Packets input or output in switch 2960
Hi all,
How to know have packets input or output of one port in switch 2960 Catalyst. Thanks you!
Thank for your answer.
I have some problems involved Cisco Catalyst 2960 Switch.
I am using a device which includes Marvell PHY chip 88E1111. The device can send and receive PTP packet to and from my PC.
Now, I want to connect the device and the PC to Cisco Catalyst 2960 Switch, which will help me trace all of packets in the network . The test scenario is below:
- Switch: Cisco Catalyst 2960
- Tracer: Wireshark software
- PC: Windows 7-64 bit, plugging in Switch port 1 (interface 1)
- Device: FPGA board, plugging in Switch port 2 (interface 2),
operating mode: 1000Mbps, Fullduplex, no auto-negotiation, no auto power efficient-ethernet.
- Interface 2 of the Switch is static set by the device’s MAC
address , which ensures the Switch known the device’s MAC.
I suffered a problem. Although the RJ45 TX status led are on, there is not any packet sent to the Switch. I have no idea in this case.
Could you give me an advise please.
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DAQmxCreateDOChan()
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If you want to be able to switch your digital line between input and output, then you will need to use the Tristate Property. There is another discussion forum with more detailed information about using this property and I believe that you have already seen it. I am going to post the link in case anyone else wants to see the other forum as well: Tristate Property Discussion Forum. Reddog's post is very informative.
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I have maybe simple question, but I can`t really understand how to figure out this problem.
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TCP/IP is not a multiplexed protocol. And why would you need threads or polling to decipher a record-oriented input stream?
Just send your images in packets with a type byte (1=command, 2=image, &c) and a packet length word. At the receiver:
int type = dataInputStream.read();
int length = dataInputStream.readInt();
byte[] buffer = new byte[length];
int count, read = 0;
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}No threads, no polling, and nuthin' up my sleeve.
Modulo bugs. -
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I'm seeing total output drops increment every now and then. We are using 3750E stack switch and are configured for WCCP L2 forward and return. Anyone know why I'm seeing out drops on the WAAS connected interface? The WAAS interfaces are setup as standby. The model is 7371...
interface GigabitEthernet1/0/4
description ****WAAS1 GIG 1/0****
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Hardware is Gigabit Ethernet, address is 0022.be97.9804 (bia 0022.be97.9804)
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Is there any GRE traffic going through this AP?
The workarounds are:
Reboot APs to bring APs back up for time being.
OR
go back to 6.0.188.0 code on WLC.
OR
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It appears that it definitely exists in your code:
12.4(21a)JHA 12.4(21a)JA01 006.000(196.000) -
Packets dropped to Access layer switch???
We have a 6509 running in Native IOS that has 2gb port channels connecting to our 7 access layer switches. About a week ago we were working with Remote span vlans and added a remote span from the 6509 to our other core (6513) which is connected via a 20Gbps portchannel. We began to notice that a lot of people were calling in reporting devices as being slow and we noticed that from the 6509 (Which was the root bridge) we were disgarding millions of packets on the transmit side of our access layer switches. We took out the remote span but it appears that we are still disgarding packets. There are no input or output errors on either side. The Remote span VLAN does not exist on the access layer switch's VLAN database. Does anybody have any idea what we should be looking for?
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create the acl's and apply them
ensure there isn't an active call
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Switch off Mac
Switch back on in Safe Mode (Hit power button, then hold Shift key until a loading bar appears at the bottom of the screen).
Switch off Mac again
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Plug power cord back in and hold down the keys alt, cmd, p, r all at the same time until you hear Starting chime for a second time and hopefully your audio will back. -
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Hello All.
Consider these bits of configuration from my ASA:
ASA Version 9.1(3)
hostname wnsk-asa
xlate per-session deny tcp any4 any4
xlate per-session deny tcp any4 any6
xlate per-session deny tcp any6 any4
xlate per-session deny tcp any6 any6
xlate per-session deny udp any4 any4 eq domain
xlate per-session deny udp any4 any6 eq domain
xlate per-session deny udp any6 any4 eq domain
xlate per-session deny udp any6 any6 eq domain
object network callhost-inside
host 10.3.2.25
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host 209.198.173.58
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network-object 175.77.48.96 255.255.255.224
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nat (server-lan,itrunk) dynamic WNSK-POOL
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nat (server-lan,itrunk) static callhost-outside
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When I check my setup with packet input, I get this:
wnsk-asa# packet input itrunk tcp 209.198.187.78 22222 10.3.2.25 3389
Phase: 1
Type: ROUTE-LOOKUP
Subtype: input
Result: ALLOW
Config:
Additional Information:
in 10.3.2.0 255.255.255.0 server-lan
Phase: 2
Type: ACCESS-LIST
Subtype: log
Result: ALLOW
Config:
access-group inbound12 in interface itrunk
access-list inbound12 extended permit tcp object-group EQUINOX host 10.3.2.25 eq 3389
object-group network EQUINOX
network-object host 175.146.14.236
network-object 175.77.48.96 255.255.255.224
network-object 209.198.187.0 255.255.255.0
Additional Information:
Phase: 3
Type: NAT
Subtype: per-session
Result: ALLOW
Config:
Additional Information:
Phase: 4
Type: IP-OPTIONS
Subtype:
Result: ALLOW
Config:
Additional Information:
Phase: 5
Type: VPN
Subtype: ipsec-tunnel-flow
Result: ALLOW
Config:
Additional Information:
Phase: 6
Type: NAT
Subtype: rpf-check
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object network callhost-inside
nat (server-lan,itrunk) static callhost-outside
Additional Information:
Result:
input-interface: itrunk
input-status: up
input-line-status: up
output-interface: server-lan
output-status: up
output-line-status: up
Action: drop
Drop-reason: (acl-drop) Flow is denied by configured rule
When I actually get on the host at 209.198.187.78 and attempt to connect to port 3389 of 209.198.173.58, it works. Packet input says it will not work. What am I getting wrong, or is the ASA tricking me?
ERMIn your packet-tracer string you direct the ASA to tell you about reachability of "10.3.2.25 3389". In your text you mention being able to get to "port 3389 of 209.198.173.58".
Which of those two are you trying to figure out? -
Analog Levels vs SPDIF Levels Input and Output in Logic Pro
Hello,
I ran a test last night for recording input and output levels from my Yamaha Motif XS8 through an Apogee Ensemble to compare Analog to SPDIF
I connected two TRS cables from the L and R outputs on the Motif XS into the Analog Inputs on the Apogee and also have the SPDIF connection from the Motif to the Apogee.
I put the master fader all the way up on the XS for the volume for analog.
The ensemble in Maestro has a +4 and -10 reference notional level option for analog inputs. i had it at +4 but changed to -10 and the analog got louder (i figured it would get louder for +10, confusing).
anyways, why is it that I can record louder levels for analog than the digital transfer?
I tracked both options at the same time then recorded vocals over it. the digital sound is too low. what's up with that?
and when I tried to bounce the recording to listen to it in ITunes, the volume levels were way lower than my cd playing through iTunes. Please enligthen me on these?
I use to record in a Roland 2480, and had similar results with loudness, but got a little louder through mastering but still....pro cds are way louder and still clear.I think there is much confusion here!
In summary, you wont be able to control the recording level of S/PDIF.
The reason is that you don't want to!
You need to think of the SPDIF connection as being more like a file transfer method. You are copying the digital data at an output to your harddrive in effect. If I send you an MP3 via email you'd never imagine that your email software is capable of changing the gain of the MP3 I send you. This might sound daft but its a useful analogy. If you need to increase the "volume" of that MP3 then you'd need to ask the sender. Its the same with your set-up.
There could well be somewhere on your synth that adjusts the instruments level, other than the master (analogue) output control. For example, make sure the midi volume of the intrument being played is set to full - ie midi vol 128. Perhaps there is somesort of virtual mixer onboard to control all the muti-timbral parts so make sure your part has its virtual fader turned up.
This is what (basically) is going on in the chain...
Your synth creates sound in its "digital brain". This sound is sent to an "output stage" which will distribute the sound to various outputs. In the case of the S/PDIF it will just send the raw digital data untouched. For the analogue side the digital signal (same as the one sent to S/PDIF) will be converted to analogue and then sent to a amplifier to get it to an appropriate "line level". This final level could well be controlled by an anogue volume control which could be adding more gain (than you think) too.
When things go to your sound card/ daw...
The purpose of a analogue gain control is to set the i/p signal so that it suitably loud to beat any noise that exists in your input circuits - so that a good signal to noise ratio is achieved. Analogue signals need to work in the right loudness zone (so to speak) as the analogue electronics will be designed to handle signal levels of a particular range. the gain control is there to make sure the signal is in that range.
Digital signals are far more predictable though and there is no advantage to your recordings if the incoming digital signal gets an increase of level at the input stage. All you are doing here is effectively adding a few zeros to the binary digital data!
Lets face it the point of recording is to get a copy of the original sound, that is as similar to the original as possible. With S/PDIF you get a perfect copy of what's coming out of your synth - so job's a good un!
If, when you come to mix in logic, you find the level of the digital recording is indeed too low for mixing/mastering purposes then just boost it in logic via a fader or via the gain plugin.
Those referrence values of -10dBV and +4dBu refer to analogue voltage levels only. they have nothing to do with the digital domain. The -10/+4 switch will be only relevant to analogue inputs and outputs. Using an analogue VU meter you should find that a sine wave that peaks at 0dBVU (totally different to 0dBFS BTW) is the equivant of a digital sine wave peaking at -18dBFS.
The analogue headroom (how loud you can go before things distort) depends on the analogue electronics and varies with different design. Analogue stuff, like mixers) often has headroom of 24dB or more. So that digital stuff can interface with analogue properly we allow for that analogue headroom to be around 18dB (usually enough in practise!)... hence -18dBFS(digital)=0dBVU(analogue).
To make your digital and analogue input signals sound similar in level you will probably have to reduce the gain of the analogue input. If you set the incoming analogue signal to peak around -14dB (or less!) or so you will probably find things more equal. If you are working in 24 bit your analogue levels can be seemingly very low before sound quality is affected. Its quite safe to record at -20 or even -30dB as shown on logic's meters for eg.
I hope all this waffle helps LOL! -
Mirroring input to output on NI usb-6259 (with or without DSP)
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I have tried looking through and modifying the example vi's (e.g. Multi-Function-Synch AI-AO.vi ) with little luck. It seems that the best I can do is to generate a square/triangle/sine wave at a frequency that is governed by my external clock trigger... No data echoing for anything complex.
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I've tried constructing this setup in all the "intuitive" ways I can think of, and have sort of struck out.
Please help?Hello Godzilla,
In order to do synchronized digital input and output you will need to route your external trigger through a "dummy" analog input task. The reason for this is that the digital I/O tasks do not support start triggers. By using the analog input sample clock and starting the digital tasks before the analog input they will be started at the same time and thus be synchronized.
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At this point you'll have a program that uses your external sample clock and you have the ability to add a digital output task that will be synchronized with your input. To add this digital output task I would suggest the Cont Write Dig Port-Ext Clk-Non Regeneration.vi example. If you copy this example into the other one and merge the two while loops you'll only have a few modifications to make.
Here are some things you'll need to change:
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Non-regeneration keeps the digital output from looping previous data. This will allow you to know if you are buffering output data fast enough. If you have trouble running this loop with your analysis then I would recommend implementing data pipelining. For an overview of what pipelining is please see this article: http://zone.ni.com/devzone/cda/tut/p/id/6425
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I hope this helps get you started, and if you'd like to talk with someone at National Instruments about Real-Time control loops feel free to contact us directly using the information at www.ni.com/contact.
Have a great day!
Brooks -
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I did a search of course, and came up with something about an aggregate. I have no idea what this is, how to do it, or if it would even work for me.
What I am trying to do is:
1) Record into Logic Express using my Tascam US-122.
2) Have playback come out of my computer sound system, not the Tascam.
If I go over to the Audio setup window, I can only record when the driver is set to Tascam US-122. Likewise, I can only listen to sound when my Built-In Audio is selected. It gets rather annoying going between the two.
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i need your help,I'm working on this issue since yesterday and have no idea how to deal with it.
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AS
BEGIN
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User::Var2Return output numeric 1 -1
but i alwas get the error
"Multiple-step OLE DB operation generated errors. Check each OLE DB status value, if available. No work was done."
Has anybody an idea how I can solve this problem?
Thanks a lot.
Kind Regards,
AliceHi Alain,
thx for your answer.
I have around 15 procedures called one after the other to calculated and modify my values. Each procedure is responsible for an other but overlapping set of variables. So i thought it would be a good idea to call them one after the other with the needed variables via a execute sql task.
So if i use a result set, how i get my stored procedure to return 10 values? I would have to use a Function instead of a procedure, wouldn't i?
As if i have 15 procedures this would be a lot of work.
But thanks a lot for the idea. I think an other idea would be to create one function which calls all stored procedures and returns all the calculated values as a result set, wouldn't it?.
Kind Regards.
Alice -
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Is there some way to make Apple Earphones be the Input and Output for Audio in the iMac
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Any help is appreciatedyou can connect the headset to something like this
http://www.ebay.com/itm/NEW-MaelineA-3-5mm-Female-to-2-Male-Gold-Plated-Headphon e-Mic-Audio-Y-Splitter-/381100440348
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Hello, I was wondering how to set-up my Mac Book so that I can use my bluetooth cellphone for audio in and audio out. Ideally, I would like to use my handset for talking on iChat and Skype. Has anyone had any luck doing this? When I bond my cellphone these optiosn do not appear. I know that this is easy to do on a Windows machine but I am not sure about how to do it on my Mac Book.
Thanks!Yes it should do what you want.
Go to "Audio Midi Setup", and go to the Audio menu and click "Open Aggregate Device Editor". The interface is pretty simple but if you do get stuck, just use the help function in Audio Midi Setup, as it has a step by step guide.
Then when you return to logic, go to the Preferences>Audio>Drivers section and select Aggregate Device as the new driver rather than either the built in sound or the tascam. Then the inputs and outputs will apply to BOTH devices. -
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Hello,
i am trying to write a new Xcontrol Element. In the data model I can create data types using the cluster to create compound types, eg an int and an int array. But how do I create an xcontrol which has multiple data inputs and outputs?
Kind RegardsLimping_Twerp wrote:
Alright: I see: An xcontrol is either an input OR an output. How do I achieve an output? Secondly: So you are saying the only Elements that can have multiple inputs and outputs are VIs?
Can you take a few steps back and explain what you are actually trying to do. SubVIs and Xcontrols have nothing in common and it is not clear why you even try to compare them side by side (e.g. in terms of the number of connectors).
Your questions about input or output tell us that you seems to have some misconceptions about xcontrols. Xcontrols are like regular controls, except they have some built-in intelligence that you can program. Most front panel object can be either controls or indicators and the same is true for Xcontrol. You create an Xcontrol, and after placing it on the front panel you can decide if it should be a control (where the code reads the value) or an indicator (were the code writes values to it). When you define the xcontrol facade, you also need to program how the visuals change if it is changed from control to indicator or vice versa.
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