Practice Lab Reflections #3

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.