CUCM System version: 7.1.3.10000-11
SIP Trunk to 3845 w/ 12.4(24)T3
I need a solution for a SIP outbound callforwarding issue. We recently changed from traditional PRI to Verizon provided SIP. Verizon requires the outbound digits be 10 and a SIP number. In order to forward an IP phone I need voice translation-rules because the phone when forwarded only sends 4 digits. I am currently using 15 sub rules and have more numbers to add. Am I doing it the most efficiant way? A dial-peer will only accept 1 rule in and 1 rule out so it looks like I am in trouble. Thanks ahead for any ideas.
Donnie
(I removed the actual area codes to stay out of trouble... lol)
voice translation-rule 6
rule 1 /^\(6[1-7]..\)$/ /XXX354\1/
rule 2 /^\(49..\)$/ /XXX463\1/
rule 3 /^\(4[4-5]..\)$/ /XXX605\1/
rule 4 /^\(22..\)$/ /XXX345\1/
rule 5 /^\(24..\)$/ /XXX354\1/
rule 6 /^\(21..\)$/ /XXX357\1/
rule 7 /^\(3[0-2]..\)$/ /XXX357\1/
rule 8 /^\(39..\)$/ /XXX357\1/
rule 9 /^\(83..\)$/ /XXX365\1/
rule 10 /^\(3[3-4]..\)$/ /XXX369\1/
rule 11 /^\(42..\)$/ /XXX877\1/
rule 12 /^\(60..\)$/ /XXX416\1/
rule 13 /^\(59..\)$/ /XXX425\1/
rule 14 /^\(13..\)$/ /XXX290\1/
rule 15 /^\(7[2-3]..\)$/ /XXX233\1/
!
voice translation-profile DIGITSTRIP-9
translate calling 12
translate called 9
translate redirect-called 6
!
dial-peer voice 100 voip
description OUTBOUND G729 Voice SIP calls to VzB
translation-profile outgoing DIGITSTRIP-9
destination-pattern 9T
voice-class codec 1
voice-class sip early-offer forced
session protocol sipv2
session target sip-server
incoming called-number .
dtmf-relay rtp-nte digit-drop
no vad
!
