For anyone working in Cisco IOS world knows the pain when an IOS gets deleted from flash by mistake or the IOS image on the flash gets corrupt. The device keeps on booting to a state known as ROMMON. The usual XMODEM procedure is painfully long. The IOS image copy process can take more than 9 hours - if the file is around 25 MB. 

To put the IOS back into the switch using USB can be pretty quick. I have done this in under an hour. Here is how to do it.

I am using Catalyst 2960X in my example below. 

​First make sure your IOS device has a USB port. If it does, we need a format a USB flash drive in a way IOS recognizes it. The file format has to be FAT16 (or FAT).
Note: FAT32 and FAT-EX are not supported by the Cisco IOS devices.

The USB drive has to be smaller or equal to 2 GB. If you don't have a USB drive smaller than 2 GB, follow this.

Here are the options I checked on when I formatted my USB drive to FAT on my Windows PC. Don't do Quick Format.

After formatting, put the IOS image on the USB drive. Now power on the switch or router. It would come to the ROMMON state.

Insert the USB flash drive in the USB port. Make sure that switch is able to see the IOS image on the USB. Issue the command dir usbflash0:

​Now enter boot usbflash0:c2960x-universalk9-mz.152-7.E2.bin so the the switch boots using the image on the USB.

For some reason, my switch came back to ROMMON after this above step as well - which it shouldn't have. So I had to repeat the above step once again. After the next time it came up fine.

Now the flash doesn't have a valid IOS image. We need to copy the image on the switch's flash.
I copied it by making the laptop as the FTP server. I use Filezilla. I gave my PC IP address of 10.10.10.2/24

On Switch

en
conf t
int vlan 1
ip address 10.10.10.1 255.255.255.0
no shut



ip ftp username cisco
ip ftp password cisco
exit

On PC

Control Panel >> Network and Internet >> Network Connections >> Ethernet >> Properties >> Internet Protocol  Version 4  (TCP/IPv4)

Make sure you are able to ping each other (from switch ping 10.10.10.2 and from laptop ping 10.10.10.1)

Once both PC and switch have IP connectivity, issue the command:

copy ftp: flash:

Provide the IP Address of FTP server (10.10.10.2 - in my example) and file name (c2960x-universalk9-mz.152-7.E2.bin - in my example)

Make sure you get the exclamations (!), and not the dots (.). If you get dots, that means either there isn't IP connectivity or (T)FTP server is not properly setup.

Issue dir to make sure that you see the IOS bin file in the switch's flash.

Make sure the config register is the right value before reloading by issuing sh ver

Reload the switch by reload command. Enter no to saving the config - this is so that on the boot, you get a fresh install of IOS.

Hopefully this will save you time just like it did to me and you can avoid xmodem IOS transfer.

Upgrade for VMWare vSphere ESXi from any 5.0 to 5.5 is pretty straight forward. However when going from 5.5 to 6.x, there are some added components like PSC (Platform Services Controller) See here.

There are various methods, however I will outline the steps by mounting a virtual DVD ISO (I used Cisco UCS Server - so my Virtual KVM was CIMC).

To mount the ISO and restarting the server and booting through the DVD, follow my other blog post (You would mount a different DVD image which would have the upgrade vSphere ISO).

Note: Check VMWare Compatibility to check if you can go from a specific source to a destination.

Here are the steps:

Once the DVD starts booting, the following screen would appear:

.....

Press Enter when its fully loaded.

Click the partition where the ESXi resides (normally its the smaller partition - the bigger partition has the virtual machines)

Note: Be careful in this step

Select what you intend on doing. I selected the Upgrade with preserving VMFS (which is the default)

Confirm

...

Once complete, the server would go through a power cycle. 

Once the server power cycles, it would go to the VMWare yellow screen. Turn on the VMs (using vSphere).

Note: vSphere on your local desktop will have to go through an upgrade once the ESXi is upgraded.

Cisco Integrated Management Controller is Cisco proprietary out-of-band management for servers. This is HP's iLO equivalent on Cisco UCS (Unified Computing System) servers.

Note: UCS are servers which are capable of running virtual machines (VMWare ESXi or other hypervisors).

Recently, I did an upgrade on the UCS server's CIMC (which include all the components mentioned in the title). I will go through the steps here. You can also upgrade individual components if needed. My server was running ESXi but that is irrelevant.

Note: Go through the Release Notes of the firmware at Cisco.com and make sure you can go from the source version to the destination version. You have to be logged in to be able to download the software.

​

Here are the steps:

​Gracefully shut down all the VMs running on the UCS server (this isn’t required but recommended) 
Open the CIMC interface by going into the CIMC IP Address (ignore any certificate errors) 

Note: This is not the same as ESXi IP. 

Note: I prefer using Internet Explorer than using Chrome for this upgrade (even though my choice of browser is Chrome). 

If the hypervisor you are running is ESXi, you will see a similar screen to the one below: ​

Press F6 on the boot screen to get in the Boot Menu. (you might have to press F6 multiple times just to make sure you hit it at the right moment) ​

Select vKVM mapped CD/DVD ​​

UCS Host Upgrade Utility will start ​​

It can a while to load the firmware files (I guess this depends on network connectivity - mine was pretty good but still it took around 10 minutes).

When you get the license agreement, click 'I Agree'.

Now the following screen is where you need to select the components you are planning to upgrade. Select the components (Now click on 'Update' or 'Update All' (I selected All). At that time the upgrade process on each component would start. 

You would get the option to select Secure Boot Option. I selected 'No' as I wasn't using this feature.

You can see the progress throughout the process (almost)

Depending on the source and target version, you might lose connection to this window while some components are being updated. Don't panic. Give it time.

​Note: Some versions where you won't lose access to the KVM Console, you would lose connection when you exit the installer when the firmware is finally applied (Adapter or BIOS firmware).

Server would go through a power cycle and would load up the hypervisor. You can go the CIMC webpage to make sure that version is actually upgraded.
Hope this helps!

Macs and PCs keyboards are different. Recently, I had purchased a Mac and was confused with some keys on how to use them on Mac. Here are some of the equivalents. I will keep on updating as I come across more keys.

Page Down = 'fn' + 'Down Arrow'
​Page Up = 'fn' + 'Up Arrow'
Delete = 'fn' + 'delete'
Backspace = 'delete'
F1, F2, F3, .... = 'fn' + top row buttons OR
F1, F2, F3, .... = 'fn' + keys on control strip (Mac Book Pros with control strip)

Other cool shortcuts

To make screenshot of the whole screen: 'shift' + 'command' + '3'
To make screenshot of a portion of the screen: 'shift' + 'command' + '4' (crosshair would appear, press and drag the mouse over the potion you make to make a screenshot)
To make screenshot of an active window: 'shift' + 'command' + '4', then press spacebar, click on any window

All above would result in a PNG (Portable Network Graphics) file on the desktop.

Even though SIP is quickly becoming the industry standard (if not already), there are issues with SIP which because the way each vendor implements it in their products. What I have seen is mostly DTMF causes most pain for the engineers.
Here are some of the ways DTMF is used in SIP environments.

IN-BAND
In In-Band, the DTMF information is carried inside the actual voice traffic. This is defined under RFC 2833.

OUT-OF-BAND
In Out-of-Band, DTMF information is carried outside the voice traffic.

CUCM Configuration
Here are the CUCM Trunk configuration parameter that can be selected:
​
DTMF Signaling Method: Choose from the following options:
No Preference (default)—Cisco Unified Communications Manager will pick the DTMF method to negotiate DTMF, so the call does not require an MTP. If Cisco Unified Communications Manager has no choice but to allocate an MTP (if the Media Termination Point Required check box is checked), SIP trunk will negotiate DTMF to RFC2833.
RFC 2833—Choose this configuration if the preferred DTMF method to be used across the trunk is RFC2833. Cisco Unified Communications Manager makes every effort to negotiate RFC2833, regardless of MTP usage. Out of band provides the fallback method if the peer endpoint supports it.
OOB and RFC 2833—Choose this configuration if both out of band and RFC2833 should be used for DTMF.

​Media Termination Point Required:
You can configure Cisco Unified Communications Manager SIP trunks to always use an MTP. Check this check box to provide media channel information in the outgoing INVITE request. When this check box is checked, all media channels must terminate and reoriginate on the MTP device. If you uncheck the check box, the Cisco Unified Communications Manager can decide whether calls are to go through the MTP device or be connected directly between the endpoints.

Note : 
If check box remains unchecked (default case), Cisco Unified Communications Manager will attempt to dynamically allocate an MTP if the DTMF methods for the call legs are not compatible.

For example, existing phones that run SCCP support only out-of-band DTMF, and existing phones that run SIP support RFC2833. Because the DTMF methods are not identical, the Cisco Unified Communications Manager dynamically allocates an MTP. If, however, a new phone that runs SCCP, which supports RFC2833 and out-of-band, calls an existing phone that runs SIP, Cisco Unified Communications Manager does not allocate an MTP because both phones support RFC2833. So, by having the same type of DTMF method supported on each phone, no need exists for MTP.

Instead of selecting Media Termination Point Required on the above option, a better option is to Insert MTP whenever is required. That is done through the SIP Trunk Profile.

​Early Offer support for voice and video calls:
This field configures Early Offer support for voice and video calls. When enabled, Early Offer support includes a session description in the initial INVITE for outbound calls. Early Offer configuration settings on SIP profile apply only to SIP trunk calls. These configuration settings do not affect SIP line side calls. If this profile is shared between a trunk and a line, only a SIP trunk that uses the profile is affected by these settings.

The Media Transfer Point (MTP) Required check box on the Trunk Configuration window, if enabled, overrides the early offer configuration on the associated SIP profile. Cisco Unified Communications Manager sends the MTP IP address and port with a single codec in the SDP in the initial INVITE.

From the drop-down list box, select one of the following three options:

Disabled (Default value) - Disables Early Offer; no SDP will be included in the initial INVITE for outbound calls.
Best Effort (no MTP inserted)

1. Provide Early Offer for the outbound call only when caller side's media port, IP and codec information is available.
2. Provide Delayed Offer for the outbound call when caller side's media port, IP and codec information is not available. No MTP is inserted to provide Early Offer in this case.

Mandatory(insert MTP if needed) - Provide Early Offer for all outbound calls and insert MTP when caller side's media port, IP and codec information is not available.

CUBE (SBC) Configuration

On the CUBE dial-peers, DTMF has to be selected.
You can actually select RFC2833 as well as a secondary DTMF if RFC2833 (in-band) is not supported.
​

​For complete dial-peer config, please refer to this blog post.

Some text in the above blog was taken was Cisco Unified Communications Manager Help Pages.

Happy new year to everyone. It's been a while since I wrote a new blog entry - it's new year, so here is the brand new blog entry for 2018.
There are couple of UCCX Script editing steps which can be used when creating menus on UCCX Scripts - they can be used interchangeably. Two of these steps are 'Switch' and 'Menu'

Switch Step

​Switch step is found under the General category on UCCX Script editor.

Switch step is like 'if, then, else' in software programming.

It would be easier to explain using an example:
Lets assume we need to create an auto attendant script where the call in user is asked to enter:
1 for Sales (redirects to extension 2001)
2 for Marketing (redirects to extension 2002)
0 for Operator (redirects to extension 2003)
or press the extension of the person you need to dial.

To start off (after the other basic steps of accepting the call, etc) I would use a Play Prompt step (under Media) to play the recording : "Press 1 for sales, press 2 for marketing, 0 to speak to an operator, or if you know the extension of the person, just dial that extension"

Once the recording has been played, the user would enter the digits on the phone. I have to capture those digits. I would use a string to capture the the digits entered by the user.
​
​I would create a parameter on script editor, where I would store the value which was entered by the user (1, 2, 3, extension). I have named this Parameter s_DigitString (s is the name distinguishes its a string, DigitString in the name distinguishes the purpose). The default value is empty.

To capture the digits, we need to create a 'Get Digit String' step to capture the digits in the s_DigitString. Get Digit String is available under the 'Media' steps on UCCX editor.

Once the digits have been captured, now comes the Switch step.

Switch step will look at the digits that are entered in the s_DigitsString and will perform the action based on the captured digit. Here are the properties of the Switch step.

Under Press 1, Press 2, Press 0, you can define where to redirect the call to (using the Call Redirect step). 'Dial by Extension' would rely on 'Default' on the above right picture (overlapping extension can be dealt with by modifying  timers on the Get Digit String, so dialing extension 2112 won't be a problem).

This is in nutshell is how a Switch step works. Now lets move to Menu step.

Menu Step

Menu step is found under Media category on the UCCX Editor.

​Menu step combines the functionality of these three steps in one:

1. Play Prompt
2. Get Digit String
3. Switch

Here are the properties. 
The second tab, Prompt on the Menu properties is where you define the Prompt that will be played.

The third tab, Filter on the Menu properties is where you define the key inputs and the action that will be taken.

Filter in Menu creates ​the same options as Cases in Switch. However, one big difference is that it allows only one digit entry. Moreover, in Menu, we don't save the user input in a variable (s_DigitString above). Hence, only one digit can be entered. This becomes an issue when the users is trying to dial an extension of a user.

To close off, I would invariably use Menu always unless I have to register more than one digit. Menu makes things lot more easier, but then the problem with Menu is that you cannot enter an extension 2112 for example. It would just register that as Option 2 with Menu.

Hope this helps.

Two commands: 'e164-pattern-map' (added from IOS 15.2) and 'server-group' (added from IOS 15.4) makes Voice Gateway/CUBE config dial-peers config very clean.

With e164-pattern-map, the digits to be matched can be groped together. So no need for multiple dial-peers (with just the changing destination-pattern). 

With server-group, all the session target (session target ipv4:x.x.x.x) can be grouped together. So no need for multiple dial-peers with just the changing session target ipv4: is required. 

Here is an example which will make things more clear:


voice class e164-pattern-map 10
​ e164 911
 e164 9[2-9]..[2-9]......
 e164 91[2-9]..[2-9]......
 e164 9[^1]T
 description OUTBOUND_CALLS

voice class e164-pattern-map 20
 e164 +1416555[12]...
 description INBOUND_CALLS

voice class server-group 10
 ipv4 140.1.4.5 preference 1
 ipv4 140.1.5.5 preference 2
 description to PSTN_SIP_PROVIDER
!
voice class server-group 20
 ipv4 10.1.100.1 preference 1
 ipv4 10.1.101.1 preference 2 
 description to CUCM

dial-peer voice 10
 description OUTBOUND_PSTN_CALLS
 destination e164-pattern-map 10
 translate-called PSTN_FORMAT
 session protocol sipv2
 dtmf-relay rtp-nte
 session server-group 10

dial-peer voice 11
 description INBOUND_FROM_PSTN
 incoming called e164-pattern-map 20
 session protocol sipv2
 dtmf-relay rtp-nte
 session server-group 10
 dial-peer voice 20

dial-peer voice 20
 description TOWARDS_CUCM
 destination e164-pattern-map 20
 session protocol sipv2
 dtmf-relay rtp-nte
 session server-group 20

dial-peer voice 11
 description INBOUND_FROM_CUCM
 incoming called e164-pattern-map 10
 session protocol sipv2
 dtmf-relay rtp-nte
 session server-group 10
 dial-peer voice 20

voice translation-rule 10
 rule 1 /^911$/ /911/
 rule 2 /^9/ /+/
 
voice translation-profile PSTN_FORMAT
 translate called 10  

I have used a translation-profile to convert the dialed-numbers to +E164 numbers (except for 911 calls). This config is not required - it depends what format the PSTN is expecting the call. I have assumed that this telco here is expecing +E164 numbering format except for 911 calls.

​By the way, under the e164-pattern-map command, other than actual e164 numbers (like in example above), url could be used. which can point towards a file on the flash or ftp somewhere which can have the numbers.

With digit manipulation on IOS, keep a few things in mind:
1.  / -- /    - It always starts with / and ends with /
2. Anything within parenthesis would be kept.
 characters are keep are like (a\)
 characters to ignore are like b\
 \1 copy the first set into the replacement number, \2 copy the second set into the replacement number, etc 
 
3. The following represents the digits/characters:
 . : Any single digit
 0 to 9,*,# : Any specific character
 [0-9] : Any range or sequence of characters
 * : Modifier—match none or more occurrences
 + : Modifier—match one or more occurrences
 ? : Modifier—match none or one occurrence

Look at the following examples. This should clarify:

Examples:

Example 1:
voice translation-rule 1
 rule 1 /123/ /456/

Will match and modify 123 anywhere in the pattern:

router#test voice translation-rule 1 123
Matched with rule 1
Original number: 123    Translated number: 456

router#test voice translation-rule 1 1234
Matched with rule 1
Original number: 1234   Translated number: 4564

router#test voice translation-rule 1 6123
Matched with rule 1
Original number: 6123   Translated number: 6456
router#test voice translation-rule 1 6123123
Matched with rule 1
Original number: 6123123        Translated number: 6456123

Example 2:
voice translation-rule 1
 rule 1 /^123/ /456/  
 
Will match only if the pattern start with 123 because carrot '^' is the starting character

router#test voice translation-rule 1 123 
Matched with rule 1
Original number: 123    Translated number: 456

router#test voice translation-rule 1 1234
Matched with rule 1
Original number: 1234   Translated number: 4564

router#test voice translation-rule 1 6123
6123 Didn't match with any of rules 

Example 3: 
voice translation-rule 1
 rule 1 /^123$/ /456/

Will only match if the pattern starts if its 123 exacly. Nothing after, nothing before.
 
router#test voice translation-rule 1 123 
Matched with rule 1
Original number: 123    Translated number: 456

router#test voice translation-rule 1 1234
1234 Didn't match with any of rules

router#test voice translation-rule 1 6123
6123 Didn't match with any of rules


Example 4:

voice translation-rule 1
 rule 1 /^40.../ /6666000/
 
Will match pattern starting with 40 and any 3 digits afterwards

router#test voice translation-rule 1 40123
Matched with rule 1
Original number: 40123    Translated number: 6666000

Example 5:

voice translation-rule 2
 rule 1 /.*/ /5554000/
 
Will replace any pattern with 555400

router#test voice translation-rule 2 123
Matched with rule 1
Original number: 123    Translated number: 5554000

router#test voice translation-rule 2 86573
Matched with rule 1
Original number: 86573  Translated number: 5554000

router#test voice translation-rule 2 ""
Matched with rule 1
Original number:   Translated number: 5554000

Example 6:

voice translation-rule 1
 rule 1 /^\(12\)3\(45\)$/ /6\1\2/

Set 1: 12
Set 2: 45
Ignore: 3

router#test voice translation-rule 1 12345
Matched with rule 1
Original number: 12345        Translated number: 61245

Example 7:

voice translation-rule 10
 rule 1 /\(32..\)$/ /416555\1/
 
Will replace 32xx to 41655532xx (used for outbound caller ID)

router#test voice translation-rule 10 3200
Matched with rule 10
Original number: 3200  Translated number: 4165553200

Example 8:

voice translation-rule 10
 rule 1 /^416555\(32..\)$/ /\1/
 
Will replace 4165553211 to 3211 (used for inbound caller ID)

router#test voice translation-rule 10 4165553211
Matched with rule 10
Original number: 4165553211  Translated number: 3211

Example 9:

voice translation-rule 7
 rule 1 /^4/ /904/ type national national
 rule 2 /^4/ /9004/ type international international
 
If number starts with 4 and type is National, it would be prefixed 90 and type will remain National
If number starts with 4 and type is International, it would be prefixed 900 and type remain International
 
router#test voice translation-rule 7 493456567 type national
Matched with rule 1
Original number: 493456567      Translated number: 90493456567
Original number type: national  Translated number type: national
Original number plan: none      Translated number plan: none
    
router#test voice translation-rule 7 493456567 type international
Matched with rule 2
Original number: 493456567              Translated number: 900493456567
Original number type: international     Translated number type: international
Original number plan: none              Translated number plan: none
 
Example 10:

voice translation-rule 8 
 rule 1 /^2\(...$\)/ /01779345\1/ type unknown national plan unknown isdn
 
This rule matches any four-digit number that starts with "2". The rule removes the "2", adds the number "01779345" as a prefix, and sets the plan to "isdn" and the type to "national".
 
router#test voice translation-rule 8 2001 type unknown plan unknown 
Matched with rule 1
Original number: 2001   Translated number: 01779345001
Original number type: unknown   Translated number type: national
Original number plan: unknown   Translated number plan: isdn 
 

 
POTS Dial-Peers
---------------

dial-peer 911 pots
 destination-pattern 911$ !Anything that is explicitly matched on POTS dial-peer is removed. This isn't true for VOIP. VOIP Dial-peer doesn't remove anything thats explicity matched.
 no digit strip <OR> prefix 911 <OR> forward-digits 3
 port 0/0/0:23 
 
dial-peer 9 pots
 destination-pattern 9[2-9]XX[2-9]XXXXXX$
 port 0/0/0:23
 ! no need for any prefix or any forward digits as only number explicitly matched is 9 which we want removed as its our PSTN code.
 
dial-peer 91 pots
 destination-pattern  91[2-9]XX[2-9]XXXXXX$
 prefix 1 <OR> forward-digits 11 ! 1 is being explicitly matched and we want it to go out as its our long distance code used by the PSTN.
 port 0/0/0:23
 
dial-peer 9011 pots
 destination-pattern 9011T ! no need for #. It would be automatically matched because of the next dial-peer.
 prefix 011 ! can't use forward-digits as we don't know the extension length that would be dialed. 
 port 0/0/0:23
 
dial-peer terminator # ! Its a default command and any call with # is already a terminator for interdigit timeout, unless its changed by someone.

Note: On CUCM, in Service Parameters "Strip # Sign from Called Party Number" is "True" by Default - Meaning it would remove # as the trailing number.


dial-peer 1 pots
 incoming called-number . ! will match incoming calls
 
dial-peer 2 voip
 destination-pattern 2...$
 session target ipv4:10.1.1.2 ! CUCM IP
 dtmf-relay h245-signal
 
dial-peer 3 voip
 incoming called-number .


Overlap Sending <-- SCCP works this way - one digit at a time.
En-Bloc <-- CUCM works this way.

A good document which I consulted when working on this post:
​http://www.cisco.com/c/en/us/support/docs/voice/call-routing-dial-plans/61083-voice-transla-rules.html

1. ENABLING THE SERVICE

GO to Cisco Unified Serviceability, Tools --> Service Activation.
Make sure Cisco Extension Mobility service is enabled. If not enabled, enable it.

2. CREATING THE SERVICE

a. GO to CUCM Administration, Device --> Device Settings --> Phone Services.
b. Add new. Enter the name such as EM or Extension Mobilty
c. Enter the URL: URL: http://<CUCM IP>:8080/emapp/EMAppServlet?device=#DEVICENAME#
d. Choose XML Service and Standard IP Phone Service as the Service Category and Service Type respectively.
e. Click Save.

3. CONFIGURING PHONES

a. Go to Device --> Phone, open the phone where you want to enable the service.
b. Related links on top of the page, select Subscribe/Unsubscribe Services and click GO.
c. Select the service created on CREATING THE SERVICE above.
d. Subscribe and Save. 
e. Make sure 'Enable Extension Mobility' is checked on the Phone.

NOW PHONES HAVE SERVICE ENABLED, BUT USERS ARE STILL NOT ENABLED FOR EM. NOW LETS ENABLE USERS

4. CONFIGURING USER DEVICE PROFILE (UDP)

a. GO to Device --> Device Settings --> Device Profile
b. Add New
c. Select Phone type (according to the phone type where the user will log into)
d. Select Device Protocol <-- You will only be presented with the Protocl option if the Phone is question supports both SIP and SCCP.
e. Select a name: such as "User1" and fill in all required fields <-- Most of these options are similar to creating a phone - only difference is there is no MAC Address option here.
f. Related links on top of the page, select Subscribe/Unsubscribe Services and click GO.
g. Select the service created on CREATING THE SERVICE above.
h. Click on 'Line [1] - Add a new DN' and enter line information (just like you would do on an IP Phone)
i. Click Save.

5. CONFIGURING USERS

a. Go to User Management --> End Users
b. Open an existing user or create a new one.
c. Make sure userid, password, pin, and phone number are configured.
d. Also under 'Extension Mobility' select "User1" (or whichever user was created on CONFIGURING USER DEVICE PROFILE) and move it down to the controlled profiles section. 

JTAPI stands for Java Telephony API. It was developed by Java for Computer Telephony Applications or CTIs. It provides a set of packages which provides basic guidelines for placing, answering and dropping a call. JTAPI is similar to TAPI (Telephony API) which was developed by Microsoft and Intel.

AXL stands for Administrative XML. AXL is used for pushing config on another system.

When integrating another system with CUCM (for example UCCX), JTAPI is used for CTI (placing call, ending call), and AXL is used to push the config onto CUCM.