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CVP 11.0 overriding Translation Profile - PCCE

Level 1
Level 1

Hi All,


Currently I have an issue where when I put the "service survivability" command under the incoming dial-peer, it seems to be overriding the translation profile on that dial-peer. This has become an issue because we are relying on the translation-profile whether or not the call is destined for Call Manager(11 digit) or CVP(10 digit). 


dial-peer voice 100 voip  <---Inbound Dial Peer
description "Inbound from PSTN"
translation-profile incoming PSTN-INBOUND <---Translation Profile to decide if number is Call Manager or CVP Bound
service survivability
session protocol sipv2
incoming uri via ATTSIPTRUNK
voice-class sip bind control source-interface Loopback1
voice-class sip bind media source-interface Loopback1
dtmf-relay rtp-nte
codec g711ulaw


voice translation-rule 100  <---Translation Profile to decide if number is Call Manager or CVP Bound
rule 1 /8183575047/ /8183575047/
rule 2 /^9496239773/ /9496239773/
rule 3 /^8189911153/ /8189911153/
rule 4 /^8183575000/ /8183575000/
rule 5 /^8186543490/ /8186543490/
rule 6 /^8183575051/ /8183575051/
rule 7 /^8183575100/ /8183575100/
rule 8 /^8059738700/ /8059738700/
rule 9 /^8666543471/ /8666543471/
rule 10 /^2133558988/ /2133558988/
rule 11 /^2138352828/ /2138352828/
rule 12 /^2138352820/ /2138352820/
rule 13 /^7148368266/ /7148368266/
rule 99 /^[2-9]..[2-9]......$/ /1\0/


voice translation-rule 200 
rule 99 /^[2-9]..[2-9]......$/ /1\0/


voice translation-profile PSTN-INBOUND
translate calling 200 <---Decide whether or not number is Call Manager(11 digit) or CVP (10 digit) bound
translate called 100 <--- Add prefix 1 to calling number making it a 11 digit


Is there another way to implement this translation-profile without interfering with CVP? From the documents I've read service survivability needs to be programmed in the incoming dial peer for Courtesy Call Back to work. 


I've seen in a PRI you can add the translation-profile command under the voice-port. Is there some kind of version of that for SIP connections?



Thank you in advance,



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