Bridging Interfaces of Windows Laptop

This will be a useful trick on Windows which you can use to give internet connection to your Cisco lab routers/servers. Basically this will allow you to connect a wired network to a wireless network through your laptop.

Go to Network Connections and select the WiFi and Ethernet interfaces you want to bridge and right click and select Bridge Connections.

Once you click Bridge Connections a new icon for Network Bridge will appear and that's it.

Now the IP address will be assigned to this new interface and will be accessible through Ethernet and WiFi. Now you can put a default route to the WiFi router in the router connected to the laptop directly to route traffic to traffic to internet.

Convert Non-Bootable Cisco CUCM/UCCX Images into Bootable Images

Most of the time you will find / download CUCM / UCCX etc images in non bootable formats which will not be installed on VMware Workstation or other platforms easily. For lab purposes you will find this post very useful. I am using UltraISO for this tutorial.

Open or extract your image to see the contents in it. Refer the following image..

(click on the image to view in full size)

You can see a [BOOT] folder in the contents. If so this is a bootable image. If that folder does not exist, it is a non-bootable image and it will show an error "Operating System not found" when you are going to install in VMWare Workstation etc.

I am converting a UCCX image here. UCSInstall_UCCX_9_0_2_UCOS_9.0.2.11002-27.sgn.iso is the image I downloaded from Cisco.

Open UltraISO and go to File > Open and open the image..

Go to isolinux folder and extract isolinux.bin to your desktop or somewhere..

Go to Bootable and select Generate Bootinfotable..

Go to Bootable and click on Load Boot File and select the extracted bin file previously.

Finally go to File > Save As and give a name like Bootable_UCCX.iso
Now its is ready to install in VMWare workstation etc.

Mapping Calls to FXO Ports on Analog Gateways

This can be found mostly nowadays with SRST configuration. But it is not only limited to SRST. If you have an ISR at home, you can do this with your home PSTN connection. I have a Cisco 2911 with FXO ports and some IP Phones which are registered to my VGW (Voice Gateway Router).

If you want to know how to register an IP Phone at a Router please click here.
If you want to know how to configure SIP SRST please click here.

Setup is simple, the RJ11 from PSTN analog line connects directly to the FXO port of the router. FXO (Foriegn Exchange Office) is a type of WAN interface card which an analog line from a PBX or a Telephone service provider can be connected to.




(1) Create a POTS dial peer for outgoing calls

VGW(config)#dial-peer voice 100 pots
VGW(config-dial-peer)#destination-pattern ..........
VGW(config-dial-peer)#port 0/0/0

Destination pattern .......... matches any number with 10 digits. port 0/0/0 is the physical analog port connected to the analog line.

(2) Configure an extension to ring for incoming calls

VGW(config)#voice-port 0/0/0
VGW(config-voiceport)#connection plar opx 1111
VGW(config-voiceport)#caller-id enable

This is configured to ring 1111 extension. Last command is to identify the caller's number.

Above configuration is enough to make calls out and receive calls in. Following command will show an active incoming call on port 0/0/0 as off-hook.

VGW# show voice port summary

Now let's say if you have 2 or more analog lines connected to several ports. You can create a trunk group and assign it to voice ports.

VGW(config)#trunk group PSTN
VGW(config-trunk-group)#hunt-scheme round-robin

VGW(config)#voice-port 0/0/0
VGW(config-voiceport)#trunk-group PSTN 1

VGW(config)#voice-port 0/0/1
VGW(config-voiceport)#trunk-group PSTN 2

In above configuration, Line choosing method is round robin which means if one line is busy it will go to the other line. 1 in the trunk-group sub command states it is the preferred line.

Note:-

show call active voice and debug voip dialpeer commands will be useful in troubleshooting.

SIP SRST Related Configuration

Survivable Remote Site Telephony (SRST) is a feature we can configure on Cisco ISR routers which will act as Call Managers when phones cannot reach CUCM. Mostly you will see this at WAN sites but for the sake of learning the technology, I am configuring it with in a LAN segment.

Following is a real world topology.

Following is the lab I am setting up..

Goal is to register the IP phones at Router when connection to CUCM is gone.
First let's look at the license capability of the Router. I am using a Cisco 2911.

Router#show license detail cme-srst

If the output looks like above, it's good to go.

(1) Create a SRST reference on CUCM

Go to CM Administration > System > SRST and Add New

IP address 192.168.137.100 is the IP address of the SRST router.

(2) Give the SRST Reference on Device Pool

Go to CM Administration > System > Device Pool and select SRST reference.

(3) Enable SIP Registrar Functionality on Router

Router(config)#voice service voip
Router(conf-voi-serv)#allow-connections sip to sip
Router(conf-voi-serv)#sip
Router(conf-serv-sip)#registrar server expires max 600 min 60
Router(conf-serv-sip)#bind all source-interface g0/0

(4) Globally Assign Characteristics to SIP Phones

Router(config)#voice register global
Router(config-register-global)#mode esrst
Router(config-register-global)#system message "SIP-SRST is Active"
Router(config-register-global)#max-dn 200
Router(config-register-global)#max-pool 20

"SIP-SRST is Active" will be displayed on screen of the IP Phones when it is registered at router.

(5) Configure SIP SRST Services for IP Phones

Router(config)#voice register pool 1
Router(config-register-pool)#id network 192.167.137.0 mask /24
Router(config-register-pool)#codec g711ulaw
Router(config-register-pool)#dtmf-relay rtp-nte cisco-rtp sip-notify

Above commands will be enough to register IP Phones at router when CUCM is down.

Router#show sip-ua status registrar will show the current registered phones.

If the router is configured correctly as an Gateway (with dial-peers etc) calls will be routed out.