SIP Dial Peer Mismatch

Hello,

Could someone help resolve issue where our SIP dial peer is ignored/mismatched and explain how to fix it without detriment to the existing call routing???
 
THE SETUP

We have an existing UK office collab setup which consists of:
 
CUCM (9.1) <-> ISR2921 (15.2(4)M4) <-> PSTN (ISDN30)
 
Currently all calls are routed from the CUCM to H323 gateway then out to PSTN.
We are in the process of testing SIP as a possible way forward (to replace PRI connection at some point).
 
I have setup everything in terms of SIP there is to be configured (as far as I can tell).
I have also configured route pattern on Call Manager to test calls to my mobile.
 
Calls come from CUCM to h323 gateway, get matched by multiple dial peers but in the end are send out the old PSTN connection. I am attaching debug dialpeer result for my test call -> test_call_debug.txt for anyone interested enough to take a look.
 
Here is my voip/sip config:

voice service voip
 ip address trusted list
  ipv4 172.0.0.0
  ipv4 46.165.252.40
  ipv4 5.150.254.205
  ipv4 72.251.241.166
  ipv4 83.222.249.39
  ipv4 54.172.60.1
  ipv4 54.172.60.0
  ipv4 54.172.60.2
  ipv4 54.172.60.3
  ipv4 35.156.191.128
  ipv4 35.156.191.129
  ipv4 35.156.191.130
  ipv4 35.156.191.131
 address-hiding
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 h323
 sip
  bind control source-interface GigabitEthernet0/0
  bind media source-interface GigabitEthernet0/0
  registrar server expires max 3600 min 600
  early-offer forced
!
voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g711alaw
 codec preference 3 g729r8
!
voice class h323 1
  h225 timeout tcp establish 3
  call preserve 
!
!
!
!
voice translation-rule 1
 rule 1 /^\(.*\)/ /9\1/ type subscriber subscriber
 rule 2 /^\(.*\)/ /90\1/ type national national
 rule 3 /^\(.*\)/ /900\1/ type international international
!
voice translation-rule 2
 rule 1 /^4236/ /6/
!
voice translation-rule 3
!
voice translation-rule 4
!
!
voice translation-profile BT-ISDN30-IN
 translate calling 1
 translate called 2
!
voice translation-profile BT-ISDN30-OUT
 translate calling 3
 translate called 4

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DIAL PEER CONFIG

dial-peer voice 1 pots
 description *** Calls to ISDN-30 Circuit ***
 preference 2
 destination-pattern 9T
 progress_ind alert enable 8
 progress_ind progress enable 8
 progress_ind connect enable 8
 port 0/0/0:15
!
dial-peer voice 2 pots
 description ** Calls from ISDN-30 Circuit ***
 incoming called-number .
 direct-inward-dial
 port 0/0/0:15
!
dial-peer voice 10 voip
 description *** Calls to CUCM Publisher ***
 preference 2
 destination-pattern [1-8]..
 progress_ind setup enable 3
 session target ipv4:172.21.100.11
 voice-class codec 1  
 voice-class h323 1
 dtmf-relay h245-alphanumeric
 ip qos dscp cs4 media
 ip qos dscp cs4 signaling
 no vad
!
dial-peer voice 11 voip
 incoming called-number 9T
 voice-class codec 1  
!
dial-peer voice 110 pots
 description *** FXS 0/1/0 ***
 preference 1
 destination-pattern 650
 progress_ind setup enable 3
 progress_ind alert enable 8
 progress_ind progress enable 8
 progress_ind connect enable 8
 port 0/1/0
!
dial-peer voice 111 pots
 description *** FXS 0/1/1  ***
 preference 1
 destination-pattern 654
 progress_ind setup enable 3
 progress_ind alert enable 8
 progress_ind progress enable 8
 progress_ind connect enable 8
 port 0/1/1
!
dial-peer voice 112 pots
 description *** FXS 0/1/2  ***
 preference 1
 destination-pattern 661
 progress_ind setup enable 3
 progress_ind alert enable 8
 progress_ind progress enable 8
 progress_ind connect enable 8
 port 0/1/2
!
dial-peer voice 113 pots
 description *** FXS 0/1/3  ***
 preference 1
 destination-pattern 668
 progress_ind setup enable 3
 progress_ind alert enable 8
 progress_ind progress enable 8
 progress_ind connect enable 8
 port 0/1/3
!
dial-peer voice 200 voip
 destination-pattern 907895438321
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte sip-kpml sip-notify
 codec g711ulaw
 no vad
!
!
sip-ua    
 authentication username uk-pstn-01 password 7 XYZ
 registrar dns:ourdomain.pstn.twilio.com expires 3600
 sip-server dns:ourdomain.pstn.twilio.com

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Thanks
Tom
StratAnAsked:
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buckethead34Commented:
Can  you post output from debug ccsip messages
0
StratAnAuthor Commented:
We figured out the issue was caused by incorrect number formatting. Twilio requires E164.

Once I have applied translation patterns I managed to make outgoing calls.

Our final config for those interested:
voice service voip
 ip address trusted list
   ipv4 54.172.60.1
   ipv4 54.172.60.0
   ipv4 54.172.60.2
   ipv4 54.172.60.3
   ipv4 35.156.191.128
   ipv4 35.156.191.129
   ipv4 35.156.191.130
   ipv4 35.156.191.131
address-hiding
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
h323
sip
  bind control source-interface GigabitEthernet0/0
  bind media source-interface GigabitEthernet0/0
  registrar server expires max 3600 min 600
  early-offer forced

Open in new window


Translation rules to convert test numbers to E.164 format:

voice translation-rule 5
   rule 1 /907888555326/ /+447888555326/
voice translation-rule 6
   rule 1 /1908444555/ /+441908444555/

voice translation-profile SIP-TWILIO-OUT
  translate calling 6
  translate called 5

dial-peer voice 200 voip
  translation-profile outgoing SIP-TWILIO-OUT
  destination-pattern 907888555326
  session protocol sipv2
  session target dns:ourdomain.pstn.twilio.com
  dtmf-relay rtp-nte sip-kpml sip-notify
  codec g711ulaw
  no vad

Open in new window

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StratAnAuthor Commented:
We figured it out on our own.
0
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