IPMA

IPMA

IPMA has two modes: proxy and shared.

Proxy Mode – Assistant and manager have two different DN’s.  A CTI route point is needed so that calls for the manager can be intercepted.  The CTI RP should have the same DN as the manager. CSS’es are used to direct traffic properly.

IPMA Proxy Line Mode

Create partition and CSS’s
It is very important that manager DNs are unreachable from all devices except the CUCM assistant RP and the managers’ proxy line on the assistant phone.  The assistant the CTI RP should be reachable for all other DN’s.

Call flow

Two partitions are required:

1       Everyone – (or whatever other PT has access to all DN’s and route patterns)
2       Manager-pt –  The manager, assistant and the CTI RP

One  CSS  is required

CSS-I-E – CTI RP and everyone

  • Configure the CTI route point
  • Add the partition of  the manager line to the CSS of the MWI
  • If using intercom, add the intercom partition, CSS, DN and translation pattern.
  • Stop/start the IPMA service
  • Add the service and phone button templates

·    http://<ipaddress&gt;:8080/ma/servlet/MAService?cmd=doPhoneService&Name=#DEVICENAME#

  1. Configure the phones
  • Assign softkey templates.  Add proxy line to assistant phone
  1. Configure the assistant application
  • Create a new manager.
  • Configure lines for manager.
  • Assign an assistant to a manager.
  • Configure lines for the assistant.
  • Configure intercom lines (optional).
  1. Configure the dial rules for the assistant
  2. Install the Assistant Console application

IPExpert (with EM)

Create manager pt and css
Create Phone services (both IPMA and EM
Create two new phone button templates for the intercoms (SCCP and SIP)
Create the CTI route point
Edit the service parameters for the IPMA (set the IP addresses mostly)
Restart the IPMA service
Create an intercom PT/CSS
Create an intercom DN

Extension Mobility

  • Ensure that the service is active
  • Create the EM service from phone services

URL: http://<IP Address of Extension Mobilityserver>:8080/emapp/EMAppServlet?device=#DEVICENAME#

  1. Create a default device profile for each phone type (not always necessary)
  2. Create the device profile for a user (device, device settings, device profile)
  3. Associate a user device profile to a user (user management, end user, extension mobility controlled profiles)
  4. Configure and subscribe the phone and user device profile to extension mobility (enable EM on phone and subscribe).
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by gmalcolm on November 25, 2011  •  Permalink
Posted in CUCM

Posted by gmalcolm on November 25, 2011

https://voice411.wordpress.com/2011/11/25/ipma/

CUC Notes

Unity Notes

CUC Admin guide is very helpful

URL that is available during the lab (support, voice,unity connection, 7.x SCCP integration)

http://www.cisco.com/en/US/docs/voice_ip_comm/connection/7x/integration/cucm_sccp/guide/cucintcucmskinny.html

Integration

Run the voicemail wizard in CM

  1. Create the name of the system
  2. Add the number of ports
  3. Configure the device information
  4. Configure the directory numbers
  5. Configure line groups
  6. Using the wizard, configure the hunt list
  7. Add a hunt pilot with the vm pilot number
  8. Configure the voice mail pilot
  9. Configure the MWI
  10. Add the voicemail profile.
  11. Make sure that the AXL application user exists with super user rights (admin)
  12. If using other than the default vm profile, make sure that the DNs have the correct vm profile.

Configure Unity

  1. Make sure all needed services are running
  2. Add a new port group under telephony integrations – device name prefix is name from UCM (CiscoUM1-VI w/o number on the end)
  3. Add ports Check telephony integration
  4. Configure AXL servers under phone system if you want to be able to import users

Add Subscribers

  1. Go to users, users. Click new user from template.  Be sure to include the extension number
  2. On the CCM server go to device phone click on the directory number configuration link and set the forward to voicemail for the busy and no answer checkboxes.  This is a very important step.

Mailbox Settings

Trivial Passwords – System Settings –> Authentication Rules –> check for trivial passwords

Users

Users can be imported from CUCM if AXL is ocnfigured.  Otherwise, users can be created within CUC.

Create Call Handler and Distribution List

URL available during the lab

http://www.cisco.com/en/US/docs/voice_ip_comm/connection/7x/administration/guide/7xcucsagx.html

  1. On the CCM server go to device CTI route point create a new route point.
  2. Click the add DN link and create a DN for calls to the extension. Ensure the DN is active and that all calls are forwarded to voicemail.
  3. On the unity server navigate to distribution lists system distribution list click add new and create a new distribution list for the phones.
  4. On the distribution list configuration page click edit distribution list members add the new user(s) here.
  5. On the unity server navigate the call management system call handlers click add new and energy extension. Configure the schedule as needed.
  6. On the call handler configuration page click edit caller input click one for key one and edit the settings to transfer the call to the phone numbers’ mailbox.

Auto Attendant

  1. Create a CTI route point in CUCM
  2. Modify the call handlers as needed under call management

Live Record

On CUC

  1. Add a new forwarded routing rule
  2. Set conversation to start live record
  3. Under routing rules condition, add new
  4. on the new forwarded routing rule condition page, click dialed number.  Select equals and enter the number of the pilot.

On CUCM

There are at least two ways to accomplish

#1

  1. Add a new DN for the pilot number of live record.
  2. In the forward all field, enter the voice mail pilot number

#2

  1. Add a CTI route point with the pilot number of live record
  2. Add CFA to point to pilot

Testing/Verification

Broadcasts

Configure the user so that they can send and update broadcasts

CUC to CME Integration Using SIP

  1. Add the phone system in CUC
  2. Add a port group.  Set the SIP security profile an the SIP transport.  The contact line name should be the pilot.
  3. Edit servers and ass the CME using ports 5060 and 5062.
  4. Configure CME:

dial-peer voice 3600 voip
destination-pattern 3600
session protocol sipv2
session target ipv4:10.10.210.12
dtmf-relay rtp-nte
codec g711ulaw
! pilot
vocie register global
voicemail 3600
!mwi
voice register global
mwi reg-e164
create profile
sip-ua
mwi-server ipv4:10.10.210.13 unsoliscited
telephony-service
mwi relay
voice register

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by gmalcolm on November 12, 2011  •  Permalink
Posted in CUC

Posted by gmalcolm on November 12, 2011

https://voice411.wordpress.com/2011/11/12/cuc-notes/

RSVP

URL for RSVP that will be available during the lab:

http://www.cisco.com/en/US/partner/docs/voice_ip_comm/cucm/srnd/7x/cac.html#wp1043984

RSVP configuration is made up of two pieces; CUCM and the IOS router

IOS Router
The router must first be configured as an MTP agent to connect to CUCM.  All routers participating in RSVP must register as MTP’s.  The router appears as a SCCP deivce to CUCM.

Here is an example configuration:

voice-card 0
 dsp services dspfarm
!
sccp local f0/0.240   
sccp ccm 10.10.210.11 identifier 1 priority 1 version 5.0.1
sccp ccm 10.10.210.10 identifier 2 priority 2 version 5.0.1
sccp                          
!
sccp ccm group 1
 bind interface f0/0.240
 associate ccm 1 priority 1
 associate ccm 2 priority 2
 associate profile 2 register br1-rsvp-agent 
 switchover method immediate
 switchback method guard timeout 7200
!
dspfarm profile 2 mtp
 codec pass-through
 no codec g711u
! this codec is used if the pass through codec negotiation fails
 codec g729r8         
 rsvp                         
 maximum sessions software 100
 associate application SCCP
 no shut

Next the router interface needs to be configured for RSVP

interface Serial0/1/1.1 point-to-point
  ip rsvp bandwidth 64

CUCM

  • Create a MTP (use the profile name from the CCM group)
  • Assign the MTP to a MRG
  • Assign the MRG to a MRGL
  • Assign the MRGL to a device pool
  • Create locations – set RSVP setting to mandatory.  Note that the hub should NOT have RSVP set to mandatory
  • Assign locations to the device pools

Verification/Troubleshooting

Make sure that routing is using the interfaces that you have specified the RSVP bandwidth on.

Show ip rsvp reservation
Debug ip rsvp resv

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by gmalcolm on November 12, 2011  •  Permalink
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Posted by gmalcolm on November 12, 2011

https://voice411.wordpress.com/2011/11/12/rsvp/

SRST

SRST

SRST is made up of a configuration on CUCM and CME.  The CUCM portion is simply configuring a SRST reference from the System, SRST page and setting this reference on the device pool.  The remainder of this document is for the CME configuration.

There are 2 modes supported; Call-manager fallback and Telephony service

Call-Manager fallback

It is very simple but limited.  There is no access to features such as call park, calling name, etc. Example

Call-manager-fallback

ip source-address 1.1.1.1 port 2000

Max-ephones 10

Max-dn 20

 

The ephones and ephone-dn’s will not appear in the configuration.  They can be modified however

Voice-port 50/0/1

Station-id name 3214-3001

Station-id number 3001

 

Dial-peer voice 20001 pots

Destination-pattern 3001

Port 50/0/0/1

Cor …

Telephony service

Telephony service provides access to all the features in CME.  Example:

Telephony-service

!determines what shows up in the config – none is similar to call-manager-fallack

srst mode auto-provision all | dn | none

srst dn line-mode octo

srst dn template <tag>

srst ephone template <tag>

dn-webedit

transfer-system full-consult

create cnf

For MGCP

ccm-manager fallback-mgcp

For H323

There might be overlap with the dial-peers (if you use ‘all’ dial-peers exist all the time).

  1. Change preference of the POTS Ephone dial-peer to a max of 9
  2. Change the behavior   to use preference instead of longest match – ‘dial-peer hunt 2’

 

Verification

Show call-manager fallback voice-port

Show call-manager fallback dial-peer

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by gmalcolm on November 3, 2011  •  Permalink
Posted in CME, CUCM

Posted by gmalcolm on November 3, 2011

https://voice411.wordpress.com/2011/11/03/srst/

IOS Voice Router Notes

CME Basic Configuration

  1. Use the telephony – service script to set up the basic configuration in call manager express

Here is a list of some of the items that can be set

  • The source IP address
  • Time Zone and date format
  • The web administrator username and password
  • The configuration files can also be created

Phone Registration and Number Assignment (SCCP)

  1. DHCP – remember to set option 150 – SCCP and 66 – SIP
  2. Create the ephone, add the MAC address and the buttons for the ephone-DN
  3. Add the Ephone-DNs

ISDN PRI

isdn switch-type primary-ni
network-clock-participate wic 0

controller T1 0/0/0
pri-group timeslots 1-3,24

dial-peer voice 1 pots
incoming called-number .
direct-inward-dial
port 0/0/0:23

Telephony-service

dialplan-pattern 1 21313… extension-length 4

MGCP

controller T1 0/0/0
pri-group timeslots 1-3,24 service mgcp

interface Serial0/0/0:23
isdn bind-l3 ccm-manager

ccm-manager redundant-host 10.10.210.10
ccm-manager mgcp
ccm-manager fax protocol cisco
!
mgcp
mgcp call-agent 10.10.210.11 service-type mgcp version 0.1
mgcp dtmf-relay voip codec all mode out-of-band
mgcp fax t38 ecm
mgcp bind control source-interface Loopback0
mgcp bind media source-interface Loopback0

IOS Call Routing

IOS call routing can utilize class of restrictions (COR) to determine dialing patterns.  Class of restrictions can then be assigned to dial peers to restrict access.  For the lab, there are two types of dial peers – POTS and VOIP.  POTS dial peers must point to a voice port.  COR on CME uses a lock and key approach.  The key is the INCOMING corlist on the ephone and the lock is the outgoing corlist on the dial-peer.

Create Class of Restrictions (base and premium)

dial-peer cor custom
name DEFAULT
name PREMIUM

dial-peer cor list PT-DEFAULT
member DEFAULT

dial-peer cor list PT-PREMIUM
member PREMIUM

dial-peer cor list CSS-DEFAULT
member DEFAULT

dial-peer cor list CSS-PREMIUM
member PREMIUM

Create dial-peers with destination patterns

dial-peer voice 999 pots
corlist outgoing PT-DEFAULT
destination-pattern 999
port 0/0/0:23
forward-digits 9

dial-peer voice 900 pots
corlist outgoing PT-DEFAULT
destination-pattern 9[1-9]…….$
port 0/0/0:23

dial-peer voice 901 pots
corlist outgoing PT-DEFAULT
destination-pattern 90[1-9]T
port 0/0/0:23
prefix 0

dial-peer voice 902 pots
corlist outgoing PT-PREMIUM
destination-pattern 900T
port 0/0/0:23
prefix 00

Assign incoming COR to EPHONE DN

Ephone-dn 1
Corlist incoming CSS-DEFAULT

Voice Translation Rules

Voice translation rules are typically made up of three parts. The first part is to create a translation rule, which uses regular expressions to perform digit manipulations. Then a translation profile is created that translates either a caller or called pattern.  Finally the translation profile is applied to a dial peer.

1a. voice-translation rule 1
rule 1 /123/ /999/
! matches 123 and replaces with 999

1b. voice translation-rule 2
rule 1 /^7…\(5555\)/ /90044232141\1/
! starting with 7 then any 3 digits, keep 5555

2.  voice translation-profile 123-999
translate called 1

3.  dial-peer voice 123 pots
translation-profile outgoing 123-999
destination-pattern 123
port 0/0/0:23

Shared line

A shared line is simply making the same ephone DN appear on both phones by using same button number.

Night service

The night service feature allows for notifications to be sent to another destination during a predefined time period.

Telephony –service
night-service code *12345
night-service everyday 18:00 09:00

ephone-dn 1  dual-line
night-service bell

After hours Call Blocking

telephony-service
after-hours block pattern 1 900
after-hours day Sun 18:00 09:00

To make a phone exempt.  Individual DNs and dial peers can also be exempted

voice register pool  1
after-hour exempt

Single number reach (SNR)

Single number reach is a new feature introduced in CME 7.1. This feature allows for a single phone number used for multiple calling destination as a users desk phone, cell phone, and home phone. In addition, this feature offers the ability to seamlessly transfer a call to multiple destinations.

Ephone-template  1
softkeys idle  Dnd Gpickup Pickup Mobility
softkeys connected  Endcall Hold LiveRcd Mobility

ephone-dn  1  dual-line
mobility
snr 921415555 delay 5 timeout 15 cfwd-noan 3002

ephone  1
ephone-template 1

Softkey Customization – SCCP
Use the Ephone template and apply the template to the ephone

Ephone-dn  10  octo-line
number 3020

Extension Mobility

ip http server

Telephony-service
url authentication http://10.10.202.1/CCMIP/authenticate.asp
! for 9.x firmware and above
service phone webAccess 0

Enabling the phone for extension mobility

ephone x
logout profile 1

ext mobility with logout profile

voice user-profile 1
user br2ph3 password adgjm
number 3102 type normal
speed-dial 1 3006
pin 1234

voice logout-profile 1
pin 1234
number 3002 type normal

Conference resources

Create the DSP farm under the voice card

voice-card 0
dsp services dspfarm

Configure the SCCP services

sccp local FastEthernet0/0.311
sccp ccm 177.3.11.1 identifier 1 version 7.0
sccp
sccp ccm group 1

associate ccm 1 priority 1
associate profile 1 register R3_CONF

Configure the dspfarm profile

dspfarm profile 1 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 2
associate application SCCP

Configure the appropriate settings under the telephony service

telephony-service
sdspfarm units 2
sdspfarm tag 1 R3_CONF
conference hardware

Transcoding resources

voice-card 1
dsp services dspfarm

Configure the SCCP services

sccp local FastEthernet0/0.400
sccp ccm 10.10.110.3 identifier 1 version 7.0
sccp
sccp ccm group 1
associate ccm 1 priority 1
associate profile 1 register br2-xcoder

dspfarm profile 1 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 2
associate application SCCP
no shut

B-ACD

Ephone hunt groups

ephone-hunt 1 longest-idle
pilot 3000
list 3001,3005
final 3002

application
service aa flash:bacdprompts/app-b-acd-aa-2.1.2.3.tcl
paramspace english index 1
param number-of-hunt-grps 2
param handoff-string aa
paramspace english language en
param max-time-vm-retry 2
param aa-pilot 3500
paramspace english location flash:bacdprompts/
param second-greeting-time 60
param welcome-prompt en_bacd_welcome.au
param call-retry-timer 15
param voice-mail 53002
param max-time-call-retry 90
param service-name queue
application
service queue flash:bacdprompts/app-b-acd-2.1.2.3.tcl
param queue-len 15
param aa-hunt10 3006
param queue-manager-debugs 1
param aa-hunt2 3210
param number-of-hunt-grps 2
service aa-drop flash:bacdprompts/app-b-acd-aa-2.1.2.3.tcl
param number-of-hunt-grps 1
paramspace english index 1
param handoff-string aa-drop
param drop-through-option 2
paramspace english language en
param max-time-vm-retry 2
param aa-pilot 3501
paramspace english location flash://bacdprompts/
param second-greeting-time 60
param welcome-prompt en_bacd_welcome.au
param call-retry-timer 15
param voice-mail 53002
param max-time-call-retry 90
param service-name queue

SRST

SCCP SRST

Call-manage- fallback
Max-conferences 8 gain –6
Transfer-system full-consult
Ip source-address 10.10.210.1 port 2000
Max-ephones 1
Max-dn 1 dual-line

MGCP Fallback
ccm-manager fallback-mgcp

SRST Dialplan
voice translation-rule 1
rule 1 /617863\(1…\)$/ /\1/

voice translation-rule 7
rule 1 /^1…$/ /863/

voice translation-rule 10
rule 1 /^1…$/ /617863/!

voice translation-profile 10digit
translate calling 10

voice translation-profile 7digit
translate calling 7

voice translation-profile strip-dnis
translate called 1

voice-port 0/0/0:23
translation-profile incoming strip-dnis

SRST Class of Restriction

dial-peer cor custom
name pt-internal
name pt-loc-ld
name pt-block

dial-peer cor list css-internal
member pt-internal

dial-peer cor list css-ld
member pt-internal
member pt-loc-ld

dial-peer cor list css-block
member pt-block

dial-peer voice 2 pots
incoming called-number .
direct-inward-dial

dial-peer voice 5000 pots
corlist outgoing css-internal
destination-pattern 5…
no digit-strip
port 0/0/0:23
prefix 212394

dial-peer voice 3000 pots
corlist outgoing css-internal
destination-pattern 3…
no digit-strip
port 0/0/0:23
prefix 011343214

dial-peer voice 911 pots
corlist outgoing css-internal
translation-profile outgoing 7digit
destination-pattern 911
no digit-strip
port 0/0/0:23

dial-peer voice 7 pots
corlist outgoing css-ld
translation-profile outgoing 10digit
destination-pattern [2-9]……
forward-digits 7
port 0/0/0:23

dial-peer voice 10 pots
corlist outgoing css-ld
translation-profile outgoing 10digit
destination-pattern [2-9]..[2-9]……
forward-digits 10
port 0/0/0:23

call-manager-fallback
cor incoming css-internal default
cor incoming css-ld 1002

RSVP

! set the dspfarm
voice-card 0

dsp services dspfarm

! config the sccp interface and set CCM
sccp local FastEthernet0/0.20
sccp ccm 10.10.210.10 identifier 1 version 5.0.1
sccp

! create a dspfarm profile
dspfarm profile 1 mtp
codec 729r8
codec pass-through
rsvp
maximum sessions software 4
associate application SCCP

! associate the dspfarm profile to CCM
sccp ccm group 1
associate profile 1 register hq-rsvp-agent

MOH

!  To invoke DSP so that multicast streams can be converted to TDM to be sent  to PSTN

ccm-manager music-on-hold

To run MOH locally from CME for fallback
ccm-manager-fallback
moh “file”
multicast moh 239.1.1.1 16384 10.10.201.1 10.10.110.2

Gatekeeper Notes

GK address resolution on ARQ
GW to GW with local zone example:

HQ

gatekeeper
zone local UCM ipexpert.com 10.10.110.1
zone prefix UCM 3* gw-priority 10 10.10.110.3
gw-type-prefix 1#* default-technology
no shutdown

dial-peer voice 3000 voip
destination-pattern 3…$
voice-class codec 1
voice-class h323 1
session target ipv4:10.10.110.3
no vad

dial-peer voice 1500 voip
destination-pattern [15]…$
voice-class codec 1
voice-class h323 1
session target ipv4:10.10.210.11
no vad

UCM

GK and h225 trunk configured
Route pattern 3XXX pointing to GK

BR2

interface Loopback0
ip address 10.10.110.3 255.255.255.255
ip ospf network point-to-point
h323-gateway voip interface
h323-gateway voip id UCM ipaddr 10.10.110.1 1719
h323-gateway voip h323-id BR2-RTR
h323-gateway voip tech-prefix 1#
h323-gateway voip bind srcaddr 10.10.110.3

dial-peer voice 2 voip
voice-class codec 1
voice-class h323 1
incoming called-number .
no vad

dial-peer voice 1500 voip
destination-pattern [15]…$
voice-class codec 1
voice-class h323 1
session target ras
no vad

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by gmalcolm on September 28, 2011  •  Permalink
Posted in CME

Posted by gmalcolm on September 28, 2011

https://voice411.wordpress.com/2011/09/28/ios-voice-router-notes/

Gatekeepers and CUBE

Overview

Gatekeeper

A Gatekeeper is a H.323 element that is used to group gateways into logical zones and perform call routing between them. Gateways are responsible for edge routing decisions between the Public Switched Telephone Network (PSTN) and the H.323 network. Cisco gatekeepers handle the core call routing among devices in the H.323 network and provide centralized dial plan administration.

Zone and Technology Prefixes

Zone Prefixes

A zone prefix is the part of the called number that identifies the zone to which a call hops off. Zone prefixes are usually used to associate an area code to a configured zone.

The Cisco gatekeeper determines if a call is routed to a remote zone or handled locally. For example, according to this configuration excerpt, gatekeeper (GK) A forwards 214……. calls to GK-B. Calls to area code (512) are handled locally.

gatekeeper

zone local GK-A abc.com

zone remote GK-B abc.com 172.22.2.3 1719

zone prefix GK-B 214…….

zone prefix GK-A 512…….

Technology Prefixes

Cisco gatekeepers use technology prefixes to route calls when there is no E.164 addresses registered (by a gateway) that matches the called number. In fact, this is a common scenario because most Cisco IOS gateways only register their H.323 ID (unless they have Foreign Exchange Station (FXS) ports configured). Without E.164 addresses registered, the Cisco gatekeeper relies on two options to make the call routing decision:

  • With the Technology Prefix Matches option, the Cisco gatekeeper uses the technology prefix appended in the called number to select the destination gateway or zone.
  • With the Default Technology Prefixes option, the Cisco gatekeeper assigns default gateway(s) for routing unresolved call addresses. This assignment is based on the gateways’ registered technology prefix.

Some notes on call routing decision process

  1. For local zones, if there is no tech-prefix or default tech-prefix configured, the target must be registered or the call will fail.
  2. For remote zones, if there is a tech-prefix but no default tech-prefix the call will fail unless hopoff is configured.  Even if the zone prefix matches, the GK will see the tech-prefix as a part of the called number.  The two GKs are not aware of the tech-prefixes configured on the opposing GK.  The format of this command is:

Gw-type-prefix 1# hopoff <gatekeeper hostname>.

There are four ways that call can be routed locally:

Hop-off prefix match

Tech prefix match and local Gateway found

Target registered

Default tech-prefix and local Gateway found

Address Resolution

 

CUBE

Cube is used to terminate and re-originate calls.  The following connection types are supported:

  • H.323 to H.323
  • H.323 to SIP
  • SIP to H.323
  • SIP to SIP

A concept called VIA-ZONES is used for Gatekeepers.  A via-zone-enabled gatekeeper is capable of recognizing via-zones and sending traffic to via-zone gateways.

In the lab, it is highly likely that the GK and the CUBE will be on the same router.

Zone local …

Invia – Use if the calls come from/to a Gatekeeper (LRQ)

Outvia – Use if the calls come from/to a Gateway (ARQ)

enable-intrazone—All intra zone calls must use the via-zone gatekeeper

In essence, CUBE is just a Gateway that registers to the Gatekeeper.  Calls are terminated on the CUBE and then re-originated.  Both incoming and outgoing dial-peers must be configured.

CUBE Example
CUBE and Gatekeeper on same Gateway
interface FastEthernet0/0.20
encapsulation dot1Q 20
ip address 10.10.200.3 255.255.255.0
ip helper-address 10.10.210.11
h323-gateway voip interface
h323-gateway voip id VIA ipaddr 10.10.110.1 1719
h323-gateway voip h323-id HQ-RTR
h323-gateway voip tech-prefix 1#

dial-peer voice 3000 voip
destination-pattern 3…
session target ras
dtmf-relay h245-alphanumeric
no vad

URL available during lab:

http://www.cisco.com/en/US/partner/docs/ios/voice/cubegk/configuration/guide/12_4t/ve_12_4t_book.html

Faststart and MTP

Faststart bypasses H.245.  Similar to SIP early offer.

Inbound example:

SIP phone (CME) –> GK –>UCM – since SIP phone is doing early offer, then H225 trunk needs to be configured for inbound faststart.

Outbound faststart –

UCM –> GK(CUBE) –> ITSP – If ITSP needs SIP early offer, faststart must be configured on UCM. Must use MTP since UCM can’t do outbound faststart.

CUBE doesn’t support empty H245 capabilities set.  Uncheck H245 capabilities set in UCM.

h323
emptycapability
h225 id-passthru
h225 connect-passthru
h245 passthru tcsnonstd-passthru

*** DONT FORGET THAT A VOIP DIAL-PEER MUST BE CONFIGURED ON THE GATEWAY FOR IT TO REGISTER TO THE GATEKEEPER ***

Some sample configurations

Gatekeeper

! config a local zone named test – it will use the highest loopback if not assigned manually

zone local TEST voiplab.com

! assign a prefix to a zone and set the primary gateway use to route the calls for the prefix

zone prefix 1… gw-priority 10 10.10.210.11

! assign a secondary Gateway

zone prefix 1… gw-priority 9 10.10.210.10

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by gmalcolm on May 10, 2011  •  Permalink
Posted in Gatekeeper

Posted by gmalcolm on May 10, 2011

https://voice411.wordpress.com/2011/05/10/gatekeepers-and-cube/

Unity SIP Integration

CUC SIP

Overview

CUC normally uses SCCP.  Here are the integration steps for SIP.

CUCM

  1. Create a SIP trunk security profile
  2. Create a SIP trunk
  3. Create a route pattern for the pilot number that points to the SIP trunk.
  4. Add the voice-mail pilot
  5. Configure the voice-mail profile
  6. Create application user (optional) – can be used to import users

CUC

  1. Create a new port Group.  Telephony Integrations > phone system > new port group
  2. From the port group basics page, add servers (for SIP).  Lowest priority preferred.
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by gmalcolm on April 17, 2011  •  Permalink
Posted in CUC

Posted by gmalcolm on April 17, 2011

https://voice411.wordpress.com/2011/04/17/unity-sip-integration/

Current Progress

My Progress (MS Excel formatted)

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by gmalcolm on April 14, 2011  •  Permalink
Posted in My Progress

Posted by gmalcolm on April 14, 2011

https://voice411.wordpress.com/2011/04/14/current-progress/

CUE

CUE Notes

Connectivity

To connect to module: service-module service-Engine 1/0 session

IP Configuration:
interface Service-Engine1/0
ip unnumbered FastEthernet0/0.311
service-module ip address 177.3.11.254 255.255.255.0
service-module ip default-gateway 177.3.11.1
!
ip route 177.3.11.254 255.255.255.255 Service-Engine1/0

Basic Configuration

Take offline – offline
Set to default – restore factory default
Cue will run you through the setup script

Changing the License

You may need to change the integration between CME and UCM.  You need to install a license to do this.  Start a FTP server with the correct license in the home path.  From the CUE module, do a ‘software install clean url’ command.  Pretty self explanatory from there.

CUE with CME Configuration

URL available during the lab :

http://www.cisco.com/en/US/products/sw/voicesw/ps5520/products_configuration_example09186a008037f2a9.shtml

Configure dial-peers for CME to CUE Voicemail

dial-peer voice 2 voip
! this is the pilot VM number
destination-pattern 3600
session protocol sipv2
session target ipv4:10.10.202.2
codec g711ulaw
no vad
! this is for MWI
incoming called-number 399[89]….

Configure Voicemail Access

telephony-service
voicemail 3600
web admin system name admin password cisco
call-forward pattern .T
! H450.2 method for transfer
transfer-system full-consult
! needed to transfer to voicemail
transfer-pattern .T

! add CFB and CFNON to DN’s
ephone-dn 1
call-forward busy 3600
call-forward noan 3600 timeout 12

Configure the HTTP server

ip http server
ip http authentication local
ip http path:/GUI

Configure MWI’s

ephone-dn 10
number 3999…. no-reg primary
mwi on

ephone-dn 11
number 3998…. no-reg primary
mwi off

Configure the Voicemail Application
ccn application voicemail
description “Cisco Voicemail”

Configure CUE via the GUI

  • Run the wizard, set the web user name and import the users
  • Set the voice mail number and the MWI method (make sure that the MWI on/off numbers show up

CUE with CME Configuration
CME Configuration

1.      Configure dial peer for communication between CME and CUE.

Dial-peer voice 2 voip
Destination-pattern (DID to voicemail)
Session protocol sipv2
Session-target ipv4:x.x.x.x
dtmf-relay sip-notify
no vad
codec g711ulaw

2.      Configure voicemail access
Telephony-service
Voicemail (dn)
Call-forward pattern .T
Web admin system name admin password cisco
Dn-webedit
Transfer-system full-consult
Transfer-pattern .T

3.      Configure ephone-dn to forward to voicemail
Ephone-dn (tag)
Call-forward busy (vm dn)
Call-forward noan (vm dn) time 10

4.      Configure MWIs
Configure ephone-dn for MWI on and one for off.  Below turns on MWI

Ephone-dn (tag)
Number (dn)
Mwi on

5.      Configure SIP

Sip-ua
Mwi-server ipv4:10.10.202.2
Voice register dn 2
Call-forward b2bua busy 3600
Call-forward b2bua mailbox 3600
Call-forward b2bua noan 3600 timeout 12
Voice register pool 2
Dtmf-relay rtp-nte

Voice register global
Voicemail 3600

Create profile

Reset

CUE Configuration from GUI

  1. Run the installation wizard

Import the users (sccp only).
Check mailbox and set cfna/cfb
Set the defaults
Configure the call handling numbers

2.  Modifiy/create users

SIP phones need manual user addition

3.  Set the MWIs

Subscribe – notify for SIP phones

CUE general delivery mailbox (GDM)

1.      Create voice hunt group in CME.  Make final number the normal voicemail DN.
2.      Add group on CUE.  Add members to group.  Enable mailbox.

CUE Configuration from CLI

1.      Configure voicemail application

Ccn application voicemail
Description “cisco voicemail”
Maxsessions (x)

2.      Configure auto-attendant
Ccn application auto-attendant
Parameter “operExtn” “(dn)”

3.      Configure SIP triggers for applications.
Ccn trigger sip phonenumber (dn) – dn should match voicemail dn
enabled

4.      Create users
Username John create
Username John phonenumber (dn)

5.      Configure mailboxes
Voice mailbox owner John
Enable
Expiration time
Mailboxsize
Messagesize

CUE with CUCM examples

CUE configuration
Use steps listed above for the CUE portion
CM Configuration

1.      Create CTI ports.  One port for each CUE port.  Device pool must match where the CUE is installed – the codec must match.
Device > phone
Add a new phone
Choose CTI port
Assign a DN to the CTI port.  DN has no correlation with what users might call.

2.      Add CTI route points.  Normally three CTI route points: voicemail, one for each AA and one for GMS to manage recorded prompts.
Device > CTI route point
Add a new CTI route point
Name the device
Configure device pool and location
Configure the CSS and ensure that this includes the partition with the DNs of the CTI ports.
Add a DN for each route point (DN is voice mail pilot)

3.      Create JTAPI user, grant CTI permissions and associate devices
User management > application user
Add new
Specify a user id
Add the user to the standard CTI enabled role. If you don’t do this or add to a different role. the CTI point will not register.

In the device information field, under available devices, select the route points and CTI ports that are associated with the ID.

In the permissions information section, click add user to group.  Select standard CTI enabled.

4.       Create a voicemail pilot and a voicemail profile

5.      Assign the voicemail profile to the DN.

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by gmalcolm on April 14, 2011  •  Permalink
Posted in CUE

Posted by gmalcolm on April 14, 2011

https://voice411.wordpress.com/2011/04/14/cue/

Basic Call Routing

Basic Routing

1.    Create partitions.  Typically a partition is created for each call type for instance 911, local, long distance and international routing will use individual partitions. Remember that a partition is a group of directory numbers with similar accessibility.

2.    Create calling search spaces. Calling search spaces determine the partitions that calling devices, Cisco IP phones and gateways can reach when attempting to complete a call.

3.    Create gateways.  Gateways are used to route off net calls that are not accessible on call manager. Gateways are typically either and MGCP or H.323.

Gateway things to remember:

  • Don’t forget to set the significant digits field in the call routing information — inbound call section
  • The calling search space also needs to be set in the call routing information section.

4.    Configure the IP phones or the correct parameters. Some of these parameters are the correct device pool, the correct calling search space, route partition and external phone number mask.  The phone number mask is set in the device phone, line number section.

5.    Configure the route group.  The route group will point to the proper Gateway.

6.    Configure the route list. The route list must point to the proper route group

7.    Configure route patterns.  Route patterns are used to match dialed digits frequently items such as 911, local and long-distance calls as well as international calls.  Here are some common examples:

9.911 – an access code of 9, a dot to separate access codes and 911.

9.1[2-9]XX[2-9]XXXXXX – long distance calls

9.[2-9]XXXXXX – local calls

Dial-peer example

HQ phone (5001)  —- UCM —- HQ RTR — PSTN WAN — PSTN phone (911)

UCM to HQ RTR is h.323, HQ RTR to PSTN is PRI

Incoming VOIP to HQ RTR

dial-peer voice 2 voip

voice-class codec 1

incoming called-number .

no vad

Outgoing to PSTN WAN

dial-peer voice 911 pots

destination-pattern 911

port 0/0/0:23

forward-digits 3

Incoming PSTN WAN

dial-peer voice 1 pots

incoming called-number .

direct-inward-dial

Outgoing from HQ RTR

dial-peer voice 5000 voip

destination-pattern 2123945…

voice-class codec 1

voice-class h323 1

session target ipv4:10.10.210.11

incoming called-number .

dtmf-relay h245-alphanumeric

no vad

Routing notes

  • The display IE delivery check box is used to set the calling name (alerting name that the directory number assigned to the phone in UCM).
  • Use calling number mask ensures that the external number mask configured on the directory number calling phone is passed to the router. Without checking this checkbox, the calling number (ANI) would be 4 digits in length.

UCM route pattern wildcards

Wildcard Pattern
X Matches a single digit
@ All NANP routes
! One or more digits
. Terminates the access code
# Terminates the inter digit timeout
[xyz] Set of matching digits [458] matches either 4,5, or 8
[x-y] Range of digits
[^x-y] Exclusion range

Local route groups

Forced Authorization Codes (FAC)

1.    Create the code and level from Call Routing, Forced Authorization Codes

2.    Apply the FAC under the Route Pattern

Client Matter Codes (CMC)

1.    Create the code from Call Routing, Client Matter Code

2.    Apply the CMC under the Route Pattern

Route Pattern Call Blocking

1.    Create a new partition for the pattern to be blocked

2.    Add the partition to the Calling Search Space

3.    Add a new route pattern.  Under the Route Option, select Block This Pattern

Time Based Call Routing

1.    Create a Time Period from Call Routing, Class of Control

2.    Create a Time Schedule from Call Routing, Class of Control

3.    Apply the Time Schedule to the Partition

Note – a route list can also be used.

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by gmalcolm on April 14, 2011  •  Permalink
Posted in Call Routing

Posted by gmalcolm on April 14, 2011

https://voice411.wordpress.com/2011/04/14/basic-call-routing/

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