Cisco Voice Guru

CUCME Integration

dial-peer voice 3600 voip
destination-pattern 3[456]00
session protocol sipv2
session target ipv4:10.10.202.2
incoming called-number 399[89]…
dtmf-relay sip-notify
codec g711ulaw
no vad

ephone-dn  4
number 3999….
mwi on

ephone-dn  5
number 3998….
mwi off

ephone-dn-template 1
call-forward busy 3600
call-forward noan 3600 timeout 10
CUE Initial Configuration

CUE Initial Configuration

site name local
phone-authentication credentials ##
site-hostname 10.10.202.1
web credentials hidden ##
!
ccn subsystem sip
gateway address “10.10.202.1”
!
ccn trigger sip phonenumber 3600
application “voicemail”
enabled   
maxsessions 6
end trigger

CUE MWI Configuration

ccn application ciscomwiapplication aa
parameter “strMWI_OFF_DN” “3998”
parameter “strMWI_ON_DN” “3999”

CUE User Creation

username UserOne create   
username UserOne phonenumber “3001”
username UserOne pin 12345
username UserOne password 12345
voicemail mailbox owner “UserOne” size 2964
no tutorial
voicemail callerid
Create a group mailbox

CUE GDM Creation

groupname Sales create
groupname Sales phonenumber "5555"
groupname Sales member "UserOne"
groupname Sales member "UserTwo"
voicemail mailbox owner "Sales"
no tutorial
voicemail callerid

TCP Port Numbers

• TCP 1720 – H.225 Call Setup / FastStart
• TCP 2000 – SCCP
• TCP 2428 – MGCP ISDN PRI Backhaul
• TCP 5060 – SIP Telephone
• TCP 11000-11999 – H.245 Capabilities Exchange / SlowStart

UDP Port Numbers

• UDP 1718 – H.225 RAS Multicast Gatekeeper Discovery
• UDP 1719 – H.225 Gatekeeper RAS Messages
• UDP 2427 – MGCP Control
• UDP 2748 – CTI/JTAPI
• UDP 5060 – SIP Trunk
• UDP 16384 – 32766 (even port numbers) – RTP
• UDP 16385 – 32767 (odd port numbers) – RTCP

CRCX

I’m on a roll tonight.  Here are some more troubleshooting/FYI notes, this time focused on SRST.  The study life of a dedicated CCIE candidate is one constant stream of information absorption occasionally interrupted by family time, eating, sleeping, and relieving oneself in the bathroom.  Sometimes the latter gets delayed by long chapters on gatekeeper behavior (as I’m experiencing right now!).

DLCX

Be aware of the following SRST restrictions:

1. Extension mobility does not work in SRST mode when using call-manager-fallback.  CME as SRST does have extension mobility, but login information is not seamlessly transferred during failure situations.
2. SRST does not have a concept of partitions or calling search spaces. CoR can be used to some extent, but it is a weak-comparison and is not automatically populated on the SRST router.
3. MWIs might be out of synch when recovering from SRST mode.
4. All forwarding (busy, no answer, and forward all) is lost in SRST mode.
5. In SRST mode, all call routing decisions are made on the SRST gateway through dial peer configuration, so anything not explicitly configured on the SRST gateway will not work.

For SRST mode, be sure to configure voicemail under “telephony-service” and to configure an “ephone-dn template” to set the necessary call-forward noan and busy behaviors.

Set your DHCP lease times to several days, or use a local DHCP server to ensure that IP Phones do not reset themselves when their lease expires during a WAN outage.

Use transfer patterns to allow transfers to non-IP Phone destinations.

Survivable endpoints: IP Phones and MGCP Gateways

Nonsurvivable endpoints: H.323 Gateways, SCCP Gateways, CTI/TAPI Endpoints

The rules for call survivability can be summarized as follows:

• If the call involves nonsurvivable endpoints, and a CallManager involved in the call fails, the call fails.
• If the call involves one nonsurvivable endpoint and one survivable endpoint, the call fails only if the CallManager that the nonsurvivable endpoint is registered to fails.
• If the call involves only survivable endpoints, and one or more CallManagers involved in the call fails, the streaming connection between the endpoints is maintained. Note, however, that the endpoints do not have supplementary services available to them after the failure.
• If a hardware-based conference bridge is involved in a call, and the CallManager controlling the conference bridge fails, all calls from nonsurvivable endpoints fail. All calls from survivable endpoints continue in the conference.

  If a software-based conference bridge is involved in a call and the CallManager controlling the conference bridge fails, all calls from the nonsurvivable endpoints fail. All calls from survivable endpoints continue in the conference until the party terminates the call voluntarily. The conference bridge reregisters with an available CallManager.

• If the call involves an MTP or a transcoder, the call fails.
• If the call is in the and a CallManager involved in the call fails, the call fails.

The IP Voice Media Streaming
Application supports only G.711 µ-law and G.711 A-law codecs. It can provide up to 128 streams of conferencing on a standalone server or 24 streams when it coresides with CallManager on the same server.onf

Conference Bridges, Transcoders, and MTPs: Best Practices

• When a media resource is assigned to any MRG, it is no longer available as part of the default MRGL. This means that endpoints that require the use of a media resource in an MRG must have an MRGL configured that contains that MRG.
• Be extremely careful when configuring your codec selections between regions. This is by far the most common cause of transcoder problems.

Troubleshooting MOH

1. Check the MOH server registration status.
2. Check the MRG and MRGL configuration.
3. Verify router configuration for multicast (if multicast is used).
   ip multicast-routing
   ip pim sparse-mode | ip pim dense-mode
4. Verify the multicast capability of the terminating voice gateway.
5. Verify the codec used by all devices involved.
6. Verify the location’s configuration if you are using the centralized call processing model.

H.225/H.245 Summary

H.225 RAS Signaling:

• RAS is the signaling protocol used between gateways and gatekeepers. The RAS channel is opened before any other channel and is independent of the call setup and media transport channels.
• RAS uses User Datagram Protocol UDP 1719 (H.225 RAS messages) and UDP 1718 (multicast gatekeeper discovery).

H.225 Call Control (Setup) Signaling

• H.225 call control signaling is used to setup connections between H.323 endpoints.
• A reliable (TCP) call control channel is created across an IP network on TCP 1720. This port initiates the Q.931 call control messages for the purpose of the connection, maintenance, and disconnection of calls.

H.245 Media Control and Transport

• H.245 handles end-to-end control messages between H.323 entities.
• H.245 procedures establish logical channels for transmission of control channel information.
• It is used to negotiate channel usage and capabilities such as flow control and capabilities exchange messages.

Outside Dial Tone Played at the Wrong Time

• If multiple patterns exist with the same first digit and at least one of the patterns is not configured to provide outside dial tone, outside dial tone is not played unless all the remaining potential matches are configured to provide it.
• Fix 1: Check the box on the 911 route pattern that tells it to play outside dial tone.
• Fix 2: Change the 911 pattern to 9.911 instead and check the box that tells it to play outside
  dial tone (although 9.911 is already included as part of 9.@).
• A good way to isolate the problem is to view the Route Plan Report in Cisco CallManager Administration (Route Plan > Route Plan Report).

Call Forward All (CFA)

• For calls within a cluster, use the Forward Maximum Hop Count service parameter (Service > Service Parameters > select a server > Cisco CallManager) to end a forwarding chain after a specified number of hops
• Use MaxForwardsToDn and StopRoutingOnUserBusyFlag to prevent routing loops that traverse other clusters or legacy TDM equipment.

Called and Calling Party Transformations

1. The External Phone Number Mask on the original calling device (when specified on a route pattern or translation pattern by the Use Calling Party’s External Phone Number Mask checkbox.)
2. A translation pattern
3. A route pattern
4. A route list using route group details
5. Transformation on the terminating device

Methodology for Resolving Call Routing Problems

1. What calling device is having the problem (phone, gateway, CTI ports, Voicemail ports, etc.)?
2. What is the dialed number as dialed by the user or sent by the gateway?
3. What calling search space (and corresponding partitions) are being searched?  Pay attention to the different line/device CSSs that come into play.
4. If a match is made (important word = IF), what pattern is matched?  What partition does that pattern belong to?
5. Is a translation pattern being us?  If so, where is it redirecting the call?
6. Are there any DDIs taking place?
7. If the route pattern goes to a route list (i.e. 99% of patterns), check the route list for digit manipulation being done at that level.
8. If you’re sure the outbound call is hitting the gateway, use my best friends:
   debug isdn q931
   debug voip dialpeer (H.323/SIP only)
   debug voice translations (H.323/SIP only)

Random Notes

• The acronym DDI refers to “digit discard instruction” and not “direct dial in” which is commonly used to refer to direct inward dial (DID) numbers in some countries.

CRCX

Dear fellow CCIE Voice candidates,

I apologize for being out-of-touch these days.  I have a few folders in my Dropbox with massive amounts of checklists and debug explanations.  However, life has been so chaotic in the past few weeks that I have not had the time to faithfully update this blog.

It’s my promise to you that I will add more information to this blog once I pass.  I want this to be a resource that people can use for different Cisco voice questions.  The most important part of the CCIE is the journey that transforms you from a puny n00b with poor troubleshooting skills and no understanding of Cisco’s documentation site to a stone-cold VoIP issue killer.  “One-way audio?  Dropped calls?  That’s nothing!”

If you have been on this journey for a while, suffered long nights, and experienced the chronic aches and pains of caffeine overdose, then you understand what I’m saying.

MDCX

Ok, changing focus…

I am 33 days away from my exam.  I feel pretty confident in my core skills.  I am weak in SIP, however.  Not in the SIP basics like dial-rules, voice register commands, and Early Offer enticements.  Instead, I’m weak in the troubleshooting skills that are required to deploy a basic SIP solution in under eight hours without any major hiccups.  I plan to be reading the old, but not outdated, “Troubleshooting Cisco IP Telephony” book by Cisco Press as well as the “SIP Trunking” book.

SIP is a very cool protocol, though I’m unsure of how or when it will replace the good ol’ ISDN PRI.  This is just one curiosity that has to be tucked away for a rainy day when there are no projects and no CCIE lab looming on the horizon.

I’ve been very tired these days.  In fact, I might not have paced myself adequately because I’m exhausted every evening.  I’m a big believer in good sleep and mental clarity before taking an exam.  I might take a few days easy before diving in for the last stretch.

Good luck to all of you out there!  I’m cheering you on!

I’ll try to post a few more times as the Day draweth nigh.

DLCX

The “Voice (soon to be certified) Guru”

Overall, I found today challenging.  I was very pleased at my speed and accuracy up through the gateways/gatekeeper section.  After 2.5 hours, I had 28 solid points and was moving on to the call routing section.  My weaknesses in this lab was IOS COR, QoS, and Messaging.  I found VPIM very easy to configure (my first time); however, I was never able to get CME-CUE-SIP MWI notifications to work.  I only made it as far as the end of section 9 and then called it quits.  I will lab this again on Thursday night.

For Thursday:

• Copy output of “show ip int bri | e una” from every router and paste into Notepad for continued reference.
• Create a matrix with digits PSTN wants to receive and ANI presentation.  See if there is some way to capture a high-level snapshot of the dial plan so I’m not burned at the end if I am asked to do SRST.

1.0 Infrastructure

• ALWAYS add the "ip helper-address" command to your routers.  Even if it is redundant, it is a safe best-practice for the lab to avoid issues.
• Remember the strings used for LDAP integration.  An example is listed below:
  • LDAP Manager Distinguished Name: cn=Administrator,cn=Users,dc=cisco,dc=com
  • LDAP User Search Base: cn=Users,dc=cisco,dc=com

2.0 UCM Basic Setup

• With locations-based CAC, you only ever configure new locations for the “spoke” sites.  Since a maximum of four G.729 calls are allowed between sites in this lab, configure a single location called LOC-BR1 and set to 96 kbps for audio.  It is not necessary to configure a LOC-HQ and set a threshold on that location as well.
• If the bandwidth values are set to a finite number of kilobits per second (kbps), CUCM will allow calls in and out of that location as long as the aggregate bandwidth used by all active calls is less than or equal to the configured values. (CUCM SRND)
• The gateways did not have locations-based CAC applied.  Even the BR1 MGCP gateway was left in Hub_none.

3.0 UCME Basic Setup

• Remember to configure the "timezone" command for SCCP and SIP phones in the IOS.  Otherwise, hours may be off.
• SIP phones receive their banner information from “voice register pool 1” | “description 3006”.
• SIP phones should be configured with “voice register pool 1” | “codec g711ulaw”.
• SCCP should be configure with caller-id using “ephone-dn 1” | “name BR2 PH1”.
• Time zone was not specified for BR2, though I’m assuming it would be the time zone for Spain.  No specific requirements were set around IP phone display presentation so I left it at default.
• Setting a time zone is probably something I want to do anyway.  If I leave the CUCME at a default time zone, my CUE module will not have the same time.
• Default dtmf-relay for CUCME SIP phones is rtp-nte.

4.0 Gateways and Gatekeeper

• Prefix outbound H.225 gatekeeper-bound calls using “dial-peer voice 10 voip” | “tech-prefix 1#”
• Got burned by omitting the command "no supplementary-service h225-notify cid-update" under "voice service voip."
• Gatekeeper commands: “show gatek end” | “show gatek gw-tech” | “show gatek zone status”
• ISDN b-channel order for MGCP gateways is set through the CUCM GUI under the T1 port.
• When receiving a gatekeeper question based off show commands, make a list of the requirements in Notepad.  I wasted precious minutes going back and rereading the outputs to determine what the question was asking.
• By default, any lab requiring gatekeeper should also demand a separate device pool and region to force intersite/gatekeeper calls to G.729.
• If CUCM handles all digit manipulation for an H.323 gateway, you can eliminate unnecessary keystrokes by using a single outbound POTS dial-peer with “destination-pattern .T”.  This would not work for sites with SRST requirements.
• If I am not allowed to use " gw-type-prefix 1#* default-technology" on HQ GK and I don’t know the Remote PSTN zone tech-prefix, how can I make my calls to India  (91 country code calls) pass on to Remote zone PSTN GK?
  • Tech Prefix is not required for routing to remote zones.
  • The configuration on the remote zone is invisible to you and hence you will have to be told EXACTLY what digits to send to the PSTN zone. If the owner of the PSTN zone requires you to send “12345” then you send “12345” and leave the method of how the call is routed to the remote gateway down to the remote gatekeeper configuration.
  • Maybe the remote gateways are registered with the tech prefix “1” or “12” or “123” or maybe he is using 1# as a default technology- we don’t really care.

5.0 Call Routing

• Every CSS has access to the <None> partition.
• If asked to create a call park range of 1100-1102 with redundancy between CUCMS, configure a range of [1100-1102] in PT-INTERNAL (CUCM_Sub) and another range of [1100-1102] in the <None> partition (CUCM_Pub).
• Stop Routing on Unallocated Number Flag set to False.  This is very important and very easy to forget.
• After hours block patterns under “telephony-service” apply to SCCP and SIP phones.  Therefore, this is the easiest way to block International outbound dialing at BR2.
  1. Global configuration: “telephony-service” | “after-hours block pattern 1 900 7-24”
  2. Device exemptions: “ephone 2” | “after-hours exempt”
• SIP Route Patterns Checklist
  1. Enterprise Parameter: Organization Top Level Domain set to “proctorlabs.com.”
  2. Enterprise Parameter: Cluster Fully Qualified Domain Name set to “proctorlabs.com.”
  3. End User: Configure telephone number as 3006@cme.com.
  4. Setup a SIP trunk from CUCM to BR2.
  5. Create SIP Route Pattern with IPv4 pattern of “cme.com.”
• Block Inbound International Calls Checklist
  1. Create translation rule using “rule 1 reject” command.
  2. Create translation profile based on calling number.
  3. Assign profile to dial-peer using “call-block translation-profile incoming” command.

6.0 QoS

• The following CUCM and CUC service parameters adjust DSCP markings for control traffic:
  • CUCM > Svc Param > IP Media Streaming App > IP DSCP to Cisco Unified Communications
  • CUCM > Svc Param > CUCM > DSCP for ICCP Protocol Links
  • CUCM > Enterprise Parameters > DSCP for Phone Configuration
  • CUCM > Enterprise Parameters > DSCP for Cisco CallManager to Device Interface
  • CUC > Advanced > Telephony > DSCP value for call signaling connections
• Verify CoS-DSCP mapping on IOS: “show mls qos map cos-dscp” | dscp: 0 8 16 26 32 46 48 56
• Trust packets from phones and servers: "interface FastEthernet 1/0" | "switchport priority extend cos 0" | "mls qos trust cos"
• You can mark SCCP packets under "sccp ccm group" using the command: "signaling dscp af31"
• When calculating WAN bandwidth, always shape to 95% of the PVC speed using: "frame-relay cir 729600" | "frame-relay bc "7296" | "frame-relay min-cir 729600" | "frame-relay be 0"
• Voice Bandwidth (without Layer 2 Overhead)
  • G.711 @ 10 ms = 80 bytes voice payload
  • G.711 @ 20 ms = 160 byte voice payload
  • G.711 @ 30 ms = 240 byte voice payload
  • G.729A @ 10 ms = 10 byte voice payload
  • G.729A @ 20 ms =20 byte voice payload
  • G.729A @ 30 ms = 30 byte voice payload
• Reasonable values for media and control traffic are 33% and 5% accordingly.
• You HAVE TO ENABLE FRTS for the policy-map (LLQ) to take effect. Running FRTS on the physical interface is mandatory unless you are using class-based shaping (nested policy-map).

7.0 Media

• In IOS, use “show voice dsp group all” to determine PVDM2 capacity on each router.
  • One PVDM2-16 = 240 MIPS
  • When calculating maximum number of transcoding sessions, deduct 15 MIPS per PRI b-channel (assuming G.711 codec out to the PSTN).  This is important to take into account.  Otherwise, your PRI calls will not have sufficient resources for transcoding.
  • G.711ulaw/alaw (low complexity) uses 15 MIPS per call = 16 calls per PVDM2-16.
  • G.729ar8/abr8 (medium complexity) uses 30 MIPS per call = 8 calls per PVDM2-16.
  • G.729r8 (high complexity) uses 40 MIPS per call = 6 calls per PVDM2-16.
• If two MoH servers are placed in the same MRG, they will load-balance between each other.  To verify, invoke MoH on two IP phones, SSH into the Subscriber, and type: show perf query class “Cisco MOH Device”.  Make sure the MOHTotalUnicastResources value = 249 on both servers.
• Sourcing MMOH from Branch Router Flash Checklist
  1. “ccm-manager music-on-hold”
  2. “call-manager-fallback” | “moh music-on-hold.au” | “multicast moh 239.1.1.1 port 16384 route 10.10.201.1 10.10.110.2” [Required: Interface for voice subnet and loopback for PSTN connections]
  3. “debug ephone moh”
  4. Media Resource Group List
  5. New device pool and region to hardcode G.711 codec.

8.0 Messaging

• Load license file and restore factory default settings on CUE early on in the lab.  This will take a while.
• Make sure to enable CUC services before configuring any other settings.
• Apply AAR Group Settings to the hunt pilot in CUCM.  Voicemail ports have no need of an external number mask.  The external number mask on the hunt pilot will be used when there is not enough bandwidth to call the internal number.
• CUC > Users > Users > Edit > Send Message Settings > User Can Send/Update Broadcast Messages to Users on This Server.
• CUC > Tools > Custom Keypad Mapping > Change keypad mapping.  This is applied under Users > Users > Phone Menu.
• CUC > Advanced > Conversations > System Broadcast Message: Default Active Days.
• CUE > To make sure that MWI for a group mailbox is sent to a specific user, use "num-exp 39993210 39993102" | "num-exp 39983210 39983102"
• CUE-CUC VPIM Checklist
  1. Check DNS server to make sure there is an appropriate MX record for the domains being used.
  2. CUC > SMTP Configuration > Server > Set "SMTP Domain" and check "Allow Connections from Untrusted IP Addresses."
  3. CUC > Networking > VPIM Locations > Create VPIM locations for CUCM/CUCME sites.  Make sure to check "Allow Blind Addressing" for remote CUCME VPIM Location.  You will also likely be asked to enter a DTMF Access ID.
  4. CUC > CLI > "set network dns primary 10.10.210.14"
  5. Reload the CUC server for the SMTP and DNS settings to go into effect.
  6. CUE > GUI > Configure > Network Locations > Location ID corresponds to DTMF Access ID in CUC
  7. CUE > GUI >System > Domain Name Settings > Add DNS server and domain name
  8. Verify operation from CUE: "no trace all" | "trace network vpim all" | "show trace buffer tail"
  9. Verify operation from phone: Call in, Press 2, Record message, ## Switches between dial by name/number, Enter remote location, Enter ### to send the message.
• CME-CUE Pre-Installation Checklist
  1. Configure Service-Engine 0/0
  2. Configure voip dial-peer for voicemail pilot: “destination-pattern 3[16]00” | “dtmf-relay sip-notify sip kpml” | “codec g711ulaw” | “incoming called-number 399[89]….” | …
  3. Configure telephony-service options: “voicemail 3600” | “call-forward pattern .T” | “web admin system name admin password cisco” | “dn-webedit” | “transfer-system full-consult” | “transfer-pattern .T”
  4. Configure web server: “ip http server” | “ip http authentication local” | “ip http path flash:/GUI”
  5. Configure ephone-dns for MWI on/off with number format set to “3999….” and “3998….”
  6. Configure sip-ua: “sip-ua” | “mwi-server ipv4:10.10.202.2”
  7. Configure voice register dn’s with call-forwarding: “voice register dn 2” | “call-forward b2bua busy 3600” | “call-forward b2bua mailbox 3006” | “call-forward b2bua noan 3600 timeout 12”
  8. Configure voice register pool rtp-nte dtmf-relay
  9. Configure voice register global options: “voicemail 3600”

9.0 High Availability

10.0 CUCM Application and Supplementary Services

11.0 Contact Center

Could anything be more disturbing than this family photograph I found online?  First of all, why is she holding a rifle with accompanying scope and bayonet?  And a green parrot, um, why?

Enjoy the 4th of July holiday weekend, get some rest, and then get back to studying H.323 and MGCP protocols!  There’s no rest for the weary…until you have a five-digit number!

1. Infrastructure

• By default, grab a copy of the initial configuration from each device before you change anything.  This is very important.
• Check for “ip helper-address” and correct DHCP pool information.
• Are the interfaces shut down?
• Is CDP enabled?
• Useful show commands: “show ip int bri” | “show vlan brief”
• We could run into a situation where CDP is disabled or broken during the lab.  These commands may help the troubleshooting process:
  • Enable/Disable CDP globally: “cdp run” | “no cdp run”
  • Enable/Disable CDP on an interface: “cdp enable” | “no cdp enable”
  • Transmission frequency of CDP updates in seconds: “cdp timer 60″ (default) | “no cdp timer” (resets to default value)
  • Amount of time receiving device should hold the information sent by your device before discarding it: “cdp holdtime 180″ (default) | “no cdp holdtime” (resets to default value)
  • Configure CDP to send Version-2 advertisements: “cdp advertise-v2″ (default) | “no cdp advertise-v2″ (resets to default value)
• LLDP  (Link Layer Discovery Protocol) is an alternative to CDP:
  • Enable/Disable LLDP globally: “lldp run” | “no lldp run” (default)
  • Enable LLDP on an interface: “lldp transmit” | “lldp receive”
  • Refer to Catalyst 3750 Switch Software Configuration Guide for more information

2. CUCM Endpoints

• Join Across Lines has always required a hardware conference bridge during my testing.

3. CUCME Endpoints

• Under “voice register global” I need to remember to add “tftp-path flash:” if I am upgrading the firmware via CUCME.
• Banner for SIP phones is configured under “voice register pool 1″ | “description +3432143002″
• What is the best dtmf-relay to use under the voice register pool section? RTP-NTE
• Would others configure additional dialplan patterns for the SIP phones to avoid timeout?  Unsure.
• When configuring SIP CUCME, you must remember to enter “mode cme”
• When configuring cBarge, you must remember to stand up a hardware conference bridge on the CME gateway, enter all the necessary sccp/dspfarm/telephony-service commands and then configure an octo-line ephone-dn with the sub-command “conference ad-hoc”

4. Call Routing

• Gatekeeper zone prefixes must be used to set the destination zone.  Otherwise, the gatekeeper will assume that the destination zone = source zone.
• OUTVIA = Calls from the gatekeeper to the destination zone go out via the VIA zone.
• INVIA = Calls coming into the gatekeeper to the destination zone come in via the VIA zone.
• By default, set any called number transformations in the Route List using NANP: PreDot.
• If the CUBE dial-peer “destination-pattern” or “incoming called-number” information is incorrect, you will see an error in “debug gatekeeper main 10″ about how it cannot find an IPIPGW in the specified via zone.  You can run a “debug voip dialpeer” on the CUBE to see what digits are being sent from the GK.
• By default, set the “dtmf-relay” on dial-peers as one of the first things you do.  Otherwise, you will always forget.
• On your H.323 gateway, set the CUCM dial-peer destination patterns to send all the digits to CUCM.  For example, if the circuit delivers 2123945002 set the destination-pattern to 2123945…$  Essentially, you are delivering all the digits to CUCM, allowing it to determine the number of significant digits.
• Useful show commands: “show call active voip brief” | “show voice call stat”
• Remember that “after-hours day Sun 7:00 7:59″ sets the bounds of 7:00:00 to 7:59:59.  If you enter something like “8:00″ for an end time, you will fail the question because that is considered by CUCME as “8:59:59″.
• I forgot to enter “clid strip name” under the H.323 dial-peer for emergency services.
• If configuring routing between two H.323 gateways, you can set this up independent of CUCM.  Why complicate a question if they do not give you that requirement?
• Dialed 90016178632123 >>T.R. /^9001/ /1/ changed number to 16178632123 >> D.P added the tech-prefix 1#16178632123 >> T.P in CUCM of 1#1617.! changed number to 98632123 after DDI predot and prefix of 9 >> R.P. was matched and set out the local route list, which is why the gatekeeper needed its own device pool with the proper local route group set.
• For device mobility, there are two important rules:
  • Different Physical Locations >> Device will use Roaming Sensitive Settings from Roaming Device Pool.
  • Same Device Mobility Group >> Device is forced to use Device Mobility Info from Roaming Device Pool
• RSVP requirements >> HQ location = LOC-HQ
• No RSVP requirements >> HQ location = Hub_none

5.0 Quality of Service and Call Admission Control

• Do QoS early on!  You have been warned!  I waited until I was already 6.5 hours in the exam.  When my QoS commands destroyed my lab, I lost a ton a points and didn’t have the time to fix it.  #FAIL
• If you are prohibited from trust traffic from endpoints, you must setup classification and marking on the switch.
• The following command will set bandwidth limitations to a certain zone: “bandwidth interzone UCM 32″.  Earlier, I had stood up a CUBE configuration with G.711 from CUCM to CUBE, which was then transcoded and then setup between CUBE and BR2-CME.  If I had set a default bandwidth limit of 32 globally, the CUBE calls would have failed.
• Standard ACLs
  • access-list 102 permit tcp any any eq 2000
  • access-list 102 permit tcp any eq 2000 any

6.0 Media

• MMOH Configuration
  • Useful show command: “debug ephone moh”
  • Configuration without “call-manager-fallback”: “telephony-service” | “moh music-on-hold.au” | “multicast moh …”
• To determine the available MIPS on a router, “show voice dsp group all”
• Voice termination: 15 MIPS for G.711, 30 MIPS for G.729
• Conference: 120 MIPS (PVDM2-16 has 240 MIPS = 2 conferences per PVDM2)
• Transcoding: 30-40 MIPS
  • G.711 consumes 15 MIPS (PDVDM2-16 has 240/15 = 16 sessions)
  • G.729a, G.729b, G.729ab consumes 30 MIPS (PVDM2-16 has 240/30 = 8 sessions)
  • G.729r8 consume 40 MIPS (PVDM2-16 has 240/40 = 6 sessions)
• To restrict the ability to initiate a Meet-Me, create separate partition PT-MeetMe-Restrict.  Add the MeetMe number to this partition and add the partion to the CSS of a particular device.  To allow others to call into the MeetMe, stand up a CTI-RP with call forward all set and a CSS that contains the PT-MeetMe-Restrict partition.

7.0 Messaging

• Call Management > Call Routing > Forwarded Routing Rules
  • Conversation > Start Live Record
  • Forwarding Stations > Equals > XXXX (i.e. CTI Route Point)
• Create a CTI Route Point and forward all calls to voicemail
• When setting up the CUE integration, I had to put in “ip http server” | “ip http path flash:GUI”
• MWI ephone-dns must be in the format 3999….

8.0 Applications

9.0 Contact Center

• Need to configure transcoders on the HQ side for the sake of G.729 calls coming from BR1.
• Make sure to add G.729r8 to the transcoding list.
• Make sure UCCX CTI Route Points have access to hardware transcoders via MRG/MRGLs.
• There is a CTI Route Point setup to transfer calls to voicemail.  There is a Call Handler setup in CUC to play recorded name and supervise the transfer.

When reading through the lab initially, be very careful to not read too much into the lab during the first few minutes.  Don’t spend great detail reading about a section that you will touch six or seven hours later.  Points to touch and understand initially: infrastructure, CUCM/CUCME phones, call routing, QoS, type of voicemail integration (avoid specific messaging questions), and presence integration.

1.0 Infrastructure

2.0 Call Routing

• For CUPC devices, you must enter the name as UPC<USERNAME> where the username is entered in upper-case (case-sensitive).
• For CUCM dial plans with sites in multiple countries (numbering plans), create route partition like PT-PSTN-NANP and PT-PSTN-ENP.

3.0 Single Number Reach

• To affect the amount of time a terminated mobility call remains at the desk phone, configure the Maximum Wait Time for Desk Pickup setting under End Users.
• Application dial rules are necessary when configuring remote destinations.  Otherwise, outbound calls to the remote destination will fail because the system is trying to dial 2123942123 instead of 9-1-212-394-2123.
• Display of caller ID for the remote destination profile will be missing by default.  When a shared line is created, only the Alerting Name is copied from the line.  Go to the DN configuration of 5002 and select RDP > Associated Devices list < Edit Line Appearance > Change caller ID name.
• Make sure to configure the Owner User ID on the Cisco IP phone used for single number reach.  After adding this ID, you will need to reset the phone.  Otherwise, you may get an error about not being a valid mobile phone user.

4.0 Messaging

• CUE Reload Process
  • Reset factory default (reload)
  • Install license (reload)
  • CLI setup (reload)
  • Finish GUI (optional, but requires reload
• The CUE pilot is created in CUCM as a CTI Route Point.
• The CUE ports are created in CUCM under DEVICES > PHONES > CTI Port
• If the CUE CTI ports in CUCM will not register, try restarting CTI Manager in CUCM.
• I had a horrible issue with my voicemail hunt list not working.  At first, I thought it was an issue with the Unity Connection to CUCM integration.   I finally realized that it was a problem separate from the integration.  How?  I called the voicemail port numbers directly.  When I dialed them, the calls went through just fine.  In the end, after resetting the hunt list, all my calls to voicemail worked correctly.
• If a CUC integration is preconfigured but the voicemail ports are not registering, it is easier to simply rip out the integration on the CUCM side, validate the CUC side, and then add the CUCM side back in.
• To allow users to call voicemail during times of network congestion, make sure to setup AAR and external number masks correctly.  You will also need to configure an alternate extension under the user configuration in Unity Connection.  Refer to the “debug isdn q931″ for the exact number that should be entered.
• To setup voicemail access through the CUPC client, there are two changes that must be made to the Class of Service in Unity Connection:
  • Allow Users to Access Voice Mail Using an IMAP Client > Allow Users to Access Message Bodies (selected)
  • Allow Users to Use Unified Client to Access Voice Mail (checked)

5.0 Quality of Service

• Before you can apply a class to a Serial sub-interface, you must go to “interface Serial 0/0/1:0″ and issue a “no fair queue” to remove the fair queue argument.
• “auto qos voip” | “auto qos voip trust” – These commands will configure QoS with FRF.12
• “auto qos voip fr-atm” | “auto qos voip trust fr-atm” – These commands will configure QoS with MLP LFI, PPP Multilink
• If you are asked to do class-based shaping, it’s best to not trust the packets in your auto qos command.  Instead, issue an “auto qos voip” and let the router add its own classification and marking.  Otherwise, you will need to touch CUCM, CUC, SCCP in IOS, MGCP in IOS, VoIP dial peers (use “ip qos dscp af31 signaling” by default).
• CoS is the default trust entered by “auto qos voip trust”.  We would need to change this on interfaces from routers or servers.  It is a layer 2 v. layer 3 issue.
• Queue-set 1 is used by all switch port by default.
• Pay attention to the different between interface- and class-based cRTP.
• Remember to mark on the ingress and queue on the egress.  In other words, you will have two separate policy-maps.
• Layer two overhead: MLP (9 bytes), FR or FRF.12 (6 bytes).
• G.729 10 ms = 10 bytes, 20 ms = 20 bytes, 30 ms = 30 bytes
• G.711 10 ms =80 bytes, 20 ms = 160 bytes, 30 ms = 240 bytes
• 20 ms sampling rate: (L2 + L3 + Payload) * 8 * 50
• 30 ms sampling rate (L2 + L3 + Payload) * 8 * 33
• Verify 3750 interface configuration using “clear counters” | “show mls qos interface FastEthernet 1/0/2 statistics”.
• Ingress 3750 queues: 2 queues, SRR shared only
  • mls qos srr-queue input priority-queue 1 bandwidth 10
  • mls qos srr-queue input bandwidth 4 4
  • mls qos srr-queue input cos-map queue 1 threshold 2 1
  • mls qos srr-queue input priority-queue 1 0 - Disables priority queue for Q1
  • mls qos srr-queue input priority-queue 2 0 - Disables priority queue for Q2
• Egress 3750 queues: 4 queues, SRR shared + shaped
  • mls qos srr-queue output cos-map queue 1 threshold 3  5
  • mls qos srr-queue output cos-map queue 2 threshold 3  3 6 7
  • mls qos srr-queue output cos-map queue 3 threshold 3  2 4
  • mls qos srr-queue output cos-map queue 4 threshold 2  1
  • mls qos srr-queue output cos-map queue 4 threshold 3  0
• interface FastEthernet 1/0/7
• srr-queue bandwidth share 10 10 60 20 – Priority 3
• srr-queue bandwidth shape 0 4 0 0  – Priority 2
• priority-queue out – Priority 1

6.0 Applications

7.0 Presence

• If I am asked to configure CUCM Presence with call lists, I will need to configure the following CallManager service parameters:
  • Default Inter-Presence Group Subscription
  • BLF for Call Lists
• Do a search in the CUCM SRND for “IPPM” in order to find the URL for the phone service configuration.
• By default, leave your Presence SIP trunk in the standard presence group so that it can see everyone’s presence information.
• Next time I do this lab, I am going to try configuring one presence group per site.  Assign this group to the phones and the lines.  You will then need to go to SYSTEM > PRESENCE and disallow presence subscription from one group to another.
• Presence information for Extension Mobility is assigned under the End User Configuration in CUCM.  You can modify the presence group and SUBSCRIBE calling search space.