All posts for the month November, 2011

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
Posted in QOS and Call Admission

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

Leave a comment
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/

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