Cisco 6807 and 6800ia Swtich QOS for Cisco ip phones
Does anyone have an example of configuring a 6800ia switch port connected to a 6807VSS parent for cisco ip phones qos. Normally we'd use auto qos voip but auto qos is not supported on 6800IA switches.
Cant find any cisco documentation of what the IA switches port config should look like for a cisco ip phone.
Any help would be appreciated.
Thanks,
Dave
I'd leave QoS alone.
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I have ios 5.1.1 and when I check for updates the phone says there are no updates
i have ios 5.1.1 and when I check for updates the phone says there are no updates
Go to Settings > General > Software Update and see if you can update it over the air.
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Roamingnome, thanks for your response. Although I have been a Bell (Canada) customer for my WiFi signal for years, the 5c and its necessary mobility plan, is very recent. Some years ago, Bell supplied its own ‘router’ for my computer, and claimed that it was possible to get their WiFi signal only with the router they supplied. When I recently received the 5c, I wondered if the plan it requires, which is within the much wider, overall Bell zone, could be used somehow to get my Air online - some sort of tethering? - that allows me to give up the Air WiFi service and router all together, but still get the Air connected. Here in Canada, Bell Mobility and other Bell services are very separate. Any suggestions appreciated.
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Stevecurtterp wrote:
I use Touchdown from Nitodesk for my work email. Its a little pricey, but it works with your companys OWA (Outlook Web Access) portal. They do have a 30 day trial that they insist that you try to make sure it will work for you before you shell out any money for the app. It syncs the Calendar, notes, tasks, emails etc.
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No I am not employed by Nitrodesk. I only use the program. There was no need to ask my IT people to set anything up for me.
Give it a try, it might work as well for you too.
One day I will have to give this app a try, I am just use to configuring my device to sync without a 3rd party app but I have heard that Touchdown seems to be the best thing going to setup a corp sync process.. -
How do I reset my secuirty questions??? I cannot remember one of the answers and cant download anything for purchase on phone
If you have a rescue email address (which is not the same thing as an alternate email address) set up on your account then you can try going to https://appleid.apple.com/ and click 'Manage your Apple ID' on the right-hand side of that page and log into your account. Then click on 'Password and Security' on the left-hand side of that page and on the right-hand side you might see an option to send security question reset info to your rescue email address.
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right now i share an icould with my sister's phone how do i log out of hers and make my own for my phone without deleting either of our apps etc on our phone?
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Cisco IP Phone 7921 message "DHCP Timeout"
hi;
i have this message "DHCP Timeout"on the cisco 7921 and the phone block in "updating firmware".
can you help me
ThanksHi,
It was due to network reachability issues.
Try configuring the IP and TFTP address statically for the IP Phone
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GKN -
Recording for Cisco IP Phones and Cisco C90 Codec
Hello
We are looking for a solution that is capable to record both Cisco IP Phones and Cisco Codec C90.
We are using CUCM 9.X for IP Phones and VCS 7.X for Cisco Codecs.
Is their any third party solution available for both the requirements or do i have to go with TCS and any other third party recording solution.
Thanks & Regards
Aniket PatilMy reply may be too late to be of any help to you, but for the benefit of others:
Be sure you understand the different types of PoE out there. The Linksys PoE switch only supports the newer IEEE 802.3af PoE standard.
The 7940, 7960, 7905 and other older Cisco phones only support Cisco pre-standard PoE and thus will not work with the 802.3af Linksys Switch.
To use this switch, you will need to make sure you are using the newer 7070, 7961, 7941 phones with support both pre-standard and 802.3af PoE.
All the best,
John -
Setting Up VLAN and QoS for VOIP on SG200-18
We recently purchased the SG200-18 smart switch to replace a Netgear unmanaged switch. We're moving our phone service to VOIP through our local ISP as well.
I've currently got the VOIP phone plugged into Port 17 on the SG200-18 (it's a Grandstream cordless VOIP phone).
I want to put the VOIP phone on a separate VLAN from the rest of the network and optimize the QoS settings so that the VOIP phone has exceptional audio quality even during intense network traffic.
Here's my questions:
1. Do I need to adjust anything on the type of port for Port 17 (since it looks like some form of Combo port)?
2. How do I go about isolating the VOIP phone on it's own VLAN (I'm seeing VLAN and Voice VLAN settings, not sure which one to use; I tried setting a VLAN and broke Internet connectivity to the phone until I went in and removed it)?
3. Do I need to adjust any QoS settings on the switch to better optimize the VOIP phone?
A couple of additional questions about the GS200-18 in general:
1. Do I need to adjust any of the System Time Settings on the switch? I'm in Central Time.
2. Do I need to adjust any of the Green Ethernet/Energy Saving settings or should I stick with the defaults?
Also, a couple of "getting started" side questions to Cisco:
1. I've registered a My Cisco account. What do I need to do to register my switch with Cisco and associate it with my My Cisco account?
2. What are the benefits of taking out a Cisco Small Business Support Contract, and about how much would it cost on the SG200-18 (I ordered it from Provantage)? I'm curious to see if it's worth the money.
Here's my "specs":
Switch: SG200-18
VOIP phone: Grandstream DP715 and 710 expandable handsets
Plugged into: Port 17 on the SG200-18
ISP: Local ISP (Direclynx)
Connection type: 3M down/500k up DSL, moving to a wireless connection coming up which will give us faster speeds
VOIP backend provider: VOIP Innovations
Router: Apple Airport Extreme AC model (I run all Macs and iOS devices and OS X Server on the network, so using the Apple router makes setup easier, since it doesn't QoS, trying to QoS and VLAN at the switch level)
Thanks everyone!Hello,
Lots of different questions here so I'll try to make sure I don't miss anything.
1. Do I need to adjust anything on the type of port for Port 17 (since it looks like some form of Combo port)?
The way the combo ports work is you can either use the SFP slot for a fiber connection or the copper ethernet port, but not both at the same time. Other then that they just function as normal network ports.
2. How do I go about isolating the VOIP phone on it's own VLAN (I'm seeing VLAN and Voice VLAN settings, not sure which one to use; I tried setting a VLAN and broke Internet connectivity to the phone until I went in and removed it)?
It sounds like you created the VLAN correctly and assigned the phone, however there wasn't anything doing any routing for that VLAN. You would need to have a VLAN capable router or a layer 3 switch so that something would act as the default gateway for the voice VLAN and route the traffic for you. Since there was nothing like this your phone lost it's connectivity to the internet when you placed it in the new VLAN. I don't think the Airport is VLAN capable, but we will come back to that.
3. Do I need to adjust any QoS settings on the switch to better optimize the VOIP phone?
Once you have a seperate VLAN setup for the phone properly you only have to tell the switch what your Auto Voice VLAN is going to be and it will automatically apply recommended QoS settings for the Voice VLAN and prioritize the voice traffic. There are ways to do this manually and even with the phone in the same VLAN however the are considerably more complicated.
1. Do I need to adjust any of the System Time Settings on the switch? I'm in Central Time.
The system time isn't always very important. You can set the correct time zone, however you should know the switch does not have a battery in it to keep track of time, so if/when it reboots or loses power the clock will reset. If you would like the switch to maintain accurate time you should setup an NTP server so the time is automatically updated from the internet. The switch will keep your timezone settings once you save them. Time is mostly important for logging and things like that, so you can configure it if you like but it is not necessary.
2. Do I need to adjust any of the Green Ethernet/Energy Saving settings or should I stick with the defaults?
Green ethernet simply reduces the power usage of the switch slightly, so unless you are having odd issues where ports are disconnecting, I would just leave them at the defaults.
1. I've registered a My Cisco account. What do I need to do to register my switch with Cisco and associate it with my My Cisco account?
There isn't really a way to associate your Small Business devices with your Cisco account. If you ever call in for technical support we will use your Cisco account and your serial number to create a support case, but even then they aren't linked together. If you decide to buy a support contract, that will be linked to your switch's S/N and your Cisco ID, so in a way that would associate them together. Devices being associated with Cisco accounts is something more common with Enterprise equipment, and mainly has to do with technical support cases.
2. What are the benefits of taking out a Cisco Small Business Support Contract, and about how much would it cost on the SG200-18 (I ordered it from Provantage)? I'm curious to see if it's worth the money.
There are a few advantages to a Support Contact. Your switch comes with a Limited Lifetime warranty that includes 1 year of technical support and return to factory hardware. With a service contract you get 3 years of technical support and next business day Advanced Replacement of the switch if it need to be replaced. I just did a quick google search, and it looks like a contract (part #CON-SBS-SVC2) costs about $50.
So there are a few other things to consider however.
As a frame of reference the average VOIP call uses about 64 - 128 kbps max.
Since you don't have a VLAN capable router or a layer 3 switch, a separate voice VLAN may not be an option. You also mention that the Apple Airport does not do QoS, meaning we will only be prioritizing the voice traffic while it is on the switch. When it is passed off to the Airport to be routed out to the internet all of the QoS settings will be lost, and normal network traffic will get the same priority as voice, since that is all up to the Airport.
With one phone the hassle of getting more equipment and setting up advanced QoS isn't really worth it, especially if the link to the internet isn't going to be participating in QoS.
One last thing I wanted to mention is you are switching to a wireless internet connection. I would ask them how their latency and jitter is, as these two network statistics greatly effect voice quality, and usually wireless performs worse when it comes to voice traffic.
I hope this information helps, if you have any more questions just let me know.
Thank you for choosing Cisco,
Christopher Ebert - Network Support Engineer
Cisco Small Business Support Center -
ASA license for Cisco IP Phone over VPN
Hi,
Are there special licenses required on the ASA to use Cisco IP Phones (Hard phone) over SSL VPN connection?
ThanksHi,
You can purchase the phone proxy license. This elimiates the need to build a VPN tunnel for voice traffic.
It is not mandatory to purchase this license however.
From the ASA configuration guide:
http://www.cisco.com/en/US/docs/security/asa/asa83/configuration/guide/unified_comm_phoneproxy.html#wp1144845
"The Cisco Phone Proxy on the adaptive security appliance bridges IP telephony between the corporate IP telephony network and the Internet in a secure manner by forcing data from remote phones on an untrusted network to be encrypted. "
Don't forget to rate all posts that are helpful. -
RTP streaming and Cisco IP phones problem
Hello,
I'm trying to write an application that should dial some numbers and play the voice message from the file into the phone line using Cisco JTAPI and Java Media Framework.
I've found some samples, that seems useful for me, but unfortunately they does not work. There are no any errors and no exceptions, I have no idea what to do.
Small brief: I make a call from one Cisco IP phone (7960) to another using Cisco JTAPI, then I catch the CiscoRTPInputStartedEv event, get the IP and port of the IP Phone and call the RTPStreamer class constuctor with them. It gives no any errors or exceptions (just a message shown below), but there is only silence in the phone line. Message:
Should b streamin'...
Encoding ok?: true
streams is [Lcom.sun.media.multiplexer.RawBufferMux$RawBufferSourceStream;@53d : 1
sink: setOutputLocator rtp://192.168.1.22:20794/audio
Please see the RTFStreamer class code below.
I set the packet size to 160 as reccomended for Cisco IP phones, I use the greeting.wav from Cisco example that properties are 8Khz 8bit mono, but it still doesn't work.
Could you help me? Thank you for any advice!
import java.io.* ;
import java.util.* ;
import java.net.* ;
import javax.media.* ;
import javax.media.control.* ;
import javax.media.format.* ;
import javax.media.protocol.* ;
import stream.*;
public class RtpStreamer
public static int PlayCounter = 0;
private RtpStreamer()
// not supported
public RtpStreamer(String IP, String Port)
PlayCounter++;
new RtpStreamer("rtp://" + IP + ":" + Port + "/");
public RtpStreamer(String CurrentMediaUrl)
PlayCounter++;
System.out.println("Should b streamin'...");
// Create a Processor for the selected file. Exit if the
// Processor cannot be created.
Processor processor = null;
StateHelper sh = null;
try
String mediaUrl = "file:\\C:\\greetings.wav";
processor = Manager.createProcessor( new MediaLocator(mediaUrl));
sh = new StateHelper(processor);
catch (IOException e)
System.out.println("Exception occured (1a): " + e);
catch (NoProcessorException e)
System.out.println("Exception occured (1b): " + e);
// for loggin purpose
//sh.setContext( getServletContext() );
// configure the processor
if (!sh.configure(10000))
System.out.println("Configuration failed!!");
// Block until the Processor has been configured
TrackControl track[] = processor.getTrackControls();
boolean encodingOk = false;
// Go through the tracks and try to program one of them to
// output ulaw data.
for (int i = 0; i < track.length; i++)
if (!encodingOk && track[i] instanceof FormatControl)
if (((FormatControl)track).setFormat( new AudioFormat(AudioFormat.ULAW_RTP,8000,8,1)) == null)
track[i].setEnabled(false);
else
encodingOk = true;
else
// we could not set this track to ulaw, so disable it
track[i].setEnabled(false);
// set packet size to 160
try
Codec codec[] = new Codec[3];
codec[0] = new com.ibm.media.codec.audio.rc.RCModule();
codec[1] = new com.ibm.media.codec.audio.ulaw.JavaEncoder();
codec[2] = new com.sun.media.codec.audio.ulaw.Packetizer();
((com.sun.media.codec.audio.ulaw.Packetizer)codec[2]).setPacketSize(160);
((TrackControl)track[i]).setCodecChain(codec);
catch (Exception e)
System.out.println("Error setting packet size in 160: " + e + " in " + e.getMessage());
System.out.println("Encoding ok?: " + encodingOk );
// At this point, we have determined where we can send out
// ulaw data or not.
// realize the processor
if (encodingOk)
if (!sh.realize(10000))
System.out.println("Realization failed!!");
// block until realized.
// get the output datasource of the processor and exit
// if we fail
DataSource ds = null;
try
ds = processor.getDataOutput();
catch (NotRealizedError e)
System.out.println("Exception occured(2): "+e);
// hand this datasource to manager for creating an RTP
// datasink.
// our RTP datasink will multicast the audio
try
//String mediaUrl= "rtp://192.168.1.12:20002/audio/1"; // it works without errors
String mediaUrl= CurrentMediaUrl + "audio";
MediaLocator m = new MediaLocator(mediaUrl);
DataSink d = Manager.createDataSink(ds, m);
d.open();
d.start();
catch (Exception e)
System.out.println("Exception occured(3): "+e);BTW is there any solution to figure out if the RTP application makes any network activity or not?
-
Hello friends!
I'm developing an application for Cisco IP Phones that whorks with JMF. Since there are some programmers in this forum that knows Cisco IP phones, I send my question here.
I was send succefully the XML message to phone (CiscoIPPhoneExecute), I was customized the paket's size to 160, etc. All works aparently. But audio stream does not reach to phone. Can anyone tell me why??
Below I write my source code.
Tanks!
Max.
import java.io.*;
import java.net.*;
import java.util.*;
import org.apache.xerces.impl.dv.util.Base64;
import javax.media.*;
import javax.media.control.*;
import javax.media.format.*;
import javax.media.protocol.*;
* @author Administrator
public class MMHTTPPost {
/** Creates a new instance of MMHTTPPost */
public MMHTTPPost() {
try {
* Envia el CiscoIPPhoneExecute al telefono
String xml = new String("<CiscoIPPhoneExecute>"+
"<ExecuteItem Priority=\"0\" URL=\"RTPRx:Stop\"/>"+"<ExecuteItem Priority=\"1\" URL=\"RTPRx:10.1.15.10:23480\"/>"+
"</CiscoIPPhoneExecute>");
String userId = "Max",
password = "12345",
basicAuth = "Basic ",
params = "XML=" + URLEncoder.encode(xml, "ISO-8859-1" );
byte[] bytes = params.getBytes();
// Create a URL pointing to the servlet or CGI script and open an HttpURLConnection on that URL
URL url = new URL( "http://10.1.15.56/CGI/Execute" );
HttpURLConnection con = ( HttpURLConnection ) url.openConnection();
// Indicate that you will be doing input and output, that the method is POST, and that the content
// length is the length of the byte array
con.setDoOutput( true );
con.setDoInput( true );
con.setRequestMethod( "POST" );
con.setRequestProperty( "Content-length", String.valueOf( bytes.length ) );
// Create the Basic Auth Header
Base64 encoder = new Base64();
basicAuth = (String)encoder.encode( ((String)(userId + ":" + password)).getBytes() );
System.out.println("Codificado: " + basicAuth);
con.setRequestProperty( "Authorization", "Basic " + basicAuth.trim() );
// Write the parameters to the URL output stream
OutputStream output = con.getOutputStream();
output.write( bytes );
output.flush();
// Read the response
BufferedReader input = new BufferedReader( new InputStreamReader( con.getInputStream() ) );
while ( true ) {
String line = input.readLine();
if ( line == null )
break;
System.out.println( line );
input.close();
output.close();
con.disconnect();
* Aca empieza la parte RTP/JMF
// Create a Processor for the selected file. Exit if the
// Processor cannot be created.
Processor processor = null;
try {
String mediaUrl = "file:\\C:\\pruebas execute\\spacemusic.au";
processor = Manager.createProcessor( new MediaLocator(mediaUrl));
} catch (IOException e){
System.out.println("Exception occured (1a): " + e);
} catch (NoProcessorException e){
System.out.println("Exception occured (1b): " + e);
// configure the processor
//processor.configure();
//Espera el estado Processor.Configured
boolean result = waitForState(processor, Processor.Configured);
if (result == false)
System.out.println("No se pudo configurar el procesador...");
// Block until the Processor has been configured
TrackControl track[] = processor.getTrackControls();
boolean encodingOk = false;
// Go through the tracks and try to program one of them to
// output ulaw data.
for (int i = 0; i < track.length; i++)
if (!encodingOk && track[i] instanceof FormatControl)
if (((FormatControl)track).setFormat( new AudioFormat(AudioFormat.ULAW_RTP,8000,8,1)) == null)
track[i].setEnabled(false);
else
encodingOk = true;
else
// we could not set this track to ulaw, so disable it
track[i].setEnabled(false);
*Aqu? se cambia el tama?o del paquete a 160
if (((TrackControl)track[i]).isEnabled()) {
try {
System.out.println("Cambiando la lista de codecs...");
Codec codec[] = new Codec[3];
codec[0] = new com.ibm.media.codec.audio.rc.RCModule();
codec[1] = new com.ibm.media.codec.audio.ulaw.JavaEncoder();
//codec[2] = new com.ibm.media.codec.audio.ulaw.Packetizer();
codec[2] = new com.sun.media.codec.audio.ulaw.Packetizer();
((com.sun.media.codec.audio.ulaw.Packetizer)codec[2]).setPacketSize(160);
((TrackControl)track[i]).setCodecChain(codec);
catch (Exception e){
System.out.println("Error al cambiar el tamano del paquete: " + e);
System.out.println("Encoding ok?: " + encodingOk );
// At this point, we have determined where we can send out
// ulaw data or not.
// realize the processor
if (encodingOk)
//processor.realize();
// block until realized.
// get the output datasource of the processor and exit
// if we fail
//Espera el estado Processor.Realized
result = waitForState(processor, Processor.Realized);
if (result == false)
System.out.println("No se pudo realizar el procesador...");
DataSource ds = null;
try
ds = processor.getDataOutput();
catch (NotRealizedError e)
System.out.println("Exception occured(2): "+e);
// hand this datasource to manager for creating an RTP
// datasink.
// our RTP datasink will multicast the audio
try
String mediaUrl= "rtp://10.1.15.56:23480/audio/1";
MediaLocator m = new MediaLocator(mediaUrl);
DataSink d = Manager.createDataSink(ds, m);
d.open();
d.start();
catch (Exception e)
System.out.println("Exception occured(3): "+e);
catch ( MalformedURLException murlex )
System.out.println( murlex );
catch ( IOException ioex )
System.out.println( ioex );
* @param args the command line arguments
public static void main(String[] args) {
new MMHTTPPost();
* Convenience methods to handle processor's state changes.
private Integer stateLock = new Integer(0);
private boolean failed = false;
Integer getStateLock() {
return stateLock;
void setFailed() {
failed = true;
private synchronized boolean waitForState(Processor p, int state) {
p.addControllerListener(new StateListener());
failed = false;
// Call the required method on the processor
if (state == Processor.Configured) {
p.configure();
} else if (state == Processor.Realized) {
p.realize();
// Wait until we get an event that confirms the
// success of the method, or a failure event.
// See StateListener inner class
while (p.getState() < state && !failed) {
synchronized (getStateLock()) {
try {
getStateLock().wait();
} catch (InterruptedException ie) {
return false;
if (failed)
return false;
else
return true;
* Inner Classes
class StateListener implements ControllerListener {
public void controllerUpdate(ControllerEvent ce) {
// If there was an error during configure or
// realize, the processor will be closed
if (ce instanceof ControllerClosedEvent)
setFailed();
// All controller events, send a notification
// to the waiting thread in waitForState method.
if (ce instanceof ControllerEvent) {
synchronized (getStateLock()) {
getStateLock().notifyAll();ignore my last post.. I didn't properly read the source code.
Anyway, I got g.711 streaming to work. I started out using the AVTransmit2 sample, but set the audio output format to what is used in the source code posted here. I've also applied the same codec chain described above, and that was it.
I could never get streaming to work using the first RTP streaming method described in the JMF Programmers guide (and this is the method used in the source posted above), but using the RTPManager, things work just fine. -
Dot1x, .1X and Cisco IP Phones
Hi,
We are busy performing dot1x tests on IP Phones. We chose the LSC approach and have generated CAPF CSRs which we have signed by our PKI infrastructure.
Once all certificates and trust have been uploaded and when we update the CUCM CTL with the Cisco CTL client tool, we received the following error message
“Could not get CAPF certificate(s).CAPF seems to be running on the CUCM Publisher but the certificate file(s) do not exist in the Certifiicate trust path on Server”
We searched Neptro with an explanation on this and found that article:
https://supportforums.cisco.com/thread/2067102
In our setup we one issuing CA in the certification path has n key of 4096 bits. This is imposed by our Security Policy and can’t be workaround from a security policy point of view.
We then had the CAPF CSR regenerated and had a test CA with an encryption key of only 2048 bit sign our certificate and Dot1x authentication. This worked just fine and test Ip Phones can now authenticate..
My question is, is that a known limitation of Cisco Callmanager which is unable to handle certificates signed by a PKI in which one of the CA has a key of more that 2048 bits. Or is this a bug related to our 8.6.2.23900-10 CUCM version.
Is there a way to bypass that limitation or a precise version of callmanager correcting it?
THanks,
AntoineYou can configure the MSFT supplicant to send an EAPOL-Logoff:
Software\Microsoft\EAPOL\Parameters\General\Global\AuthMode -- REG_DWORD
0: Machine authentication mode in Windows XP Client RTM. When a user logs in, if the connection has already been authenticated with Machine credentials, the users credentials are not used for authentication.
1: Machine authentication with re-authentication functionality. Whenever a user logs in, 802.1X authentication is performed using the users-credentials.
2: Machine authentication only Whenever a user logs in, it has no effect on the connection. 802.1X authentication is performed using machine credentials only.
In the wired-Ethernet case you should set (SupplicantMode = 3) AND (AuthMode = 0) AND (disable Machine-Authentication OR ensure that there are no machine credentials on the client). This will ensure that when a user logs off, an EAPOL-Logoff will be sent out. So, AFAIK, this is the bad news .. you lose machine-auth.
Actually, stay tuned for the ability for our IP Phones to be able to do this on behalf of a PC very soon. What will happen is when an IP Phone senses EAPOL through it, it will know who the supplicant is, and what port they're on (the phone's PC port). Assuming 2 conditions above, if link to phone's PC port goes down, IP Phone will transmit EAPOL-Logoff to PC immediately (on PCs behalf).
Hope this helps. -
Using 802.1X and non-Cisco IP Phones
Hi there,
Having some questions about an 802.1x/non-Cisco ip phone setup and was hoping to find some answers/user-experience with this setup.
Main questions i'm facing:
1) When using non-Cisco ip phones (eg Nortel or Siemens) and a previous authorized client connected behind this ip phone gets disconnected. What will this action do with the authorized state of 802.1X on the switch port? WIll it stay authorized until the reauth timer expires or does it reject communication from any other device?
2) What about EAPOL-Logoff messages from the ip phone to the switch. Are these only used by Cisco phones when they experience a link-status change on data ports?
Thanks for sharing your thoughtsOverall, you need to try and deal with the fact that a machine can disappear from the network and the network may not know about it directly (i.e. Link doesn't go down).
I have no idea what other phones do, but Cisco phones send an EAPOL-Logoff when something is unplugged. This lets the switch know directly, and 1X session start is torn down immediately, closing what would be a security hole.
Fundamentally, re-auth is a workaround only, and this is not the reason to enable re-auth to begin with.
If your phone doesn't send an EAPOL-Logoff in this case, the switch might be left thinking an attack is underway when someone else tries to plug in (with presumably a different MAC). You do NOT want this to occur.
Hope this helps,
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