Sip signaling with supervised transfer possible?

I'm doing a supervised transfer (call replacement), between two established calls that my UCMA app have control of. But after the transfer has been initiated and succesfully ended, I'm getting a Terminating > Terminated state change event of the call
I used BeginTransfer on, and no more notifications.
The problem with that is, that the 2 endpoints are now talking together, and I ain't getting any message when they hang up that call, which means I don't know if they are available or not.
Is it possible to get notifications of call states after call replacement?? I can live with tha I ain't in control of the call anymore, but its mandatory that I know state changes after transferred.

If I understand what you're saying, then no. Once you've transferred the calls so that the endpoints are talking to each other, your UCMA application isn't involved in the call anymore, and therefore won't get any notifications. Unfortunately there's no
(easy*) way to monitor state changes of a call you're not part of.
If it's really important to continue to receive notifications, you could create a conference with those two endpoints in it (so they can talk to each other), but your app can stay in the call, monitoring state changes.
Hope that helps.
-tom
*OK OK, it technically IS possible - you could use MSPL and listen for state changes, and then match that up with the conversation IDs of the calls you transferred. It's a lot of effort though.
Tom Morgan Developing Lync

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With supervised ipads, what is the best way to transfer photos and videos from the ipad to the PC? Wireless transfer is not an option in our circumstances

With supervised ipads, what is the best way to transfer photos and videos to the PC? Wireless transfer is not an option in our circumstances.

There are only two ways of transfering things on the iPad.
- via WiFi
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From what I have read, you need to connect to iTunes on the supervising PC.
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Is it possible to condition a load cell signal with a 1520 SCXI and then use it as the primary analog feedback for a 735X motion control board?

My system
PXI 1052 chassis
PXI 8196 RT controller
Flexmotion 7356 motion controller
Flexmotion 7354 motion controller
6259 M series DAQ
1520 SCXI
1520 SCXI
1314 SCXI frount mounting terminal block
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The problem:
The motion controller boards (7356 and 7354) can be configured for force feedback control via analog feedback from a load cell. However the boards implicitly expect the load cell signal to be amplified and conditioned with and external signal conditioner rather then using an NI SCXI module. Currently I’m am reading the SCXI signals via the 6259 M seriers DAQ and consequentially have no way to route the signal into the 735X boards for force feedback control. I can use softmotion to control the motors but then I loose the fast PID update rates possible with the 735X boards.
The question:
Is it possible to condition a load cell signal with a 1520 SCXI and then use it as the primary analog feedback for a 735X motion control board?
Thanks
Patrick Aubin
Ph.D Candidate
University of Washington &
VA MedicalCenter of Excellence for
Limb Loss Prevention and Prosthetic Engineering

Hi GG,
Thank you for your feedback.
>create your own external circuitry to amplify the signal.>
True a few op-amps could amplify and filter the signal externally but the performance of such an amateur built system will never match the accuracy and functionality of the 1520 SCXI. That’s why we spent the $2500 for the professional version.
>One thing that you could do is to use a UMI with your motion controller and then wire your signal from your SCXI to the UMI.>
As I understand it the UMI is basically a break out connectivity interface for third party drives. After conditioning and amplifying the analog force signal with the SCXI module the analog signal isn’t available to wire into a UMI. To my knowledge the conditioned signal of the SCXI resides only in the PXI backplane (i.e. there is no physical access to the conditioned analog signal). Can you explain how the UMI could be help here.
Again thanks for your input GG.
Patrick

Is it possible to send and recieve audio signal with AI / AO in NI-6229 which is DAQ board.

I'd like to test MP3 Player.
The procedure is below.
1. Send a audio signal of 1kHz from mp3 file in PC through AO in NI-6229.
2. Record it with MP3 player.
3. Play a recorded audio signal with MP3 player.
4. Recieve the signal through AI in NI-6229 and measure frequency and voltage of the signal.
5. Compare the signal with original signal.
Do you think it is possible?

Hello Blue Sky80,
It is possible to measure audio signals using LabVIEW and Sound an Vibration toolset. Please take a look at this Developer Zone article on Audio measurements using LabVIEW to get more information on this subject.I hope this helps!
Regards,
Ima
Applications Engineer
National Instruments
LabVIEW Introduction Course - Six Hours
Getting Started with NI-DAQmx

Is it possible to detect low frequency signals with a high sampling rate?

Hello everyone,
I'm having an issue detecting low frequency signals with a high sampling rate. Shouldn't I be able to detect the frequencies as long as the sampling rate is at least 2 times the highest frequency I will measure? The frequency range I am measuring is 5-25 Hz, and I use Extract Single Tone.vi to measure the frequency. The sampling setting I am using is 2 samples at 10 kHz. Is there a method I can use to make this work?
Attached is the vi.
Attachments:
frequencytest.vi ‏21 KB

You are sampling at 10Ks/S, but only taking two samples. What do you expect to see? If your signals are binary (On or Off) you would only see either an on or an off, or if the rise/fall time was fast and you were Extremely lucky, one of each. If you want to see a waveform you have to sample for at least the period of a waveform. So you should take samples for at least 0.2 seconds to capture an entire waveform at 5Hz, ideally longer. Think of looking at a tide change at a dock. If you want to see the entire tide change you will probably have to measure repeatedly over 24 hours, not just run out on the dock, measure the height twice and leave. That wouldn't tell you anything other than at that precise moment the tide height was X, but not that it was at high tide, low tide, in between, etc.
I type too slowly, I see that a more technical answer has been given, so mine will be the philosophical one!
Putnam
Certified LabVIEW Developer
Senior Test Engineer
Currently using LV 6.1-LabVIEW 2012, RT8.5
LabVIEW Champion

QOS configuration - Is there a way to prevent marking SIP signaling as media

I have configured QOS for Lync 2013 per all the documenation, blogs, etc.
While reviewing network captures to make sure it is working, I noticed something: normal SIP signaling traffic from the Lync servers to port 5067 uses random source ports, and when those source ports fall into the ranges defined for media, gets marked
as if it was media traffic. This is over-prioritizing: marking SIP signaling traffic with a higher DSCP than is warranted.
Any way to prevent this?

The two main methods are marking source ports or marking anything coming from the service executables with a DSCP value. Sometimes SIP signaling traffic will fall into the source range, and it will always come from the service executable. You
can't force only signaling to use a set range source ports, so there's not too much you can do. That being said, SIP signaling traffic is generally negligible compared to the media traffic so I generally just let it be.
Please remember, if you see a post that helped you please click "Vote As Helpful" and if it answered your question please click "Mark As Answer".
SWC Unified Communications

How to send a ttl signal with a good exactitude less than 10ms?

i have a pci 6013 (8 input channel) and pci 6711 (4 outpout channel).Then i need to send a ttl signal to a camera ccd to take a photo when a waveform sinus is equal at zero (rising), but when i send this TTl signal with dasylab using generator pulse or combi trigger and readind this signal, i have difference between dasylab pulse(that is exactly) and measurement pulse that is approximate 40ms after.But i need to have less than 10ms is it possible?
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Yes, it's possible.
You just have to reduce your block size to 10 !
For more informations, visit Forum Dasylab en français at http://forum.aceboard.net/?login=128657
Frédéric Miqueau
MD Electronic, France
http://www.mdelectronic.fr
French Dasylab Forum : http://www.dasylab-forum.de/forum/index.php?showforum=43

How to measure a current signal with NI 6211?

How to measure a current signal with NI 6211?

Hi Clecio,
This is a charge accelerometer. It is different from the IEPE accelerometers that outputs a voltage signal.
You need a charge amplifier. Then according to the charge amplifier output we can select an appropriate module to measure the signal.
It is not possible to measure charge accelerometers with NI 9234 directly.
Best Regards,
Rodrigo Schneiater
Application Engineering
National Instruments Brazil
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Boosting o2 Wifi signal with Airport Express

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I had read that one could boost our o2 Wifi signal with the Airport Express. I don't seem to be able to do it! Can anyone help.
Detsky

It is possible if the wireless network is created by another AirPort Express (AX), an AirPort Extreme base station (AEBS), or a Time Capsule. Otherwise it is not possible.

Better estimation of phase difference between two signals with variable frequency!

Hello LabView Gurus,
Being a power engineer and having just a little knowledge of signal processing and labview, I have been pulling my hair out for the last couple of days to get a better estimation of phase difference between two signals.
We have two analog voltage signals; 1. sine wave (50Hz ± 1Hz) and 2. a square wave with exactly half of sine wave frequency at any time.
At the starting point of operation (and simulation/acquisition) both signals will have no phase difference. However, the square wave's frequency changes unpredictably for a just a few millisecond but then it gets synchronized with sine wave's frequency again. This means that the square wave will be phased out from its original position. The task of the labview is to find the phase difference between the two signals continuously.
My approach to determine the phase difference is to measure the time when sine wave crosses zero amplitude and the time when the very next square wave changes amplitude from zero volts to +ve voltage (I have a 0.5volts threshold just to avoid any dramas from small line noise). The difference between these times is then divided by the time period and multiplied by 360 to get this phase difference in angles.
As this part is just a small block of a big project, I can only allow 5000Hz sampling rate each for both signals. I read 500 samples (which means I read data from 5 cycles of sine wave and 2.5 cycles of square wave).
Now the problem is, as long as the frequency of sine wave stays constant at exactly 50Hz, I get a good estimation of the phase difference but when the frequency changes even a little (and it will happen in the real scenario i.e 50Hz ± 1Hz and the square wave's frequency is dependent of sine wave's frequency), the estimation error increases.
I have attached my labview program. From front panel, you can set the phase of square wave (between -180 and 0) and you should see the labview's calculated phase in the indicator box named 'Phase'. Then you can press 'Real Frequency' switch that would cause the frequency to change like it would in real operation.
You can observe that the estimation error increases after you push the button.
All I need to do is to reduce this estimation error and make it as close to the actual phase difference as possible. Any help would be greatly appreciated.
I am using LabView 2009 for this task.
The application is for electric machines and the stability/performance of machines under different faults.
Thank you for reading this far!
Regards,
Awais
Attachments:
v603.png ‏320 KB
v603.vi ‏186 KB

Jeff Bohrer wrote:
Basic math gives me a bit of pause on this approach. You are sampling at 50 times the frequency of interest so you get 50 samples per cycle. your phase resolution is 1/50th cycle or 7.2 degrees +/- noise. You will need to samlpe faster to reduce phase resolution or average multiple readings (at a time cost that is signifigant)
Jeff- (Hardly Working)
I am sampling at 100 times the sine wave's frequency and 200 times the square wave's frequency. Increasing the sampling rate completely solves my problem. But since I am acquiring several other inputs, I cannot afford a sampling rate higher than 5kHz.
F. Schubert wrote:
I'm not a signal processing expert, but here my basic understanding.
If you simulate sampling with 5kHz and a frequency of 50 Hz (and both are 'sync' by design), you always get an exact 5 periods. Any variation of your signals frequency gives you a propability to get 4 or 6 'trigger' events. That's an up or down of 20%!
The one measure to reduce such problems is using 'window functions'. They don't fit your current approach (counting instead of a DSP algorithm), so this needs to be reworked as well.
My approach would be to use the concept of a Locki-In amplifier. You need to phaseshift your ref-signal by 90°. Then multiply your measurement signal with the ref signal and the phase shifted ref signal. The obtained values for x/y coordinates of a complex number. Calculate the theta of the complex number (with the LV prim). Feed this in a low pass filter.
The trick on this is, that the square wave has harmonics in it, in this you are interested in the second harmonic which is the sine wave.
To get rid of the effect that the sync between sampling rate and ref signal frequency gives an error, you then can use the window I mentioned above (place it before the lock-in).
For a design that really plays well, use a producer-consumer design pattern to get the calculations done in parallel with the DAQ.
I suggest you to check on wikipedia for some of the keywords I mentioned. Go also for the external links which lead to great tutorials and AppNotes on the signal processing basics.
Sorry, it's not a simple solution I offer and we will have quite some conversation on this forum if you follow this path. Maybe someone else knows a simpler way.
Felix
www.aescusoft.de
My latest community nugget on producer/consumer design
My current blog: A journey through uml
An interesting view. the sine wave can indeed be looked as a second harmonic of the square wave. I will implement your idea and get back to you as soon as I get some results. But since I have very limited knowledge of signal processing, it might take me a while to get my hear around the solution you mentioned.

Error in Unity Supervise Transfer

Hi all,
Currently, I am trying to enable call queuing on a subscriber account through supervise transfer but cannot make it work in Spanish. At the beginning, I considered this as a misconfiguration issue but turning the language into English made it work. The fact is that Unity tries to transfer the call and prompts this well, "wait while I transfer your call", if the call is busy, we would expect, "extension YYY is on the phone now, if you would like to hold press 1, to live a message press 2,..." (that works perfectly in ENU language). However, in Spanish prompts "The system cannot complete the call in this moment, try again later" after trying to transfer the call.
I am working with CCM6.0 and Unity 4.2 and have searched some similar TAC case but had no luck.
When the failure happens, the same 2 errors appears in Event Viewer,
EventID: 531
Cisco Unity's telephony component has encountered a serious error.
EXPLANATION:
A serious failure has occurred on port 1 while trying to open a wav file or stream. Depending upon the severity of the failure, parties on this call could experience unexpected behavior or be disconnected. In some cases, further calls on this port will not be handled correctly.
TECHNICAL DETAILS:
Thread 0x00001138 had a failure on port 1 in method CAvMiuWave::GetLength()
DESCRIPTION: AvWav WavOpenEx2 failed with 0x00000000.
DETAILS:
FileName: ForceToHoldExt.
CALLINFO:
CallerID: 800
CalledID: 716
RedirectingID: 716
Origin: Internal
Reason: FwdBusy
Trunk: 0.
CALL SEQUENCE:
[11:20:46:984 - 0x0000087C] Drop() - S_OK
[11:22:39:250 - 0x000017DC] LINE_APPNEWCALL(0x00010097)
[11:22:39:250 - 0x000017DC] SetEvent(NewCall)
[11:22:39:250 - 0x000017DC] SetEvent(CallInfoArrived)
[11:22:39:296 - 0x00001138] Answer() entered
[11:22:39:296 - 0x00001138] lineAnswer(0x00010097) - 0x000100AB
[11:22:39:296 - 0x000017DC] LINE_REPLY(0x000100AB | 0x00000000)
[11:22:39:421 - 0x000017DC] LINECALLSTATE_CONNECTED(0x00010097 | Active)
[11:22:39:421 - 0x00001138] WaitFor(0x000100AB | CONNECTED) - CONNECTED
[11:22:39:421 - 0x00001138] Answer() - S_MIU_CONNECTED
[11:22:39:421 - 0x00001138] Answer() entered
[11:22:39:421 - 0x00001138] Answer() - S_MIU_CONNECTED
[11:22:39:468 - 0x00001138] GatherDigits(FLUSH) entered
[11:22:39:468 - 0x00001138] GatherDigits(FLUSH - <>) - S_OK
[11:22:39:500 - 0x00001138] Answer() entered
[11:22:39:500 - 0x00001138] Answer() - S_MIU_CONNECTED
[11:22:39:500 - 0x00001138] Play() entered
[11:22:39:500 - 0x00001138] WavOpen(AvPHGreetESP005.wav) - 0x00000C87
[11:22:39:500 - 0x00001138] WavGetLength(0x00000C87) - 2742
[11:22:39:500 - 0x00001138] WavClose(0x00000C87) - 0x00000000
[11:22:39:500 - 0x00001138] WavOpen(AvPHGreetESP005.wav) - 0x00000C88
[11:22:39:500 - 0x00001138] WavSetChunks(0x00000C88) - 0x00000000
[11:22:39:500 - 0x00001138] WavSetPosition(0x00000C88 | 0) - 0
[11:22:39:500 - 0x00001138] WavSetVolume(0x00000C88 | 50) - 0x00000000
[11:22:39:500 - 0x00001138] WavSetSpeed(0x00000C88 | 100) - 0x00000000
[11:22:39:500 - 0x00001138] WavPlay(0x00000C88) - 0x00000000
[11:22:42:265 - 0x00000E28] SetEvent(WavStopped)
[11:22:42:265 - 0x00001138] WaitFor(StopEvents) - WavStopped
[11:22:42:265 - 0x00001138] WavGetLength(0x00000C88) - 2742
[11:22:42:265 - 0x00001138] WavClose(0x00000C88) - 0x00000000
[11:22:42:265 - 0x00001138] Play(Item 1 of 1 | AvPHGreetESP005.wav) - 0 to 2742
[11:22:42:265 - 0x00001138] Play(1 Item | PlayLength 2742) - S_OK
[11:22:42:265 - 0x00001138] GatherDigits(FLUSH) entered
[11:22:42:265 - 0x00001138] GatherDigits(FLUSH - <>) - S_OK
[11:22:42:265 - 0x00001138] Transfer(WAITCONNECT to 717) entered
[11:22:42:265 - 0x00001138] lineSetupTransfer(0x00010097) - 0x00010050
[11:22:42:265 - 0x000017DC] LINE_REPLY(0x00010050 | 0x00000000)
[11:22:42:281 - 0x000017DC] LINECALLSTATE_ONHOLDPENDTRANSFER(0x00010097)
[11:22:42:281 - 0x00001138] WaitFor(0x00010050 | ONHOLD) - ONHOLD
[11:22:42:281 - 0x00001138] lineDial(0x000100CF | 717) - 0x000100D7
[11:22:42:875 - 0x000017DC] LINE_REPLY(0x000100D7 | 0x00000000)
[11:22:42:875 - 0x000017DC] SetEvent(CallInfoArrived)
[11:22:42:890 - 0x000017DC] SetEvent(CallInfoArrived)
[11:22:42:890 - 0x000017DC] LINECALLSTATE_DISCONNECTED(0x000100CF | Busy)
[11:22:42:890 - 0x00001138] WaitFor(0x000100D7 | CONNECTED) - BUSY
[11:22:42:890 - 0x00001138] Transfer(WAITCONNECT to 717) - E_MIU_DEST_BUSY
[11:22:42:890 - 0x00001138] TransferComplete(REJECTED) entered
[11:22:42:890 - 0x00001138] lineDrop(0x000100CF) - 0x0001006F
[11:22:42:890 - 0x000017DC] LINE_REPLY(0x0001006F | 0x00000000)
[11:22:42:890 - 0x000017DC] LINECALLSTATE_IDLE(0x000100CF)
[11:22:42:890 - 0x000017DC] StopIO(TERM_CONSULT_DISCONNECT) entered
[11:22:42:890 - 0x000017DC] StopIO(TERM_CONSULT_DISCONNECT) - S_OK
[11:22:42:890 - 0x00001138] WaitFor(0x0001006F | IDLE) - IDLE
[11:22:42:890 - 0x00001138] lineUnhold(0x00010097) - 0x000100B5
[11:22:42:890 - 0x000017DC] LINE_REPLY(0x000100B5 | 0x00000000)
[11:22:42:953 - 0x000017DC] LINECALLSTATE_CONNECTED(0x00010097 | Active)
[11:22:42:953 - 0x00001138] WaitFor(0x000100B5 | CONNECTED) - CONNECTED
[11:22:42:953 - 0x00001138] lineDeallocateCall(0x000100CF) - 0x00000000
[11:22:42:953 - 0x00001138] TransferComplete(REJECTED) - S_OK
[11:22:42:953 - 0x00001138] GatherDigits(FLUSH) entered
[11:22:42:953 - 0x00001138] GatherDigits(FLUSH - <>) - S_OK
[11:22:42:953 - 0x00001138] Play() entered
[11:22:42:953 - 0x00001138] WavOpen(AvPHGreetESP021.wav) - 0x00000C89
[11:22:42:953 - 0x00001138] WavGetLength(0x00000C89) - 1003
[11:22:42:953 - 0x00001138] WavClose(0x00000C89) - 0x00000000
[11:22:42:953 - 0x00001138] WavOpen(AvCommonESP207.wav) - 0x00000C8A
[11:22:42:953 - 0x00001138] WavGetLength(0x00000C8A) - 720
[11:22:42:953 - 0x00001138] WavClose(0x00000C8A) - 0x00000000
[11:22:42:953 - 0x00001138] WavOpen(AvCommonESP201.wav) - 0x00000C8B
[11:22:42:953 - 0x00001138] WavGetLength(0x00000C8B) - 582
[11:22:42:953 - 0x00001138] WavClose(0x00000C8B) - 0x00000000
[11:22:42:953 - 0x00001138] WavOpen(AvCommonESP053.wav) - 0x00000C8C
[11:22:42:953 - 0x00001138] WavGetLength(0x00000C8C) - 806
[11:22:42:953 - 0x00001138] WavClose(0x00000C8C) - 0x00000000
[11:22:42:953 - 0x00001138] WavOpen(ForceToHoldExt) - 0x00000000.
EventID: 807
Cisco Unity's multi-media component has enountered a serious error.
EXPLANATION:
A serious failure has occurred on port 1 while opening an audio stream. Depending upon the severity of the failure, parties on this call could experience unexpected behavior or be disconnected. In some cases, further calls on this port will not be handled correctly.
TECHNICAL DETAILS:
Thread 0x00001138 had a Failure on Port 1 in AvWav
DESCRIPTION:
File: E:\Views\CU4.2.0.110\un_Miu\UnityAvWav\WAV.C(1094)
Method: WavOpenEx
Failure: call to mmioOpen(ForceToHoldExt) failed(101).
I have checked the spanish wav files involved and are ok. I can play all of them locally.
On the other hand, I attach an extract of a microtrace displaying the error.
I dont know what I am missing and Its a bit frustating.I will really appreciate your help.
Thanks in advanced.

it_lines TYPE STANDARD TABLE OF tline WITH HEADER LINE.
insfile TYPE text ,

Poor signal WITH Network extender.

Hoping to get some help here so i dont have to call verizon.
Im just recieved my verizon network extender, the SCS-2u01 model.
I installed it, waited for it to boot and and restricted it to just my wife and I phone numbers.
We are for sure connected to it, *48 confirms and get the beeps when placing calls, but im getting mostly 1 bar of service within 30feet and literally even holding the phone right up against the thing gets usually 2 bars, sometimes 3, never 4.
We just switched from sprint and with sprint we had and airave which appears to be 100% identical device, setup identical and would would get full signal
with in 40 feet(all downstairs and even outside) and 2-3 bars upstairs and way out in the yard.
Ideas?The only thing i can think of is it it 2 feet from our wireless router, although no different then the airave.

I recently started using the Sprint Airave so I should be able to contribute to this.
Network extenders only have a few pieces to them. They have a broadband connection port which needs to be hooked up to your modem, some ethernet ports that hook up to your router/computer, an antenna for broadcasting the wireless signal and a GPS antenna for 911 emergency situations when you cannot place it close to a window.
If your signal is not strong inside your house then normal wireless troubleshooting steps should apply. I'd suggest starting with a site survey and moving your NE around the house until you can find a spot that performs better. Try locations close to windows and away from anything metal/electric if possible. There might be electrical equiptment or competing wireless signals that are interfering with your reception in its current location. If your NE is still not performing after relocating it, then contact VZW for support and a possible replacement device.
Site Survey walkthrough:
http://3gstore.com/page/13_will_an_antenna_help_me.html
Verizon lets you manage your NE online through the My Verizon account. Once you sign into the account you should be able to make any additional config changes that you want. I'd look into changing the wireless channels or broadcast mode to see if there is an improvement.
More details available here:
http://www.verizonwireless.com/b2c/store/accessory?action=gotoSetup

Measuring DC signal with ac coupling

I'm TOTALLY new to Labview. I was trying to use NI DAQ 4472 for measuring DC signal with large DC offset. I would like to see an example that sets AC coupling for filtering the unwanted DC offset. I found an example with "DAQmx Channel Property", but I need help in connecting it with the block diagram. I have attached my block diag. and and help in connecting the DAQmx Channel Property Node is highly appreciated.
Thanks.
Attachments:
Sensor_Configuration_Test.vi ‏60 KB

Good afternoon MiamiVipul,
Thanks for contacting National Instruments with your issue, we'll try and get you up and running as quickly and efficiently as possible.
Look like you've got the right approach to your problem.
In order to integrate a property node into your code, you simply have to wire the task reference and error cluster in line with the rest of your code. I.e. after you create the task, you will have the purple task reference out as well as the pink error cluster out, wire those to the corresponding intputs on the property channel node and then wire the task and error out of the property node into the DAQmx Create Channel vi.
Have a look at the examples by going to Help > Find Examples (Browse Tab) > Hardware Input/Output > DAQmx > Analog Measurements. There are a variety of different measurement types which you can start from to give you a strong idea of the general DAQmx coding principles. Several of these also use Property Nodes. For example Acq&Graph Voltage - Int Clk-Analog Start w Hyst.vi
You mentioned you want to measure "DC signal with large DC offset", which I'm assuming is a typo, as AC coupling is of course designed to remove the DC offset so you can measure the AC signal.
Best of luck with your project.
Sincerely,
Minh Tran
Applications Engineering
National Instruments

Read encoder signal with 9201

Hi
I have a quite uncommon problem. I need to acquire the encoder signal with a NI-9201 (analogue module).
Is this even possible? Of course I saw I that I should use a NI-9401 but my chassis is full...
I already tryed to acquire impulse signals with such a module but they were slower.
Thanks for any help.
Yves

Hi Yves,
when you acquire data from an encoder, do you mean a quadrature encoder or rotary encoder?
This encoder generates a binary or digital signal.
Of course, you can always acquire a digital signal with an analog input module. This isn't recommended and necessary when you can use a pre-built api in the palettes for digital modules.
When there is no free slot you can think about an expension chassis or a single carrier. This depense on your application.
How did you acquired the slower signal until now?
Kind Regards,
Philipp K. | NI-Germany

Sip signaling with supervised transfer possible?

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