When you are building a converged network you are integrating Data, Voice and Video. The voice network has to be connected the telephony network of your Telco. The most common way how to do it, is just to use some telephony interfaces (FXO, E&M, T1/E1, ...) on your ISR router which will be connected directly to a Telco's endpoint. By using this way you have to have additional cards in your router, the management may be more difficult and the costs are usually higher.
Other approach how to connect your VoIP based telephony network to a Telco is directly with VoIP trunk. Why should you get one Internet connection and other different telephony connections if you can integrate all those services also on your network edge, just by reusing your data (Internet) connection. In this way you can save your company some money by lovering costs for different network accesses, your network is more easily managed, and you can implement all of your required converged network functionality in one device - in an ISR router with Unified Communications Manager Express.

What you will need is a Telco operator which provides VoIP services. Nowadays almost every Telco has also VoIP services in his list.
You have to get some information about the provided VoIP service by your Telco - which signalizing protocol they use - usually SIP and in some cases it may be H323. Then you should ask about supported codecs - only uncompressed G.711 or also a better choice for a WAN link - G.729.
The next step is to configure your CME service with an external SIP trunk. This can be accomplished in many different ways.
Depending on your Telco - he may provide a "real" SIP trunk which is a predefined voice route to the IP address of your router. For this kind of service you must have a public static IP address on the WAN side. If your operator provide this kind of service, you should get it. It's the best choice. Then the configuration of CME is really simple - consist just in creating a VoIP dial peer to PSTN numbers directing to your Telco's VoIP server.

! this configuration uses by default the G.729 codec. If your telco has no support for this codec, you must configure also a codec parameter for this dial peer
dial-peer voice 1 voip
 destination-pattern 9.
 session protocol sipv2
 session target dns:myvoicesp.dot

A bit more complicated is a situation when your Telco provides the VoIP service in a "true" SIP mode - in which your device has to register itself to the VoIP server of Telco. Usually the Telco assign your an username and password which is then used in the registration process. The advantage of this solution is that you may have a dynamic IP address on the WAN side, or even in some situations you may have a private IP address behind NAT. The CME's configuration for this solution may a a bit tricky and sometimes you may spend few minutes with a debug lines. Anyway, after configuring it works with the same quality like the previous solution.

sip-ua
authentication username VoIPUSERNAME password VoIPPASSWORD
registrar dns:myvoicesp.dot expires 300
sip-server dns:myvoicesp.dot

... you can find out here.

Yes, you can put colors into your CLI of IOS. The keys for this feature are the escape characters and the terminal support for colors.
For example to create a banner like on the picture below you have to enter a banner command like this:

swr(config)#banner motd #
Enter TEXT message.  End with the character '#'.
< CTRL+V >< ESC >[1;33m

     _______.____    __    ____ .______
    /       |\   \  /  \  /   / |   _  \
   |   (----` \   \/    \/   /  |  |_)  |
    \   \      \            /   |      /
.----)   |      \    /\    /    |  |\  \----.
|_______/        \__/  \__/     | _| `._____|

Cisco Unified Application Designer is a tool which helps you to rapidly develop applications that converge voice and video with enterprise applications and data. Without learning the complex details of telephony protocols and real-time media processing, you use drag-and-drop techniques to visually construct applications containing communications business logic. This approach will help you reduce development time for new applications to days or weeks instead of months, while producing more full-featured applications.
An integrated component of the Cisco Unified Application Environment, the Cisco Unified Application Designer offers these features:

Finally I've got a new Cisco IP Communicator 2.1.
Some weeks ago I wrote an article about the new Cisco IP Communicator 2.1. In that time there was just in some non-official version and a public whitepaper, but yesterday I found it on CCO (you have to be logged in). It looks like it was released on 16-MAY-2007.
I am very happy because the new version has finally support for the Realtec HD Audio chip which is used in my notebook. The previous version (2.0) of IP Communicator didn't detect my sound card so I wasn't able to create voice calls from my notebook :-(  But now with 2.1 it works pretty well :) Big thanks for Cisco devel guys :-)

If you are not satisfied with prebuilt ringtones in your Cisco IP Phone, then you can quite easily get your favorite song as your ringtone. Here is the procedure how to create your own ringtone:

1. Get your favorite song as a mp3 or a wav file.
Maybe Google with help you to find it :)

2. Convert it into a RAW PCM file with some specified requirements.
You can use any kind of sound editing software in which you can create a RAW PCM file.

During the CZ&SK NetAcad conference some guys asked me how could they find which IOS contains the Cisco Unified Communications Manager Express (formerly Cisco Call Manager Express). The answer is quite easy, you can use Cisco Feature Navigator web page. Click on the "Search by feature", then select "Call Manager Express" in version that you want and click on "Add" and "Continue". Then select your "Hardware plaform" and there you are :) You will get a list of IOS images for your HW platform which contains CCME.

During the last week I was again presenting on the CZ&SK NetAcad Conference 2007 our experiences with teaching IP Telephony with NetAcad IPT materials. The presentation was great (at least I hope it was :)). Maybe the most important part of my presentation was the summary of feedbacks that we've got from our students on IPT trainings. I will just briefly mention some of most wanted things that we should include in our training on IPT:

• Transformation of IPT(x) into independent course like the FWL
  • Unfortunately not yet, not in this year and not even in the next year :(
• SIP
  • CCME supports SIP trunk and from CCME 4.0 it also supports SIP endpoints registration
• Unity Express
  • This is important if you want to do the IPTX certification exam
• IP IVR
  • It's just nice :)
    

Cisco has one of the biggest IP Telephony network (if not the biggest one). But it wasn't built just from one day to another. It took some years to build it into its present state. If you want to learn more about how has Cisco migrated its old telephony network into a one fully converged network with IP Phones and unified communications, then click on the links bellow:
IP Telephony Deployment (PDF - 2.34 MB)

In March 2006, Cisco introduced a new portfolio of products, the Cisco Unified Communications system of products.
Cisco continues the evolution of the portfolio in March 2007 by further refining several product names within the Cisco Unified Communications system. Cisco is changing the name of Cisco Unified CallManager and Cisco Unified CallManager Express to Cisco Unified Communications Manager (Express). The change from "CallManager" to "Communications Manager" allows the names of these products to more accurately reflect the functions they represent: world-class call processing, plus video, mobility, and presence applications and more. Existing mobility applications Cisco Unified Mobile Connect and Cisco Unified MobilityManager have been renamed Cisco Unified Mobility, and Cisco Unified Presence Server has been renamed Cisco Unified Presence.

The upcoming version of Cisco IP Communicator in version 2.1 will support Microsoft Windows Vista and there will be also support for SIP  (I think only for CCM 5.x) as well as SCCP protocols.
For more information please refer to the attached document.

I have been looking at some youtube videos and I've found out that there are many interesting videos also about VoIP and Cisco.
So I am posting here some videos of the best I've found:

Welcome to a World, where being here means being there:

The power of communication in a daily life:

Many times you need to explain what is the VoIP all about. I've found a nice flash animation from Cisco which explains in basic what is VoIP and what are the parts of the VoIP network. It's a quite old animation but maybe it will be useful:
http://www.cisco.com/warp/...kbs/voip_kb.html