lee