Showing posts with label CUCM - Call Routing And Plan. Show all posts
Showing posts with label CUCM - Call Routing And Plan. Show all posts

Saturday, October 6, 2012

CUCM - Service Parameters

     Service Parameters are global settings on multiple CUCM services (CallManager, IP Voice Media, Extension Mobility, etc.) that may apply to all servers (clusterwide parameters) or to only specific nodes (server parameters).

     Keep in mind that for Service Parameters, there is an "Advanced" button at the top of the page which presents additional Service Parameter options!


Service Parameters

Parameter Name
Suggested Value
Parameter Value
Description
CDR Enabled Flag
True
False
Enables Call Detail Records
CDR Log Calls with Zero Duration Flag
False
False
Enables CDR for calls that never connected or that lasted for < 1 sec
Station KeepAlive Interval
30
30
Interval between KeepAlives from IP phones to CUCM
Station and Backup Server KeepAlive Interval
60
60
KeepAlive interval between phone and secondary CUCM
T301 Timer (Alerting Timeout)
180000
180000
How long CUCM waits to receive the Alerting message
T302 Timer (Interdigit Timeout)
10000
15000
Timer after each digit is pressed before pushing the call through
Strip + on Outbound Calls
False
False
CUCM strip + on outbound for MGCP GWs and SIP Trunks. H.323 ALWAYS strips the + sign.
Built-in Bridge Enable
On
Off
Enables builtin bridge to save on conferencing resources. G.711 supported only.
Device Mobility Mode
On
Off
Allows roaming phones to dynamically associate with the local device pool
Display Device Mobility Location During Phone Registration
True
True
Displays the Device Mobility location when booting up
Off-hook to First Digit Timer
15000
15000
Time CUCM waits for the first digit to be dialed
Privacy Setting
True
True
Enables (True) or disables (False) the Privacy feature
Allow Peer to Preserve H.323 Calls
True
False
Allows H.323 GW to preserve an active call during a loss of connection to CUCM
SIP Min-SE Value
1800
1800
Minimum Session Expiration timer for SIP
SIP Session Expires Timer
1800
1800
Interval between session refresh requests
Call Park Display Timer
15
10
Time that Call Park number is displayed
Call Park Reversion Timer
60
60
Call reverts back to original phone if not answered
Party Entrance Tone
False
True
Determines if a tone will play when a party joins or exits a call with more than two parties (barge, cbarge, conference, join, meetme)
Block Offnet to Offnet Transfer
False
False
Blocks the ability to transfer an offnet call to another offnet number
Suppress MOH to Conference Bridge
True
True
Don’t play MOH on conference bridges
Drop Ad Hoc Conference
Never
Never
Ad Hoc Conference bridges stay up as long as two users are on the conference
Maximum Ad Hoc Conference
16
4
Maximum Ad Hoc Conference Participants
Maximum MeetMe Conference Unicast
16
4
Maximum MeetMe Conference Participants
Advanced Ad Hoc Conference Enabled
True
False
Conference Chaining and control Enabled
Forward Maximum Hop Count
8
12
Maximum times a single call can be forwarded or diverted. For Q.SIG calls, the maximum value is 15.
Forward No Answer Timer
16
12
The Global FNA timer setting
Max Forward Hops to DN
8
12
Max number of forwards allowed for a DN. Prevents call routing loops within CUCM
G.722 Codec Enabled
Enabled
Enabled
Enabled G.722 advertisement
Automated Alternate Routing Enable
False
False
Enables CAC Automated Alternate Routing
Single Button Barge/CBarge Policy
CBarge
Off
Sets the Single Button Barge feature to use barge, CBarge or disable
Allow Barging When Ringing
False
False
Allows users to barge before the phone is answered
Stop Routing on Out of Bandwidth Flag
False
False
Call routing behavior for calls through trunks when not enough bandwidth is available. Used for subsequent route lists and AAR/Locations.









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Wednesday, November 2, 2011

Block Toll Fraud Numbers – Route Filter

Toll Fraud List

        There are area codes that can be reached from within the United States or Canada without dialing an international code. This is where Toll Fraud is a huge issue and should be blocked on all systems. You may want to create an “Executive” Calling Search Space which will allow any/all calls to go through unfiltered, but only at the client’s request.

242     Bahamas
246     Barbados
264     Anguilla (split from 809)
268     Antigua and Barbuda
284     British Virgin Islands
340     US Virgin Islands: St Thomas, St John
345     Cayman Islands
441     Bermuda
473     Grenada
649     Turks and Caicos Islands
664     Montserrat
670     Northern Mariana Islands
671     Guam
758     St. Lucia
767     Dominica
784     St. Vincent and Grenada
787     Puerto Rico
809    Caribbean, Bermuda, Puerto Rico, Virgin Islands
868    Trinidad and Tobago
869    St. Kitts/Nevis
876    Jamaica
900    Pay_Per_Call Numbers
939    Puerto Rico
976    Pay_Per_Call Numbers



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CUCM - Automated Alternate Routing (AAR)

Automated Alternate Routing (AAR) is a high-availability feature that takes a call which would normally traverse the WAN, but is re-routed through the PSTN because the set bandwidth has been reached between two locations.

To be honest, I haven't deployed AAR at many sites. I guess this has a lot to do with the fact that I'm either really lazy, or that the cost of bandwidth has gone down so much that bandwidth and call admission control (CAC) hasn't been much of a customer requirement, or it's probably a combination of both.

If you would like to deploy AAR, the following steps and options are below, but keep in mind AAR ties directly to Call Admission Control (CAC) which means you'll have to configure and/or pay attention to Location and Region settings.


Configuring Automated Alternate Routing (AAR)
  1. Turn the service on!!
    1. System > Service Parameters
    2. Set “Automated Alternate Routing Enabled” to True. It is False by default.
    3. (Optional) Change “AAR Network Congestion Rerouting Text” to alert users that call is being re-routed. By default it is set to “Network Congestion. Rerouting.”
    4. If you don't turn this on, callers will experience a re-order tone
  2. Call Routing > Automated Alternate Routing Group
  3. Create XXX-AAR for each site
    1. You can actually create a single AAR group for NANP sites, but it depends on a few things. I would recommend creating an AAR Group for each site to keep everything clean and less confusing for administrators.
  4. Set Prefix Digits Within to 9 (Local) or 91 (Long Distance) for AAR calls between phones within this group
  5. Set Prefix Digits Between to 9 (Local) or 91 (Long Distance) for AAR calls between phones from AAR Group A to AAR Group B. (this field does not show up if you have ONLY 1 AAR group)
  6. AAR Group and CSS
    1. The AAR Group and CSS can be set on the Device Pool or the Phone, gateway, or endpoint.
  7. Extension AAR Settings
    1. Make sure that you have the correct External Phone Number Mask (EPNM) for each phone. This is how AAR figures out what number and how many digits to dial after prepending 91 or 9 within the AAR Group. 
    2. If, for some reason, you want AAR triggered calls to go to voicemail or to a different number/group altogether, you can configure these options within "AAR Settings" on the extension
    3. The AAR Settings options (voicemail or AAR Destination Mask) take precedence over the External Phone Number Mask
  8. Lastly, the AAR triggered number is matched to a route pattern using the AAR Calling Search Space. If an outbound route is matched, the call is pushed out through the PSTN.


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