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inbound call routing

zrunner626
Level 1
Level 1

Can anyone point me to a good document on inbound call routing?

Also, if you have any other thoughts on the scenerio below, would be appreciated.

 

We are adding another number to our circuit and I'm trying to figure out how to get it to route.

 

Today we receive 10 digits from the telco:

dial-peer voice 100 pots
 incoming called-number .
 direct-inward-dial

 

Translate them to 3 or 4 digits

voice translation-rule 1  (bogus numbers used here)
 rule 1 /^1234567\(...\)$/ /\1/
 rule 2 /^2234567\(...\)$/ /\1/
 rule 3 /^3234567\(...\)$/ /\1/
 rule 4 /^4234567\(...\)$/ /\1/
 rule 5 /^5234567\(...\)$/ /\1/
 rule 6 /^6123456\(5...\)$/ /\1/
 rule 7 /^712345\(9...\)$/ /\1/
 rule 8 /^723456\(4...\)$/ /\1/
 rule 9 /^733456\(6...\)$/ /\1/
 rule 10 /^743456\(...\)$/ /\1/
 rule 11 /^753456\(1...\)$/ /\1/
!
voice translation-profile DID
translate called 1

 

Send them to Call Manager

dial-peer voice 1113 voip
 preference 2
 destination-pattern [0-9]...$
 session target ipv4:10.X.14.11
 incoming called-number 9T
 voice-class codec 1  
 voice-class h323 1
 dtmf-relay h245-alphanumeric h245-signal
 fax-relay ecm disable
 fax rate 14400
 ip qos dscp cs3 signaling
 no vad
dial-peer voice 11131 voip
 preference 10
 destination-pattern [0-9]..$
 session target ipv4:10.X.14.10
 incoming called-number 9T
 voice-class codec 1  
 voice-class h323 1
 dtmf-relay h245-alphanumeric h245-signal
 fax-relay ecm disable
 fax rate 14400
 ip qos dscp cs3 signaling
 no vad
dial-peer voice 11132 voip
 preference 1
 destination-pattern [0-9]..$
 session target ipv4:10.X.25.11
 incoming called-number 9T
 voice-class codec 1  
 voice-class h323 1
 dtmf-relay h245-alphanumeric h245-signal
 fax-relay ecm disable
 fax rate 14400
 ip qos dscp cs3 signaling
 no vad
dial-peer voice 111322 voip
 tone ringback alert-no-PI
 preference 1
 destination-pattern [0-9]...$
 progress_ind setup enable 3
 progress_ind alert enable 8
 progress_ind connect enable 8
 session target ipv4:10.X.25.11
 incoming called-number 9T
 voice-class codec 1  
 voice-class h323 1
 dtmf-relay h245-signal h245-alphanumeric
 ip qos dscp cs3 signaling
 no vad
dial-peer voice 111311 voip
 tone ringback alert-no-PI
 preference 5
 destination-pattern [0-9]...$
 progress_ind setup enable 3
 progress_ind alert enable 8
 progress_ind connect enable 8
 session target ipv4:10.X.14.10
 incoming called-number 9T
 voice-class codec 1  
 voice-class h323 1
 dtmf-relay h245-signal h245-alphanumeric
 ip qos dscp cs3 signaling
 no vad
dial-peer voice 11133 voip
 preference 2
 destination-pattern [0-9]..$
 session target ipv4:10.X.14.11
 incoming called-number 9T
 voice-class codec 1  
 voice-class h323 1
 dtmf-relay h245-signal h245-alphanumeric
 ip qos dscp cs3 signaling
 no vad

 

CUCM then has a translation pattern

21xxx that points to endpoint or

219441 (and others such as)

 

Since the new number had overlapping last 4 digits, I tried adding a translation rule for 6 digits

 rule 12 /^4235\(......\)$/ /\1/

 

adding an additional dial-peer to CUCM

dial-peer voice 11134 voip
 preference 10
 destination-pattern [0-9].....$
 session target ipv4:10.X.25.11
 incoming called-number 9T
 voice-class codec 1  
 voice-class h323 1
 dtmf-relay h245-alphanumeric h245-signal
 ip qos dscp cs3 signaling
 no vad

 

And a translation pattern in CUCM for the last 6 digits

123456

 

However I get Cause i = 0x8081 - Unallocated/unassigned number

4 Replies 4

R0g22
Cisco Employee
Cisco Employee
Your translations are stripping down the number to 4 digits. You have VoIP dial-peers that match on 3 or 4 digits and then route to the destination, but where do you have the translation profile applied ?

voice-port 1/0:23
 translation-profile incoming DID
!         
voice-port 1/1:23
 translation-profile incoming DID

 

The 3 and 4 digit translations are not the problem. They route fine. it's when I tried adding the new number with the 6 digit translation. I posted those at the end of my first post.

It should be because you are overlapping your dial plan. Keep it consistent with 4 digits.
Also, send a log with "debug isdn q931" , "debug voip translation" and "debug voice cc inout" enabled for this failed call.

I can send those debugs as I was testing with the telco previously.

 

Ok, Guess that is going to be a problem since the 4 digit conflicts with another number. -Thanks