VOIP_Providers

VoIP Providers

If you wish to make calls over the internet, then you will need to sign up to a VoIP provider.

The largest list of VoIP providers in Australia (or VoIP providers that service Australia) is provided by Market Clarity. A smaller list is provided by VoIP Choice & Internet Choice which all list the plan prices and call rates.
Although these websites are updated regularly, but be sure to check with any provider that you intend to sign up to in case of price changes.

To get the best value, take a good look around before you decide (especially at the VoIP Forum) to see if there are any limited offers available.

It is also important to note that you can use multiple VoIP providers if your hardware supports it (eg SPA3000). If your hardware doesn't support it, you can use a service called Voxalot to do your call routing for free, so that calls to different destinations go out through different VoIP providers, achieving the best rates for each destination.

There are many reasons to use an Australian provider (easier to handle billing issues, better latency, and quite often quality!), but if you call (or receive calls from) international destinations, then dont just limit yourself to Australian providers. There are hundreds of providers out there around the world – with anything from free accounts, to paid accounts with indial numbers in the most unusual of countries. (You can get an Indial number in an international city of your choice, thus saving your friends or relatives money, as when they call that local Indial number, it comes straight through to your VoIP phone and only costs them a local call!)

Some popular Australian Providers have their own Wiki pages. Follow the links below.

- Ace Communications Group
- Agile Networks
- Antratel
- ATP
- Aussiewide
- BBPGlobal
- Broadband Solutions
- Brown Bear Internet
- CentrixPhone
- ClicknCallnow.com
- Comcen Internet Services
- Delacon
- Engin
- Exetel
- eVoIP
- Faktortel
- foneAnyWare
- Fonebox
- Freshtel
- FWD
- Globechat
- GoTalk
- iiNet
- Internode (NodePhone)
- iPrimus
- Maxo Telecommunications
- MondoTalk
- MyNetFone
- Mytel
- Nehos Communications
- NetFuse
- Nettalk
- NuFone
- OneNetwork
- Oztell/Net2Max
- Resonance Networks
- PennyTel
- Saco Technology
- Freecall (Siphone)
- SIPME
- Sipphone (Gizmo Project)
- SmartVoice
- Spin Internet Services
- TPG
- VoNEX – Hosted PBX
- Vocaltone
- VCall
- VoIP Australia
- Voipbuster (voipstunt & voipdiscount, The Betamax Group)
- VOISE
- VoXaLot
- WorldDialPoint
- Your Phone

Blank Form for adding extra providers if required

VoIP Service fault monitoring websites:

• OzVoIPStatus – VoIP Monitoring Website (Also has some customer opinions/comments)

The following are not providers but are worth mentioning:

- Australian 1300 VoIP Gateway (Dial into many VoIP providers from the PSTN with this number)
- Sipbroker (VoIP Peering system)

FREE International DID Providers
Here's a list of some Free international DID providers

How Voice over IP Works

How Voice over IP works is a mystery to some small business owners and employees. But if you understand how Voice over IP works, you can more easily use the technology to your company’s advantage.

How Voice over IP Works: At a Glance

• With VoIP, analog voice calls are converted into packets of data. The packets travel like any other type of data, such as e-mail, over the public Internet and/or any private Internet Protocol (IP) network. This is the essence of how Voice over IP works.
• Using a Voice over IP service, you can call landline or cell phones. You can also call computer-to-computer, with both parties speaking into a computer microphone and listening through computer speakers or headsets.
• When evaluating how Voice over IP works, it’s worth noting that you can make or receive calls using landline telephones. All you need is an analog telephone adapter connected to your network. Also, to ensure the best voice quality and security, consider using your VoIP or other communications system on a private IP network.

How Voice over IP Works for Your Business

How Voice over IP works for your business is simple: By adding voice to a data network, you’ll reduce costs, improve productivity, and enhance collaboration.

• You’ll save money by having one network to manage instead of two. You can easily add, move, or change phone extensions and locations, which saves money and gives you more flexibility.
• Your workforce can use your communications system from home or on the road. Also, wireless IP phones connect users to your communications system and data resources, such as customer information, while they’re in the warehouse, on the sales floor, or anywhere they can access your data network wirelessly.
• Unified communications solutions for small businesses go beyond basic VoIP capabilities in enhancing collaboration. With a unified communications solution, workers can easily collaborate through voice, video chat, Web conference, and instant messaging. Employees can collaborate using each technology individually or all of them simultaneously, and from a single, easy-to-use interface.

Find out about Cisco Unified Communications solutions tailored specifically for small businesses.

Contact a Cisco partner who can help you determine the best communications solution for your business.

Other articles you may be interested in:

• Voice Over IP FAQ
• Why VoIP?

How to remove a Temp License

There is a tool on the UCCX server called CET.bat which is a sort of utility tool that allows you to do some VERY complex and powerful things on the server. To access the tool you simply need to open up a command prompt, or browse to the c:\program files\wfavvid folder and launch the CET.bat file. See the screenshot below:

Now this might look a little strange, however, just type CET.bat from the command line and you will see the black DOS syntax go flying across the screen. Eventually you will be brought to the GUI based menu below, once you are into the GUI you need to select “com.cisco.crs.cluster.config.LicenseConfig. This brings up a list of CRS Licenses. You need to delete the temporary license.

To delete the temporary license simply highlight the row containing the license file by clicking on it with your mouse, then right clicking and selecting “Delete”, see below:

Once you have deleted the key, you need to restart the CRS Node Manager. This is often a place where most people mess up, because they think the Node Manager can be restarted via the Control Center in CRS’s Appadmin page. The best and easiest place to restart the Node Manager is straight from the Services.msc. *** NORMALLY, I would never recommend restarting any CRS services this way, but for this top level service this is the place to be:

Once you click on Restart it may take several seconds/minutes for all CRS services to restart. The best way to double check and make sure you are okay is to always refresh the screen here, but to also log into the Appadmin page and visit the Control Center to make certain RMCM and Unified CM Telephony at a minimum restarted successfully. You can also place test calls into the applications.

Hope that helps!

VMware ESXi setup of Vlans

open up the vmware infrastructure client
select the server
select configuration from the tabs in the header
select networking from the hardware list on the left
select add networking in the top right
select "Virtual Machine" and click next
select "Create a virtual switch" and click next
Enter a name for the switch (don't worry about vlan IDs)

rinse and repeat for any other subnets you want to have.

when you build the virtual machines that you want to route with make sure you have more then one virtual interface per machine and connected to more than one virtual switch.

physical interfaces come in to play is if you want to have these virtual switches connect to your physical lan

Enterprise parameters

Refer 3.3.3 SR4a

a. Automated Alternate Routing Enable – False by default, to enable set it to True

b. Peer Mode Enable Code —? Entarapeeee – This enables the code to enable Peer mode Database. Default is 0000, Maximum length – 4 — Check this out!!!!

c. Sync between ADP and Phone config – Default is True. If set to false Cm will not update ADP when updating speed dials, ip phone services, directory numbers etc.

d. Max no of device level trace – default is 12, max is 256, min is 0

e. URL Help – help hyperlink in the gateway web pages. (/help”>http:///help). Once you add a gateway like 6608, you should see a help hyperlink which takes you to product specific help.

f. Dependency records is set to False (disabled) in 3.3.3 sr4. Its enabled in 3.3.3

g. Show Ring settings ——- This will enable whether user can change ring settings or not. By default its set to False. If set to true, user can change ring settings from user webpage (http://cmserver/ccmuser)

Ring settings mean, how to ring when phone is idle and when phone is in use.

ex: ring normally, ring once, flash only, do nothing.

h. CDR parameters (6 parameters) – study these when you study CDR. ***REMEMBER***

i. Default Network Locale and Default User Locale – If a phone or device pool is not configured with a locale, it chooses what ever locale that is configured here.

j. URL Authenticate, URL Directories, URL Idle, URL Idle Time (sec), URL Information, URL Messages

Find out what is URL Authenticate ********REMEMBER*********

URL Directories – for showing corporate directory on phone. Remember to change all Enterprise paramater server names to ip address.

Make sure all users in CD have telephone field defined. Otherwise people wont be able to dial through a search in CD.

URL IDLe is used to display an image like logo of the company. Use IP Phone productivity services to perform this task. ********REMEMBER TO DO TASK**********

URL Idle timer = 0 (disables IDL url feature)

URL Information – press I page will invoke this web page.

URL Services – When services button is pressed, this url is invoked.

IP Phone Proxy address – To access all the above URLS a proxy can be specified.

k. Enable All user search – This if set to true, will allow blind search from Corporate directory in phone. Default is true.

l. User search limit. This is the number of users that are returned for a blind search from CD. Default is 64. Maximum is 500. (minimum is 1)

UCSniff (VoIP and IP Video Security Assessment Tool)

UCSniff is a VoIP & IP Video Security Assessment tool that integrates existing open source software into several useful features, allowing VoIP and IP Video owners and security professionals to rapidly test for the threat of unauthorized VoIP and Video Eavesdropping. Written in C/C++, and available on Linux and Windows, the software is free and available for anyone to download, under the GPLv3 license. Some useful features of UCSniff that have been combined together into a single package:

• Allows targeting of VoIP Users based on Corporate Directory and/or extensions
• Support for automatically recording private IP video conversations
• Automatically re-creates and saves entire voice conversation to a single file that can be played back by media players
• Support for G.729, G.723, G.726, G.722, G.711 u-law, and G.711 a-law compression codecs
• Support for H.264 Video codec
• Automated VLAN Hop and VLAN Discovery support
• A UC Sniffer (VoIP and Video) combined with a MitM re-direction tool
• Monitor Mode
• Sniffs entire conversation if only one phone is in source VLAN
• Gratuitous ARP Disablment Bypass support
• TFTP MitM Modification of IP Phone features
• Realtime VoIP and Video Monitor

You can download and try out UCSniff from their website.

Understanding SIP at UC Expo 09

An interesting day at the Unified Communications Expo in London, Olympia on 12th March (I kept away from day 1 on the 11th). I spent most of it wandering around looking at other seminars and also talking to people about SIP ~ especially if they have any training requirements for their technical staff and what I found was that overall, people are still confused about SIP and even unsure if it's something they need training on at all..! This confuses me as a lot of products on display rely on SIP and please don't tell me that everything works ok yet! Or maybe it does as long as customers stay 'locked-in' to one supplier that has only tested SIP on their own kit and not looking to honor the principles of SIP by opening up networks and solutions so that the customer has more choice... If this is so, they may as well carry on using their own proprietary protocols. But SIP is not going away, it IS being built into all PBX's and IP Phones and when you add SIP Trunking into the equation - it's a protocol that has got to be understood properly.

Anyway, our Seminar (in the VoIP business theatre sponsored by Siemens) on 'Understanding SIP' seemed to go down well to a packed house. The main points I tried to get across in the seminar were:

1. SIP is an evolving protocol so make sure suppliers can support what they sell.

2. Look at manufacturers and how they are supporting SIP now and what their roadmaps for development are.

3. Get help by looking at and even getting involved with working groups such as BLISS.

4. If you are a manufacturer, get involved with the SIP Forum by looking at their SIP Connect program and even attending their SIPit events.

5. That The SIP School is the best place to learn about SIP.

6. You can get accredited (SSCA™) and be recognised by companies such as Mitel, Panasonic, Ingate etc.

7. In a 'Down Economy' try eLearning as an alternative to classroom training - you don't have to cut out training altogether as people have a tendency to do in tough times.

8. eLearning is the 'Green' alternative.

9. Use the FREE "Interactive Directory" at http://wiki.thesipschool.com to advertise your VoIP/SIP products and services and help to grow this 'global' directory.

10. Ask tough questions of any supplier when looking at new Unified Comms solutions and listen closely to their answers. The last thing you need is your UC solution 'breaking' when you apply Windows updates, upgrade firmware in IP Phones or upgrade your Firewall etc.

The slides for the show are available here (scroll down the profile 1st - though they don't make much sense without Audio)

That's it for this blog and I'll be attending the Convergence Summit (North) next week and be on a panel talking about SIP Trunking with a focus on the selling! I'll let you know how it goes.

SIP in Unified Communications

Do you fire up your Gmail client to send an email to another Gmail user? And would that email fail to arrive if you used Outlook instead? Of course not and it's all down to the standard email protocol called SMTP. Every email client supports it so you don't have to worry what the other person is using. Likewise, HTTP delivers web content regardless of the browser being used.

This has always been the promise of SIP (the Session Initiation Protocol). Designed to establish, manage and 'tear down' communication channels between any media device, anywhere. Sound good?

SIP in Unified Communications

I'm not going to give you a definition of Unified Communications (UC) here, just hit Google and wade through all the interpretations that pop up. I do however, want to talk about how SIP 'underpins' UC.

It does this by ensuring that all elements of a business UC solution can establish connections to work together. It also allows for 'mid session' information to be passed between these elements to enhance the communications experience. A good example of this is starting up a whiteboard application with a colleague to give your existing conversation another dimension.

SIP/SIMPLE

Now UC isn't just about Instant Messaging, but I need to start somewhere.

The IETF SIMPLE (SIP for Instant Messaging and Presence Leveraging Extensions) working group focuses on the application of SIP to Instant messaging (IM) and Presence services. Invariably their work will affect any UC solution.

Now you may not know this, but when you use IM you become a 'Presentity'. That is, you are an 'entity' with a presence state i.e. online, not available etc. So now, if someone wants to see your presence state, their IM client sends a SIP SUBSCRIBE message to your IM server to 'follow' you. If you change your presence state i.e. off to lunch your IM client will send a NOTIFY message to your 'followers'. All of this is defined in RFC 3265, overseen by the SIMPLE working group.

They have also overseen the development of a number of RFCs that cover how presence information is coded in XML documents. These documents are carried in the body of a SIP message and the use of XML here can be taken advantage of by applications to facilitate interoperability and automation of business processes.

SIP/SIMPLE in Action

So, in the background SIP/SIMPLE is delivering (to you) presence state updates for the people you are following. Click on a 'buddy' to start a chat and SIP will carry the message. If you now want to speak with your friend, 'click to call' and SIP sets up the channel through an INVITE. Whilst you are talking you can fire up your Webcam and SIP will establish this connection for you without affecting your voice connection. Don't forget you can bring another person in on the conversation by initiating a conference; SIP can take care of that as well. In effect, it's SIP that connects everything together though it's important to remember that it's the applications being used that are making people more effective in their work.

Open Sesame

Because SIP is an open standard, all UC solutions should be able to work with each other seamlessly. Unfortunately, although everyone has their own presence capability there is no way to 'federate' (or join) them.

Today, if you are on a Cisco system you can't see who is present on an Avaya system. Maybe they'd just like you to buy their solution.

The push for Federations

But hold on! Microsoft and IBM announced (at Voicecon Orlando) that they will support Intra-domain federation so that a company with UC solutions from these two vendors will be a step closer to full UC support regardless of platform and client. Will this 'shake up' other manufacturers into doing the same? And how long will it be until SIP gives us the same 'transparent' interoperability that SMTP and HTTP do? Only time will tell.

CCIE Voice Lab Equipment and IOS Version

Passing the Voice lab exam requires a depth of understanding difficult to obtain without hands-on experience. For your convenience, I have researched the equipment list that Cisco recommend's for CCIE Voice track. Early in your preparation you should arrange access to the equipment covered on the exam, and listed below.

• MCS 7835 Call Manager/CRS Servers
• MCS 7835 Unity Server
• Catalyst 6503
• 3725 Routers (WAN, Gatekeeper, Gateway, PSTN Emulation)
• Cisco Unity Express
• 2651XM H.323 Gateway
• VG248 Analog Phone Gateway
• ATA186 Analog Telephone Adapters
• Catalyst 3550 with inline power
• IP Phones
• Analog Phones
• FAX

Software Versions

Any major release of Call Manager, Unity or CRS that has been generally available for six months is eligible for testing in the CCIE Voice lab exam. You may see more recent IOS versions installed in the lab, but will not be tested on those features unless indicated below.

• All routers use IOS version 12.4 Main Train
• Catalyst 6503 series switches use Switch OS version 7.6(X)
• Catalyst 3550 series switches use IOS version 12.1EA
• CCM version 4.1(3) with latest SR
• Unity version 4.0(5)
• CRS version 4.0(1)

Network Interfaces

• Fast Ethernet
• Frame Relay

Telephony Interfaces

• FXS
• FXO-M1
• T1
• E1

CCIE Voice Lab v3.0 Equipment and Software List

Passing the CCIE Voice Lab Exam requires a depth of understanding difficult to obtain without hands-on experience. Early in your preparation, you should arrange access to the equipment listed below:

Lab Equipment:

• Cisco MCS-7845 Media Convergence Servers
• Cisco 3825 Series Integrated Services Routers (ISR)
• Cisco 2821 Series Integrated Services Routers (ISR)
• ISR Modules and Interface Cards+ VWIC2-1MFT-T1/E1
  - PVDM2
  - HWIC-4ESW-POE
  - NME-CUE
• Cisco Catalyst 3750 Series Switches
• IP Phones and Soft Clients

Software Versions

Any major software release which has been generally available for six months is eligible for testing in the CCIE Voice Lab Exam.

• Cisco Unified Communications Manager 7.0
• Cisco Unified Communications Manager Express 7.0
• Cisco Unified Contact Center Express 7.0
• Cisco Unified Presence 7.0
• Cisco Unity Connection 7.0
• All routers use IOS version 12.4T Train.
• Cisco Catalyst 3750 Series Switches uses 12.2 Main Train

Network Interfaces

• Fast Ethernet
• Frame Relay

Telephony Interfaces

• T1
• E1

IPexpert Vol 1 – Lab 1a Overview

Over the last 2 weeks I have been working my way through the IPX Vol 1. Reason I have decided to stick to IPX for technology focussed labs is that they are more aligned with the actual blueprint.

Lab 1a covers the Implementation of Campus Infrastructure:

• VLAN
• DHCP
• TFTP
• NTP

IPX suggests that you complete the lab in 3 hours but I took my time going through the solutions over and over again and looking up the tasks on CUCM SRND and CUCME SRND. I did skip basic phone autoregistration here since its covered in detail in Lab 2a and 3a.

VLAN:

Coming from R&S background, this was routine work for me but I still wanted to read through and understand everything in VLANs. Basically it shows if you don't have any knowledge of the physical topology you can use CDP (Cisco Discovery Protocol) to discover the neighbors and configure the basic VLANs. This section asks you to configure trunk, create vlans on all the sites (HQ, BR1, BR2) and put the ports in data and voice vlans. Worth going through over and over again to get familiar with VLANs esp Voice VLANs. It will also help you under the difference between 3750 VLANs and 4ESW VLANs.

Important Commands:

• show cdp neighbors
• vlan (number)
• vlan (number) -> name NAME
• switchport mode trunk
• switchport trunk encapsulation dot1q
• switchport trunk native vlan (number)

DHCP & TFTP:

For DHCP/TFTP we can use the CUCM and any Cisco IOS Router or Switch. Vol 1 makes you configure both so that you are familiar with the options. CUCM Subscriber will be used as the DHCP/TFTP server for HQ and BR1 Phones and BR2 router will be used as a DHCP/TFTP server for the BR2 Phones.

Important Commands:

• ip helper-address x.x.x.x
• ip dhcp excludd-address x.x.x.x.x x.x.x.y
• ip dhcp pool
• option 150 ip x.x.x.x
• default-router x.x.x.x
• show ip dhcp binding

NTP:

NTP (Network Time Protocol) is very important in voice because you want to make sure that all your timestamps are accurate. As far as NTP is concerned, you want to make sure that all your servers and routers/switches are getting the time from a centralized source in this case PSTN router.

Important Commands:

• ntp server x.x.x.x
• show ntp status
• show ntp associations

TEHO – Tail End Hop Off

Tail End Hop Off (TEHO) refers to routing long-distance calls across the VoIP network and dropping them off to the Public Switched Telephone Network (PSTN), as a local call, at a remote gateway. In TEHO situations, you can reduce the configuration complexity from the need to configure N2 entities to needing only N entities. The following assumptions for TEHO apply:

• Each site has a different route pattern and route list for each of the other N-1 sites.
• For a given site, S, each of the N-1 route lists to another (remote) site has, as first preference, a route group of one or more gateways that are local to that other site followed by, as second preference, a route group that is local to S. Therefore, when sufficient trunking resources are available to honor the first preference, a long-distance call uses a gateway at the remote site to go offnet and thus bypass any tolls; otherwise, the call defaults to a local gateway and incurs toll charges.

Again, Cisco Unified Communications Manager has an identical routing policy for all sites. The second preference of routing a call through the local PSTN of a site (if the system fails to drop off the call as a local call at the remote PSTN) forces the customer to provision separate instances of all routing information for each site.

Cisco Unified Communications Manager Features and Services Guide, Release 7.0(1)

Implementing TEHO with PSTN Backup in Cisco Unified Communications Manager Learning Module:

Watch "Implementing TEHO with PSTN Backup in CUCM" module to cover basics for TEHO. By the end of this module, you will be able to:

• Describe the purpose and basic operation of TEHO
• Identify considerations when implementing TEHO
• Describe how TEHO is implemented
• Configure TEHO between two sites with PSTN Backup

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Cisco 7941/7961 Stuck at Upgrading Screen

I ran into this issue the other day when I had to do a factory reset on my Cisco 7961 and after that it was stuck at the upgrading screen while displaying the MAC address on the bottom left.

Looking through various lists, it appears that others have ran into this issue before. Here are a few things you need to check for this to work

• DHCP ip helper-address
• Phone Load
• DHCP Scope

Most common issue is that folks forget to have the correct helper-address. In my case I was pointing the phone to the SUB whereas the DHCP server was the PUB. I try NOT to get firmware from my CME router and in all cases get it from the CUCM since that's the easiest way to achieve this task.

I hope this was helpful!

Recovering CUE Module from an Errant State

I was getting ready to start my Cisco Unity Express lab and ran into this issue where the CUE module was stuck in errant state.

Once I sessioned into the CUE module this is what I saw:

BR2-RTR#service-module service-Engine 1/0 session

Trying 10.10.202.1, 2066 ... Open

EXCEPTION: /sw/info/ps: -5: out of memory for event sources

Waiting for IOS to register IP address.

- waited 15400 seconds...

WARNING:: IOS communication appears delayed!

WARNING::

WARNING:: Please verify the Service Engine IP Address

WARNING:: and Default Gateway are configured correctly

WARNING:: on the service engine interface in IOS.

EXCEPTION: /sw/info/ps: -5: out of memory for event sources

EXCEPTION: /sw/info/ps: -5: out of memory for event sources

EXCEPTION: /sw/info/ps: -5: out of memory for event sources

EXCEPTION: /sw/info/ps: -5: out of memory for event sources

EXCEPTION: /sw/info/ps: -5: out of memory for event sources

EXCEPTION: /sw/info/ps: -5: out of memory for event sources

BR2-RTR#service-module service-Engine 1/0 session Trying 10.10.202.1, 2066 ... OpenEXCEPTION: /sw/info/ps: -5: out of memory for event sourcesWaiting for IOS to register IP address. - waited 15400 seconds...WARNING:: IOS communication appears delayed!WARNING::WARNING:: Please verify the Service Engine IP AddressWARNING:: and Default Gateway are configured correctlyWARNING:: on the service engine interface in IOS.
EXCEPTION: /sw/info/ps: -5: out of memory for event sourcesEXCEPTION: /sw/info/ps: -5: out of memory for event sourcesEXCEPTION: /sw/info/ps: -5: out of memory for event sourcesEXCEPTION: /sw/info/ps: -5: out of memory for event sourcesEXCEPTION: /sw/info/ps: -5: out of memory for event sourcesEXCEPTION: /sw/info/ps: -5: out of memory for event sources

This out of memory message is caused by the bug CSCeg66388 which means that the Cisco Unity Express module has tried to establish connection with the IOS router multiple times and was not successful.

You can go ahead and reset the module to fix this issue:

BR2-RTR#service-module service-Engine 1/0 reset
Use reset only to recover from shutdown or failed state
Warning: May lose data on the the NVRAM, nonvolatile file system or unsaved configuration!
Do you want to reset?[confirm]
Trying to reload Service Module Service-Engine1/0.
Mar 4 01:15:49.835: %LINEPROTO-5-UPDOWN: Line protocol on Interface Service-Engine1/0, changed state to down
Mar 4 01:15:59.835: %LINEPROTO-5-UPDOWN: Line protocol on Interface Service-Engine1/0, changed state to up
Mar 4 01:16:00.839: %OSPF-4-NO_IPADDRESS_ON_INT: No IP address for interface Service-Engine1/0

This should take care of the issue but if you were reusing an existing module that had configuration and wanted to wipe it clean then proceed with the following command:

se-10-41-1-16# offline
!!!WARNING!!!: If you are going offline to do a backup, it is recommended
that you save the current running configuration using the 'write' command,
prior to going to the offline state.
Are you sure you want to go offline[n]? : y
se-10-41-1-16(offline)# restore factory default
!!!WARNING!!!: This operation will cause all configuration and data
on the system to be erased. This operation is not reversible.
Do you wish to continue[n]? : y
Restoring the system. Please wait .....done
System will be restored to factory default when it reloads.
Press any key to reload:
System reloading ....

I am glad I ran into this issue during my prep because this is something you don't want to see for the first time in the lab. I hope this was helpful to you!

Building your own CCIE Voice Home Lab

Building a home lab is never easy or cheap as most of you already know. In order to prepare for ANY CCIE you must have access to a full rack.

I am one of those who cannot work on remote racks. So once I decided to go for CCIE Voice, I started looking for the equipment. One major issue was figuring out which vendor's lab equipment list to follow since there are a few out there. I decided to go with IPexpert's BLS and thus my home rack is similar to IPX Racks. Problem is that no one lists the details so you really have to dig in and ask others. Here is a list of the equipment I have and you can modify it as you need. This will allow me to work on IPX labs without changing anything.

Update:10/14/2009

I also purchased Internetwork Expert's Voice Vol 1 and 2 and after looking at their topology it appears that my rack fits both IPX and IE So far IE Vol 1 looks better but I will keep you posted.

• HQ
  • Cisco WS-3750-24PS-E
  • Cisco 2821
  • VWIC2-2MFT-T1/E1
  • PVDM2-64
• HQ Phones
  • Cisco 7961
  • Cisco 7961
  • Cisco 7961
• BR1
  • Cisco 2811
  • PWR-2811-AC-IP
  • VWIC2-2MFT-T1/E1
  • PVDM2-64
  • HWIC-4ESW-POE
• BR1 Phones
  • Cisco 7961
  • Cisco 7961
• BR2
  • Cisco 2811
  • PWR-2811-AC-IP
  • VWIC2-1MFT-T1/E1
  • WIC-1DSU-T1-V2
  • PVDM2-64
  • HWIC-4ESW-POE
  • NM-CUE
• BR2 Phones
  • Cisco 7961
  • Cisco 7961
• PSTN/FR Router/Terminal Server
  • Cisco 3725
  • AIM-VOICE-30
  • VWIC2-2MFT-T1/E1
  • VWIC2-1MFT-T1/E1
  • VWIC2-2MFT-T1
  • WIC-1DSU-T1
  • NM-16A
  • PWR-2811-AC-IP
• PSTN Phone
  • Cisco 7961
• Server and Applications
  • Dell T100 Server Xeon 3.0 8GB RAM
  • VM for CUCM PUB
  • VM for CUCM SUB
  • VM for CUC
  • VM for UCCX
  • VM for CUPS
  • VM for Windows 2003 for AD intg

Installing UCCX 7 on VMware ESXi

Installing UCCX is NOT easy because of the fact that it is a Windows based install and Cisco does not let you install it on a VM. I will start with first creating a new VM and then how to get the UCCX up and running. I will only go as far as the activation because Integration into CUCM 7 is part of the process you must learn

I did a tutorial earlier about installing Guest OS on VMware ESXi earlier. Follow the tutorial to install a Windows 2003 Enterprise Server 2003 32-Bit on your ESXi. Keep the following in mind for UCCX related install:

• 2048mb or 2GB Memory
• 80gb Hard Disk

Everything else should be the same and you should be able to install the Windows OS using my tutorial.

Once installed, go ahead and reboot the VM. If you don't it will ask you to reboot anyways.

Now comes the tricky part, Cisco UCCX WILL NOT install on a machine other than the actual Cisco Server. So you have to play around with the registry file to get this going. Make sure you REBOOT the Server before changing the registry otherwise it WILL NOT WORK.

Here are the parameters you need in your registry:

• KEY - Hkey_local_machine/software/Cisco Systems/Model
• String Value - Hardware 7835H04
• String Value - Memory 2048
• String Value - Speed 3400

Once you modify the registry, you can go ahead and reboot the server.

Installing UCCX 7:

Now that you have a clean Windows machine patched for UCCX 7 install, go ahead and run the setup for UCCX 7. Few things to keep in mind:

• Make sure you choose CUCM rather than CUCME on the initial screen. I don't know what I was thinking and went with the 2nd option of CUCME and had to redo the VM along with installing the UCCX 7. So don't make that mistake.
• Also make sure that IIS is installed and enabled BEFORE installing UCCX 7 otherwise you WILL run into issues.
• Cisco Security Agent must also be disabled before starting the Install.
• Otherwise install is pretty basic and you should be able to click through options and once completed it will reboot.

Initial Setup and Activation Process:

1. Once the system starts up, go to http://177.1.10.40/appadmin (This is INE Toplogy). Use the default user name Administrator and password ciscocisco.

   
   
   

2. For the CCIE Voice Lab, we will be using Single Node deployment.

   

3. Now enter the CUCM Publisher IP and AXL User and Password. An AXL_User group should be created in your CUCM Pub for Standard API access. Then go to USER MANAGEMENT > APPLICATION USER and create an AXL user and assign it to the AXL Group created earlier. I have created an AXL user "admin" with password "ciscouccx". Just to clarify, AXLaccount is used by UCCX to access backend API of the CUCM.

   

4. You can get the license from http://www.cisco.com/go/license and enter your PAK. Cisco also has a Demo license but you have to work for a partner to get that

   

5. At this point the basic Activation is complete.

   

6. You have to Integrate UCCX 7 with CUCM 7 but that's something that I will let you figure out because you might be required to do that in the actual lab.

I hope this has been helpful to you. It took me a while to figure this out but after reading these steps you should be able to install UCCX 7 on a VMware without any issues. Don't forget to leave feedback

How To Fix CUCM Database Replication Issues

So if you have been working with Cisco Unified Communications Manager for a while then you are pretty familiar with the Database Replication Issues.

I ran into this a few times and thought it would be a good idea to have a post about it.

There are a few reasons for this and it seems to happen a lot in a VMware CCIE Voice environment.

So let's get straight to the resolution steps:

1. Remove CUCM Subscriber from the Group
2. Disable all services on the Subscriber
3. Login to your Publisher and issue "utils dbreplication repair all" command

This process can take some time so sit back or just work on some other tasks before coming back to your CUCM cluster.

Hope this helped!

CCIE R&S 4.0 Lab Blueprint

The following topics are general guidelines for the content likely to be included on the lab exam. However, other related topics may also appear on any specific delivery of the exam. In order to better reflect the contents of the exam and for clarity purposes, the guidelines below may change at any time without notice

Exam Sections and Sub-task Objectives

1.00	Implement Layer 2 Technologies	?
1.10	Implement Spanning Tree Protocol (STP)
	(a) 802.1d
	(b) 802.1w
	(c) 801.1s
	(d) Loop guard
	(e) Root guard
	(f) Bridge protocol data unit (BPDU) guard
	(g) Storm control
	(h) Unicast flooding
	(i) Port roles, failure propagation, and loop guard operation
1.20	Implement VLAN and VLAN Trunking Protocol (VTP)
1.30	Implement trunk and trunk protocols, EtherChannel, and load-balance
1.40	Implement Ethernet technologies
	(a) Speed and duplex
	(b) Ethernet, Fast Ethernet, and Gigabit Ethernet
	(c) PPP over Ethernet (PPPoE)
1.50	Implement Switched Port Analyzer (SPAN), Remote Switched Port Analyzer (RSPAN), and flow control
1.60	Implement Frame Relay
	(a) Local Management Interface (LMI)
	(b) Traffic shaping
	(c) Full mesh
	(d) Hub and spoke
	(e) Discard eligible (DE)
1.70	Implement High-Level Data Link Control (HDLC) and PPP
2.00	Implement IPv4
2.10	Implement IP version 4 (IPv4) addressing, subnetting, and variable-length subnet masking (VLSM)
2.20	Implement IPv4 tunneling and Generic Routing Encapsulation (GRE)
2.30	Implement IPv4 RIP version 2 (RIPv2)
2.40	Implement IPv4 Open Shortest Path First (OSPF)
	(a) Standard OSPF areas
	(b) Stub area
	(c) Totally stubby area
	(d) Not-so-stubby-area (NSSA)
	(e) Totally NSSA
	(f) Link-state advertisement (LSA) types
	(g) Adjacency on a point-to-point and on a multi-access network
	(h) OSPF graceful restart
2.50	Implement IPv4 Enhanced Interior Gateway Routing Protocol (EIGRP)
	(a) Best path
	(b) Loop-free paths
	(c) EIGRP operations when alternate loop-free paths are available, and when they are not available
	(d) EIGRP queries
	(e) Manual summarization and autosummarization
	(f) EIGRP stubs
2.60	Implement IPv4 Border Gateway Protocol (BGP)
	(a) Next hop
	(b) Peering
	(c) Internal Border Gateway Protocol (IBGP) and External Border Gateway Protocol (EBGP)
2.70	Implement policy routing
2.80	Implement Performance Routing (PfR) and Cisco Optimized Edge Routing (OER)
2.90	Implement filtering, route redistribution, summarization, synchronization, attributes, and other advanced features
3.00	Implement IPv6
3.10	Implement IP version 6 (IPv6) addressing and different addressing types
3.20	Implement IPv6 neighbor discovery
3.30	Implement basic IPv6 functionality protocols
3.40	Implement tunneling techniques
3.50	Implement OSPF version 3 (OSPFv3)
3.60	Implement EIGRP version 6 (EIGRPv6)
3.70	Implement filtering and route redistribution
4.00	Implement MPLS Layer 3 VPNs
4.10	Implement Multiprotocol Label Switching (MPLS)
4.20	Implement Layer 3 virtual private networks (VPNs) on provider edge (PE), provider (P), and customer edge (CE) routers
4.30	Implement virtual routing and forwarding (VRF) and Multi-VRF Customer Edge (VRF-Lite)
5.00	Implement IP Multicast
5.10	Implement Protocol Independent Multicast (PIM) sparse mode
5.20	Implement Multicast Source Discovery Protocol (MSDP)
5.30	Implement interdomain multicast routing
5.40	Implement PIM Auto-Rendezvous Point (Auto-RP), unicast rendezvous point (RP), and bootstrap router (BSR)
5.50	Implement multicast tools, features, and source-specific multicast
5.60	Implement IPv6 multicast, PIM, and related multicast protocols, such as Multicast Listener Discovery (MLD)
6.00	Implement Network Security
6.01	Implement access lists
6.02	Implement Zone Based Firewall
6.03	Implement Unicast Reverse Path Forwarding (uRPF)
6.04	Implement IP Source Guard
6.05	Implement authentication, authorization, and accounting (AAA) (configuring the AAA server is not required, only the client-side (IOS) is configured)
6.06	Implement Control Plane Policing (CoPP)
6.07	Implement Cisco IOS Firewall
6.08	Implement Cisco IOS Intrusion Prevention System (IPS)
6.09	Implement Secure Shell (SSH)
6.10	Implement 802.1x
6.11	Implement NAT
6.12	Implement routing protocol authentication
6.13	Implement device access control
6.14	Implement security features
7.00	Implement Network Services
7.10	Implement Hot Standby Router Protocol (HSRP)
7.20	Implement Gateway Load Balancing Protocol (GLBP)
7.30	Implement Virtual Router Redundancy Protocol (VRRP)
7.40	Implement Network Time Protocol (NTP)
7.50	Implement DHCP
7.60	Implement Web Cache Communication Protocol (WCCP)
8.00	Implement Quality of Service (QoS)
8.10	Implement Modular QoS CLI (MQC)
	(a) Network-Based Application Recognition (NBAR)
	(b) Class-based weighted fair queuing (CBWFQ), modified deficit round robin (MDRR), and low latency queuing (LLQ)
	(c) Classification
	(d) Policing
	(e) Shaping
	(f) Marking
	(g) Weighted random early detection (WRED) and random early detection (RED)
	(h) Compression
8.20	Implement Layer 2 QoS: weighted round robin (WRR), shaped round robin (SRR), and policies
8.30	Implement link fragmentation and interleaving (LFI) for Frame Relay
8.40	Implement generic traffic shaping
8.50	Implement Resource Reservation Protocol (RSVP)
8.60	Implement Cisco AutoQoS
9.00	Troubleshoot a Network
9.10	Troubleshoot complex Layer 2 network issues
9.20	Troubleshoot complex Layer 3 network issues
9.30	Troubleshoot a network in response to application problems
9.40	Troubleshoot network services
9.50	Troubleshoot network security
10.00	Optimize the Network
10.01	Implement syslog and local logging
10.02	Implement IP Service Level Agreement SLA
10.03	Implement NetFlow
10.04	Implement SPAN, RSPAN, and router IP traffic export (RITE)
10.05	Implement Simple Network Management Protocol (SNMP)
10.06	Implement Cisco IOS Embedded Event Manager (EEM)
10.07	Implement Remote Monitoring (RMON)
10.08	Implement FTP
10.09	Implement TFTP
10.10	Implement TFTP server on router
10.11	Implement Switch-module Configuration Protocol (SCP)
10.12	Implement HTTP and HTTPS
10.13	Implement Telnet

How to Install a Guest OS on VMware ESXi

In this post I will try to capture all the steps required to install a Guest Operating System on a VMware ESXi Server. This method should also work on the VMware ESX Server. I am assuming a few things here:

• You already have VMware ESXi installed on a capable Server (If you don't have a server you can buy it from eBay)
• You have installed a VMware Vsphere Client on your client Machine.
• You own a legal Copy of a Windows 2003 Enterprise Server (.ISO File)
• You have enough disk space on your server

Creating a New Virtual Machine

1. Open your VMware Vsphere Client to access your VMware ESXi Server. Right click on the ESXi Server and click New Virtual Machine.

   

2. Proceed with creating a Typical Virtual Machine since it will install all the common settings.

   

3. Now choose a name for the new Virtual Machine. In my case this is the CUCM Subscriber Server so I named it SUB.

   

4. Select the DataStore where you want this new Virtual Machine to be installed.

   

5. Select the Guest Operating System you are going to install on your new Virtual Machine.

   

6. Select the amount of storage for your new Virtual Machine. I selected the minimum of 8gb but also selected thin provisioning so if more space is needed it will be available.

   

7. Now you can review all the settings before the final OK. If you are satisfied with the settings, click FINISH.

   

8. Verify that your new Virtual Machine was successfully installed by looking in the left column for your Vsphere Client.

   

Locating & Installing the Guest Operating System

1. Select the newly created Virtual Machine (SUB) and then click Edit Virtual Machine Settings.

   

2. In the Virtual Machine properties, you will select the CD/DVD Drive. I had an ISO file stored on the ESXi's local datastore so I just clicked on the Datastore ISO file and browsed to my ISO file's location. Make sure to check the Connect at power on box otherwise it will never boot from the ISO file.

   

3. Now is the time to power on the newly created Virtual Machine. Since we made sure that installation media was connected at power on, you will see the Windows Setup Screen. Press ENTER to continue.

   

4. Follow the normal Windows Installation procedures.
5. After the installation is finished, windows will restart. Here our new Windows Virtual Machine is ready to be used! Make sure you install all the VMware Tools for better performance Please leave comments if you think this was helpful.