SIP dialpeer to CUE IVR

Hi,
I have a SIP trunk configured in VG to land incomming calls. Calls from outside are able to receive directly on DID extensions successfully.
Want to configure the DID number to be forwarded to IVR so i can hear welcome greeting.
like if someone calling on 7700(DID) the call should be forwarded to IVR (CUE-1119).
Quick response is highly appreciated.
Thank You
Sam

Hello Suresh,
I checked incomming & outgoing dial-peers. Outgoing is 100 & incomming 5001. Below configuration example.
voice translation-rule 1
rule 1 /7700/ /1119/
voice translation-profile for_7700_1119
translate called 1
dial-peer voice 100 voip
description *** VOIP DIAL-PEER To IP-PHONES ***
translation-profile outgoing for_7700_1119
destination-pattern 22877..
session protocol sipv2
session target ipv4:172.22.240.100
voice-class codec 1
dtmf-relay h245-alphanumeric h245-signal rtp-nte
fax protocol none
no vad
supplementary-service pass-through
dial-peer voice 5001 voip
description **Incoming calls to STC SIP Trunk**
translation-profile incoming for_7700_1119
session protocol sipv2
incoming called-number .T
voice-class codec 1
dtmf-relay rtp-nte
no vad
Could you please help me with sample templates of below:
- Configuration in CUCM to accpet the call 1119 & send to IVR.
Thank You for reply.
Sam

Similar Messages

AIM-CUE IVR

Hi,
I have a AIM-CUE with a 1GB Flash. When I try to add on a 4 port ivr license, I get the following error message:
Downloading ftp cue-vm-license_4port_ivr_7.0.2.pkg
Bytes downloaded : 3319
Validating package signature ... done
compatibility mode
Validating installed manifests .............complete.
The system will be brought to offline state for a brief period
and will be brought back to online state automatically
No work order produced.
The system is back in online state
55296+0 records in
108+0 records out
ERROR - Hot Installation failed.
I am not sure why the installed failed.
Any help appreciated.
Thanks,
Paul

Hi,
The AIM is in the 2821 router running 12.4 AES T train IOS.
CUE will allow me to installed the zero ivr port package but not the 2 or 4 port.
CUE config :
se-10-2-1-2# show software packages
Installed Packages:
- Installer (Installer application) (7.0.2.0)
- Thirdparty (Service Engine Thirdparty Code) (7.0.2)
- Bootloader (Primary) (Service Engine Bootloader) (2.1.14)
- Infrastructure (Service Engine Infrastructure) (7.0.2)
- Global (Global manifest) (7.0.2)
- Service Engine license (License for the Service Engine) (2.1.2.0)
- Auto Attendant (Service Engine Telephony Infrastructure) (7.0.2)
- Voice Mail (Voicemail application) (7.0.2)
- Bootloader (Secondary) (Service Engine Bootloader) (2.1.14.0)
- Core (Service Engine OS Core) (7.0.2)
- GPL Infrastructure (Service Engine GPL Infrastructure) (7.0.2)
Installed Plug-ins:
- CUE Voicemail Language Support (Languages global pack) (7.0.2)
- CUE Voicemail US English (English language pack) (7.0.2)
se-10-2-1-2# show software licenses
Installed license files:
- voicemail_lic.sig : 25 MAILBOX LICENSE
Core:
- Application mode: CCM
- Total usable system ports: 6
Voicemail/Auto Attendant:
- Max system mailbox capacity time: 840
- Default # of general delivery mailboxes: 10
- Default # of personal mailboxes: 25
- Max # of configurable mailboxes: 35
Interactive Voice Response:
- Max # of IVR sessions: Not Available
Languages:
- Max installed languages: 2
- Max enabled languages: 2

Calls from Sip Trunk to UC540 and then to CUE returned ** Service Unavailable**

Hi to all
i have something strange here and i need your assistance
Call Flow:
Sip trunk-->UC540--> CUE
When calls coming to UC540 from outside and then going to cue then we send back service unavailable.I made a translation and i sent directly the incoming calls to CUE
The same behavior is also if i send the calls to dummy number and then from there set forward all to voice mail.
Incoming voicemail is working fine
Incoming calls to phones also ok
Uc540: 8.6
CUE: 8.6.5
A number: 99999999
B number: 22777777
Voice Mail Number:111
Attached is the trace
i see that we hit the correct dial peers .
I have enable only trancoder since MTP is not register ( don't know why , but i don't think also that is necessary..
voice service voip
ip address trusted list
ipv4 172.16.80.0 255.255.255.0
ipv4 172.16.81.0 255.255.255.0
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
supplementary-service media-renegotiate
sip
no update-callerid
dial-peer voice 1000 voip
description **SIP TRUNK**
translation-profile incoming SIP-INCOMING
translation-profile outgoing SIP-OUTGOING
destination-pattern 9T
modem passthrough nse codec g711alaw
session protocol sipv2
session target sip-server
incoming called-number .T
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
fax-relay ecm disable
no fax-relay sg3-to-g3
fax rate 9600
fax protocol pass-through g711alaw
no vad
dial-peer voice 2001 voip
description ** cue voicemail pilot number **
destination-pattern 111
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
incoming called-number 111
no voice-class sip outbound-proxy
dtmf-relay sip-notify
codec g711ulaw
no vad
Regards
chrysostomos

Hi
Interface                  IP-Address      OK? Method Status                Protocol
FastEthernet0/0            unassigned      YES NVRAM up                    up
FastEthernet0/0.10         192.168.0.10    YES DHCP   up                    up   ----> For internet
FastEthernet0/0.20         10.151.5.130    YES NVRAM up                    up ------> For sip trunk
In0/0                      10.1.10.2       YES unset up                    up    --------> default gw for cue
Vlan1                      unassigned      YES unset up                    up
Vlan100                    unassigned      YES unset up                    up
Vlan200                    unassigned      YES unset up                    down
Vlan300                    unassigned      YES unset up                    down
NVI0                       10.1.10.2       YES unset up                    up
BVI1                       192.168.20.1    YES NVRAM up                    up
BVI100                     10.1.1.1        YES NVRAM up                    up   ---------> ip for cme
Loopback0                  10.1.10.2       YES NVRAM up                    up   ------> default gw for cue
dial-peer voice 2001 voip
description ** cue voicemail pilot number **
destination-pattern 111
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
incoming called-number 111
no voice-class sip outbound-proxy
voice-class sip bind control source-interface BVI100
voice-class sip bind media source-interface BVI100
dtmf-relay sip-notify
codec g711ulaw
no vad
interface FastEthernet0/0.10
description **FOR INTERNET**
encapsulation dot1Q 10
ip address dhcp
ip access-group 105 in
ip nat outside
ip inspect SDM_LOW out
ip virtual-reassembly in
interface FastEthernet0/0.20
description **FOR SIP TRUNK WITH ISP**
encapsulation dot1Q 20
ip address 10.151.5.130 255.255.255.240
ip route 10.1.10.1 255.255.255.255 Integrated-Service-Engine0/0
ping 10.1.10.1 source bvi100
Type escape sequence to abort.
Sending 5, 100-byte ICMP Echos to 10.1.10.1, timeout is 2 seconds:
Packet sent with a source address of 10.1.1.1
Success rate is 100 percent (5/5), round-trip min/avg/max = 1/1/1 ms
I have bind the interface of cme ( 10.1.1.1) but the call fails again
Attached is the trace
Anything to advice?

CME SIP issue - Cisco 7821 phone not registering

Hi
I am having issues with getting a Cisco 7821 phone to register.
Current deployment is with Cisco 6921 phones SCCP registration
SIP integration with CUE
SIP integration with Mitel system
c2951-universalk9-mz.SPA.154-3.M1.bin (CME 10.5)
In flash:
rootfs78xx.10-1-1SR1-4.sbn
kern78xx.10-1-1SR1-4.sbn
sboot78xx.10-1-1SR1-4.sbn
sip78xx.10-1-1SR1-4.loads
The 7821 phone gets IP address but fails to register. Please could somebody let me know why phone is not registering.
Configuration below (10.245.226.132 is CME address) .
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol pass-through g711ulaw
modem passthrough nse codec g711ulaw redundancy maximum-sessions 5
h323
sip
registrar server expires max 600 min 60
options-ping 90
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8
voice register global
mode cme
source-address 10.245.226.132 port 5060
max-dn 30
max-pool 10
load 7821 sip78xx.10-1-1SR1-4
authenticate register
authenticate realm all
timezone 22
date-format D/M/Y
voicemail 590
tftp-path flash:
create profile sync 0061443538560005
network-locale GB
voice register dn 1
number 1010
name user1
label user1
mwi
voice register pool 1
busy-trigger-per-button 2
id mac F09E.636E.63F2
type 7821
number 1 dn 1
presence call-list
dtmf-relay rtp-nte
username 1010 password 123
codec g711ulaw
no vad
dial-peer voice 391 voip
description *** Auto Attendant ***
destination-pattern 399
session protocol sipv2
session target ipv4:10.245.226.131
dtmf-relay sip-notify
codec g711ulaw
no vad
dial-peer voice 392 voip
description *** Administration Via Telephone ***
destination-pattern 392
session protocol sipv2
session target ipv4:10.245.226.131
dtmf-relay sip-notify
codec g711ulaw
no vad
dial-peer voice 393 voip
description *** Extension Assigner ***
service ea out-bound
destination-pattern 393
session target ipv4:10.245.226.132
dial-peer voice 590 voip
description *** Voice Mail Pilot ***
destination-pattern 590
b2bua
session protocol sipv2
session target ipv4:10.245.226.131
dtmf-relay sip-notify
codec g711ulaw
no vad
dial-peer voice 1 pots
description ** Match all incoming POTS calls **
translation-profile incoming IncomingPSTNcalls
incoming called-number .
direct-inward-dial
dial-peer voice 899 voip
description Call to Mitel
translation-profile incoming Prefix9
translation-profile outgoing rem44
destination-pattern [23]..
session protocol sipv2
session target ipv4:192.168.114.2
voice-class codec 1
dtmf-relay rtp-nte
no vad
interface GigabitEthernet0/0
description *** Connection to Mitel Phone System ***
ip address 192.168.114.5 255.255.255.248
duplex auto
speed auto
interface ISM0/0
description *** Connection to Cisco Unity Express ***
ip unnumbered GigabitEthernet0/1
service-module ip address 10.245.226.131 255.255.255.128
!Application: CUE Running on ISM
service-module ip default-gateway 10.245.226.132
interface GigabitEthernet0/1
description *** Connection to IP Phone LAN ***
ip address 10.245.226.132 255.255.255.128
duplex auto
speed auto
ip http server
ip http authentication local
ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
ip http path flash:
ip route 0.0.0.0 0.0.0.0 10.245.226.129
ip route 10.245.226.131 255.255.2
tftp-server flash:apps37sccp.1-4-4-0.bin
tftp-server flash:sip78xx.10-1-1SR1-4.loads
tftp-server flash:rootfs78xx.10-1-1SR1-4.sbn
tftp-server flash:sboot78xx.10-1-1SR1-4.sbn
sip-ua
mwi-server ipv4:10.245.226.131 expires 3600 port 5060 transport udp
registrar ipv4:10.245.226.132 expires 600
gatekeeper
shutdown
telephony-service
authentication credential cmeadmin c4p1ta2012
xml user xmladmin password xmladmin 15
extension-assigner tag-type provision-tag
max-ephones 104
max-dn 299
ip source-address 10.245.226.132 port 2000
auto assign 101 to 105
no service directed-pickup
timeouts interdigit 5
system message CFGS
url services http://10.245.226.131/voiceview/common/login.do
url authentication http://10.245.226.132/CCMCIP/authenticate.asp
cnf-file location flash:
cnf-file perphone
load 7931 SCCP31.9-2-1S
load 6921 SCCP69xx.9-2-1-0
time-zone 22
date-format dd-mm-yy
voicemail 590
max-conferences 8 gain -6
call-forward pattern .T
moh enable-g711 "music-on-hold.au"
web admin system name cmeadmin secret 5 $1$QmIK$46fDKVSudMxzI2bRp/Ef7/
time-webedit
transfer-system full-consult
transfer-pattern .T
secondary-dialtone 9
create cnf-files version-stamp Jan 01 2002 00:00:00
ephone-dn 298
number 598...
mwi on
ephone-dn 299
number 599...
mwi off

Page 7 of the following link recommends that you use option 150 with the Cisco 7800 series phones and use option 66 if you cannot use option 150
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cuipph/7821_7841_7861/10_1/english/admin_guide/PA2D_BK_AB3F74DA_00_admin-7821-7841-7861-10_0/PA2D_BK_AB3F74DA_00_admin-7821-7841-7861-10_0_chapter_01.pdf
Dynamic Host Configuration Protocol (DHCP)
DHCP dynamically allocates and assigns an IP address to network devices.
DHCP enables you to connect an IP phone into the network and have the phone become operational without your needing to manually assign an IP address or to configure additional network parameters.
DHCP is enabled by default. If disabled, you must manually configure the IP address, subnet mask, gateway, and a TFTP server on each phone locally.
Cisco recommends that you use DHCP custom option 150. With this method, you configure the TFTP server IP address as the option value. For additional supported DHCP configurations, go to the "Dynamic Host Configuration Protocol" chapter and the "Cisco TFTP" chapter in the Cisco Unified Communications Manager System Guide.
Note
If you cannot use option 150, you may try using DHCP option 66.

CUE 3.x to 7.x - Upgrading - need new license?

When upgrading from CUE 3.x to 7.x, is a new license file required if there are no mailbox count changes, etc.?
For exampled - show software licenses:
Installed license files:
- ivr_lic.sig : NULL IVR LICENSE
- voicemail_lic.sig : 12 MAILBOX LICENSE
If mailbox count is unchanged, will the upgrade process convert the licenses when doing a software install upgrade?
Cheers.
-Robert

Hi Robert,
Beginning with Cisco Unity Express 7.1, a new type of license called CSL licensing is supported. With CSL licenses, the mailbox license count includes both personal mailboxes and GDMs. The type of the mailbox is determined when it is configured. Also, the call-agent is no longer specified using licenses and can be configured either as part of post-install process or during bootup.
Obtaining Migration Licenses
Note There is no cost to obtain migration licenses for CSL.
Required Information
To obtain CSL migration licenses you need to have the following information:
•The SKU for the features that you need.
This is the information that you determined in the "Selecting which Features You Want" section.
•The Product ID (PID)
•The Serial Number (SN) from the device.
Using the Migration Portal
Note You must have a CCO password to access some of the URLs in the following procedure.
Follow these steps to obtain a migration license for your existing Cisco Unity Express 7.1 features:
Step 1 Go to the licensing portal at www.cisco.com/go/license and click the link at the bottom of the page to go to the license migration portal.
Step 2 On the router that is running the features that you want to migrate, enter the show inventory command to see the corresponding product IDs and serial numbers.
Step 3 Enter the appropriate product ID and the serial number and click Continue.
The product ID is the type of Network Module that you are using, such as AIM-CUE.
Step 4 Use the drop-down menus to select the product family, product, and feature (SKU), then click Continue.
For the product, use the SKU that you determined in the previous section for the features that you want to migrate.
Enter your registration information, including your company's name and e-mail address, then click Continue.
Step 5 Verify all of your license information and click Submit to receive the free CSL upgrade license file by e-mail.
Step 6 Repeat the appropriate steps above for each device that is running features that you want to migrate.
Step 7 Copy the license file(s) to a FTP or TFTP server.
Adding More Mailboxes, Ports, IVR Sessions, and TimeCardView Users
When you buy licences for additional features on Cisco Unity Express 7.1, the licences are available only in the increments shown below. These licenses are the same for all supported devices. See the Release Notes for Cisco Unity Express 7.1 for a list of supported devices. The following types of licenses are available:
•Mailboxes (available only in increments of five)
•Ports (available only in increments of two)
•IVR (available only in increments of two)
•TimeCardView users (available only in increments of one)
The corresponding SKUs for these licenses are:
•FL-CUE-MBX-5
•FL-CUE-PORT-2
•FL-CUE-IVR-2
•FL-TCV-USER-1
To determine the total quantity of licenses you need, multiply the incremental quantity from the name of the CSL license with the explicit quantity of increments that you want to purchase. For instance, FL-CUE-MBX-5 represents an incremental quantity of five mailboxes. If you select an explicit quantity of four for the FL-CUE-MBX-5 feature, the total quantity will be five times four, or 20.
Using CLI Commands to Install Licenses
After you obtain a license file from Cisco's licensing portal and copy it to a FTP or TFTP server, use the following CLI commands to install the license. Only the use of FTP is shown below.
SUMMARY STEPS
1. license install ftp://username:password@ip_address/path/license_file
2. reload
3. enable
4. show license all
5. show license in-use
6. show license status application
From this excellent doc;
http://www.cisco.com/en/US/docs/voice_ip_comm/unity_exp/rel7_1/Licensing/Using_CSL.html#wp1102792
Cheers!
Rob

IVR Based Campaign Error

Dear All,
We are using SIP dialer for a IVR based campaign. On the event viewer we are getting the following messages.
Aug 12 09:55:04 192.168.1.1 160402: ccmpg1a: Aug 12 2013 04:55:03.669 +0000: %ICM_BADialer_BlendedAgent-6-1438044: %[comp=BADialer-A][pname=baDialer][iid=prod][mid=1438044][sev=info]: [Skill group (ID=5750) in the IVR_Camp ] has insufficient records [581] in the last one minute on the dialer machine [CCMPG1A]
we are using ICM 8.5(4). I have performed some R&D but there is no information available on the support forum.
Regards,
Amir

Would be able to restart the dialer ? just a try not sure if this can re-sync the data.

How to download VoiceXML Web Application from CUE?

Hi All
Our customer has already had CUE IVR working with VoiceXML function.
what we need to do is to study their existing IVR with VXML, make certain change base on
current requirement.
however after a couple days dig through their CUE system and googling, I didnt find a way to get the
original "war" file mentioned in link below.
http://www.cisco.com/en/US/partner/docs/voice_ip_comm/unity_exp/rel3_0/vxml/developer/guide/vxmljsp.html#wp1060138
I understand that after deploying the web application to CUE, it will create a folder under "IVR > VXML Applications"
my questions are
1. how can we download all deployed WAR files from CUE?
2. or how can we get the original VXML and JSP files?
3. any way to reverse engineering the deployed files back to WAR file?
thanks
Yuke

Hi Gergely
thanks for ur reply. I downloaded the 7.0.6 version of sdk due to my CUE is at version 7.0.
1. I tried to build it a few time with no success. is there any trick on building the war file?
C:\ant\bin>echo off
C:\ant\bin>C:\ant\bin\build.bat -v
C:\ant\bin>REM 56789012345678901234567890123456789012345678901234567890123456789
012345
C:\ant\bin>REM -----------------------------------------------------------------
C:\ant\bin>REM build.bat -- Build Script for WEBAPP
C:\ant\bin>setlocal EnableDelayedExpansion
C:\ant\bin>echo off
C:\ant\bin>REM Execute ANT to perform the requested build target
C:\ant\bin>"C:\Program Files (x86)\Java\j2re1.4.2"\bin\java -classpath c:\ant\li
b\ant.jar;c:\ant\lib\optional.jar;"C:\Program Files (x86)\Java\j2re1.4.2"\lib\to
ols.jar;c:\webapp\webserver\lib\commons-logging-api.jar;c:\webapp\webserver\lib\
container_util.jar;c:\webapp\webserver\lib\core_util.jar;c:\webapp\webserver\lib
\crimson.jar;c:\webapp\webserver\lib\etomcat.jar;c:\webapp\webserver\lib\facade2
2.jar;c:\webapp\webserver\lib\jakarta-regexp-1.3.jar;c:\webapp\webserver\lib\jas
per-runtime.jar;c:\webapp\webserver\lib\jasper.jar;c:\webapp\webserver\lib\mx4j-
jmx.jar;c:\webapp\webserver\lib\mx4j-tools.jar;c:\webapp\webserver\lib\servlet.j
ar;c:\webapp\webserver\lib\stop-tomcat.jar;c:\webapp\webserver\lib\tomcat-coyote
.jar;c:\webapp\webserver\lib\tomcat-http11.jar;c:\webapp\webserver\lib\tomcat-jk
2.jar;c:\webapp\webserver\lib\tomcat-util.jar;c:\webapp\webserver\lib\tomcat.jar
;c:\webapp\webserver\lib\tomcat33-coyote.jar;c:\webapp\webserver\lib\tomcat33-re
source.jar;c:\webapp\webserver\lib\tomcat_core.jar;c:\webapp\webserver\lib\tomca
t_modules.jar;c:\webapp\webserver\lib\xalan.jar;c:\webapp\lib\ivrjsp.jar -Dtomca
t.home=webserver org.apache.tools.ant.Main
Buildfile: build.xml
prepare:
compile:
     [jspc] Compiling 9 source filesC:\ant\bin\jsp-gen
     [java] java.lang.NoClassDefFoundError: org/apache/jasper/JspC
     [java] Exception in thread "main"
BUILD FAILED
file:C:/ant/bin/build.xml:50: Java returned: 1
Total time: 0 seconds
C:\ant\bin>echo off
C:\ant\bin>
it throws some error, but I have no idea where to start the troubleshooting
2. what do u mean by browse through the cue system? using gui? or cli?
from gui, I didnt find anywhere I can download the "extracted" war files.
under cli, I can use "ccn copy url", only if I know the directory and file name.
e.g.
ccn copy url "http://localhost/vbrowser/vxml/vxml.vxml" document generic test.vxml
do u happen to know that from cli, how do I list all files under localhost/vbrowser/vxml?
thanks
Yuke

Upgrading CUE 7.0 to CUE 8.6 on Cisco 2821 NME-CUE

Hi guys,
I read that migration form CUE 7.0 to 8,6 if possible, but I dont undersantd for license, this link tell that I contact with license page for migrate the license of mi actualy CUE 7.0, the question is, this migration has a cost? o is free i only need contact to team license.
http://www.cisco.com/en/US/docs/voice_ip_comm/unity_exp/rel7_1/Licensing/CUELicense.pdf
Thanks for your help.

Hi Roberto,
There is no cost (free) to migrate to CSL Licenses
Migrating to CSL Licenses
When you upgrade to version 7.1 or later, you must replace your pre-CSL licenses with CSL licenses. You are required to migrate your licenses for mailbox features and port features. If you purchased the optional IVR features, you must also migrate your licenses for the IVR features. The migration procedure for each of these types of licenses are described in a separate section below.
Note If you are upgrading to the same of licensed features as before, the CSL licenses are free.
When you obtain licenses for features for Cisco Unity Express 7.1 or later, the licenses are available only in the increments shown below. The following types of licenses are available:
•Mailboxes (available only in increments of five)
•Ports (available only in increments of two)
•IVR (available only in increments of two)
Note CSL licenses are the same for all supported devices. See the Release Notes for Cisco Unity Express for a list of supported devices.
The corresponding SKUs for these licenses are:
•FL-CUE-MBX-5
•FL-CUE-PORT-2
•FL-CUE-IVR-2
•FL-TCV-USER-1
To migrate to CSL licenses, use the following procedure:
1. Use the show software license command to determine which pre-CSL licenses you have.
The name of the pre-CSL licenses include the incremental quantity of the license. For example SCUE-LIC-25CME.
2. For mailboxes, add the number of GDMs (Group Delivery Mailboxes) that you will be using because they were not included in the pre-CSL counts but are now being represented in the CSL counts.
3. Divide the total number of pre-CSL licensed features by the incremental quantity represented in the name of the appropriate CSL SKU.
For mailboxes, divide by five. For ports, divide by two. For IVR sessions, divide by two.
If the result is not a whole number, round up. This will give you the explicit quantity of CSL incremental licenses that you need.
As an example, if you have the following pre-CSL licenses and you have 10 GDMs:
•SCUE-LIC-100CCM
•8 instances of SCUE-LIC-PORT-2
•SCUE-IVR-S16
You will need to choose:
•22 instances of FL-CUE-MBX-5
•8 instances of FL-CUE-PORT-2
•8 instances of FL-CUE-IVR-2
Using the Migration Portal
Note You must have a Cisco.com password to access some of the URLs in the following procedure.
Follow these steps to obtain a migration license for your existing Cisco Unity Express 7.1 features:
Step 1 Go to the licensing portal at www.cisco.com/go/license and click the link at the bottom of the page to go to the license migration portal.
Step 2 On the router that is running the features that you want to migrate, enter the show inventory command to see the corresponding product IDs and serial numbers.
Step 3 Enter the appropriate product ID and the serial number and click Continue.
The product ID is the type of Network Module that you are using, such as AIM-CUE.
Step 4 Use the drop-down menus to select the product family, product, and feature (SKU), then click Continue.
For the product, use the SKU that you determined in the previous section for the features that you want to migrate.
Enter your registration information, including your company's name and e-mail address, then click Continue.
Step 5 Verify all of your license information and click Submit to receive the free CSL upgrade license file by e-mail.
Step 6 Repeat the appropriate steps above for each device that is running features that you want to migrate.
Step 7 Copy the license file(s) to a FTP or TFTP server.
From this good doc;
http://www.cisco.com/en/US/docs/voice_ip_comm/unity_exp/rel7_1/Licensing/Using_CSL.html#wp1102792
Cheers!
Rob

CUCME + CUE port license

Do you need a port license in integrating CUCME and CUE?

Hi Ramon,
Yes you would require Port licenses for this integration (increments of 2)
The following types of licenses are available:
•Mailboxes (available only in increments of five)
•Ports (available only in increments of two)
•IVR (available only in increments of two)
•TimeCardView users (available only in increments of one)
The corresponding SKUs for these licenses are:
•FL-CUE-MBX-5
•FL-CUE-PORT-2
•FL-CUE-IVR-2
•FL-TCV-USER-1
To determine the total quantity of licenses you need, multiply the incremental quantity from the name of the CSL license with the explicit quantity of increments that you want to purchase. For instance, FL-CUE-MBX-5 represents an incremental quantity of five mailboxes. If you select an explicit quantity of four for the FL-CUE-MBX-5 feature, the total quantity will be five times four, or 20.
From this excellent doc;
http://www.cisco.com/en/US/docs/voice_ip_comm/unity_exp/rel7_1/Licensing/Using_CSL.html#wp1102792
Cheers!
Rob
Please support CSC helps Kiva
https://supportforums.cisco.com/blog/12122171/cisco-support-community-helps-kiva

Remove digits from incoming DID number

How translation rule can i use to remove 6 digits of DID number received from ITSP and pass on 4 digits to the PABX (integrated with ISDN pri ) on cisco voice gateway.
The incoming dial peer is voip dial peer because call is coming from ITSP using SIP.
For eg DID number range is 0419137101 - 0419137599

If I follow, I believe you could use something like this:
voice translation-rule 1
rule 1 /.+$....$/ /\1/
voice translation-profile strip6
translate called 1
dial-peer voice 1 voip ! Inbound SIP dialpeer
translation-profile incoming strip6

CUE 8.6.7 - Can't sync users from SIP phones

Hello, I'm trying to Synchronize information from CUCME version 10.0 to create new users and assign a mailbox, I don't have any problems with phones using SCCP, but I can't see phones running SIP, is there any configuration that I'm missing??
voice register global
mode cme
source-address 172.16.10.129 port 5060
timeouts interdigit 5
max-dn 200
max-pool 42
load 3905 CP3905.9-2-1-0
authenticate register
timezone 9
time-format 24
date-format D/M/Y
hold-alert
no dst auto-adjust
voicemail 290
create profile sync 0005396071474939
network-locale ES
user-locale ES
ntp-server 172.16.10.129 mode directedbroadcast
conference hardware
voice register dn 1
number 253
name User1
huntstop channel 3
voice register pool 1
busy-trigger-per-button 2
id mac F41F.C267.ECC7
type 3905
number 1 dn 1
template 1
dtmf-relay sip-kpml
username user1 password user1
description user1
codec g711ulaw
Regards,
Juan Carlos Arias

Did you ever find an answer to your question? Building CUCME with 7841 SIP only phones and trying to research if there is a resolution to SIP users not synchronizing in CUE.

Integrating CCM 4.1(3) w/ a SIP IVR

I have set up a SIP trunk to move calls between SCCP endpoints/MGCP gateways and a SIP IVR. I have assigned it to a Route Group/Route List and assigned a Route Pattern. The IVR is reachable, but the SIP trunk is always sending DNIS as the route pattern assigned to the SIP trunk (8000), rather than the original called number (7000) (the CCM-side device dials a number and is redirected across the SIP trunk). Is there a way to send original dialed number rather then the redirected number here?
Also, the SIP trunk's destination is set as the IP Address of the IVR, and not a proxy server. If this IVR is the only SIP device I need to support, will I have any trouble with this configuration?

David
I'm not sure I fully understand your configuration. I have a SIP trunk connected to a SIP Server providing call center duties.
I defined a SIP Trunk with the server name = Mesa. I have the fields Connected Line ID and Name = Allowed, Redirecting Number Delivery-Inbound/Outbound checked. I assigned a DN 0f 6000 to the SIP server. Then I built a route pattern 6000 with the Gateway or Route List field = Mesa. All of the fields on the route pattern are set to default. When someone on the CCM side dials 6000 the call is routed across the SIP trunk to the SIP server and the Caller ID is displayed correctly. When one of the SIP call agents makes an oubound call it routes across the SIP trunk and the Caller ID displays correctly. Hope that provides some help.
Larry

CME/CUE SIP Phones DTMF-Relay

Hi all,
Just looking for some clarification on this one. I'm seeing some conflicting advice about setting the DTMF-Relay on SIP Phones registered to CME with a CUE Module. I've read some documentation indicating that rtp-nte RFC2833 is the only dtmf-relay supported for SIP Phones registered to CME, however I've also read some documents indicating that sip-notify must be configured as the dtmf-relay on SIP phones when they are communicating to a CUE module. I'm assuming I'm going to need to configure an MTP on the CME, but just wondering what the official DTMF config should be under the voice register pool for SIP phones.
Thanks!

Hi logan
When doing lab with cme 7.0 and sip phones .sip phones are not recognizing the "sip-notify" dtmf-relay method .It can only recognize "rtp-nte" method and it does not matter weather you are using sip-notify or rtp-nte for a dial-peer pointing to cme .
i configured on cue
ccn subsystem sip
dtmf-relay sip-notify
end
on cme i configured a dial-peer pointing to cue
dial-peer v 3888 voip
destination-pattern 3888
session target ipv4:177.3.11.10
codec g711ulaw
no vad
session protocol sipv2
dtmf-relay sip-notiy
on my sip phones
voice register pool 1
dtmf-relay sip-notify ------> now in this case cue wont recognize dtmf tones
when i change this dtmf-relay method to rtp-nte it recognizes dtmf tones to when recording a message

CUE giving a fast busy

I am facing a weird issue and wondering if someone have seen this before...
Plateforme is 2801 with CME 8.6, IOS 15.1 and CUE 7.4. I believe the version of CUE is compatible and should work with CME 8.6.
What's happening is I get an immediate fast busy when hitting the VM button on a phone.
The dial-peer is below for the VM pilot number:
dial-peer voice 7000 voip
destination-pattern 7000
session protocol sipv2
session target ipv4:10.1.150.2
dtmf-relay sip-notify
codec g711ulaw
Debug voip ccapi inout shows disconnection error code 38 - Network out of order
Not output from debug ccsip messages
Debug ccsip all shows CME creating a sip connection to the CUE module but it immediatly fails:
*Aug 17 15:07:40.983 CST: //6/FA0D0B74800E/SIP/State/sipSPIChangeState: 0x6A5BAEC0 : State change from (STATE_DISCONNECTING, SUBSTATE_NONE) to (STATE_DEAD, SUBSTATE_NONE)
*Aug 17 15:07:40.983 CST: //6/FA0D0B74800E/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x6A5BAEC0
State of The Call : STATE_DEAD
TCP Sockets Used : YES
Calling Number : 4444
Called Number : 7000
Source IP Address (Sig ): 3.3.3.3
Destn SIP Req Addr:Port : 10.1.150.2:5060
Destn SIP Resp Addr:Port : 10.1.150.2:5060
Destination Name : 10.1.150.2
Tthe CUE module is up, licenses are installed and doesn't show any errors. Except dealing with a defective CUE module not sure why it is not accepting incoming SIP connections.
I would appreciate any inputs.

I reloaded the CUE multiple times. No output when doing debusg ccsip messages.
I removed dial-peer 102 but no change. Debug voip dialpeer inout shows 7000 being matched.
Also no change after setting
ccn subsystem sip
default gateway address
default gateway port
All I can see is CUE not accepting a SIP connection and CCAPI inout debug shows Disconnection code 38 - Network out of order.

CUE 8.6.3 ringing busy

Hi,
We are having an issue where CUE has stopped working. It had been working just fine but now when we call it we get a busy signal. I tried to reset the module but still have the same issue. When I do a service-module integrated-service-engine 0/0 status it shows it is running fine however when I try to do a session I get the error "Connection refused by remote host".
Here is some of the config
voice service voip
sip
bind control source-interface Vlan90
bind media source-interface Vlan90
interface Integrated-Service-Engine0/0
ip unnumbered Vlan90
ip nat inside
ip virtual-reasembly in
service-module ip address 10.1.10.1 255.255.255.252
service-module ip default-gateway 10.1.10.2
interface GigabitEthernet0/1/3
switchport access vlan 90
service-module ip address 10.1.10.1 255.255.255.252
service-module ip default-gateway 10.1.10.2
interface Vlan90
ipaddress 10.1.10.2 255.255.255.252
ip access-group 103 in
ip nat inside
ip vertual-reassembly in
ip route 10.1.10.1 255.255.255.255 Vlan90

Hi Dustin,
You and Paolo are certainly on the right track here +5 "P"
Here's a clip from the licensing guide;
Note Each IVR session consumes one port and thus reduces the number of ports available for use by voicemail or auto-attendant applications. Please carefully consider how many IVR session licenses you need and only install or activate licenses for that number of IVR sessions.
If the number of IVR session licenses is greater than or equal to the number of ports, then voicemail and auto-attendant applications will be disabled due to the lack of available ports. The
show license status application
command will indicate this condition by stating the error message "voicemail disabled, ivr session quantity (x) is equal to or exceeds available ports (Y)" where X is the number of IVR sessions and Y is the number of ports. To change this condition, either reduce the number of IVR sessions via the command
license activate ivr sessions N
(where N is less than the number of ports and may be zero to deactivate all IVR sessions), or remove the IVR session licenses, or add more port licenses, and then issue the
reload command.
http://www.cisco.com/en/US/docs/voice_ip_comm/unity_exp/rel7_1/Licensing/Using_CSL.html#wp1065047
You will need to deactivate the IVR sessions and you should be good to go.
Cheers!
Rob
"And if I should fall behind
Wait for me" - Springsteen

SIP dialpeer to CUE IVR

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