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Cube - Redirecting calls to ITPS based on Calling number

gfolens
Level 4
Level 4

‎10-11-2021 07:08 AM

I need to configure Cube on a ISR that is also using ISDN PRI to route calls to PSTN.

I would like to route calls of specific DN's (calling number) via the new SIP trunk of the ITSP.

I have these existing dial-peers:

 

! Incoming DP from CUCM to ISR:

dial-peer voice 1 voip
preference 2
session protocol sipv2
session transport tcp
incoming called-number .T
dtmf-relay rtp-nte
no vad

 

! Outgoing DP from ISR to PSTN via E1

dial-peer voice 200 pots
preference 1
destination-pattern 0T
progress_ind alert enable 8
progress_ind progress enable 8
progress_ind connect enable 8
port 0/1/0:15

 

I've created new dial-peers for calls to the ITSP:

! Incoming DP from CUCM to ISR:

dial-peer voice 1000 voip
preference 1
session protocol sipv2
session transport tcp
destination dpg 4000 ! linked to DP 4000
incoming called-number .T
incoming calling e164-pattern-map 2000  ! contains list calling numbers to use ITSP
voice-class codec 1 offer-all
dtmf-relay rtp-nte
no vad

 

! Ougoing DP from ISR to ITSP:

dial-peer voice 4000 voip
translation-profile outgoing E164-Orange
preference 3
destination-pattern .T
session protocol sipv2
session transport udp
session server-group 2000
voice-class codec 2
voice-class sip profiles 2001
voice-class sip tenant 2000
dtmf-relay rtp-nte

 

But still the outgoing DP 200 is selected for all calls from CUCM.

In the debug dial-peer I see DP 1000 is selected first but then DP 1 is selected based on DNIS...

I've also tried to but incoming called-number .T on DP 4000 and adjust preference to 1 but no change in behaviour.

Any advise? 

Labels:
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4 Replies 4

Steven L
Spotlight
Spotlight

‎10-11-2021 07:34 AM

remove the incoming called-number line on DP 1000

 

 

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gfolens
Level 4
Level 4
In response to Steven L

‎10-11-2021 07:42 AM

Already tried but the behaviour remains the same. Incoming called-number (DNIS) on DP1 has highest priority.

0 Helpful

Steven L
Spotlight
Spotlight
In response to gfolens

‎10-11-2021 07:47 AM

it needs to be removed. dial peers should only have one matching rule for incoming and one for outgoing. i would even separate further and say incoming and outgoing should never be on the same peer.

 

please remove and run:

 

debug voip ccapi inout

debug voip dial-peer

 

 

post those results.

0 Helpful

Roger Kallberg
VIP
VIP

‎10-11-2021 08:31 AM - edited ‎10-11-2021 09:34 AM

Try with this

voice class e164-pattern-map 1
 description ** Inbound calls to all Number patterns **
  e164 +<number range 1>
  e164 +<number range 3>
  e164 +<number range 2>
  e164 +<number range 4>
 !
!
voice class e164-pattern-map 2
 description ** Outbound calls from SIP Number patterns **
  e164 +<number range 1>
!
voice class e164-pattern-map 3
 description ** Outbound calls from ISDN Number patterns **
  e164 +<number range 3>
  e164 +<number range 2>
  e164 +<number range 4>
  e164 .
!
voice class e164-pattern-map 2000
 description E164 Pattern Map for called number to SIP
  e164 0T 
 !
!

!
voice class dpg 1
 dial-peer 210
!
voice class dpg 2
 dial-peer 110
!
dial-peer voice 1000 voip
 description Outbound calls from CUCM outbound to ISDN
 translation-profile incoming NOPLUS-IN
 modem passthrough nse codec g711ulaw
 session protocol sipv2
 destination dpg 2
 incoming calling e164-pattern-map 3
 voice-class codec 1  
 voice-class sip bind control source-interface GigabitEthernet0/0/0
 voice-class sip bind media source-interface GigabitEthernet0/0/0
 dtmf-relay rtp-nte sip-kpml
 fax-relay ecm disable
 no fax-relay sg3-to-g3
 fax rate 14400
 fax nsf 000000
 fax protocol t38 version 0 ls-redundancy 2 hs-redundancy 1 fallback pass-through g711ulaw
 no vad
!
dial-peer voice 1001 voip
 description Outbound calls from CUCM to CUBE outbound SIP
 translation-profile incoming NOPLUS-IN
 modem passthrough nse codec g711ulaw
 session protocol sipv2
 destination dpg 1
 incoming calling e164-pattern-map 2
 voice-class codec 1  
 voice-class sip bind control source-interface GigabitEthernet0/0/0
 voice-class sip bind media source-interface GigabitEthernet0/0/0
 dtmf-relay rtp-nte sip-kpml
 fax-relay ecm disable
 no fax-relay sg3-to-g3
 fax rate 14400
 fax nsf 000000
 fax protocol t38 version 0 ls-redundancy 2 hs-redundancy 1 fallback pass-through g711ulaw
 no vad
!
dial-peer voice 1010 voip
 description Inbound calls to CUCM subscribers
 modem passthrough nse codec g711ulaw
 session protocol sipv2
 session server-group 1
 destination e164-pattern-map 1
 voice-class codec 1  
 voice-class sip options-keepalive profile 1
 voice-class sip bind control source-interface GigabitEthernet0/0/0
 voice-class sip bind media source-interface GigabitEthernet0/0/0
 dtmf-relay rtp-nte sip-kpml
 no fax-relay sg3-to-g3
 fax rate 14400
 fax nsf 000000
 fax protocol t38 version 0 ls-redundancy 2 hs-redundancy 1 fallback pass-through g711ulaw
 no vad
!
dial-peer voice 100 pots
 tone ringback alert-no-PI
 description Inbound calls from PSTN
 incoming called-number .
 direct-inward-dial
!
dial-peer voice 110 pots
 trunkgroup PRI_PSTN 1
 tone ringback alert-no-PI
 description Outbound calls to PSTN
 destination-pattern 0T
 progress_ind setup enable 3
 progress_ind alert enable 8
 progress_ind progress enable 8
 progress_ind connect enable 8
 progress_ind disconnect enable 8
 no digit-strip
 no sip-register
!
dial-peer voice 200 voip
 description Inbound calls from PSTN - SIP
 redirect ip2ip
 session protocol sipv2
 incoming uri via PSTN
 voice-class codec 10  
 voice-class sip early-offer forced
 voice-class sip bind control source-interface GigabitEthernet0/0/3
 voice-class sip bind media source-interface GigabitEthernet0/0/3
 dtmf-relay sip-kpml rtp-nte sip-notify
 fax-relay sg3-to-g3
 fax rate 9600
 fax nsf 000000
 fax protocol t38 version 0 ls-redundancy 2 hs-redundancy 1 fallback pass-through g711ulaw
 no vad   
!
dial-peer voice 210 voip
 description Outbound calls to PSTN - SIP
 translation-profile outgoing PSTN-OUT-SIP
 session protocol sipv2
 session server-group 2000
 destination e164-pattern-map 2000
 voice-class codec 10  
 voice-class sip profiles 10
 voice-class sip options-keepalive profile 2000
 voice-class sip bind control source-interface GigabitEthernet0/0/3
 voice-class sip bind media source-interface GigabitEthernet0/0/3
 voice-class sip audio forced
 dtmf-relay sip-kpml rtp-nte sip-notify
 fax-relay sg3-to-g3
 fax rate 9600
 fax nsf 000000
 fax protocol t38 version 0 ls-redundancy 2 hs-redundancy 1 fallback pass-through g711ulaw
 no vad
!

We’re doing the very same thing as of a few weeks as we are in the process of migration from ISDN based PSTN to SIP at one of our US based locations. 



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