Thursday, January 15, 2009

CUCME-CUE Lab 4 – CUCME POTS & VoIP Dial Peers

So far, Fuzzy Dunlop is impressed with the Cisco Unified Communications Express pilot in the Baltimore and New York offices. However, he has now instructed you to configure the two CUCME routers to interface with the PSTN as well as providing VoIP calls between the two locations over the frame relay link. Your tasks are as follows below.

Lab 4.1 – CUCME POTS and VoIP Dial Peer Tasks
1. Fuzzy is a cheap CIO – he has only provisioned the PRI for twelve channels. Configure an inbound POTS dial peer to support this.
2. Both Arliss Michaels and Ari Gold require a special inbound POTS line on top of their DID number associated with the PRI circuit. In Baltimore, make sure 410-555-5001 is routed to Arliss’s phone and 212-555-6001 is routed to Ari’s phone in New York. Ensure that Arliss and Ari always know who is calling inbound on these numbers.
3. Internal users must dial “9” to place and outside call. When dialing “9”, a different dial tone should be heard on the internal phones.
4. The PRI circuits at both locations should take precedence over the POTS connections for outbound calls.
5. Outbound destination patterns should be configured in such a way to allow outbound calls to be placed as quickly as possible, without any unnecessary delay.
6. Create appropriate inbound dialing for support the DID range and dial plans depicted in the diagram in Lab 3.1.
7. The security phones depicted in the diagram in Lab 3.1 should be mapped appropriately 410-555-5002 and 212-555-6002.
8. Internal users should be able to dial “H-E-L-P” on their IP Phones, which will ring the analog security phones at each location.
9. Configure an appropriate VOIP dial peer to allow four-digit dialing over the Frame Relay link. Maintain the G.722 codec (we’ll cover transcoding in a subsequent lab).
10. Configure each CUCME to route internal four-digit dial calls out the PSTN in the event of a WAN failure. Make sure the IP path is preferred first; the PRI circuit second; the POTS lines third.
11. You may not use any sort of digit translation on the Baltimore router to achieve task #10.
12. You must use digit translation on the NY router to achieve task #10.
13. Verify calling, including routing calls to the PSTN during a WAN failure.

Lab 4.2 – CUCME POTS and VoIP Dial Peer Task Verification
Most of the tasks in the lab are fundamentally similar, and in some instances, the same, as some of the VOIP and POTS dial peer tasks in the VOIP Fundamentals Labs. Therefore, rather than repeating myself, I will cover the tasks that are either new or different from the previous VOIP Fundamentals labs.


1. When configuring the MFT on the CUCME routers, I only configure twelve (12) B-Channels. The main reason is simply to conserve DSP resources for later configuration tasks, such as transcoding and conferencing.

!
controller T1 0/0/0
cablelength short 110
pri-group timeslots 1-12,24
!

2. To accomplish the task of routing one of the inbound POTS (FXO) lines directly to either Arliss or Ari, simply configure those FXO ports as a plar obx connection. You’ll also do the same for the security phone at each location.

!
voice-port 0/1/0
connection plar opx 2005
description inbound 212-555-6001
caller-id enable
!

3. To allow secondary dial-tone for external calls, simply add the “secondary-dialtone” command under the telelphony service. However, I could not get secondary dial-tone to work on the SIP CUCME router. The CUCME Administration Guide is extremely vague as to whether SIP CUCME supports this. There is no “secondary-dialtone” equivalent command under voice register global mode.

telephony-service
secondary-dialtone 9

User error (me) perhaps? Anyone else out there get secondary dial-tone to work on their SIP CUCME router?

4. There are at least two ways to configure the “H-E-L-P” requirement. One fairly simple approached is to configure another POTS dial-peer for destination-pattern 4357 (HELP), mapping it to the FXS with the attached analog device.

A second possibility is to create a translation rule, and then map this rule to each ephone-dn. For example:

!
translation-rule 1
Rule 0 ^4357 1611
!
!
ephone-dn 1 dual-line
translate called 1


5. I added the call fallback to each router. This significantly speeds up the process of falling back to a PSTN dial-peer in the event of a VOIP/WAN link failure. Below is an output of a call being routed over the PSTN (PRI) during a WAN failure.

##Shutting down the Frame Relay link##

Baltimore(config)#int s0/2/0
Baltimore(config-if)#shut

NewYork#sh call fallback cache

Fallback Mode : Active

Type Id IP Address Codec Delay Loss ICPIF Reject Accept
---- ----- ---------- ----- ----- ---- ----- ------ ------
ICMP 1 172.16.1.1 No Codec N/A 100 N/A 0 0

1 active probes


##placing a call from 2005 to 1005##

NewYork#
Jan 15 05:03:04.578: %ISDN-6-CONNECT: Interface Serial0/0/0:11 is now connected to 4105551005 N/A
Jan 15 05:03:04.578: %ISDN-6-CONNECT: Interface Serial0/0/0:11 is now connected to 4105551005 N/A
NewYork#


6. One additional comment/observation before providing the pertinent “show run” edits for each CUCME router. I was having no luck getting the SIP phones to display their local directory numbers (NYC Router). After reviewing the various SIP XML and systems files, and comparing them against their SEP counterparts, something caught my attention.

I noticed that that the while the SEP XML and systems files where telling the SCCP phones to point to the router’s local director via HTTP, I did not observe this on the SIP CUCME router SIP XML and systems files. Therefore, under voice register global, I add a URL for the directory on the local router. This seems to have rectified the issue.

voice register global
url directory http://10.1.22.1:80/localdirectory

7. Below are the pertinent “show run” edits for the Baltimore and New York CUCME routers.

Baltimore CUCME

Baltimore#sh run
!
card type t1 0 0
!
network-clock-participate wic 0
!
isdn switch-type primary-ni
!
controller T1 0/0/0
cablelength short 110
pri-group timeslots 1-12,24
!
!
translation-rule 1
Rule 0 ^4357 1611
!
!
ip sla responder ##for the call fallback##
!
!
call fallback active
!
!
voice-port 0/0/0:23
description PRI ciruit to PSTN
!
voice-port 0/1/0
connection plar opx 1005
description inbound 410-555-5001
caller-id enable
!
voice-port 0/1/1
connection plar opx 1611
description inbound 410-555-5002
caller-id enable
!
voice-port 0/3/0
description Baltimore-Security
caller-id enable
!
voice-port 0/3/1
caller-id enable
!
!
dial-peer voice 1 pots
description outbound via pri
preference 1
destination-pattern 9[2-9].........
port 0/0/0:23
!
dial-peer voice 2 pots
description outbound via 410-555-5001
preference 2
destination-pattern 9[2-9].........
port 0/1/0
!
dial-peer voice 3 pots
description outbound via 410-555-5002
preference 2
destination-pattern 9[2-9].........
port 0/1/1
!
dial-peer voice 4 pots
description inbound PRI dial-peer
incoming called-number .
direct-inward-dial
port 0/0/0:23
!
dial-peer voice 1611 pots
description Baltimore-Security
destination-pattern 1611
port 0/3/0
!
dial-peer voice 2000 voip
description VOIP Peer to NYC
preference 1
destination-pattern 20..
monitor probe icmp-ping 172.16.1.2
session protocol sipv2
session target ipv4:172.16.1.2
dtmf-relay rtp-nte
codec g722-64
!
dial-peer voice 2001 pots
preference 2
destination-pattern 20..
port 0/0/0:23
forward-digits all
prefix 212555
!
dial-peer voice 2002 pots
preference 3
destination-pattern 20..
port 0/1/0
forward-digits all
prefix 212555
!
dial-peer voice 2003 pots
preference 3
destination-pattern 20..
port 0/1/1
forward-digits all
prefix 212555
!
!
ephone-dn 1 dual-line
number 1001
label Cal Ripken 1001
description 410-555-1001
name Cal Ripken
translate called 1
!


New York CUCME

NewYork#sh run
!
card type t1 0 0
!
network-clock-participate wic 0
!
isdn switch-type primary-ni
!
!
voice register global
url directory http://10.1.22.1:80/localdirectory
!
voice translation-rule 1
rule 1 /\(10..\)/ /410555\1/ ##for the PSTN fallback translation##
!
!
voice translation-profile prefix
translate called 1 ##for the PSTN fallback translation##
!
!
controller T1 0/0/0
cablelength short 110
pri-group timeslots 1-12,24
!
!
ip sla responder
!
call fallback active
!
!
voice-port 0/0/0:23
description PRI ciruit to PSTN
!
voice-port 0/1/0
connection plar opx 2005
description inbound 212-555-6001
caller-id enable
!
voice-port 0/1/1
connection plar opx 2611
description inbound 212-555-6002
caller-id enable
!
voice-port 0/3/0
description NewYork-Security
caller-id enable
!
voice-port 0/3/1
caller-id enable
!
ccm-manager fax protocol cisco
!
mgcp fax t38 ecm
!
!
!
dial-peer voice 1 pots
description outbound via pri
preference 1
destination-pattern 9[2-9].........
port 0/0/0:23
!
dial-peer voice 2 pots
description outbound via 212-555-6001
preference 2
destination-pattern 9[2-9].........
port 0/1/0
!
dial-peer voice 3 pots
description outbound via 212-555-6002
preference 2
destination-pattern 9[2-9].........
port 0/1/1
!
dial-peer voice 4 pots
description inbound PRI dial-peer
incoming called-number .
direct-inward-dial
port 0/0/0:23
!
dial-peer voice 2611 pots
description NewYork-Security
destination-pattern 2611
port 0/3/0
!
dial-peer voice 1000 voip
description VOIP Peer to Baltimore
preference 1
destination-pattern 10..
monitor probe icmp-ping 172.16.1.1
session protocol sipv2
session target ipv4:172.16.1.1
dtmf-relay rtp-nte
codec g722-64
!
dial-peer voice 1001 pots
translation-profile outgoing prefix
preference 2
destination-pattern 10..
port 0/0/0:23
forward-digits all
!
dial-peer voice 1002 pots
translation-profile outgoing prefix
preference 3
destination-pattern 10..
port 0/1/0
forward-digits all
!
dial-peer voice 1003 pots
translation-profile outgoing prefix
preference 3
destination-pattern 10..
port 0/1/1
forward-digits all
!
dial-peer voice 4357 pots
description NewYork-Security
destination-pattern 4357
port 0/3/0
!

7 comments:

kyrios said...

Hi Mark... i've seen your blog, please let me tell you, it's great, god job..

I have a question, i hope you can help me..

How can i register ephones on CUCME from WAN?.. SIP no problem, but SCCP none..

I try with..
ip source-address 192.168.10.1 port 2000 secondary "public IP Address"
but nothing happens..

On my cipc this show..
http://www.imagechile.net/img/img9_1236645609w.JPG

Greetings!

Mark G. Reyero said...

Hi Kyrios,

Are you connecting across a "WAN" simulated in your own lab, or are you actually trying to connect via the public Internet?

If the public internet, your ISP is likely filtering the 192 RFC 1918 address.

If you are working in your own lab, have you looked at the Layer 3 routing? For example, you may want to troubleshoot the routing via some extended pings. You may also want to check the tftp bindings in your CUCME.

Hope these ideas help!

kyrios said...

Hi Mark,
My WAN connection is INTERNET.. yes i know, i must ensure with ACL... i have that in mind.

There's no filtering by my ISP.

Both routers have a public address, meanwhile i've been created a vpn to register ephones via internet.

but, without this (vpn) is possible do it?..

Thanks again
kyrios

Mark G. Reyero said...

You would need to create an IPSEC tunnel and route your RFC 1918 addressed through that tunnel.

anand said...

Hi Mark,

Thanks for all the information.I have a question for you.I have the routers in H323 mode.I tested pstn fallback in H323 by looking at the probes in wireshark and the voice quality is good.

I want to do PSTN fallback in SIP.I configured the originating router with call fallback active command and ip sla responder on destination router.PSTN fallback is not working in SIP and i can't see any probes in wireshark.

I tried imapiring with abnormal values and even then it is trying to go through WAN and the voice quality is degraded.From this i think PSTN Fallback is not working.Can you help me with this.

Rehan said...

Mark excellent blog.
I have a question;

I want to for outbound calls split certain endpoints to specific POTS.

e.g. Redirect ext 6900 to 213-342-1111

Redirect ext 6800 to 714-212-1234

Any help will be greatly appreciated.
Thanks

Ray

Rehan said...

Mark excellent blog.
I have a question;

I want to for outbound calls split certain endpoints to specific POTS.

e.g. Redirect ext 6900 to 213-342-1111

Redirect ext 6800 to 714-212-1234

Any help will be greatly appreciated.
Thanks

Ray