blindhog.net

As most of you may or may not know, Cisco currently only provides non-bootable Unified Communications ISO images on CCO for download. These ISOs are listed as being provided for “upgrades” and hence will not boot.
 
The only difference between these non-bootable images and a bootable disc is that the bootable disc includes a boot sector file.   Fortunately, this boot sector file can easily be extracted from a bootable disc and then injected into your non-bootable ISO.  To accomplish this, you will need to use some sort of ISO image software such as UltraISO, MagicISO, PowerISO, etc.  I prefer to use UltraISO for its ease of use.
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Cisco IP Communicator (CIPC) requires a sound card be installed on a PC in order to install or open. Under normal circumstances, this completely makes sense. However, in other circumstances (demos/labs), PCs are actually virtual machines … and do not have sound cards. In these situations, the engineers/trainers/etc are more concerned with dialing than actually hearing audio.

In order to use CIPC in a virtual machine, install Virtual Audio Streaming – a virtual sound card tool from ShiningMorning.com prior to installing CIPC. This software will install virtual audio drivers on your VM which allow CIPC to install and have dialing capabilities. Unfortunately, CIPC will still only allow one instance to run at a time…unlike IP Blue. I have tested this with VMWare ESXi but it may also work with other VMWare products.
 
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This tutorial will show you how to create or convert an existing audio file and save it in CCITT u-law – 8KHz 8bit Mono format for use in Cisco Unified Call Manager Express (CUCME), Cisco Unified Contact Center (UCCX) or Cisco Unity Express (CUE). This has always been an easy task with the Windows Sound Recorder that came with Windows XP. However, the sound recorder that comes with Windows 7 does not work the same.
 
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This tutorial will show you how to make config changes with a simple tcl script and invoke the script from an IOS menu. I was recently presented with a scenario where the FXO ports on a customer router were not receiving the supervisory disconnect from the PBX.
 
Until I find a way to auto reset the ports with EEM and the long call duration flag, the temporary solution is to create a menu where the customer can view call durations and reset voice-ports with a menu. The tcl script performs a ‘shut’ and ‘no shut’ on the voice port.
 
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Bit-Twist is a simple yet powerful libpcap-based Ethernet packet generator. It is designed to compliment tcpdump or wireshark packet captures. In this tutorial, you will learn how to edit and replay network packet captures files on a live lab network.
bittwist.sourceforge.net
 
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This tutorial is a continuation of the virtual voice lab series. In this tutorial, you will add a dial-plan to the PSTN router that was added in the first part of the PSTN simulation lab. (more…)

Occasionally, I have run into the issue when turning up a PRI that ring back is not heard when placing an inbound call. Caller only hears silence until the called party answers. This seems to be more prevalent with smaller Telco’s who don’t have a lot of experience with PRIs.

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Last week, I was part of the Gestalt IT Network Field Day in San Jose, CA. You may have noticed a lot of buzz on twitter and blogs. The purpose of the Tech Field day is to bring independant people active in the networking community together with vendors to discuss products, ideas, technologies and most important Networking. Not only Networking in a technical sense but also networking in terms of people. (more…)

This tutorial is a continuation of the virtual voice lab series. Unfortunately, the virtual voice lab has been a “work in progress” much longer than I had hoped it would be when I started it two years ago. In this tutorial, you will add a new router to your GNS3 topology. It will connect it to the HQ, BR1 and BR2 routers to simulate an Internet connection and the Public Switched Telephone Network (PSTN).

There is one caveat to this PSTN simulation… GNS3/Dynamips is not capable of simulating traditional analog or digital circuits. We will be simulating an Internet Telephony Service provider (ITSP). ITSPs have become increasingly popular in the past ~5 years. Basically, the PSTN router will route incoming calls to the branch routers and received outgoing calls from the branch routers over a SIP or H323 trunk.
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