GNS3 setup and outside connections with VoIP example
I don’t want to write another guide to setup GNS3, there is a great tutorial which (in my opinion) covers all necessary steps. I just want to summarize a few things which may shorten the time to get a router in GNS3 to run, make it talk via the ethernet of your PC and connect to a physical router. Further I added an example for some simple VoIP configs to make a CME (with an 79xx phone registered) in the virtual router talk to a FXS interface (analog phone) on another router.
First of all you need to install GNS3, which in turn takes the necessary steps to install the underlying dynamips. Special thanks for this excellent work goes to the folks around the dynamips and GNS3 team. Have a look at the GNS3 site ( www.gns3.net ) and download the “all-in-1″ package. For the installation and initial etup, there is an excellent tutorial here: http://sourceforge.net/projects/gns-3/files/GNS3/0.5/GNS3-0.5-tutorial.pdf/download
I am running GNS3 in an XP64 environment on an AM2 platform with 8GB ram. I tried a few different router models and for my part the 2691 router runs rock solid. The 2621XM had its odds and crashed every now and then, but I cannot say what caused this. The downside of the 26xx series is the limited availability of up to date IOS if you want to test new features. Currently 12.4-15(T10) is the newest IOS which does not include some new SIP-related features. For testing and practicing standard VoIP features this will do the job in any case. I’m using an “Advanced IP-services” feature set. If you need to use the newest IOS-T-Versions, you need to simulate a 7200 series router for whichyou can download a 12.4-24(T1) version.
After the installation and initial setup, I just built a simple network with 2 routers (1 virtual and 1 physical) connected to each other via the PC’s ethernet interface. After starting GNS3, placing a 2691 router and a “cloud” symbol on the workspace, my simple network looks like this:
I switched on the display for the interface names. This is why you see the “nio_gen_eth:\….” entry unter the cloud. This is the physical PC interface. Regardless what IP-address the PC-interface has, the router address is independent (f0/0). If the router address is in the same subnet as your PC, you will be able to access the router from your PC. Lets assume a physical router with its f0/0 interface is reachable from the PC as well and has a FXS-VIC interface 0/1/0. The configuration of R0 (virtual router) and R1 (physical router) could look like this (assuming a very basic H323 connection to make the FXS interface reachable from CME):
R0:
interface FastEthernet0/0
ip address 192.168.16.3 255.255.255.0
speed 100
half-duplex <– we are sharing the PC-port like a HUB
!
dial-peer voice 100 voip
destination-pattern 1… <– H323 connection to the physical router
session target ipv4:192.168.16.100 <– IP-address of the physical router
!
telephony-service <– Switch on CME
max-ephones 2
max-dn 5
ip source-address 192.168.16.3 <- accept registration from IP-Phones at this address
!
!
ephone-dn 1 <– the DN for the CIPC or 79xx phone
number 2000
!
ephone 1
description CME-Phone-1
mac-address 0000.0000.0001 <– put the real mac-addres of the phone here
type CIPC <– phone type CIPC, 7940, 7960 etc.
button 1:1 <– first line gets DN #1 (2000)
!
If you use a 79xx phone you will also need to get the tftp server address (option 150)handed out to the phone via DHCP. A sample configuration would look like this:
ip dhcp excluded-address 192.168.16.1 192.168.16.150
!
ip dhcp pool IP-Phones
network 192.168.16.0 255.255.255.0
default-router 192.168.16.3
option 150 ip 192.168.16.3
This should let the IP-phone register with the CME. The dial-peer with the destination pattern 1… will send all dialed numbers with 4 digits starting with 1 to the .100 IP-address. The configuration of R1 with the FXS-port in 0/1/0 will look like this:
R1:
interface FastEthernet0/0
ip address 192.168.16.100 255.255.255.0
speed 100
half-duplex <– we are sharing the PC-port like a HUB
!
dial-peer voice 200 voip
destination-pattern 2… <– H323 connection to the virtual router
session target ipv4:192.168.16.3 <– IP-address of the virtual router
!
dial-peer voice 100 pots
destination-pattern 1000 <– DN of the analog phone
port 0/1/0 <– physical FXS-port
The destination-pattern 1000 assigns DN 1000 to the analog port and is used as CLID on outgoing calls.
This should serve just as a starter configuration. Dive into the VoIP configurations and test any szenario with multiple routers – the sky is the limit. For demo purposes I connected the physical router with a FXO-port to my landline and so I got a PSTN integration. Further possibilities could be setting up a VPN to one of your routers and register the CIPC on your Laptop via the VPN at the CME…..
An integration with UCM will be possible in a similar way. The router needs a Dial-peer pointing to the Callmanager and in turn the Callmanager needs to know a gateway entry with the routers IP-address…..
by Patrick Geschwindner, Ascolta
GNS3 (dynamips) and Voice-Labs
I hear often that it is hard to practice/demo voice related configurations with GNS3. This is only partially true.
It is correct that GNS3 cannot substitute Multiservice routers like 28xx etc because the DSP resources cannot be emulated. For using T1/E1 or FXS/FXO connections you need the real hardware interfaces and of course another PBX/router interface to connect your T1 to. If you have a double T1 you could use a T1 x-over to let the router do both ends.
For all other configurations like Dial-peers, translation patterns, SIP-UA configurations, CME, ephone registrations, CUBE etc, GNS3 can do the job. Just an example from my own lab: I have several accounts at SIP-Providers in US and Europe to use with a standard sip-phone like X-Lite. The router can log in this SIP-account and let CME or UCM handle the call. Via this way I provide a local german PSTN-number for my german friends which makes my phones in Minnesota ring. The thing has of course a downside… they keep forgetting that I am 7h behind, but for this purpose Unity will take the call.
Where is GNS3 now coming into the ballpark? For each router you can specify only one SIP-registrar server, but I like to have all my SIP-accounts registered at the same time to route calls between them. GNS3 can now run the virtual routers, one for each SIP-provider, register your PSTN-numbers and UCM/CME can do the call handling.
by Patrick Geschwindner, Ascolta
Convert Cisco IP-Phone from SIP to Skinny (SCCP) and reverse
I have often been asked for a quick way to convert a Cisco IP-Phone between SIP and SCCP. People get their IP-Phones often from E-Bay or other 2nd hand resource for testing and practicing and they may receive them with the ”other” phone load they didn’t want. The easiest way to convert is via Autoregistration. With CME this is a litle more complex, but with UCM (v6 or higher) this is an easy task.
First of all, switch on Autoregistration. If you need help with that, there is another post by Brandon in this blog how to switch on Autoregistration.
Second, switch the Autoregistration Protocol to SIP or SCCP in the Enterprise Parameters (System -> Enterprise Parameters) . If the phone had been registered before, it needs to be deleted, otherwise it will not be considered as a “new” one and Communicationmanager will just reregister the phone.

Save the new setting and delete the phone and connect it if not yet done. The phone now will download the default configuration file and the new phone loads. It will reload multiple times, so just be a little patient.
One remark for phones being connected to the UCM behind a WAN: TFTP-downloads via a WAN can be pretty slow. A download can be between 20 and 30 minutes for all phones based on Java which are literally all but the older 7940′s and 7960′s
by Patrick Geschwindner, Ascolta
First mobile post
Well I can’t type that fast yet but I broke down and got an iPhone. I got my wife one as well and I got her a new MacBook also. I figure that she deserved it since she puts up with my many hours of CCIE study.
So far I love it.
Do you have any suggestion on how to use it for cisco studies? How about apps? Let me know. I want to make a top ten list.