Recently, I  had a request from my friend asking for a help to divert the CME reception calls at 5pm to a pre-defined mobile number and UN-divert the call at 8:30am when the receptionist started working. I was going to post it last month but due to being busy in project work and other commitments I couldn’t possible get even a minute to post this !!

Anyhow, It took me about 30 minutes to think about the best solution and here is what I came up with and it works perfectly.

If you have any better way to achieve this please let me know.

required: CME router version 4 or 7x (didn’t test on older), any model of cisco router.

Here are the steps:

Steps #1
———————————————————————————
Copy below call-divert.tcl and call-un-divert.tcl in a notepad and save them as .tcl extension

User Access Verification

Username: push

Password:

HQ-CME>en
HQ-CME#dir *.tcl
Directory of flash:/
94  -rw-         108  Jul 17 2009 14:28:02 +10:00  frog.tcl
95  -rw-         481  Jul 17 2009 14:40:20 +10:00  call-divert.tcl
96  -rw-         484  Jul 17 2009 14:40:30 +10:00  call-un-divert.tcl

512065536 bytes total (443867136 bytes free)

HQ-CME#

——————- call-divert.tcl ——————————

HQ-CME#more flash:call-divert.tcl

#
# Copyright (c) 2009 FROG silly billy

# All rights reserved.
# by:            Push Bhatkoti 28 Mar 2009 / CCIE# voice 21569

# title:          Call divert
# name:       call-divert.tcl

# desc:     This script runs in conjunction with IOS KRON which  diverts CME DID number         #                  office phone to a Mobile phone after 5pm

#

ios_config “ephone-dn 50″ “call-forward all 00412733020″
ios_config “end”
ios_config “do wr mem”

HQ-CME#

——————–call-un-divert.tcl————————–

HQ-CME#more flash:call-un-divert.tcl
#
# Copyright (c) 2009 FROG silly bill
# All rights reserved.#
# by:       Push Bhatkoti 28 Mar 2009/ CCIE Voice#21569
# title:    CME router Call UN-divert at 8:30am
# name:    call-un-divert.tcl
# desc:     This script runs in conjunction with ours IOS KRON, which UN-diverts CME DID
#               so that when receiptionist starts in the morning will be able to attend the calls
#
#           * download the file into flash:call-divert.tcl
#

ios_config “ephone-dn 50″ “no call-forward all  004127492820″
ios_config “end”
ios_config “do wr mem”

HQ-CME#

HQ-CME#

Now you’d be thinking what the hell ephone-dn is doing here.  Actually ephone-dn 50 is a receiption octoline here is the sample config of DN.

ephone-dn  50  octo-line
number 2000 secondary 94232000
pickup-group 88
label Nice-Dolls – 3002
description 02 23233002
name Reception
call-forward busy 4222
call-forward noan 4222 timeout 50
corlist incoming INTL-COR
no huntstop
hold-alert 120 originator
transfer-mode consult

IN above ephone-dn, 94232000 is the main DID number which hits CME router and receptionist picks it up and then transfers to the phones.

Step #2:
———————————————————————————

Put the above 2 files in a TFTP server and then copy them into the flash:

HQ-CME#dir *.tcl
Directory of flash:/
95  -rw-         481  Jul 17 2009 14:40:20 +10:00  call-divert.tcl
96  -rw-         484  Jul 17 2009 14:40:30 +10:00  call-un-divert.tcl
512065536 bytes total (443867136 bytes free
HQ-CME#

Step #3:

Final step, Kron about two scripts to run them in a required time. BTW, cron used to be a unix/linux worlds keyword but it seems Cisco has  adapted it by using a fancy word like “Kron” duh!

The original requirement was 5pm calls divert to a mobile phone and 8:30 am call un-divert.

Here is how they should be cron’d:

First two cron policy lists and reference the two .tcl scripts into them:

kron policy-list call-divert
cli tclsh call-divert.tcl  ! for call divert

!
kron policy-list call-un-divert
cli tclsh call-un-divert.tcl ! for call un-divert

Then create 2 kron occurance and put above policy list with required divert / undivert time in them.

kron occurrence call-divert at 14:59 recurring !
policy-list call-divert Divert receiption call at 4:59pm

!
kron occurrence call-un-divert at 8:29 recurring
policy-list call-un-divert ! Undivert receiption call at 8:29AM
!
If anyone has better solution, please feel free to provide your feedback.

-Push Bhatkoti

CCIE voice#21569

We spent a lot of time in debugging and troubleshooting DSP issue. Recently, I used this DSP test (a hidden one yes) command and it gave me immediately result that the DSP on the router has gone alley and as a result the calls from one location to another one were having issue.

This handy tool has been tested on ISR modules.

Router# test voice driver
After typing out above command, you will see below menu driven options:

Enter VPM or HDV or ATM AIM or NM-HD-xx or HDV2 slot number : 1

C2800 Debugging Section;

1 – FPGA Registers Read/Write

2 – 5510 DSP test

3 – DSPRM test

4 – HDLC32 test

5 – Quit
Select option : 2
5510 DSP Testing Section:

1 – Reset ALL DSPs

2 – Reset 1 DSP

3 – Download DSPware

4 – CHPIR Enable/Disable

5 – Display c5510 ring

6 – Show HPI RAM

7 – Show eHPI memory thru Relay command

8 – Show Controller

9 – c5510 Keepalive Enable/Disable

10 – Download bootloader using indirect slave access

11 – Write HPI RAM

13 – DSP application download

14 – faked dsp crash

15 – Wait in Firmware Restart Indication

16 – Display DSP Keepalive Status

17 – QUIT
Select option : 16
DSP Keepalive Status Display:

=============================

DSP 1 is not UP, State: 5, Keepalive Sent: 0, Skip 0

DSP 2 is not UP, State: 5, Keepalive Sent: 0, Skip 0

DSP 3 is not UP, State: 5, Keepalive Sent: 0, Skip 0

DSP 4 is not UP, State: 5, Keepalive Sent: 0, Skip 0

DSP 5 Not Exist

DSP 6 Not Exist

DSP 7 Not Exist

DSP 8 Not Exist

5510 DSP Testing Section:

1 – Reset ALL DSPs

2 – Reset 1 DSP

3 – Download DSPware

4 – CHPIR Enable/Disable

5 – Display c5510 ring

6 – Show HPI RAM

7 – Show eHPI memory thru Relay command

8 – Show Controller

9 – c5510 Keepalive Enable/Disable

10 – Download bootloader using indirect slave access

11 – Write HPI RAM

13 – DSP application download

14 – faked dsp crash

15 – Wait in Firmware Restart Indication

16 – Display DSP Keepalive Status

17 – QUIT

-Push

CCIE Voice#21569

hehe… not in 10 mintues!! it’s in 20 minutes

It has been a long time since my previous posting on this blog. I have been busy doing all sort of bits and pieces!!

It seems that people are finding issues with integrating Cisco Call manager with Cisco Presence server. I have to agree to it because there are very limited documentation available for the public on the net. So I thot to spare some time and post it for the benifit of all.

I’m assuming you have already installed and configured cisco Call manager 7x. Two phones are already registered to it.

Also assuming that you have done basic installation of Cisco presence server. As you know Installing Cisco presence is like installing Yahoo messenger in your windows XP.

The next .. next.. next and finish strategy . It’s just the same as you add Call manager subscriber to a Publisher (in 5/6/7x), same secret, same Pub Ip address will be asked during the installation.

Here are the summary steps of integrating Cisco call manager 7x with Cisco presence 7x.

Step#1: Enable presence globally on Cisco Call manager

By default presence subscription is disable on CCM.

System>Service parameter>Cisco Call Manager>

search for “Inter-presence” key word and set “Allow Subscription”

Step#2: Create SIP trunk Security Profile in CCM

Special setting is required for SIP trunk which runs from CCM to Presence.

Copy “non Secure SIP Trunk Profile” to “Presence non-secure SIP trunk Proifle”

Modify below parameters:

1. Device security mode: Non-Secure

2. Incoming Transport type: TCP+UDP

3. Outgoing Transport Type: TCP

4. Incoming Port 5060 (untick Enable digest authentication)

5. Enable application Level Authentication UNTICK

6. Accept Presence Subscription TICK

7. Accept Out-of-Dialogue REFER TICK

8. Accept Unsoliciliated Notification TICK

9. Accept Replace header TICK

10. Transforms security status UNTICK

Save it

Step#3: Add a SIP trunk now from CCM to Presence

Device>TRUNK>SIP-trunk>

Protocol = SIP

fill below:

1. Device Name : PRESENCE-TRUNK

2. Description : blah blah

3. Device Pool : DP_HQ

4. Common Dev conf : None

5. call classification : On-Net

6. Media resource Grp : MRG_HQ

7. Location : HQ_LOC

8. AAR GROUP : HQ_AARG (if not using AAR leave empty)

9. Packet Capture mode : None

10. Packet Capture duration: 0

11. MTP required : TICK

12. Retry Video call as audio : TICK

13. SIP information – Desitnation Add: 142.2.64.51

DST is a SRV : UNTICK

1. Destination port : 5060

2. SIP PROFILE : Presence non-secure SIP trunk Proifle

Save above.

Step#3: Make your IP Phone presence capable

1. Register a phone 2001 name it HQ-Phone1

2. Create end user “test” and associate HQ-Phone1/2001 with the “test” user

3. Make sure test user is a part of “Standard CCM End User” and “standard CTI enable”

4. Make sure Primary extension “2001” is selected when you create the above “test” user

Step#3: Add an application user for IPPM and MOC CTI ports

This will be used by Presence server to initiate IP Phone services:

A) Go to > User Management>Application User>

1. User ID : IPPM

2. pass : blah

3. Presence Grp : Standard

4. Groups : Standard CCM End User

save it

Repeat above “A” steps for moc_user as well. moc_user will be used by MOC CTI user in Presence. All user who want presence using Microsoft MOC client will be associated to this user.

Make sure all “accept” tick boxes are TICKED on moc_user.

B) Go to > SYSTEM>Application Server> Add NEW

add Presence server IP address here I..e 142.2.64.51

save this as well. Damn too many things to save

Step#4: Create IP Phone service URL

Go to> Device>Device Settings> IP Phone Service

1. Service Name : IP PhoneMSG

2. ASCII Service Name : IP PhoneMSG

3. Service Description : Blah

4. Service URL : http://142.2.64.11:8081/ippm/default?name#Device blah

5. Service Category : XML Service

6. Service Type : Standard IP Phone Service

7. Blank

8. Blank

9. Enable : TICK

It’s standard Phone URL we create and subscribe in CCM. Nothing new!!

Make sure you copy the correct URL from the DOC CD.

Save above as well.

****Then subscribe above service to HQ phone1/2001*****

Step#5: Enable presence Licensing for each user

Go to> System>License>Capability Assignment>

Then Find the end user you want to assign the presence license.

Tick the user and hit <Bulk Assignment>

a new pop up window with pop-up. Tick both check-boxes in that and save.

1. Enable CUP – TICK
2. Enable CUPC - TICK

Step#6: Add CUPC client in CCM for HQ 2001 phone

The trick here is, this is a dummy phone which will control provide HQ Phone1/2001’s presence information to Presence server. Add this dummy presence client and add a HQ2001 DN to it.

Go to> Device>Phone> add NEW

Phone Type : Unified Personal Communicator then hit <NEXT>

1. Device Name : XXXCISCO

2. Device Pool : DP_HQ

3. Phone Button Templ : Personal Communicator SIP blah

4. CSS : Blah select all common things

5. Owner user ID : test ← a must

6. leave everything else default

7. Device security profile : Unified Personal Communicator Standard

8. SIP profile : Standard SIP profile

9. Digest User : test ← a must ** read more about this

   Save everything above so far so good , well done

   Now add a DN to this above device: (same as HQ phone1 shared one)

10. Directory Number : 2001

11. everything else … default or your customizeble >> Save above

Once above dummy device is added, associate this device with “test” user we created previously. Now you remember we have 2 device associated with this user: A) 2001 phone and B) XXXCISCO

Also Make Physical phone DN2001 has “test” user associated with it. This is the last option in line 2001’s setting before “save” button. If this has not been done and you run presence diagnostic it will keep telling you that “No line appreance existed in CCM blah blah”

That s all we needed to do on Call Manager. Now Jump on the Presence BOX.

Step#7: Presence box general configuration:

After installing basic presence, you’ll see presence post install setup screen on your web browser by typing presence Server IP address on your browser and supplying credentials to the login screen.

(hehe, I call it doggie screen, sitting like a dog waiting for your fingers to feed it like dog wait for food )

So you’ll see “Post Install Setup” screen with below options:

1. CUCM Publisher IP address : 142.2.64.11 (default, not changeble)

2. AXL User : Administrator

   1. 1. 1. 1. 1. (I’m too lazy to create a new one, for production server you must create a new AXL user for security reason.)

3. Axl password : blah blah..

4. Confirm password : blah blah <then hit the “NEXT”>

5. Security password : blah blah (whatever you supplied during installation)

6. Then hit the “CONFIRM” (Ignore the warning)

   Finally you will get 3 options:

   A) Home B) Status C) TOPOLOGY

7. Click on “HOME” you’ll see you are in a new home i.e. preseence main admin page.

Step#8: Upload License and Activate presence Services

1. first upload the license if you haven’t done that so far.

2. GO to > Cisco Unified Servicebility>>Tools>Activate services
   Activate all services, it will take 2-3 minutes.

Step#9: Configure Presence

Jump straight on Presence Admin page>>Diagnostic>System Troubleshootor

Pay attention to RED crossed balls and yellow exclamation ! Signs and fix them one by one.

1. Under Presence Engine: Click on FIX under “no commnication presence” this will take you to add presence gateway:

   Add NEW>

   Presence Gateway type : CUCM
   description : blah

   Presence Gateway : 142.2.64.11 ← CCM IP

   Double check the settings under below menus:

2. SYSTEM> CCM Publisher : Check all parameter under this

3. SYSTEM> Application Listioner>Default class SIP TCP Listioner (make sure its what you have defined in the SIP trunk on CCM – transport method TCP or UDP, both should have the same protocol/port) we are using:
   Protocol = TCP
   PORT = 5060

4. SYSTEM> Security>INCOMING ACL
   Add NEW> description=blah/all address pattern=all

Step#10: Tune the Presence Engine’s Service parameter (same as we do with CCM)

SYSTEM>> Service Parameter>Select active CUPS Server> Select Presence Engine

1. Search “Proxy Domain” and set it to : 142.2.64.51 (or domain name)

2. Search “Transport Preferred Order” and set it to : TCP/UDP/TLS

Step#10: Iconfigure P Phone Messenger on Presence server

Application>IP Phone> Setting

1. IPPM Application Status : ON

2. Application user Name : IPPMSG (created in step 3A)

3. Application Password : blah…

4. confirm password : Blah

5. Max Instant message : 25 default

6. Subscription timeout : 3400 default

7. Publish timeout : 3600 default

   Hit “SAVE”

Step#11: Select a SIP trunk between Presence to CCM

Tell presence which SIP trunk should be used for pumping calls to CCM.

Presence>>Setting>

1. CUP CVP Support : UNTICK

2. MAX Contact List Size : 200

3. Enable Instalt messeging : TICK

4. Enable SIP Publish on CUCM TICK

5. CUCM SIP Publish Trunk : <Select_Your_Trunk><– A MUST

Don’t forget to save after above. Above SIP trunk will be automatically listed in above “5”. This we is the one we created on CCM.

Step#12: Set TFTP address for IP COMMUNICATOR Clients

Application>Unified IP Personal Communicator>Settings

1. Proxy Listenor : Default Cisco SIP proxy TCP Listenor

2. Primary TFTP : 142.2.64.11 (CCM pub tftp)

3. Backup TFTP : 142.2.64.12 (sub tftp) or whatever

LDAP – if you are using LDAP put LDAP parameters there. Else disable it.

Step#13: For MOC client define CTI Gateway

Application>>CUCM CTI Gateway>Settings

1. Application Status : ON

2. Application Username : moc_user (make sure its created on CCM as app usr)

3. Application Password : blah

4. Confirmed Password : blah

5. CUCM Address : 142.2.64.11 (CCM address)

Now time to run the Presence troubleshooter again. This will tell you whats remaining and how to fix it. Once those are done, activate the presence and other services and bingo.. you are ready to rocck!! on hang on, oh no!! Still remaining:

1. MOC integration

2. Creating users and testing presence

3. Voicemail integration with Presence

I’m sleepy now, will continue remaining part next day!!

————————————————————————————————————————————

How to integrate Cisco Presence with Call manager

————————————————————————————————————————————

Push Bhatkoti (CCIE voice#21569)

]]> http://pushkarbhatkoti.wordpress.com/2009/06/08/cisco-presence-integration-with-ccm-7x-in-20-minutes/feed/ 7 pushkarbhatkoti http://pushkarbhatkoti.wordpress.com/2009/02/16/how-to-check-call-manager-6x-5x-7x-dhcp-lease/ http://pushkarbhatkoti.wordpress.com/2009/02/16/how-to-check-call-manager-6x-5x-7x-dhcp-lease/#comments Mon, 16 Feb 2009 09:39:40 +0000 pushkarbhatkoti http://pushkarbhatkoti.wordpress.com/?p=197 ]]>

How to check DHCP leases status on cisco CCM 6x

Cisco CCM 5/6/7x still have built-in DHCP server.

Cisco call manager : Obviously, this is a big enterprise level IPTELEPHONY product which really “WORKS”. No product is perfect as every product has its cavete and room for improvement.

One of the issue I came across was the issue with “How to check” what address is allocated to which IP phone.
It was long due in my TODO list but I have been flat out recently at work and didn’t have time to do blog:

Today I just got home and as usual I  was checking my personal email and found an email message from a guy who came through google and hit my  blog asking for the same thing
Thanks Leonardo for reminding this.

After reading above request, I decided to flock through the CCM and cross this off my “todo list”

In the past, so far I have done 12-16 call manages 5/6x installation but most of the customer were using their dedicated DHCP and a few of them were using Call manager’s built-in DHCP server. Back to CCM 4x days, Cisco used to use Microsoft DHCP server (WIN3k or WIN2k) which was full flag and can tell who have what Address at any given time.

In CCM 5/6/7x there is no easy way to check the DHCP lease(es) allocated to IP phones by the DHCP server.

CCM new boxes are Linux based and the DHCP server implementation is also opensources software.

It’s basically opensource DHCPD daemon (/etc/dhcpd.conf style) which is good and stable but what a shame, this BIG commercial product and it doesn’t give us a GUI or CLI interface to see what address are leased!
I believe Cisco will give it a go and improve it in the future.

For the time being, here is my workaround to find out the leases (not easy method though as i said above).

There is also some other method to know the status of DHCP leases in cisco CCM DHCP server

Basic information can be collected via SNMP or Syslog server:
- CCM sends DHCP alarm/trap alerts to a SNMP platfarm (HPopenview, Ciscoworks etc)
- CCM also sends same alarms/alerts to a preconfigured Syslog server.

Above information are very basic and that still doesnt’ really tells the status of the leases. Its noticeble and I have observed that the SNMP/SYSLOG alert only triggers  when there is something reallly wrong with DHCPD. For example DHCP pool gets full or there are no IP addresses to allocate to DHCP clients.

Above methods are useless as nobody will wait when there is really problem. Everyone want to see the leases when they want and when they feel. Just think about  stressing alert telling “Hey bobby the DHCP server has no IP address”
was sent to you when you were in a night club.

Another way to see the lease alerts could be RTMT tool. but it still doesn’t do the good job. My method 100% shows DHCP leases. Its a bit hard work but really works.

Here is my method which really works, just need to get the root access to the box. To get access to root shell you have 2 ways:

1. Create a remote account from “disk recovery” menu and ask TAC to provide you password.

2. Risky method: the following steps detail the process on how GRUB is reinstalled on the master boot record:

• Boot the system from an installation boot medium.
• Type linux rescue at the installation boot prompt to enter the rescue environment.
• Type chroot /mnt/sysimage to mount the root partition.
• Then create username as u create in regular LINUX. ‘useradd’ passwd
• edit /etc/passwd file and set created users userID and GUID to 0:0
• note: 0:0 is for root
• after that u can simply ssh it using putty.exe or ssh user@ipaddresofccm

Once you got access to CCM root have a look at this file:

cat /etc/dhcp3/dhcp.conf <———-this is the main DHCPD config file

Here in that file you will not see anything about where lease file will be pointing to. This is by default pointed to

Now create a file which is visible through the operating system CLI (damn CLI).

type touch /common/download/dhcpleases.hehe

now link default dhcpleases.hehe file to dhcpd.lease file which is in /var/lib/dhcp/dhcpd.leases. /var/lib/dhcp/dhcpd.leases is the default file where DHCPD daemon writes all leaes status.

alternately if you are conforteble with Linux u can just simply do this:

Edit the /etc/dhcpd.conf file and just add this line:

lease-file-name “/common/download/dhcpd.leases”

Restart the DHCP daemonn by issueing ’service dhcpd restart or service dhcp3-server restart’

It will write all status of dhcp leases to /common/download/dhcp.leases file.

Now jump on your CCM friendly (heydid i say friendly?) GUI and download the dhcpdlease.hehe file and u can see who have what leases. So thats all about it.

Those brainless programmers did not think about doing this little job from Administrator point of view. I know there is a RTMT tool but it would be nice to see it somewhere on the GUI.

Another tip on DHCP server:
—————————————–

How to make sure that whatever is configured via CCM GUI under DHCP server is 100% correct and working?
Yes you can confirm this using SQL query commands:

STEP#1: Login using ssh to the CCM server

STEP#2:
- To check DHCP servers: issue “run sql select * from dhcpserver”

- To check DHCP scopes: issue “run sql select * from dhcpsubnet”

Good luck and happy learning…

PS: if you found this article useful, don’t forget to send me a postcard!! hehe..

CCIE voice#21569

Cisco recommends purging CCM 4x CDR database before running DMA tool to migrate CCM 4x to 5x/6x.

You may use either method shown below to reduce the CCM 4 CDR database.

Method#1 (My favourite)

Shrink the database via the SQL Enterprise Manager.

Note: This is done on the Publisher.

1. Select Start > Programs > MS SQL Server > Enterprise manager.
2. Choose the Publisher server.
3. You can either go to the ART database or CDR database, based on which file is large.
4. Right-click the database.
5. Select All Tasks > Shrink database and click Files.
6. Select the database file CDR and CDR_log on the new window.
7. Shrink each of them.This process takes some time.If the file is still large or SQL is not able to shrink the transactional file, use Solution 3.

Method#2:

For SQL 2000 (Cisco CallManager 3.3 and 4.0), complete these steps to reduce the size of the CDR_log.LDF. This is located at C:\Program Files\Microsoft SQL Server\MSSQL\Data.

Use these commands at the command prompt:

ART
C:\>osql -E
1>use art
2>go

1>backup log art with no_log
2>go

1>dbcc shrinkdatabase (art)
2>go

CDR
C:\>osql -E
1>use cdr
2>go

1>backup log cdr with no_log
2>go

1>dbcc shrinkdatabase (cdr)
2>go

Cheers
-Push CCIE#21569

Step#1 List number of node in the cluster using CLI:

admin: run sql select name,nodeid from ProcessNode

name               nodeid

================== ======

EnterpriseWideData   1

222.22.2.22          4

222.22.2.25          2

admin: show perf query class "Number of Replicates Created and State of Replication"

==>query class :

- Perf class (Number of Replicates Created and State of Replication)

has instances and values:

ReplicateCount  -> Number of Replicates Created   = 344

ReplicateCount  -> Replicate_State                = 2

Noticed 2 means replication status between two serves are okay up and running.

admin: utils diagnose module validate_network

Log file: /var/log/active/platform/log/diag1.log

Starting diagnostic test(s)

===========================

test - validate_network    : Passed                      

Diagnostics Completed

Recovering CCM Administrator and Security Passwords

This section replaces the section Recovering the Administrator Password in the “Log In to Cisco Unified Communications Operating System Administration” chapter” of the Cisco Unified Communications Operating System Administration Guide for releases 5.0(4), 5.1(1), 6.0(1), and 6.1(1a).

If you lose the administrator password or security password, use the following procedure to reset these passwords.

Note During this procedure, you must remove and then insert a valid CD or DVD in the disk drive to prove that you have physical access to the system. He he .. you must walk to the Server room or a datacenter to do this. Use KVM switch or similar thing to access VGA of CM.

Procedure

Step 1 Log in to the system with the following username and password:

•Username: pwrecovery

•Password: pwreset

The Welcome to platform password reset window displays.

Step 2 Press any key to continue.

Step 3 If you have a CD or DVD in the disk drive, remove it now.

Step 4 To continue, press any key.

The system tests to ensure that you have removed the CD or DVD from the disk drive.

Step 5 Insert a valid CD or DVD into the disk drive.

The system tests to ensure that you have inserted the disk.

Step 6 After the system verifies that you have inserted the disk, you get prompted to enter one of the following options to continue:

•To reset the administrator password, enter a.

•To reset the security password, enter s.

•To quit, enter q.

Step 7 Enter a new password of the type that you chose.

Step 8 Reenter the new password.

The password must contain at least 6 characters. The system checks the new password for strength. If the password does not pass the strength check, you get prompted to enter a new password.

Step 9 After the system verifies the strength of the new password, the password gets reset, and you get prompted to press any key to exit the password reset utility.

Caution The security password on all nodes in a cluster must match. Change the security password on all machines, or the cluster nodes will not communicate.

Software Feature License Information Omitted from

[root@ccm5 bin]#
[root@ccm5 bin]# pwd
/usr/local/platform/bin
[root@ccm5 bin]#

[root@ccm5 bin]# ./pwreset

********************************************************
********************************************************
**                                                    **
**   Welcome to Platform password reset               **
**   Admin and Security password reset are possible   **
**                                                    **
********************************************************
********************************************************

You will be required to remove, then insert any valid CD/DVD media
in order to prove you have physical access to the system.

To begin you will need to remove any media from the CD/DVD drive.
You may press Control-C at any time to abort.

Remove any media from the CD/DVD drive and press any key when ready…
testing for removal of CD/DVD media
Please insert any valid CD/DVD media.
Press any key when ready…

You must insert the CD/DVD media to continue
Press any key when ready…

Thank you, you may now proceed with Platform password reset.

Enter a for admin password reset.
Enter s for security password reset.
Enter q to Quit.

Source: www.cisco.com

no! not possible – its very secure appliance!

http://www.hackingalert.com/hacking-articles/hacking-tools.php

Note: This article is pulled from:

Source: http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_configuration_example09186a00808ff666.shtml
All credit goes to Cisco.com
(Cisco keep moving the pages here and there so, I thot to keep a copy of it for benifit of everyeone.)

Introduction

Today, the telecommunications industry is in the process of making the transition from long establishing switching and transport techonologies to IP-based transport and edge devices. The IP communication revolution has started to create a tremendous commercial impact in small and medium businesses. These small and medium businesses are realizing that the use of IP is very efficient because IP can use Voice, Video, and Data capabilities over a single network, instead of using three separate special-purpose networks. Figure 1 shows an IP telephony deployment trending towards IP trunking.

Figure 1 – IP Telephony System

IP PBXs are starting to predominate in the business of the Voice technology, and the TDM PBXs are no longer the primary source as the crossover going between two Voice networks. The usage of the TDM PBXs has decreased in the last couple of years, and the use of the IP PBX is becoming a good investment in IP LANs and WANs. In order to connect to the PSTN, PBXs need some sort of trunking such as TDM (T1/E1) or analog lines. IP PBXs can access the PSTN using these types of trunks, but need a media gateway that converts the IP voice traffic to traditional PSTN, which sometimes can result in successive translation from IP domain to TDM domain. These successive translations increase the maintenance costs of the gateways, increases latency, and reduces voice quality.

In order to avoid these problems, the IP PBXs use protocols for session initiation and management, the most prominent of which is Session Initiation Protocol (SIP). This document provides a description on SIP trunking and Cisco CallManager Express (CME), and a configuration to implement an IP-based telephony system with CME using SIP trunking for inbound and outbound calls.

Prerequisites

Requirements

Ensure that you meet these requirements before you attempt this configuration:

• CME release 4.1 is installed
• An image of Cisco IOS® Software Release 12.4(11)XJ or IOS 12.4(6th)T is on the router
• An NM-CUE module is installed with CUE release 2.3.4

Components Used

The information in this document is based on these software and hardware versions:

• Cisco 3825 Router on Cisco IOS Software Release 12.4(11)XJ
• Cisco Catalyst 3550 Switch on Cisco IOS Software Release 12.4
• Cisco IP 7960 Phone
• Cisco CallManager Express 4.1
• Cisco Unity Express 2.3.4

The information in this document was created from the devices in a specific lab environment. All of the devices used in this document started with a cleared (default) configuration. If your network is live, make sure that you understand the potential impact of any command.

Conventions

Refer to the Cisco Technical Tips Conventions for more information on document conventions.

SIP Protocol

SIP is an ASCII based, application-layer control protocol that can be used to establish, maintain, and terminate calls between two or more endpoints. SIP has rapidly emerged as the standard protocol used in IP communications, because it is a multimedia protocol that can be used for video sessions and instant messaging in addition to voice. Also, SIP can handle conference sessions and broadcasts, as well as one-to-one sessions. SIP has great potential in transforming and developing the way people communicate. For this reason, Cisco has and continues to play an important role in taking a leadership to create new technologies that make SIP and its applications the standard of IP communications.

SIP trunks are similar to a phone line, except that SIP trunks use the IP network, not the PSTN. In addition, SIP trunks permit the convergence of voice and data onto common all-IP connections. In order to access the IP network using an SIP trunk, it is necessary that configurations be made on the service provider, as well as on the customer side. Customers need to set and configure CME, which is the PBX that will interpret the SIP signal adequately and pass traffic successfully. The service provider needs to configure an SIP Proxy Server. However, SIP trunks are more complicated to establish than regular PSTN trunks. The reason is that a customer faces challenges in handling different interpretation and implementations of SIP by equipment vendors, delivering security, managing quality of service (QoS), enabling Network Address Translation (NAT) and firewall traversal, and ensuring carrier-grade reliability and continuity of service.

These points describe why SIP trunks are becoming so apparent in small and medium businesses:

• Quick and Easy Deployment
• Improved Utilization of Network Capacity
• Potential for Consolidating and Lowering Telephony Costs
• Economical Direct Inward Dial (DID)
• Business Continuity

CME SIP Trunk Support

Cisco CME is an IP telephony solution that is integrated directly into Cisco IOS software. CME permits small and medium businesses to deploy voice, data, and video on a single platform. An IP telephony network is simple to set because CME runs on a single router, which delivers a PBX functionality for businesses. Therefore, by using CME, small and medium businesses can deliver IP telephony and data routing using a single converged solution with minimal costs.

DTMF Relay for SIP Trunks

CME started to support SIP trunking when CME 3.1 was released. However, some problems existed when an SIP phone called an SCCP phone or tried to access voicemail. The problem is that SCCP phones connected to CME require the use of out-of-band DTMF relay to transport DTMF (digits) across VoIP connections, and SIP phones use in-band tranports. A DTMF distortion existed between the two devices. When CME 3.2 was released, support was added to the DTMF relay. DTMF digits from SCCP could be converted to in-band DTMF relay mechanism through RFC2833 or Notify methods.

CME currently supports this list of DTMF internetworking for SIP to SIP calls:

• Notify <—> Notify since 12.4(4)T
• RFC2833 <—> Notify since 12.4(4)T
• Notify <—> RFC2833 since 12.4(4)T
• Inband G711 <—> since 12.4(11)T [Requires Transcoder]

CME currently supports this DTMF internetworking for SIP to SCCP calls:

• SCCP out-of-band—SIP Notify / RFC2833 since 12.4(4)T

Codec Support and Transcoding

Another important aspect to consider when you set up an SIP trunk is the codecs supported. Codecs represent the pulse-code modulation sample for signals in voice frequencies. SIP trunks support these codecs: G.711 and G.729. However, for different features such as Cisco Unity Express (CUE) and Music on Hold (MOH), only codec G.711 is supported. This means that voice calls that use SIP trunks using codec G.729 cannot access CUE, unless a transcoder exists to permit the compression and decompression of voice streams to match the CUE capabilities. MOH can also use codec G.729 to save bandwidth, but the codec does not provide adequate quality MOH streams. This is due to the fact that G.729 is optimized for speech. Therefore, you must force MOH to use G.711.

Call Forward

When a call comes in on an SIP trunk and gets forwarded (CFNA / CFB / CFA), then the default behavior is for the CME to send the 302 “Moved Temporarily” SIP message to the Service Provider (SP) proxy. The user portion of the Contact Header in the 302 message might need to be translated to reflect a DID that the SP proxy can route to. The host portion of the Contact Header in the 302 message should be modified to reflect the Address of Record (AOR) using the host-registrar CLI under sip-ua and the b2bua CLI under the VoIP dial peer going to the CUE.

Some SIP proxies might not support this. If so, then you need to add this:

Router(config)#voice service voip

Router(conf-voi-serv)#no supplementary-service sip moved-temporarily

Figure 2 shows the behavior of the CME system when the 302 message is disabled.

Figure 2 – Call Forward Busy (CFB) flow with 302 message disabled

This method will allow hairpinning of the 302 SIP messages for call forwards on the CME. The above is also required if there are certain extensions that have no DID mapping as the SP proxy might not know how to route such calls. If you disable the 3xx response, the calling-number initiator can be used to preserve the caller ID of the original calling party.

Call Transfer

When a call comes in on an SIP trunk to an SCCP Phone or CUE AutoAttendant (AA) and is transferred, the CME by default will send a SIP REFER message to the SP proxy. Most SP Proxy Servers do not support the REFER method. This needs to be configured in order to force the CME to hairpin the call:

Router(config)#voice service voip

Router(conf-voi-serv)#no supplementary-service sip refer

Figure 3 shows the behavior of the CME system with the REFER method disabled.

Figure 3 – Transfer with REFER disabled

If REFER is supported on the SIP proxy, the user portion of the Refer-To and Referred-By must be translated to a DID that the SP proxy understands. The host portion of the Refer-To and Referred-By fields must be an IP address or DNS that the SP proxy can route to as well (this occurs by default on CME 4.1).

Call Hold

If an SCCP phone places a call from PSTN on HOLD, the CME locally changes the media. No SIP messages are sent across on the SIP trunk. Music on Hold will be played to the user across the SIP trunk based on the CME configuration.

Configure

In this section, you are presented with the information to configure the features described in this document.

Note: Use the Command Lookup Tool ( registered customers only) to obtain more information on the commands used in this section.

Network Diagram

This document uses this network setup:

Configurations

These configuration elements provide an outline of the steps required to configure your CME with SIP trunks:

• Infrastructure Elements: Interfaces, TFTP and DHCP services, NTP, etc
• Telephony-service: Enables IOS “PBX” call control on the CME platform including elements of phone management
• Ephones an Ephones-dns: Define IP phones and their telephone numbers
• Dial Plan: Dial-peers, extensions, voice-translation rules
• IOS SIP Configuration: Enables SIP, phone registration with SIP proxy, call routing over trunks, etc
• Voicemail Support: Cisco Unity Express
• Switch Catalyst Configuration: IP address, Interfaces, etc

This is the complete configuration needed to deploy a CME system with SIP trunks:

!
AUSNML-3825-01#show run
Building configuration...

Current configuration : 8634 bytes
!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname AUSNML-3825-01
!
boot-start-marker
boot-end-marker
!
enable secret 5 $1$vBU1$MCMG1rXM5ejME8Wap6W0H1
!
no aaa new-model
clock timezone central -8
clock summer-time central recurring
ip cef
!

!--- DHCP Configuration ---

ip dhcp pool Voice
   network 172.22.100.0 255.255.255.0
   option 150 ip 172.22.1.107
   default-router 172.22.100.1
!
ip dhcp pool Data
   network 172.22.101.0 255.255.255.0
   option 150 ip 172.22.1.107
   default-router 172.22.101.1
!
!
ip domain name cisco.com
ip name-server 205.152.0.20
multilink bundle-name authenticated
!
voice-card 0
 no dspfarm
!
!
!
!

!--- Voice Class and Service VoIP Configuration ---

voice service voip
 allow-connections sip to sip
 no supplementary-service sip moved-temporarily

!---Disable 302 sending

 no supplementary-service sip refer

!---Disable REFER sending

 sip
  registrar server expires max 3600 min 3600
  localhost dns:domain.test.com
!
!
voice class codec 1
 codec preference 1 g711ulaw
!
!
!
!
!
!
!
!
!
!
!

!--- Voice Translation Rules ---

voice translation-rule 1
 rule 1 /5123781291/ /601/

!--- An inbound rule for AA pilot "601

 rule 2 /5123781290/ /600/

!--- An inbound rule for the voicemail pilot "600"

!
voice translation-rule 2
 rule 1 /^911$/ /911/

!--- An outbound rule to allow "911"

 rule 2 /^9(.*)/ /\1/

!--- An outbound rule to strip "9" from PSTN calls

!
voice translation-rule 3
 rule 1 /^.*/ /5123781291/

!--- An outbound rule to change calling-number CLID to a
!--- "main" number

!
voice translation-rule 4
 rule 1 /^9(.......)$/ /512\1/

!--- An outbound rule to add areacode for local calls

 rule 2 /600/ /5123788000/

!--- An outbound rule to present the voicemail pilot extension as DID

 rule 3 /601/ /5123788001/

!--- An outbound rule to present the AA pilot extension as DID

 rule 4 /^2(..)$/ /51237812\1/

!--- An outbound rule to support transfers and call-forwards

 rule 5 /^9(.*)/ /\1/

!--- An outbound rule to strip "9" from "9+" transfers and call-forwards

!
!
voice translation-profile CUE_Voicemail/AutoAttendant

!--- Applied to the inbound dial-peers for CUE

 translate called 1
!
voice translation-profile PSTN_CallForwarding

!--- Applied to CUE dial-peers

 translate redirect-target 4
 translate redirect-called 4
!
voice translation-profile PSTN_Outgoing

!--- Applied to all outbound dial-peers

 translate calling 3
 translate called 2
 translate redirect-target 4
 translate redirect-called 4
!
!
!
!
!
!
!
vlan internal allocation policy ascending
!
!
!
!

!--- Internet Connection Configuration ---

interface GigabitEthernet0/0
 no ip address
 duplex auto
 speed auto
 media-type rj45
 no keepalive
!
interface GigabitEthernet0/0.1
 encapsulation dot1Q 1 native
 ip address 172.22.1.71 255.255.255.0
!
interface GigabitEthernet0/0.20
 encapsulation dot1Q 20
 ip address 172.22.101.1 255.255.255.0
!
interface GigabitEthernet0/0.100
 encapsulation dot1Q 100
 ip address 172.22.100.1 255.255.255.0
!
interface GigabitEthernet0/1
 no ip address
 shutdown
 duplex auto
 speed auto
 media-type rj45
 no keepalive
!
interface Service-Engine1/0
 ip unnumbered GigabitEthernet0/0.1
 service-module ip address 172.22.1.253 255.255.255.0
 service-module ip default-gateway 172.22.1.71
!
ip route 0.0.0.0 0.0.0.0 172.22.1.1
ip route 172.22.1.253 255.255.255.255 Service-Engine1/0
!
!
ip http server
no ip http secure-server
!
!
!

!--- TFTP Server Configuration  ---

tftp-server flash:P0030702T023.bin
tftp-server flash:P0030702T023.loads
tftp-server flash:P0030702T023.sb2
tftp-server flash:P0030702T023.sbn
!
control-plane
!
!
!
!
!
!
!

!--- SIP Trunk Configuration ---

dial-peer voice 1 voip
 description **Incoming Call from SIP Trunk**
 translation-profile incoming CUE_Voicemail/AutoAttendant
 voice-class codec 1
 voice-class sip dtmf-relay force rtp-nte
 session protocol sipv2
 session target sip-server
 incoming called-number .%
 dtmf-relay rtp-nte
 no vad
!
!
!
dial-peer voice 2 voip
 description **Outgoing Call to SIP Trunk**
 translation-profile outgoing PSTN_Outgoing
 destination-pattern 9........
 voice-class codec 1
 voice-class sip dtmf-relay force rtp-nte
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 no vad
!
!
!
dial-peer voice 3 voip
 description **Outgoing Call to SIP Trunk**
 translation-profile outgoing PSTN_Outgoing
 destination-pattern 9[2-9]..[2-9]......
 voice-class codec 1
 voice-class sip dtmf-relay force rtp-nte
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 no vad
!
!
!
dial-peer voice 4 voip
 description **Outgoing Call to SIP Trunk**
 translation-profile outgoing PSTN_Outgoing
 destination-pattern 9[0-1][2-9]..[2-9]......
 voice-class codec 1
 voice-class sip dtmf-relay force rtp-nte
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 no vad
!
!
!
dial-peer voice 5 voip
 description **911 Outgoing Call to SIP Trunk**
 translation-profile outgoing PSTN_Outgoing
 destination-pattern 911
 voice-class codec 1
 voice-class sip dtmf-relay force rtp-nte
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 no vad
!
!
!
dial-peer voice 6 voip
 description **Emergency Outgoing Call to SIP Trunk**
 translation-profile outgoing PSTN_Outgoing
 destination-pattern 9911
 voice-class codec 1
 voice-class sip dtmf-relay force rtp-nte
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 no vad
!
!
!
dial-peer voice 7 voip
 description **911/411 Outgoing Call to SIP Trunk**
 translation-profile outgoing PSTN_Outgoing
 destination-pattern 9[2-9]11
 voice-class codec 1
 voice-class sip dtmf-relay force rtp-nte
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 no vad
!
!
!
dial-peer voice 8 voip
 description **International Outgoing Call to SIP Trunk**
 translation-profile outgoing PSTN_Outgoing
 destination-pattern 9011T
 voice-class codec 1
 voice-class sip dtmf-relay force rtp-nte
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 no vad
!
!
!
dial-peer voice 9 voip
 description **Star Code to SIP Trunk**
 destination-pattern *..
 voice-class codec 1
 voice-class sip dtmf-relay force rtp-nte
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 no vad
!
!
!

!--- Voicemail Configuration ---

dial-peer voice 10 voip
 description **CUE Voicemail**
 translation-profile outgoing PSTN_CallForwarding
 destination-pattern 600
 b2bua

!--- Used by CME to send its IP address to SP proxy instead of CUE

 session protocol sipv2
 session target ipv4:172.22.1.155
 dtmf-relay sip-notify

!--- This can also be RFC2833 going to CUE

 codec g711ulaw

!--- CUE only supports G711ulaw as the codec

 no vad

!--- With VAD enabled, messages left on CUE could be blank or poor quality

!
!
!
dial-peer voice 11 voip
 description **CUE Auto Attendant**
 translation-profile outgoing PSTN_CallForwarding
 destination-pattern 601
 b2bua
 session protocol sipv2
 session target ipv4:172.22.1.155
 dtmf-relay sip-notify
 codec g711ulaw
 no vad
!
!

!--- SIP UA Configuration ---

sip-ua
 authentication username 5123781000 password 075A701E1D5E415447425B
 no remote-party-id
 retry invite 2
 retry register 10
 retry options 0
 timers connect 100
 registrar dns:domain.test.com expires 3600
 sip-server dns:domain.test.com
  host-registrar
!
!

!--- CME Telephony Service Configuration ---

telephony-service
 no auto-reg-ephone
 load 7960-7940 P0030702T023
 max-ephones 168
 max-dn 500
 ip source-address 172.22.1.107 port 2000
 calling-number initiator

!--- Preserves the caller-id of a call when transferred or forwarded

 dialplan-pattern 1 51237812.. extension-length 3 extension-pattern 2.. no-reg
 voicemail 600
 max-conferences 12 gain -6
 call-forward pattern .T
 call-forward system redirecting-expanded

!--- Enables translation rule features for call-forwarding

 moh music-on-hold.au
 transfer-system full-consult dss
 transfer-pattern 9.T
 secondary-dialtone 9
 create cnf-files version-stamp Jan 01 2002 00:00:00
!
!

!--- Ephone and Ephone-dn Configuration ---

ephone-dn  11  dual-line
 number 201 secondary 5123781201 no-reg both

!---"no-reg both" means do not try to register either extension with SP SIP Proxy

 name John Smith
 call-forward busy 600
 call-forward noan 600 timeout 15
!
!
ephone-dn  12  dual-line
 number 202 secondary 5123781202 no-reg both
 name Enrique Zurita
 call-forward busy 600
 call-forward noan 600 timeout 15
!
!
ephone-dn  13
 number 5123788000
 description **DID Number for Voicemail**
!
!
ephone-dn  14
 number 5123788001
 description **DID Number for Auto Attendant*
!
!
ephone-dn  15
 number 8000... no-reg primary
 mwi on
!
!
ephone-dn  16
 number 8001... no-reg primary
 mwi off
!
!
ephone  1
 mac-address 0008.A371.28E9
 type 7960
 button  1:11
!
!
!
ephone  2
 mac-address 0008.A346.5C7F
 type 7960
 button  1:12
!
!
!
!
line con 0
 stopbits 1
line aux 0
 stopbits 1
line 66
 no activation-character
 no exec
 transport preferred none
 transport input all
 transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
line vty 0 4
 password ut69coe
 login
!
scheduler allocate 20000 1000
ntp server 172.22.1.107
!
end

se-172-22-1-253#show run

Generating configuration:

clock timezone America/Chicago

hostname se-172-22-1-253

ip domain-name localdomain

groupname Administrators create
groupname Broadcasters create


!--- Users ---

username Enrique create
username John create
username Enrique phonenumberE164 "5123781202"
username John phonenumberE164 "5123781201"
username Enrique phonenumber "202"
username John phonenumber "201"


!--- AutoAttendant ---

ccn application autoattendant
 description "**AutoAttendant**"
 enabled
 maxsessions 4
 script "aa.aef"
 parameter "busOpenPrompt" "AABusinessOpen.wav"
 parameter "operExtn" "601"
 parameter "welcomePrompt" "AAWelcome.wav"
 parameter "disconnectAfterMenu" "false"
 parameter "busClosedPrompt" "AABusinessClosed.wav"
 parameter "allowExternalTransfers" "false"
 parameter "holidayPrompt" "AAHolidayPrompt.wav"
 parameter "businessSchedule" "systemschedule"
 parameter "MaxRetry" "3"
 end application


!--- MWI ---

ccn application ciscomwiapplication
 description "ciscomwiapplication"
 enabled
 maxsessions 8
 script "setmwi.aef"
 parameter "CallControlGroupID" "0"
 parameter "strMWI_OFF_DN" "8001"
 parameter "strMWI_ON_DN" "8000"
 end application


!--- Voicemail ---

ccn application voicemail
 description "**Voicemail**"
 enabled
 maxsessions 4
 script "voicebrowser.aef"
 parameter "uri" "http://localhost/voicemail/vxmlscripts/login.vxml"
 parameter "logoutUri" "http://localhost/voicemail/vxmlscripts/mbxLogout.jsp"
 end application


!--- SIP ---

ccn subsystem sip
 gateway address "172.22.100.1"

!--- Must match the "ip source-address" in telephony-service

 dtmf-relay sip-notify
 mwi sip outcall

!--- Subscribe / Notify and Unsolicited Notify have not been tested

 transfer-mode blind bye-also

!--- Testing with REFER method on CUE has caused certain call flows to break

 end subsystem


!--- Trigger Phones ---

ccn trigger sip phonenumber 600
 application "voicemail"
 enabled
 maxsessions 4
 end trigger

ccn trigger sip phonenumber 601
 application "autoattendant"
 enabled
 maxsessions 4
 end trigger

service phone-authentication
 end phone-authentication

service voiceview
 enable
 end voiceview


!--- Voicemail Mailboxes ---

voicemail default mailboxsize 21120
voicemail broadcast recording time 300

voicemail mailbox owner "Enrique" size 300
 description "**Enrique_Mailbox**"
 expiration time 10
 messagesize 120
 end mailbox

voicemail mailbox owner "John" size 300
 description "**John'sMailbox**"
 expiration time 10
 messagesize 120
 end mailbox

end

!--- Interface Connected to CME/CUE Router ---

interface FastEthernet0/2
 description Trunk to 3825
 switchport trunk encapsulation dot1q
 switchport mode trunk
 no ip address
 duplex full
 speed 100


!--- Interfaces Connected to the IP Phones ---

interface FastEthernet0/7
 switchport trunk encapsulation dot1q
 switchport trunk native vlan 20

!--- Data Traffic ---

 switchport mode trunk
 switchport voice vlan 100

!--- Voice Traffic ---

 no ip address
 spanning-tree portfast

interface FastEthernet0/8
 switchport trunk encapsulation dot1q
 switchport trunk native vlan 20
 switchport mode trunk
 switchport voice vlan 100
 no ip address
 spanning-tree portfast


!--- IP Address ---

interface Vlan1
 ip address 172.22.1.194 255.255.255.0
!
ip classless
ip route 0.0.0.0 0.0.0.0 172.22.1.1
ip http server

Verify

There is currently no verification procedure available for this configuration.

Troubleshoot

This section provides information you can use to troubleshoot your configuration.

The Output Interpreter Tool ( registered customers only) (OIT) supports certain show commands. Use the OIT to view an analysis of show command output.

Note: Refer to Important Information on Debug Commands before you use debug commands.

Troubleshooting Registration

Troubleshooting the SIP trunk on CME involves the same commands you use for IOS SIP GW troubleshooting and CME troubleshooting. Use these commands in order to check if your DN is registered:

• show sip-ua register status—Use this command to display the status of E.164 numbers that a SIP gateway has registered with an external primary SIP registrar.
• debug ccsip message—Enables all SIP SPI message tracing, such as those that are exchanged between the SIP user-agent client (UAC) and the access server.

Troubleshooting Call Setup

Commands for troubleshooting calls over SIP trunks are essentially the same as you use for regular SIP GW and CME troubleshooting.

Show commands:

• show ephone registered—Verifies ephone registration.
• show voip rtp connection—Displays information about RTP named-event packets, such as caller-ID number, IP address, and ports for both the local and remote endpoints.
• show sip-ua call—Displays active UAC and user agent server (UAS) information on SIP calls.
• show call active voice brief—Displays active call information for voice calls or fax transmissions in progress.

Debug commands:

• debug ccsip message—Enables all SIP SPI message tracing, such as those that are exchanged between the SIP UAC and the access server.
• debug voip ccapi inout—Traces the execution path through the call control API.
• debug voice translation—Checks the functionality of a translation rule.
• debug ephone detail mac-address <mac of phone> —Sets detail debugging for the Cisco IP phone.
• debug voip rtp session named-events—Enables debugging for Real-Time Transport Protocol (RTP) named events packets.
• debug sccp message—Displays the sequence of the SCCP messages.

Push Bhatkoti

CCIE voice#21569

One of the our customer was upgrading his CCM from 5 to  6 but wanted a fresh install of CCM6.x music on hold server (MOH). He installed CCM6 but didn’t have any clue of how to download the MOH file from the older server and put in the new server.

Therefore, he called the EXPERT [of courese me] for help. I never did this in the past so, I had to dig through google and fouund the solution for him and that really worked. I thot this will be helpful to the folks who are looking for downloading a moh file from ccm5/6/7 servers.

Its as easy as drinking a beer in a BAR.

Step#1: SSH to CCM 5/6/7x publisher

Unfortunately, there is no way we can pull moh file via GUI interface. The only way I know is via CLI.

You can use putty.exe if you are a windows  fane or I use ubuntu so i just open a terminal windows and simply use ’ssh root@ccmip’ to log into call manager publisher. After ssh you will get a prompt similar below:

admin:

Step#2  Setup a FTP server to put  file from moh server to a ftp server.

Setup a SFTP server. I use SSHD as a SFTP server. For windows you can download filezilla sftp server from below url:http://filezilla-project.org/

Create a username and password in sftp. We will use this username and password in step #$

Step#3  List MOH related filesin the call manager

Use a command called “file list activelog mohprep/*” to list all moh files in the mohprep directory.
You will see output similar to below:
admin:file list activelog mohprep/* ← note * means all files
SampleAudioSource.alaw.wav

SampleAudioSource.g729.wav

SampleAudioSource.ulaw.wav

SampleAudioSource.wb.wav

SampleAudioSource.xml

CiscoMOHSourceReport.xml

Step#4 Download the moh files to a SFTP server

Use a command called “file get activelog mohprep/filename”to download above listed file(s).

example:

admin:file get activelog mohprep/ SampleAudioSource.alaw.wav
Enter SFTP server IP: 4.2.2.2 ← SFTP server IP
Enter SFTP username: test
Enter SFTP password: test

Thats all, your file is transferred to the FTP Server and you can use GUI interface to put the file to a new/other call manager publisher.

————————snippet————–

admin:file list activelog mohprep/*
CiscoMOHSourceReport.xml                SampleAudioSource.alaw.wav
SampleAudioSource.g729.wav              SampleAudioSource.ulaw.wav
SampleAudioSource.wb.wav                SampleAudioSource.xml
dir count = 0, file count = 6
admin:
admin:
admin:file list activelog mohprep/*
CiscoMOHSourceReport.xml                SampleAudioSource.alaw.wav
SampleAudioSource.g729.wav              SampleAudioSource.ulaw.wav
SampleAudioSource.wb.wav                SampleAudioSource.xml
dir count = 0, file count = 6
admin:
admin:file get
file get activelog
file get inactivelog
file get install
file get tftp

admin:file get active
admin:file get activelog ?
Syntax:
file get activelog file-spec [options]
file-spec   mandatory   file to transfer
options     optional    reltime months|weeks|days|hours|minutes timevalue
abstime hh:mm:MM/DD/YY hh:mm:MM/DD/YY
match regex
recurs

admin:file get activelog mohprep/SampleAudioSource.xml
Please wait while the system is gathering files info …done.
Sub-directories were not traversed.
Number of files affected: 1
Total size in Bytes: 606
Total size in Kbytes: 0.5917969
Would you like to proceed [y/n]? y
SFTP server IP: 172.16.16.111
SFTP server port [22]:
User ID: push
Password: ********
Download directory: /home/push

This is the script when you add moh file and that gets replicated to all nodes:

[root@ccm5 bin]# more moh_do_backup.sh
#!/bin/bash
#
# moh_do_backup.sh script will make a tar ball for backup targets
#
# SYNOPSIS
# moh_do_backup.sh <log path> <status file path> <target node>
#

#
# Checking parameters
#
if [ $# -lt 3 ]
then
echo “Usage: $0 <log path> <status file path> <target node>”
exit 1
fi

LOGFILE=$1
STATUS_FILE=$2
NODE=$3

echo “0″  > $STATUS_FILE

echo “Starting MOH backup” >> ${LOGFILE}

sudo -u root /bin/tar cvfpP – /usr/local/cm/sftp/mohprep/* 2>>${LOGFILE} | sudo
-u drfuser ssh drfuser@${NODE}

RESULT=$?

if [ $RESULT -gt 0 ]
then
echo “MOH: Backup failed ($RESULT)”  >> ${LOGFILE}
exit 106
fi

#
# update status file to 100%
#
echo “100″ > $STATUS_FILE

echo “MOH: Finished backup” >> ${LOGFILE}
exit 0

[root@ccm5 bin]#
[root@ccm5 bin]#
—————————–end of snippetts——————-

Push

CCIE voice#21569
blog: pushkarbhatkoti.wordpress.com

]]> http://pushkarbhatkoti.wordpress.com/2009/01/08/how-to-download-music-on-hold-moh-file-from-ccm5x-6x-7x-server/feed/ 8 pushkarbhatkoti