cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
2253
Views
25
Helpful
27
Replies
Beginner

CUCM troubleshooting, can i tell if CUCM is receiving a call?

Hello,

I'm trying to better understand and troubleshoot my CUCM environment. I have CUCM 11.5, a CUBE router and an ITSP SIP provider as my connection to the PSTN. 

I have outbound calls working, internal numbers can call and voice calls work fine, however I cannot get incoming calls to work. 

I believe my CUBE router is forwarding calls to CUCM because when I do debug voice ccapi inout the output shows the correct inbound and outbound dialpeers being matched.

So I believe my cube is telling me it is forwarding the calls correctly but when I call from my mobile (cell) phone inwards the phone rings and eventually goes to the the SIP providers voicemail.

Are there any commands I can run on the CUCM to troubleshoot this? To see if CUCM is receiving the call? And then if it is receiving the call, why its not connecting it?

Many Thanks,

Tom

Everyone's tags (4)
1 ACCEPTED SOLUTION

Accepted Solutions
Rising star

Great it worked out ,

Great it worked out , Outbound proxy command is used to route  SIP Messages to the outbound proxy . You can even use outbound proxy on individual dial-peers.

Please rate if found applicable.

Thanks

Haris

27 REPLIES 27
Highlighted
Cisco Employee

you need to pull the CCM

you need to pull the CCM traces in order to verify this ..other than you can take debugs on CUBE router where you can have pretty much good idea about the call issue however for detail you need to see the CCM traces in order to find out actual root cause.

if there is SIP connection between CUBE and CUCM

debug ccsip message

debug ccsip error

debug ip tcp transcation

debug voice ccapi inout

Br, Nadeem Please rate all useful post.
VIP Advisor

Hi,

Hi,

From CUCM, try using RTMT > Session Trace > Real- Time and see if you can find your incoming call or not. You can also run real time trace from CLI but this will be complex due to amount of messages.

From CUBE using debug ccsip messages command will help you to judge if CUCM received the message or not. In case CUCM receives the message from CUBE will send a response. If you see a response from CUCM then it received the message from CUBE else not.

Rising star

Please attach ccsip debug

Please attach ccsip debug output from CUBE Router, we can find out reason why call got disconnected from it, which will set us in direction for troubleshooting  the issue.

Beginner

Hi,

Hi,

Many thanks for the offer, that'd be great. I'm not getting very far at the moment.

Which debug output would be most helpful? messages, calls or just all?

Thanks,

Tom

Beginner

Hi,

Hi,

Attached is the show ccsip messages command. I've sanitised it, removing the calling mobile phone number and replacing it with 07000111222 and also the SIP number which I've renamed to 01000111222.

my setup is:

CUCM                -> CUBE                 -> ITSP

172.16.127.70    -> 172.16.127.15    -> phone.tel2.co.uk

Any insight you can provide would be very useful.

Thanks,

Tom

I am seeing in debug output,

I am seeing in debug output, In response to invite from CUBE cucm is sending SIP/2.0 401 Unauthorized back.

Looks like you have

Or you have SIP realm configured.
Please check both.

*Aug 22 19:52:54.915: //54/D9EF5ACA8037/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.16.127.15:5060;branch=z9hG4bKE15D8;received=172.16.127.15;rport=50331
From: "07000111222" <sip:07000111222@phone.tel2.co.uk>;tag=229F0C-688
To: <sip:201@172.16.127.70:5060>;tag=as554ed9c5
Call-ID: D9F1CC03-67D811E6-803D890E-8347BEBD@172.16.127.15
CSeq: 101 INVITE
Server: 2talk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="voip", nonce="3240a9f4"
Content-Length: 0


Timestamp: 3554740374915
UTC Timestamp:3554740374915
Source Filename: rbt13B8.tmp

Beginner

Hi Moshfiqur,

Hi Moshfiqur,

Many Thanks for your response.

That is interesting, because I assumed this message didn't matter. Currently the CUBE is setup with a SIP-UA with the relevant details for the SIP trunk provider so the CUBE router authenticates with the ITSP provider and it can place calls.

The number 201 is an internal number which I'm trying to get all external calls forwarded to. I assumed the 201 number wouldn't need to be configured to authenticate with the service provider because that's what the CUBE was doing?

I currently don't have any message digest setup on the CUCM (that i know of :) ) and the realm configured is between the CUBE and the ITSP as part of their setup instructions. 

Should I need to configure message digest on CUCM to authenticate the SIP provider? Or should I need to configure the realm on CUCM? I assumed the CUBE handled both of these elements.

Many Thanks,

Tom

Frequent Contributor

A CUBE has two parts: The SIP

A CUBE has two parts: The SIP connection to the provider and the SIP connection to the CUCM. BOTH connections have to be configured correctly.

For example when an IP phone makes a call, the CUCM sends an INVITE to the CUBE, the CUBE (depending on the CUBE configuration) can modify this INVITE and then sends an INVITE to the provider. Normally you must do some number normalization, most providers (if not all) want to see E164 numbers.

The same when you receive a call from the provider, the CUBE receives an INVITE from the provider and the CUBE sends an INVITE to the CUCM

One of your call legs might not be correctly configured (number, codec, etc)

Jan

Beginner

Hi,

Hi,

thinking about it I dont have a dial peer from the CUM to my cube, I assume that means it's using the dial peer 0? would this cause an issue?

dial-peer voice 1000 voip
 description CUBE-TO-CUCM
 destination-pattern 2..
 session protocol sipv2
 session target ipv4:172.16.127.70
 codec g711ulaw
 no vad
!
dial-peer voice 2 voip
 description CUBE-TO-ITSP
 destination-pattern 0..........
 session protocol sipv2
 session target dns:phone.tel2.co.uk
 voice-class sip outbound-proxy dns:phone.tel2.co.uk:5060
 dtmf-relay cisco-rtp h245-alphanumeric h245-signal rtp-nte sip-kpml sip-notify
 codec g711ulaw
 no vad
!
dial-peer voice 100 voip
 description ITSP-TO-CUBE
 translation-profile incoming INCOMING
 session protocol sipv2
 incoming called-number 441252493259
 voice-class sip dtmf-relay force rtp-nte
 no voice-class sip early-offer forced
 codec g711ulaw
 no vad
!

Could this be the issue?

Do I need a dial peer with incoming called-number 2.. ?

what would be the destination pattern? or would I not need a destination pattern? I'll try this now.

Many Thanks,

Tom

Beginner

Hi,

Hi,

Just a quick update. I've just tried adding this dial peer and it hasn't seemed to make a difference:


!
dial-peer voice 50 voip
 description CUCM-TO-CUBE
 session protocol sipv2
 session target sip-server
 incoming called-number 2..
 codec g711ulaw
 no vad

Regards,

Tom

Rising star

Double check is the End

Double check is the End Device 201 also in the Location Hub None along with the SIP Trunk.

Have you configured any Location Bandwidth for Intra Location settings.

Thanks

Haris

Beginner

Hi,

Hi,

The 201 jabber client is also in location hub_none, see attached.

Regarding location bandwidth, I've configured region bandwidth but I don't think I've configured location bandwidth. I'm looking now, where is this configured?

Thanks,

Tom

Beginner

Hi,

Hi,

I've just noticed the location in my device pool was set to "none" rather than hub_none. I've now changed it, it hasn't appeared to have made a difference but I just wanted to update you.

I assume the device pool location should also be set to hub_none?

Thanks,

Tom

Rising star

Just add incoming dial-peer

Just add incoming dial-peer with number which you will dial out.

Also have you implemented any CAC Method on router as per the document 

https://supportforums.cisco.com/document/71326/call-admission-control-cac-implementation-cube

CreatePlease to create content
Content for Community-Ad
July's Community Spotlight Awards