Options
Incoming Dial-Peer-Translation pattern question
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?
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?
Comments
-
Options
shodown
Member Posts: 2,271
So call calls come in to unity express first?Currently Reading
CUCM SRND 9x/10, UCCX SRND 10x, QOS SRND, SIP Trunking Guide, anything contact center related -
OptionsLegacy User
Unregistered / Not Logged In Posts: 0 ■□□□□□□□□□
Yes all inbound calls designated for the 201 office number are translated to pilot #2003 which forwards to the AA.
-
Optionsshodown
Member Posts: 2,271
so then they calls are being sent from the AA to the phones. In that cause you have the virtual dial peers that CME makes for the ephones which you can't manipulate. So you are kinda stuck where are you are at unless you hand out DID'sCurrently Reading
CUCM SRND 9x/10, UCCX SRND 10x, QOS SRND, SIP Trunking Guide, anything contact center related -
OptionsLegacy User
Unregistered / Not Logged In Posts: 0 ■□□□□□□□□□
Yea, thats what I figured earlier that it should work using DIDs. I just wanted to make sure there wasn't a way to rig it up before I drop the idea. Thanks for taking the time though, your still an O.G.
