Due to the enormous response to my posting Monday we've setup an automated method to request access to the new beta material.  To access the new material login to your account and click on the upgrade link to the right of the "R&S Lab Workbook Volume I v4.1" link.  You'll be sent an email with instructions on how to use our new locklizard pdf reader, and you'll see a new course subscription in the members site for the new content access.  The documents can be printed twice each, but viewed an unlimited amount of times.

If you are going to use our rack rentals for configuring the labs there are some major feature enhancements that we've been working on.  You can now load initial configs, save your configs, and reload your saved configs on demand to any rack session that you have.  Next to your active rack session you'll see a link that says "Beta CP" for the new control panel.

Please let me know what you think of the new content format and any ways that we can improve it.


Brian McGahan


A voice lab rack usually utilizes dedicated piece of hardware to simulate PSTN switch. Commonly, you can find a Cisco router in this role, with a number of E1/T1 cards set to emulate ISDN network side. It perfectly suits the function, switching ISDN connections between the endpoints. Additionally, it is often required to have an “independent” PSTN phone connected to the PSTN switch, in order to represent “outside” dialing patterns - such as 911, 999, 411 1-800/900 numbers. The most obvious way to do this is to enable a CallManager Express on the PSTN router, and register either hardware IP Phone or any of IP Soft-phones (such as IP Blue or CIPC) with the CME system.

However, there is another way to accomplish the same goal using IOS functionality solely. It relies on the IP-to-IP gateway feature, called “RTP loopback” session target. It is intended to be used for VoIP call testing, but could be easily utilized to loopback incoming PSTN calls to themselves. Let’s say we want PSTN router to respond to incoming calls to an emergency number 911. Here is how a configuration would look like:

voice service voip
allow-connections h323 to h323
interface Loopback0
ip address
dial-peer voice 911 voip
destination-pattern 911
session target ipv4:
incoming called-number 999
tech-prefix 1#
dial-peer voice 1911 voip
destination-pattern 1#911
session target loopback:rtp
incoming called-number 1#911

The trick is that only IP-to-IP calls could be looped back. Because of that, we need to redirect the incoming PSTN call to the router itself first, in order to establish an incoming VoIP call leg.

While this approach permits VoIP call testing, it lacks one important feature, available with the “real” PSTN phone: placing calls from the PSTN phone to the in-rack phones. However, you can always use “csim start” command on the PSTN router to overcome this obstacle. Have fun!

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