Need help Creating a translation pattern that adds dial out digits to incoming calls

[sam saeed]

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?

[MUDIT MATHUR]

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

[sam saeed]

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.