show sysdb /sw/info/filesys — will show how much disk space is being used for voicemail. It will also show disk usage in percentage
To compact the database.
# offline
database compact
Archive for the ‘Cisco Unity Express’ Category
One of the current limitations is that the JTAPI (CTI-quick buffer encoding [QBE]) signaling packets are unmarked (TOS = 0) when the Cisco Unity Express AIM transmits them. In order to correct this, use an access control list (ACL) on the router that has the Cisco Unity Express AIM installed to mark and prioritize the traffic.
The JTAPI signaling from the Cisco CallManager is correctly marked with a differentiated services code point (DSCP) value of CS3.
The JTAPI signaling protocol uses TCP port 2748. Dedicate 20 kbps per Cisco Unity Express site for this traffic.
All Real-Time Protocol (RTP) audio traffic from the Cisco Unity Express AIM or the IP phone is correctly marked with a DSCP value of 0xEF.
This example shows a sample configuration for this on the router where a.b.c.d is the IP address of the Cisco Unity Express AIM:
access-list 101 permit tcp host a.b.c.d any eq 2748 ! class-map match-all cti-qbe match access-group 101 ! policy-map cti-qbe class cti-qbe set dscp af31 bandwidth 20 ! interface Serial0/1 service-policy output cti-qbe
Why CUE integration with CCM doesn’t need MWI numbers?
Posted: July 20, 2006 by cciestudy in Cisco Unity ExpressSince Cisco Unity Express uses JTAPI to interface with the Cisco CallManager, no MWI on and off numbers need be configured.
EAG is implemented as a system script called checkaltgreet.aef. To activate EAG is a script, insert a Call Subflow checkaltgreet.aef into the script.
EAG is activated by the existence or absence of the greeting. It is stored as AltGreeting.wav. If this file exists on the system, the EAG is automatically activated and the special greeting is played whenever the script directs it to be played.
The number to call to record the greeting (to turn on the EAG feature) or to delete it is the Cisco UE pilot number associated with greeting management system (GMS).
Create a Public distro list and assign it a number.
You can choose individual phones as members of the PDL.
Or you can create a Group, make this group a member of the PDL and make individual phones as members of the group. You can also add a GDM as a member.
MWI doesn’t seem to work for the line appearance for GDM extensions.
CUE autoattendant – how to dial an extension at any time during the prompt?
Posted: July 14, 2006 by sankar in Cisco Unity ExpressThe following script allows to dial an extension at any time during the prompt, along with giving an option to press 1 or 2 to do different tasks (like dial by name or reach an operator etc).
CME-CUE on two different routers
Posted: July 14, 2006 by cciestudy in Callmanager Express, Cisco Unity ExpressIf you need to integrate CME with a CUE on a different chassis, you would need the following.
1. Create a h323 dial-peer in CME gateway and point it to the gateway that hosts CUE.
2. Create a SIP dial-peer in the gateway that hosts CUE to point to CUE ip address.
You cannot create a SIP dial-peer on the CME gateway and point it directly to the CUE IP Address.
How to mark traffic coming in and out of CUE ?
class-map match-all SipControl
match protocol sip
!
policy-map SipMark
class SipControl
set ip dscp cs3
class class-default
interface Service-Engine0/0
ip unnumbered FastEthernet0/0
ip route-cache flow ——————————This command was used solely for the purpose of Netflow. Nothing to do w/marking
service-module ip address 10.1.11.5 255.255.255.0
service-module ip default-gateway 10.1.11.4
service-policy input SipMark ——————– This policy map marks packets from CUE to CME (thats why its applied inbound). Thanks to Ted for correcting this. I had applied this service policy outbound in my earlier post
dial-peer voice 20000 voip
destination-pattern 3600
session protocol sipv2
session target ipv4:10.1.11.5
dtmf-relay sip-notify
codec g711ulaw
ip qos dscp cs3 signaling — This marks SIP control packets going from CME to CUE.
no vad
Netflow output:
Protocol Total Flows Packets Bytes Packets Active(Sec) Idle(Sec)
——– Flows /Sec /Flow /Pkt /Sec /Flow /Flow
TCP-WWW 447 0.0 25 335 0.0 0.4 1.8
UDP-NTP 218 0.0 1 76 0.0 0.3 15.4
UDP-other 34 0.0 260 201 0.0 9.1 15.5
ICMP 25 0.0 16 217 0.0 21.8 15.5
Total: 724 0.0 28 273 0.0 1.6 7.0
SrcIf SrcIPaddress DstIf DstIPaddress Pr SrcP DstP Pkts
Se0/0 10.1.11.5 Local 10.1.11.4 01 0000 0303 6
Se0/0 10.1.11.5 Local 10.1.11.4 01 0000 0303 27
Se0/0 10.1.11.5 Local 10.1.11.4 11 8003 42BC 901
Se0/0 10.1.11.5 Local 10.1.11.4 11 13C4 13C4 3
10.1.11.5 is CUE, 10.1.11.4 is CME. Port 13C4 in hex translates to 5060. THis is the port on which SIP protocol listens for control packets.
Output of show policy-map interface
PSTNSw#sh policy-map int
Service-Engine0/0
Service-policy input: SipMark
Class-map: SipControl (match-all)
7 packets, 2911 bytes
5 minute offered rate 0 bps, drop rate 0 bps
Match: protocol sip
QoS Set
dscp cs3
Packets marked 7
Class-map: class-default (match-any)
125 packets, 51142 bytes
5 minute offered rate 0 bps, drop rate 0 bps
Match: any
PSTNSw#
PSTNSw#
1. Call redirect
This step does a release to switch type of transfer. Eventhough it has four options (Successful, Busy, Invalid, Unsucessful), none of the steps under these options take effect.
CUE – CME – CCM Integration
Posted: May 2, 2006 by cciestudy in Callmanager Express, Cisco Unity Express, Media ResourcesWhen CME and CUE are integrated and you place a call from a CCM to CME using G729 and the call rolls over to CUE, then you need to have transcoding enabled in CME for this to work. This is needed because CME is controlling the call leg to CUE via SIP. There is no point in having transcoder at CCM and this will not be used.
CUE understands only G711 codec.
If you want to use g729 over the WAN, you need to setup transcoders for the CME.
In CME, each call takes 2 transcode session (one for each leg of the call). So if you setup 2 sessions, you can place only 1 call.
###Activate DSP farm services####
voice-card 0
dspfarm
dsp services dspfarm
### Enable SCCP and select the interface that will be used to register
sccp local FastEthernet0/0
sccp ccm 10.1.11.4 identifier 1
sccp
!
##Define DSP profiles, use mac-address of the interface selected above.
sccp ccm group 1
associate ccm 1
associate profile 1 register mtp0013c3443150
!
dspfarm profile 1 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec gsmfr
codec g729r8
maximum sessions 2
associate application SCCP
###Tell CME To use this trancoder. Note that the units command has to be specified before the tag command. With out the sessions command transcoding will not work. All 3 commands below are necessary.
telephony-service
sdspfarm units 1
sdspfarm transcode sessions 4
sdspfarm tag 1 mtp0013c3443150
PSTNSw#sh sccp
SCCP Admin State: UP
Gateway IP Address: 10.1.11.4, Port Number: 2000
IP Precedence: 5
User Masked Codec list: None
Call Manager: 10.1.11.4, Port Number: 2000
Priority: 1, Version: 3.1, Identifier: 1
Transcoding Oper State: ACTIVE – Cause Code: NONE
Active Call Manager: 10.1.11.4, Port Number: 2000
TCP Link Status: CONNECTED, Profile Identifier: 1
Reported Max Streams: 4, Reported Max OOS Streams: 0
Supported Codec: g711ulaw, Maximum Packetization Period: 30
Supported Codec: g711alaw, Maximum Packetization Period: 30
Supported Codec: g729ar8, Maximum Packetization Period: 60
Supported Codec: g729abr8, Maximum Packetization Period: 60
Supported Codec: gsmfr, Maximum Packetization Period: 20
Supported Codec: g729r8, Maximum Packetization Period: 60
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
PSTNSw#sh sdspfarm sessions
Stream-ID:1 mtp:1 10.1.11.4 19468 Local:2000 START
usage: Ip-Ip
codec:G729 duration:20 vad:0 peer Stream-ID:2
Stream-ID:2 mtp:1 10.1.11.4 19552 Local:2000 START
usage: Ip-Ip
codec:G711Ulaw64k duration:20 vad:0 peer Stream-ID:1
Stream-ID:3 mtp:1 10.1.11.4 16748 Local:2000 START
usage: Ip-Ip
codec:G729 duration:20 vad:0 peer Stream-ID:4
Stream-ID:4 mtp:1
a. Define a local location ID (Define a location number, Location name and abbreviation ID.Specify the Domain name/IP address to be ip address of local CUE module Phone prefix and VPIM Broadcast ID are not required for CUE to CUE networking.
b. Voicemail encoding may be set to dynamic. If a CUE detects that the destination is another CUE It uses g711. If its Unity, it uses g726.
c. Make sure Networking is Enabled for this location by checking the "Enabled" checkbox.
d. Add a remote location ID (Define a location number, Location name and abbreviation ID.Specify the Domain name/IP address to be ip address of remote CUE module Phone prefix and VPIM Broadcast ID are not required for CUE to CUE networking.
e. Add a remote user who falls in the remote location.
1. To send a message to a remote user, you need to enter the remote location ID followed by the extension of the remote user. (Blind addressing)
If the location ID is 200 and the user is at extension 1000, then you need to enter 2001000
2. Abbreviation defined for a location is used when playing back a message (as part of the envelope). It cannot be used to address messages.
3. Broadcast message can be send between locations. The MWI for broadcast message depends on the settings at the remote location CME.
4. When Spoken Name Caching is enabled, the system will cache the spoken name of the remote user when it receives a message from the remote user and will use it for future directory searches.
5. You can record spoken names for remote locations.
To do a live record, conference the call with Unity Express.
1. Hit Conf key
2. Dial your own extension so that the call will be forwarded to your own mailbox or Dial the extension of a general mailbox.
3. Hit Conf key again
Transfering a call directly to a Subscriber’s mailbox
Posted: April 29, 2006 by sankar in Cisco Unity ExpressIf your users fall in the range 3XXX, create an ephone-dn as follows.
ephone-dn 1
number 13…
call-forward-all 8888 <— forward all calls to voicemail.
Set the E164 number on every mailbox in CUE to 13XXX (where XXX is replaced by the appropriate users extension)
When the receptionist receives a call for extension 3001, she will need to transfer to 13001 to send the call directly to 3001's mailbox.
For more info, visit http://www.cisco.com/en/US/partner/products/sw/voicesw/ps5520/products_tech_note09186a00802ab979.shtml
CUE uses the last redirected number to send the call to the correct voicemail box.
X 2000 CFA to X 2001 and X 2001 CFNA to Voicemail
When you call X 2000 and it rolls over to voicemail, you will hear the greeting for X 3001. This is the opposite of what happens with Unity.
Note the Diversion header in "Debug ccsip messages" command:
INVITE sip:8888@10.1.11.5:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.11.4:5060;branch=z9hG4bK4D6ED7
Remote-Party-ID: <sip:2007@10.1.11.4>;party=calling;screen=no;privacy=off
From: <sip:2007@10.1.11.4>;tag=7D469D78-79A
To: <sip:8888@10.1.11.5>
Date: Wed, 03 May 2006 21:32:39 GMT
Call-ID: 2E82CA5F-DA2311DA-8673A220-2692CE36@10.1.11.4
Supported: 100rel,timer,resource-priority
Min-SE: 1800
Cisco-Guid: 780283447-3659731418-2255594016-647155254
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIF
Y, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1146691959
Contact: <sip:2007@10.1.11.4:5060>
Call-Info: <sip:10.1.11.4:506
Diversion: <sip:2001@10.1.11.4>;privacy=off;reason=no-answer;counter=2;screen=noExpires: 180
This does not apply for call transfers.
Language customization only affects the voicemail and AA system prompts. The GUI, CLI and system monitoring and debugging tools, such as log file messages, are always supported in English only.
The CUE GUI login username is case sensitive. You cannot enter your username in CAPS, while your password in Lower case.
CUE describes COS as Capabilities (unlike Unity)
There are 5 COS settings –
a. Super User – Administrator rights to the CUE system
b. Voicemail Broadcaster – Capability to send broadcast
c. Private List Viewer – Ability to view other user's private lists (cannot modify them)
d. Public List Manager – Ability to manage public distro lists (can modify them)
e. AVT – Administration via Telephone – This gives rights to modifying Prompts, greetings via GMS.
DLists can be of two types.
a. Public Distro List
b. Private Distro List
A member of a PDL can be another PDL, GDM or a regular mailbox.
To send a message to a PDL, you have to signin and select option 2. You cannot send it through an AA script.
A public list is created by the admin for systemwide use. There is a default Public list called Everyone that is generated by CUE system. This list is automatically populated every time a new user is added. You cannot edit or delete the Everyone public list. You can define up to 15 public distro. lists. You can define a total of 1000 distribution list entries per CUE system. Each PDL can be assigned an extension, so that people may send a message to the PDL by dialing the number.
A private list can be administered only by the subscriber himself. YOu can create up to 5 private lists. Each private list can be assigned a single digit extension (1 through 5). You may use this number to send a message to the private list. If a user is deleted from the system his private lists are also deleted. You can have up to 50 private distribution list entries per user.
A user can be given voicemail broadcast rights. He will need to subscribe to a group (newly created one or the system default Broadcasters group). Set the Voicemail broadcaster option checked in the group setting. To send a voicemail broadcast, you will have to call in to the prompt management number (GMS) and select option 3 to administer a voicemail broadcast. Choose option 1 to send voicemails to users in the local CUE module.
To enable MWI for Broadcast messages
Defaults > Voicemail > Use MWI for broadcast messages
Broadcast messages are not stored as seperate messages on each mailbox. It is stored only once and all users will have a reference to that message. A user cannot delete a Broadcast message, but can save or skip the message.
Prompt Management Application does two things
1. Administration via Telephone:
2. Voice Mail Broadcaster:
To get access to Administration via Telephone, you have to dial the number for "Prompt Management" Application.
This can be found at
Administration > Call-in Numbers
To login to Administration via Telephone, you need to be a member of a group that has Administration via Telephone capability. Use your own mailbox id and pin to login
1. Configure > Groups > Administration via Telephone
1. Defaults > Mailbox > Maximum Caller Message Size
Voicemail > Mailbox > Maximum Caller Message Size
Applies to voice messages left by external callers or subscribers when they get forwarded to voicemail.
2. Defaults > Voicemail > Maximum subscriber recording size
Applies when subscribers signin and then try to send a message
CUE stores messages and greeting in G711 format.
1 min of stored voice = 60 seconds * 64000 bits = 3840 Kb = 480 KB
On a AIM with 1 Gig Flash, 400 Meg will used for storing messages and 600 Megs used for OS, scripts, greetings etc
Primary E.164 number can be used as equivalent to Alternate Extension of Unity
A mailbox associated with a group is called a General Delivery Mailbox.
A group member can log into the GDM and manage its voice message content
A group owner can make changes to the group membership
You need to be a mailbox user before you can login to a GDM, because GDM option (option 9) is available only after you signin to your mailbox.
CME – CUE – MWI setting
Posted: April 27, 2006 by cciestudy in Callmanager Express, Cisco General, Cisco Unity ExpressFor CUE you need to define mwi on and off as two different ephone-dn's. You cannot enable the mwi on-off using the same ephone-dn like you do with Unity.
Set up the mwi number following by dots equal to the extension length
ephone-dn 10
number 1599….
mwi on
ephone-dn 11
number 1598….
mwi off
The default ccn application for mwi uses the extensions 8000 for ON and 8001 for OFF. You can change this from CLI
ccn application ciscomwiapplication
description "ciscomwiapplication"
enabled
maxsessions 4
script "setmwi.aef"
parameter "strMWI_OFF_DN" "1598"
parameter "strMWI_ON_DN" "1599"
parameter "CallControlGroupID" "0"
end application
1. When the extension appears on line 1, red lamp MWI and envelope is used.
2. When the extension appears on any line other than line 1, only envelope is used.
Unity Express – Changing IP Address
Posted: April 27, 2006 by cciestudy in Cisco General, Cisco Unity ExpressAfter changing the IP Address in the interface service-module, calls to voicemail are getting answered, but callers does not hear anything.
The show call active voice brief shows that the call leg to unity express is getting connected to the old IP address.
Resetting the CUE fixed the issue.
CCIE Voice notes 