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]

Hi Aok,

It just because of your help its working. Next step is to test outbound calls. Many thanks once again

[sarwarm123]

Hi Aok,

Now I am having an issue with digit translation, basically I want to translated DNIS 02476012665 to internal 5 digit VOIP number 88000. I tried num-ex command but thats not working. I also tried to apply voice translation rule but getting Called Number not translated

Called Number=02476012665(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,

I think I am doing worng translation, can you please help me in this. Also do I need to add outgoing dial-peer to CUCM for 5 digit VOIP numbers?

[Ayodeji Okanlawon · Accepted solution]

Hi,

Try this,

voice translation-rule 1

rule 1 /^02476012665/ /88000/

voice translation-profile CLI

translate called 1

then apply it to the dial-peer

dial-peer voice 20 voip

translation-profile outgoing CLI

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

[sarwarm123]

Brilliant, you are a star mate that works, I need to read more about voice translation. I was applying rules as incoming

I also tried to make outbound call which works only when Router Pattern is

0[1-7]XXXXXXXXXX and outgoing dial peer is matching destination-pattern 0[1-7].........

But I want to make call with 9 for example 91785830000 I changed Route Pattern  9[1-7]XXXXXXXXXX and configured Discard Digit None with following dial peers. But it fails I think I also need to apply 9 DIGIT Strip translation rule.

I have also tried this with destination-pattern 9T to strip digit without using translation rule but that doesn’t work either. Need your help please

dial-peer voice 2 voip

description *** Outbound calls to ITSP ***

destination-pattern 90[1-7].........

session protocol sipv2

session target sip-server

dtmf-relay rtp-nte

codec g711alaw

[Ayodeji Okanlawon]

Yes you need to strip the 9. because your provider does not want the 9

You can do that from CUCM..

configure 9.0[1-7]XXXXXXXXXX and the do discard digits predot.

You then need to ensure your destination-pattern in the gateway does not have 9 inform of it because cucm will send the calls without 9

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

[sarwarm123]

Hi Aok,

I tried that way 9.0[1-7]XXXXXXXXXX and the do discard digits predot but some reason this does not discard 9 and send it sends all digits with 9

From: <86860>;tag=2653607~867df168-c5c9-4b90-8b3a-75c7400ce77b-44046791

To: <901785830000>;tag=876B708-125D

However if I configure 91.0[1-7]XXXXXXXXXXX it or any other digit with apart from 9 it strip predot for example 77.0[1-7]XXXXXXXXXXX. Is that any BUG??? Can we strip digit through translation profile?

Route Pattern with 91.0[1-7]XXXXXXXXXXX

INVITE sip:01785830000@10.116.143.151:5060 SIP/2.0

Via: SIP/2.0/TCP 10.102.243.13:5060;branch=z9hG4bK8176776429784

From: <>;tag=2653934~867df168-c5c9-4b90-8b3a-75c7400ce77b-44047317

To: <01785830000>

Date: Wed, 27 Mar 2013 12:58:57 GMT

Route Pattern with 9.0[1-7]XXXXXXXXXX

INVITE sip:901785830000@10.116.143.151:5060 SIP/2.0

Via: SIP/2.0/TCP 10.102.243.13:5060;branch=z9hG4bK817f21e48fd6a

From: <86860>;tag=2654256~867df168-c5c9-4b90-8b3a-75c7400ce77b-44047797

To: <901785830000>

Date: Wed, 27 Mar 2013 13:03:01 GMT

[Ayodeji Okanlawon]

I want the CUBE config. I dont htink there is a problem with CUCM.

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[Ayodeji Okanlawon]

UPDATED:

OK,

Thats strange..

Lets do this..

voice translation-rule 100

rule 1 /^9/ //

translation-profile STRIP9

translate called 100

dial-peer voice 101 voip

translation-profile incoming STRIP9

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

[sarwarm123]

Hi Aok,

I tried this but still its not stripping the 9 number.

RoutePattern 9.0[1-7]XXXXXXXXXX and the Nodigits discard predot

I have attached CUBE config and CCSIP debug All logs

[Ayodeji Okanlawon · Accepted solution]

I updated the post to say that you should apply it to dial-peer 101

dial-peer voice 101 voip

translation-profile incoming STRIP9

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

[sarwarm123]

Yep that works Can you please suggest me docuementation about voice translation to understand how it works.

One Last thing currently ANIS is showing numbers starting with +4407XXXXXXXX (sip:+447967999481) which I want to replace with 9XXXXXXXXXX. I mean nation type, internation type.

Can you please help me what translation rule I will use for this and on which dial-peer it will apply?

[Ayodeji Okanlawon · Accepted solution]

Hi you can add this to the existing voic transaltion-rule 1

voice translation-rule 1

rule 2 /^+44\(...........\)/ /9\1/  ---------NB th enumber of dots should = to the number of digits after +44

voice translation-profile CLI

translate calling 1

You can read more about translation rules here

http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a0080325e8e.shtml

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

[sarwarm123]

Hi Aok,

I am testing SIP calls, most of the tests are ok apart from two following issues

1) CLI display sometime and somtime doesn't

2) The time CLI display on the called handset (mobile phone) if I try to reject call from the called handset it disconnects on called handset (mobile phone) butI can hear VOIP handset still ringing which ends after couple of minutes.

I have treid to fix CLI by adding SIP profies but it doesn't work on the outgoind dial-peer to ITSP

voice class sip-profiles 1
request INVITE sip-header P-Asserted-Identity add "P-Asserted Identity:<02476012665>"

[Ayodeji Okanlawon]

Muhammad,

1. CLI issues..What is the requirement? Do you need CLI displayed on all calls?

In what direction do you have CLI issues? Is it inbound calls or outbound calls

You need to do some test call and send  "debug ccsip messages" for scenarios where CLI is not displayed..I need to see what you are sending to your ITSP if it is affecting outbound calls

2. The second question is a know behaviour of SIP trunks. When you reject a call, it rings for a while on the phone before it terminates. We may be able to work on it..You can also do a test call and send "debug ccsip messages"

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

[sarwarm123]

Hi Aok

• •1) There is no CLI issues for inbound calls. Issue is for only outbound calls which display sometime on called number and sometime doesn’t. Don’t know is that ITSP issue or any configuration issue.
  

If outbound CLI display issue resolves then I want to display outbound CLI from specific VOIP numbers and block outgoing CLI for specific numbers.

• •2) Call termination behaviour is quite weird, if outbound CLI display then call termination does not complete. However the time outbound CLI doesn’t display it terminates straightway.
  

Please see ccsip messages debug