11-06-2014 07:01 AM - edited 03-17-2019 12:49 AM
I came across an article yesterday and it showed the steps how to fix Missed Call/Received Call numbers so that you can dial them from the menu correctly (auto-add a 9, etc.)?
I tried it this morning and came up with this translation pattern:
voice translation-rule 6
rule 1 /^201\(.*\)/ /8\1/
rule 2 /\(..........\)/ /81\1/
voice translation-profile filter_Incoming
translate calling 6
This translation pattern rule 1 adds the dial out character 8 and strips 201 for local calls. Rule 2 adds dial out character 8 and adds 1 for long distance. The purpose of this translation rule is when the ephone receives the phone call the characters 8 and 1 are added so when you quickly need to redial you do not have to edit the number and add 8 for each call.
I tested the translation-rule:
ROUTER-2911#test voice translation-rule 6 9082121231
Matched with rule 2
Original number: 9082121231 Translated number: 819082121231
Original number type: none Translated number type: none
Original number plan: none Translated number plan: none
ROUTER-2911#test voice translation-rule 6 2019121231
Matched with rule 1
Original number: 2019121231 Translated number: 89121231
Original number type: none Translated number type: none
Original number plan: none Translated number plan: none
ROUTER-2911#
Issue is I am not sure with my inbound call leg if it can even work. We dial out through the SIP Trunk and the incoming is translated to the AutoAttendant on Cisco Unity Express.
voice translation-rule 1
rule 1 /2015552100/ /2003/
voice translation-profile CUE_Voicemail/AutoAttendant
translate called 1
dial-peer voice 9 voip
description **Incoming Call from SIP Trunk**
translation-profile incoming CUE_Voicemail/AutoAttendant
call-block translation-profile incoming BLOCKED-INCOMING
call-block disconnect-cause incoming call-reject
session protocol sipv2
session target dns:nd01-04.fs.SIPPROVIDER.net
incoming called-number .%
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
Can what I am trying to do be done with my current setup?
11-07-2014 01:16 AM
Hi patldmart012,
The dial-peer 9 that you have attached will not be affected by following config
voice translation-rule 6
rule 1 /^201\(.*\)/ /8\1/
rule 2 /\(..........\)/ /81\1/
voice translation-profile filter_Incoming
translate calling 6
Because you have not applied the translation profile "filter_incoming" on the dial-peer.
Could you please provide the exact call flow?
Along with that, If you are facing issue with calls on SIP Trunk, please collect following debugs in a logging buffer and attach the file. I will analyse it and will get back to you.
debug voip ccapi inout
debug ccsip message
debug voice translation
Debug h225 asn1 (If H323 involved)
Debug h245 asn1 (If H323 involved)
Debug MGCP Packets (If MGCP involved)
Also provide the running config of the GW.
These are verbose debugs, so please collect them in the following manner:
Router(config)# service sequence
Router(config)# service timestamps debug datetime msec
Router(config)# logging buffered 30000000 7
Router(config)# no logging con
Router(config)# no logging mon
Router# Clear log
Router# term no mon
<Enable debugs, then wait for issue to occur.>
Router# term len 0
<Enable session capture to txt file in terminal program.>
Router# Undebug all
Router# sh log
Once i have the logs, i will analyse it and will get back to you.
Regards,
Mudit Mathur
11-07-2014 07:06 AM
Hello there,
I have not applied it but as is I do not think what I am trying to do will work. I just wanted to make sure there was not a way to tweek it to get it to work.
Call flow:
All inbound calls designated for the 201 office number are translated to pilot #2003 which forwards to the AA.
This is the rule applied to the DP going to the AA.
voice translation-rule 4
rule 1 /^8(.......)$/ /201\1/
rule 2 /2000/ /2015553000/
rule 3 /2003/ /2015553000/
rule 4 /^8(...)$/ /2015553\1/
rule 5 /^8(.*)/ /\1/
voice translation-profile PSTN_CallForwarding
translate redirect-target 4
translate redirect-called 4
dial-peer voice 7 voip
description **CUE Auto Attendant**
translation-profile outgoing PSTN_CallForwarding
destination-pattern 2003
b2bua
session protocol sipv2
session target ipv4:10.10.10.3
dtmf-relay sip-notify
codec g711ulaw
no vad
The AA then calls the internal extensions.
I am thinking it may be possible if I had a few DIDs but at this point I am thinking this idea is a lost cause.
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