Polycom SoundStructure module

tnguyen

Hi guys,

I am new to AMX. I have some problems with Polycom SoundStructure duet module(non-rpm version). I upload the demo program in the package to NI-3100 and try to make a phone call. It doesn't work properly. For example, I press all phone digits and press Dial but the Polycom SS only goes off hook and doesn't dial out.

Do you have any ideas what goes wrong with the module? and how do I fix it ?or what is the best practice to work with this module?

Thank you very much.

[Deleted User]

Are you using a VOIP card or traditional phone line?

tnguyen

It is just an analog phone line. I changed the input and output array from VOIP to Phone In and Out accordingly. I used serial connection between controller and sound structure.

I also had a lifesize duet module running on NI-3100 and it seems to be OK.

[Deleted User]

As a test for communications to the device and potential custom programming issue, can you telnet to the master and do (change the virtual device number to match that of your module's duet virtual)

device status 41001:1

Look at the feedback channels. Do you see 251 and 252? If so, that is positive but not all inclusive. If they are not there, it suggests communications problems - baud, cabling, config, etc.

If you do have 251 and 252, I would move into evaluating my code and usage of the duet module. To rule out your implementation of the duet module, I suggest throwing together a quick RPM file. Load the supplied sound structure file, and the RPM project file. Test with the RPM gui and see if the outcome is the same.

If the RPM test succeeds, and you can control the device, you have vetted the cable and have pointed squarely at how you have set up the module in your custom code. I have not looked at this module in a while, but have you read through the interface document that came with the module's zip file? If not, start there. Review the programmer notes and file inclusion notes.

Hedberg

tnguyen wrote: »

Hi guys,

I am new to AMX. I have some problems with Polycom SoundStructure duet module(non-rpm version). I upload the demo program in the package to NI-3100 and try to make a phone call. It doesn't work properly. For example, I press all phone digits and press Dial but the Polycom SS only goes off hook and doesn't dial out.

Do you have any ideas what goes wrong with the module? and how do I fix it ?or what is the best practice to work with this module?

Thank you very much.

If the Polycom is going off hook when you try to dial, it would seem that you are successfully communicating with it and that you likely have the virtual devices and the names of the Polycom objects to be controlled correct. Setting these things up correctly is non-trivial for everybody, not just those new to AMX. Study the module documentation and the sample axs file to see how this stuff works. Make sure the names of the objects you want to control match the names used by whoever programmed the Polycom. As I said, if you are getting the Polycom to go off hook, you likely have the phone out object properly configured.

So, when it goes off hook, do you get a dial tone? Some dialers won't dial if a dial tone is not detected. I've programmed a couple of these Polycom devices but can't recall if they are among those.

If you get a dial tone but no numbers are being sent, check the dialog between the module and the Polycom to see if the strings are actually being sent. Try sending the proper strings directly to the Polycom bypassing the module and see what happens.

But, if the module is causing the Polycom to go off hook, either there is a problem with the dial tone or you're not triggering the proper channels in the module.

The modules will work, if everyuthing is set up properly. We did one earlier this month using Polycom_SoundStructure_Comm_dr1_0_0.jar

tnguyen

I don't think I have any connection issues between NI-3100 and SS C16 as I press "Send DTMF Directly On" and press "Off Hook" then I can hear the dial tone and it connects to other line just fine. However, the system will miss numbers if I press numbers fast. I have to press one by one 1s each . Additionally, when "Send DTMF Directly On" is set, the entered number isn't displayed on the text box. It dials out as soon as I press "Off Hook" though.
So I can dial out as following steps:
+ Set DTMF Directly On
Either:
+Off Hook and press numbers one by one slowly
Or:
+Press all numbers blindly (no text displays on the text box)
+Press Off hook


Since I had a Lifesize Room 220 module taking the 41001:1:0 to 41001:11:0 so I changed the Polycom virtual channel port to "41101:1:0 - 41101:16:0". I kept all of the including part the same as in the example. I use the same jar Polycom_SoundStructure_Comm_dr1_0_0.jar as Hedberg mentioned.

I didn't check the virtual port 251 and 252. Would you please show me the correct steps to check the feedback channel correctly?

dvPolycomSoundStructure = 5001:3:0 //It is connected to Com 3
dvTPMain	= 10001:1:0 //I am using TPC-Ipad
dvPolycomSoundStructureTp = 10001:12:0 

DEFINE_MODULE 'PolycomSoundStructureAudioConferencerComponent' conferencer(vdvDev, dvPolycomSoundStructureTp, dvTPMain, nAudioConferencer, nAudioConferencerPages)
DEFINE_MODULE 'PolycomSoundStructureDialerComponent' dialer(vdvDev, dvPolycomSoundStructureTp, dvTPMain, nAudioConferencer, nDialerPages,OutArray)
DEFINE_MODULE 'PolycomSoundStructureModuleComponent' module(vdvDev, dvPolycomSoundStructureTp, dvTPMain, nAudioConferencer, nModulePages)
DEFINE_MODULE 'PolycomSoundStructureVolumeComponent' volume(vdvDev, dvPolycomSoundStructureTp, dvTPMain, nAudioConferencer, nVolumePages,OutArray)
DEFINE_MODULE 'PolycomSoundStructureAudioMixerComponent' mixer(vdvDev, dvPolycomSoundStructureTp, dvTPMain, nAudioConferencer, nAudioMixerPages,PresetArray)
DEFINE_MODULE 'PolycomSoundStructureGainComponent' gain(vdvDev, dvPolycomSoundStructureTp, dvTPMain, nAudioConferencer, nGainPages,InArray)
DEFINE_MODULE 'PolycomSoundStructurePreAmpComponent' preamp(vdvDev,dvPolycomSoundStructureTp,dvTPMain,nAudioConferencer,nPreAmpPages,sSurroundMode,InArray,OutArray)
// Define your communications module here like so:
DEFINE_MODULE 'Polycom_SoundStructure_Comm_dr1_0_0' comm(vdvPolycomSoundStructure, dvPolycomSoundStructure)

I changed the default VOIP to Phone In and Out.

char InArray[][][50] = {
	{'Phone In','PHONEIN'},
	{'Mics','INPUT'},
	{'Table Mic','INPUT'}, 
	{'Table Mic 1','INPUT'},
	{'Table Mic 2','INPUT'},
	{'Table Mic 3','INPUT'},
	{'Table Mic 4','INPUT'},
	{'Table Mic 5','INPUT'},
	{'Table Mic 6','INPUT'},
	{'Table Mic 7','INPUT'},
	{'Sig Gen','SIGGEN'}

	
}
//-----------------------//
char OutArray[][][50] = {
	{'Phone Out','PHONEOUT'},
	{'Bannon Amp','MONO'},
	{'Amplifier 1','STEREO'}
}

I imported all the modules to Module Folder and import the mainInclude to Include folder. I checked the Module subpage. "Reboot" is blinking green. "Device not Online". "Data Not Initilized". Pressing "Reinitialize" button doesn't do anything. "Query Firmware version" isn't displayed but "Query module version" is displayed.

tnguyen

Case Solved

Guys,
Thank you very much for your support. The problem is the RS-232 connection isn't working well with Polycom even though it is able to send out commands but not properly. So as soon as I switch to ethernet connection, it is working like a charm now.

Thanks again. I look forward to the next support.

Teo