CUBE configuration with CUCM Call Manager

[sarwarm123]

Our company recently decided to have a SIP trunk in our Cisco VOIP environment. I have not configured CUBE router before so I have read number of documents related to CUBE configuration with CUCM. However I still have confusion so I need your help. Please see proposed network diagram below.

My Question is, do we connect one gig port to company LAN where CUCM is connected and second gig interface will connect to SIP Provider customer end router? For example

interface GigabitEthernet0/0

description Connection to LAN

ip address 172.102.243.10 255.255.255.0

duplex auto

speed auto

!

interface GigabitEthernet0/1

description connection to SIP Provider

ip address 99.102.153.24 255.255.255.0

duplex auto

speed auto

I have seen some other sample configurations where people configured only one gig port for example (I have also attached full sample configuration)

interface GigabitEthernet0/0

description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$

no ip address

shutdown

duplex auto

speed auto

service-policy output VOIP-Policy

!

interface GigabitEthernet0/1

description $ES_LAN$

ip address 146.191.201.41 255.255.255.0

duplex auto

speed auto

h323-gateway voip interface

h323-gateway voip bind srcaddr 146.191.201.41

service-policy output VOIP-Policy

[sarwarm123]

Aok,

10.160.132.98 is CUBE WAN ip address. Actually they given us wrong ip addresses intially. I am going to do test as you said and reply you shortly

[sarwarm123]

Aok,

As you said I have attached show log file after configured

logging buffered 10000000

no logging rate-limit

no logging monitor

[Ayodeji Okanlawon]

Ok..I can see whts going wrong..

Can you change the codec on the dial-peer to cucm, 20, 21, to use codec g711alaw

dial-peer voice 20 voip

codec g711alaw

!

dial-peer voice 21 voip

codec g711alaw

Test again

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[sarwarm123]

Aok,

As you said I have changed codec but still no luck. I have attached updated CUBE config and Log file

[Ayodeji Okanlawon]

OK,

S now we see the call is sent to CUCM..But CUCM does not respond to the INVITE. The call was sent to .12 and then to .13, but we do not get any response from both CUCM servers.

Sent:

INVITE sip:02476012665@10.102.243.12:5060 SIP/2.0

Via: SIP/2.0/UDP 10.116.143.151:5060;branch=z9hG4bK71631

From: <>;tag=4BED078-10A5

Sent:
INVITE sip:02476012665@10.102.243.13:5060 SIP/2.0
Via: SIP/2.0/UDP 10.116.143.151:5060;branch=z9hG4bK820BB
From: <>;tag=4BEDE2C-1DCB

We need to look at CUCM logs and see whats happening there..WHy are we not getting any response from CUCM.

Can you enable CUCM logs, set it to detailed and send the logs to me?

UPDATED: Do you have the CUCM group with both servers assigned to the device pool of the sip trunk?

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[sarwarm123]

Thats correct, please see the screenshots attached. Also, could you please guide me how can I collect traces of CUCM

[Ayodeji Okanlawon]

Muhammad,

Use this link to know how to setup and collct the traces

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a0080094e89.shtml

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[sarwarm123]

Hi Aok,

What traces do you want me to collect, for example Cisco Call Manager etc

[Ayodeji Okanlawon]

Sorry, I want Callmanager traces (only the SDI logs)

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[sarwarm123]

Aok,

Please see attached SDI logs

[Ayodeji Okanlawon]

Muhammad,

when you collected the logs did you do another test call. I cant find the trace for the call. I assumed you didnt set traces to detailed initially. Can you do another test call and send me the traces again..Please include the calling time..Ie let me know what time you did the test call

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[sarwarm123]

Hi Aok,

Yes I made four to five test calls and time was in between 21:44 to 21:46. Unfortunately I also could not find called and calling number in trace file. Is that because SIP trunk to CUBE has some issue?

CallFlow is

ITSP--->ASA----->CUBE---->CUCM

I dont think its ASA blocking anything as I was already filtering the traffice and dont see any deny messages. Do you want to try once again?

[Ayodeji Okanlawon]

That means the INVITE is not getting to CUCM, thats why we are not getting any response to the INVITE from CUCM.

You can do a test call again, before you send the logs check that the calling or called number is present in the trace. If it is not then we have issues some where...

Have you tested an outbound call over the SIP trunk? Maybe we should try that and then send the traces so we see if CUCM sends outbound INVITE to CUBE

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[sarwarm123]

Hi Aok,

Good news its working, that was ASA blocking. I really appreciate your help, you done a great job Many thanks. Incoming calls are working, now I need to test outbound calls. Hopefully this part will go smoothly but if encountered any issue will ask your help.

As I mentioned above I also need help in SIP normalization

Basically I need to send 10 digits to ITSP excluding 9 and 0. ITSP also wanted to add +44

ex

+447967999481

Tomorrow I will test outbound calls please see my outboud dial-peer is that ok?

dial-peer voice 22 voip

description *** Outbound Calls to Backup CUCM***

destination-pattern 0[2-9].........

session protocol sipv2

session target ipv4:192.168.132.3

dtmf-relay rtp-nte

codec g711alaw

Can do that way?

dial-peer voice 22 voip

description *** Outbound Calls to Backup CUCM***

destination-pattern 9T

session protocol sipv2

session target ipv4:192.168.132.3

dtmf-relay rtp-nte

codec g711alaw

[Ayodeji Okanlawon · Accepted solution]

Glad I could help. You can just use voice translation rules to strip the 9. The destination pattern 9T should be pointing to your ITSP not CUCM..as you have on dial-peer 2. You can use 9T on dial-peer 2 also

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