Cisco Support Community
cancel
Showing results for 
Search instead for 
Did you mean: 

Incoming call from Telco that is forwarded to PSTN number is failing

Problem


The call flow is:  Telco--(SIP)--CUBE--(SIP)--CUCM--Phone
 Call forwarding to PSTN line is not working when the call is originated from PSTN.

Following are the  challenges for incoming calls from Telco

When a call come into the CUBE from Telco to CUCM, you want dialed number and calling number

to be transformed before send to CUCM.  
a) From International caller: dialed number looks like 00966-3-813-xxxx,
but Telco will send 3813xxxx and this must become xxxx before CUBE route the call to CUCM  
b) From country caller (different city): dialed number looks like 03813xxxx, but Telco will send 3813xxxx
and this must become xxxx before CUBE route the call to CUCM.  
c) From country caller (same city): dialed number looks like 813xxxx. Telco will send 813xxxx
and this must become xxxx before CUBE route the call to CUCM.  
d) For calling number, you may add prefix 900 (to international number) and prefix 90 (to national number) for redial purpose.


Solution

 
To achieve these challenges a sample configuration on CUBE is given below.

1) Translation rule to transform called number sent by Telco  (3813xxxx and 813xxxx) to xxxx as below
!
voice translation-rule 1
 rule 4 /^3813\(....\)/ /\1/  
 rule 5 /^813\(....\)/ /\1/
!
2) Translation rule to add prefix 900 to international number, and prefix 90  to national number as below
!
voice translation-rule 5
rule 1 /\(.*\)/ /90\1/ type national national
rule 2 /\(.*\)/ /900\1/ type international international
!

3)Translation rule to remove prefix 9 from the dialed number (sent by CUCM) as below

!
voice translation-rule 2
rule 3 /^9\(.*\)/ /\1/


4) Translation rule to ADD prefix 8136 or 8133 to calling number (sent by CUCM) as below

!
voice translation-rule 3
rule 1 /^6\(...\)$/ /8136\1/
rule 2 /^3\(...\)$/ /8133\1/
!


  • Leave your translation profiles OutCalls and inboundcalls as below
 !
voice translation-profile OutCalls
translate calling 3
translate called 2
!
voice translation-profile inboundcalls
translate calling 5
translate called 1
! 


  • Add below incoming dial-peers for calls coming from Telco and for calls coming from CUCM 


 !         
dial-peer voice 1 voip
description +++ INCOMING CALL FROM TELCO  +++
translation-profile incoming inboundcalls
rtp payload-type cisco-codec-fax-ack 103
rtp payload-type nte 97
session protocol sipv2
incoming called-number .
voice-class codec 1  
 dtmf-relay rtp-nte
no vad   
!

!
dial-peer voice 2 voip
description +++ INCOMMING CALL FROM CUCM +++
rtp payload-type cisco-codec-fax-ack 103
rtp payload-type nte 97
session protocol sipv2
incoming called-number 9T
voice-class codec 1  
 voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
!       

! dial-peer voice 110 voip description +++ OUTGOING CALL TO TELCO +++ translation-profile outgoing OutCalls destination-pattern 9T rtp payload-type cisco-codec-fax-ack 103 rtp payload-type nte 97 session protocol sipv2 session target ipv4:10.205.20.50 voice-class codec 1  voice-class sip dtmf-relay force rtp-nte dtmf-relay rtp-nte no vad   ! ! dial-peer voice 112 voip description +++ Outgoing Direct Lines To UC +++ destination-pattern ....$ preference 0 rtp payload-type cisco-codec-fax-ack 103 rtp payload-type nte 97 session protocol sipv2 session target ipv4:172.19.0.4 voice-class codec 1  voice-class sip dtmf-relay force rtp-nte dtmf-relay rtp-nte no vad ! ! dial-peer voice 113 voip description +++ Outgoing backup Lines To UC destination-pattern ....$ preference 2 rtp payload-type cisco-codec-fax-ack 103 rtp payload-type nte 97 session protocol sipv2 session target ipv4:172.19.0.5 voice-class codec 1  voice-class sip dtmf-relay force rtp-nte dtmf-relay rtp-nte no vad !         
Related Information

Version history
Revision #:
1 of 1
Last update:
‎10-12-2012 02:44 AM
Updated by:
 
Labels (1)
Everyone's tags (5)
Comments

use incoming calling-number .T in dial-peer voice 1 voip

to check if the call is reaching you and what dial-peer it using please post the output of debug voip ccapi inout