Ive been struggling with certain elements of wildcards and digit matching.
This is an excellent overview of all special characters used in matching digits with CUCM:
It also contains examples next to each special characters making the concept easier to understand.
Cisco Unified Mobility: simple concept, COMPLEX configuration! In this Nugget, you’ll learn about
…Cisco Mobile Connect
(aka Single Number Reach)
…Cisco Mobile Voice Access (MVA)
(Call into the phone system, then out to the PSTN with the company Caller ID)
IP Phone is single point of contact, can call out to mobile phone or other PSTN number. This eats up your trunks! Potentially 3 outgoing lines are ‘eaten’ up. Cisco Unified Mobile Voice Access (MVA) Call into corporate number and place a call. Hiding your number! Configuration
- Add Mobility Softkey – Template and Mobility Key to On Hook and Connected states **Hit softkey and it can transfer the call to your nominated number! 🙂
- Configure End User – Enable Mobility/Enable MVA
- Configure IP Phone – Change Softkey Template for Mobility User/Owner User ID – Look to User Profile for Mobility Section
- Add Remote Destination Profile – (Shared Line) Device -> Device Settings -> RDP
- Add Remote Destinations – Device Remote Destination – Individual Numbers to Associate with RDP/Shared Line *Answer Too Soon Timer/Answer Too Late Timer/Delay Before Ringing Timer *Mobile Phone = Use Softkey/Enable Mobile Connect = Allows it to ring *A schedule can also be set! 🙂
- Configure Service Parameters/Activation -> CUMVAS – Turn on! *System-> SP -> Matching Caller ID with Remote Destination/Number of Digits for Caller ID Partial Match *Mobile Voice Access Number *Media Resources -> Mobile Voice Access Information *Outside world calls into the DDI for MVA -> Blah Blah….. An application is required on the VG for MVA *Requires a ‘service’ string on the VG – Google – Not in scope of CIPT1
- Add Access Lists (Opt…) -Who can call you? Call Routing -> Class Of Control -> Access Control List -> Allowed/Blocked – Add End User to ACL – Filter Mask/DN Mask *Apply the ACL to the Remote Destination Configuration
Wow a lot of configuration for such a simple feature!!!
Notes – 210715
RDP = Virtual Phone/RDs = Virtual Lines
Add a RD to the RDP with the DN of the user
CSS for MVA Access (Call comes into Organisation, then back out with Caller ID withheld or changed)
Rerouting CSS – Affects Mobility Connect/SNR
Once your users know they are being monitored by a SIP “watcher,” they may not be as comfortable picking up their phone. In this Nugget, you’ll learn two methods of access control for Presence: Partitions and Subscribe CSSs and Presence Policies.
1. Partition and Subscribe CSSs
Create a Subscribe CSS and assign to the phone as SUBSCRIBE CSS (Not the normal privileges CSS) This is for who you want to watch.
2. Presence Policies
If you use both methods, both of them must agree to obtain watcher privileges.
All phones by default in the standard Presence group. If you want to use this feature, then dont use the standard Presence group!
System -> Presence Group
Setup relationships between groups and then put phones in them. When you put phones in THE SAME presence group, they automatically can see each other. Allow or Disallow Subscriptions)
Assign Phone to Presence Group.
Service Parameters -> Publisher -> Subscribe (System Default = Disallowed)
Instant messaging is such a handy form of communication because you can see the status of other users before you ever message them! It’s the same way with Presence: you’re now able to see the status of another user of the phone system before you ever call them. In this Nugget, Jeremy illustrates how to configure Presence enabled speed dials, call directories, and SIP Trunks
Extension of SIP protocol (Vendor to Vendor compatibility)
‘Watcher’ + ‘Presence Entity’ SECURITY!
Device -> Device Settings -> Phone Button Template -> Change Speed Dial to Speed Dial BLF (This is enabling Presence)
System -> Enterprise Parameters – BF for Call Lists – ENABLED
Enable feature on SIP Trunk -> SIP Trunk Security Profile -> Accept Presence Subscription
This Nugget covers three features to enhance your Cisco IP Telephony system: Call Back (notifies you when a user is available), Call Park (placing a call on hold at an assigned extension), and Call Pickup (answering another ringing phone from your own device).
If you call a user and they don’t answer, press the Callback button, as soon as they make a call and hang up, you are advised that the user is available and a speed-dial button appears to call the user back. **This feature is part of a modified softkey template**
Configure Softkey Layout -> On Hook (Cancel Callback)
Configure Softkey Layout -> Ring Out -> Callback
On the Phone itself, apply the Softkey Template.
‘Put calls into a parking lot!’ – A call is parked to an extension number (5511 example). Receptionist sees this and advises user to dial 5511 to pickup the call. Can be dialed from any phone.
Call Routing -> Call Park -> Setup your range of DNs for parking
Hit the park button on the phone, system will park the call onto the first parking DN available.
Directed Call Park = Enables the Receptionist to transfer the call manually into a parking spot.
*Busy Lamp Field is compatible with Directed Call Park
BLF – You can add a Call Park BLF to the Phone Button Template
Retrieval Prefix = Number to collect call from
Used everyday in my job! Nothing else needed here.
Just like the Apple iPhone, Cisco IP Phone supports applications too (albeit not as entertaining as the Apple iPhone)! In this Nugget, you’ll see the types of services supported by Cisco IP Phones and walk through the necessary configuration to provision these services.
We normally change the CUCM servers to use their IP addressed and not DNS.
System -> Enterprise Parameters
Phone URL Parameters: change the DNS name to an IP Address
2 Major types of services – XML and Java MIDlet
Services Button on the IP Phone accesses the Services URL
Device -> Device Settings – > IP Phone Services
Add new Service with Services URL
Open a Phone -> Related Links -> Subscribe/Unsubscribe Services
Add your new custom Service
Check the Phone and it should see the new service!
CUCM treats the media resources you add to the cluster like a bowl full of goldfish: it simply grabs the first one it can from the bowl when they are needed (this is known as the NULL Media Resource Group). In this Nugget, Jeremy walks through the methods of categorizing, grouping, and prioritizing which media resources CUCM should use FIRST when they are requested.
MRG + MRG
MRGL is assigned to a Device or at a Device Pool level.
Media Resource Group – (All available Media Resources are listed)
Media Resource Group List – A list of groups
The name of the Nugget says it all. Come here to learn how to add Music on Hold (MOH) Audio Sources (including the Fixed Audio Source), manage unicast/multicast MOH, and assign MOH sources to IP phones and Common Configuration Profiles.
- MoH Resource Placement (Centralized?)
- Unicast VS Multicast – One to One and One To All (One to One – You will never start during a song, you will start at the beginning. Unicast is more resource intensive, Multicast is easier on resource. Everyone is tuning into the radio as such! 1 stream per MoH source)
- 16 Bit PCM WAV File
- 8,16,32,48 khz sample rates
- MoH Configuration
SRST Gateway can be set as a MoH device
1-51 different audio sources, the 51st/last source is a fixed audio source
Media Resources -> Music On Hold Server
Maximum Half Duplex Streams – 250 (One way only)
Maximum Multicast Connection: 250000
Change Multicast IP address
Media Resources -> Music On Hold Server Audio Sources
Use Default Source, or add new file!
Once configured, go to a Phone and select MOH User and Network Hold Sources.
User Hold is when User presses Hold
Network Hold is when someone is put on Hold by any other method than the Hold button
Common Device Configuration
Device-> Devices Settings -> Common Device Configuration
Apply batch/common configuration to phones
G.711 mulaw is default MoH CODEC, although when you upload a WAV it converts to all CODECs
QoS Values can also be set for MoH traffic! (EF DSCP is default)
While the CUCM Voice Media Streaming App is the primary source of software-based conferencing, CUCM supports DSP-based hardware resources as well. In this Nugget, Jeremy show how to manage both styles of conference resource and discusses tuning the conferencing service parameters.
- Voice Media Streaming App
- Service Parameters
- Adding software and hardware resources
Default in CUCM is 48 participants in overall conferencing up to 256
Media Resources -> Conference Bridges
Trusted Relay – Allow devices in from untrusted networks
DSPFARM – From CVOICE – Essentially Groups of DSP resources
There are many features that just “happen” on the VoIP network that we take for granted: hold, transfer, conference, music on hold…the list goes on. ALL of these features require resources to support them: Media Termination Points (MTP), Conference Bridges (CFB), Music on Hold (MOH), Transcoding (XCODE), and Annunciator (ANN). This Nugget provides an over view of each service detailing is function and placement in the VoIP network.
Media Resources Defined…. “Hardware or Software that support the function of the VoIP network”
- Conference Bridge: Handles Conference Calls – (G.711 Only with CUCM – Software based resource)
- Media Termination Point: Hold Calls On Hold! (CUCM)
- Annunciator: Plays recorded announcements (CUCM) (Accessed via TFTP as below)
- Transcoder: Converts one CODEC to another (DSPs needed and not performed in CUCM – VG for example. Can also be done between phones as DSP chips installed locally into the phone itself!)
- MoH Server: Plays Music on Hold (CUCM)
To access Annunciator messages/wav files:
Cisco OS Admin -> Software Upgrades -> TFTP File Management
All files in the TFTP server
How does CUCM do all of this…….?
Serviceability-> ‘Cisco IP Voice Media Streaming App’ -> Enables all of the above!
System>Service Parameters>Cisco IP Voice Media Streaming App -> You can decide what roles each CUCM server has.
Voice Termination (DSP required) Unable to do with CUCM. Call from PSTN, converted to another CODEC, not possible with CUCM. VG can do this for us as it has a foot in the PSTN and IP world.