ANI Based Routing

juan.anzola · ‎04-26-2012

Hi Guys,

Consider the following scenario:

I have a CUCM Cluster with the Following Extensions:

2XXX: Sales

3XXX: Engineering

I have a single H.323 or SIP PSTN GW.

I have 2 different ITSP: 10.2.2.2 and 10.3.3.3

Right now, All calls are routing properly throug 10.2.2.2

I want calls from 2XXX to be routed out 10.2.2.2 and calls from 3XXX to be routed out 10.3.3.3

The requirement is to do this without modifying anything in CUCM.

Thanks in advance,

Gajanan Pande · ‎04-26-2012

Juan,

Not very sure if the " match calling number " type of dial-peer config can be done for internal numbers but the easiest way to do it using CUCM, I wonder why such restriction ( as mentioned in the last line of your requirement ).

I would simply give them different PSTN access code ( e.g. 9 & 8 etc ) & route them using 2 route patterns to 2 diff gateways.

GP.

Pls rate helpful posts !!

Yogesh Singh · ‎04-27-2012

You may Use COR ..+ Answer Address

Use answer- address 1234567 following example is for your cube ,

(1002) Phone ---- CUCM ---------- CUBE ===================== ITSP 1- Phone 2002)

(1003) Phone---- CUCM ---------- CUBE ===================== ITSP 2- Phone 2002)

Issue : When Ext 1002 makes outbound call, Call should go out through First

1-10 channels

and when Ext 1003 makes call, channels 1st ITSP should be used. 1002 should not

be able to use 2nd ITSP .

Following cmds needs to be issued on the Gateway :

!

dial-peer cor custom

name 1002

name 1003

!

dial-peer cor list 1002

member 1002

!

dial-peer cor list 1003

member 1003

!

dial-peer voice 2000 pots

DESC 1st ITSP

session Target Ipv4:X.X.X.X

corlist outgoing 1002

destination-pattern 2...

codec/voice/class codec

!

dial-peer voice 1002 voip

corlist incoming 1002

answer-address 1002--------------------------This we need to make more

specific according

to your dial-plans .

!

dial-peer voice 1003 voip

corlist incoming 1003

answer-address 1003--------------------------This we need to make more

specific according

to your dial-plans .

!

dial-peer voice 2222 pots

desc 2nd ITSP

session target ipv4:x.x.x.x

corlist outgoing 1003

codec / voice class codec

!

Hope this helps..

juan.anzola · ‎04-27-2012

Hi Yogesh,

Thank you for your time! My first concern, was that using Incoming Dial-Peers with "answer-address" will prohibit the use of a global "incoming called-number ." in any other Dial-Peer. I will have to use a more specific "incoming called-number" por incoming PSTN Calls. I will try this in my lab and let you know.

Thanks!

Yogesh Singh · ‎05-02-2012

Did it work ?

juan.anzola · ‎05-02-2012

Hi Yogesh,

Yes, it worked fine without the need of using COR lists. his was my Testing Scenario:

-------------------------------------------------------------------------------------------------------------------------------------------------------------

IP Phone with 2 Extension
- 5001 (DID: 2123945001)
- 5010 (DID: 2123945010)
PSTN Test Number: 911
CUCM <----H323-----> GW <----SIP-----> ITSP1(10.10.101.1)

SIP -----> ITSP2(10.10.102.1)

Calls from 5001 should route to ITSP1
Calls from 5010 should route to ITSP2

-------------------------------------------------------------------------------------------------------------------------------------------------------------

SOLUTION

-------------------------------------------------------------------------------------------------------------------------------------------------------------

Create 3 different Incoming Dial-Peers (For this to work, the "incoming called number" should be specific to the digits delivered by the ITSP's)
- Incoming Dial-Peer for ANI=5001
- Incoming Dial-Peer for ANI=5010
- Incoming Dial-Peer for DNIS=212394XXXX
Create 2 "voice translation profiles" for the following:
- Prefix "#" at the incoming call leg when ANI=5010
- Strip "#" at the outgoing call leg when DNIS=#911

-------------------------------------------------------------------------------------------------------------------------------------------------------------

CONFIG

-------------------------------------------------------------------------------------------------------------------------------------------------------------

voice translation-rule 1

rule 1 /#/ //

!

voice translation-rule 2

rule 1 /.*/ /#\0/

!

voice translation-profile STRIP#

translate called 1

!

voice translation-profile PREFIX#

translate called 2

!

dial-peer voice 100 voip

description CALLS-TO-CUCM

destination-pattern 2123945...

voice-class codec 1

voice-class h323 1

session target ipv4:10.10.210.11

dtmf-relay h245-alphanumeric

no vad

!

dial-peer voice 101 voip

description CALLS-TO-CUCM

preference 1

destination-pattern 2123945...

voice-class codec 1

voice-class h323 1

session target ipv4:10.10.210.10

dtmf-relay h245-alphanumeric

no vad

!

dial-peer voice 911 voip

description OUTGOING-LEG-ITSP1

destination-pattern 911

session protocol sipv2

session target ipv4:10.10.101.1

dtmf-relay rtp-nte

codec g711ulaw

!

dial-peer voice 9911 voip

description OUTGOING-LEG-ITSP2

translation-profile outgoing STRIP#

destination-pattern #911

session protocol sipv2

session target ipv4:10.10.102.1

dtmf-relay rtp-nte

codec g711ulaw

!

dial-peer voice 95001 voip

description INCOMING-LEG-ITSP1

answer-address 5001

voice-class codec 1

voice-class h323 1

dtmf-relay h245-alphanumeric

no vad

!

dial-peer voice 95010 voip

description INCOMING-LEG-ITSP2

translation-profile incoming PREFIX#

answer-address 5010

voice-class codec 1

voice-class h323 1

dtmf-relay h245-alphanumeric

no vad

!

dial-peer voice 999 voip

description INCOMING-LEG-FROM-PSTN

session protocol sipv2

incoming called-number 2123945...

codec g711ulaw

!

-------------------------------------------------------------------------------------------------------------------------------------------------------------

HTH

Yogesh Singh · ‎05-02-2012

Great ! Nice alternative :-)

Thanks

Regards

Yogesh