vcruntime140.dll - Dialogic HMP on VM with Win Server 2008 R2 Standard.

[Guest Illy]

Hi, I'm trying to install Dialogic HMP on VM with Win Server 2008 R2 Standard. I installed it but when I tried to start Manager I got message vcruntime140.dll is missing. 

I tried to uninstall to try again but that also is not possible.When _I try to uninstall from CP it gives ncmapi.dll failed to load. I have Win Visual c++ redistributable 2015.

Earlier I installed it on my PC with Win10 and installation of Dialogic and Voiceguide passed ok. 

Thanks in advance for your help. 

BR

[SupportTeam]

We just confirmed that the HMP drivers installed and started OK on a brand new clean install Win 2008 system.

We used a new Win2008R2 SP1 (build 7601) install, and the HMP drivers downloaded from VoiceGuide's website. VM environment was used.

The HMP drivers installed with no problems. The HMP installer showed that it was installing the Visual C++ 2015 Redistributable Packages at the very beginning of install, and then, after those completed, proceeded to install HMP itself.

Windows was restarted after HMP install as per the message box displayed by the HMP installer after HMP install completed.

After Windows restart, both the Dialogic Configuration Manager and the Dialogic License Manager were opened with no problems.

No "vcruntime140.dll is missing" errors were encountered.

Was your Win2008 a new clean install?

[SupportTeam]

Microsoft is stopping support for Windows 2008 soon (January 2020) so would recommend using a more recent Windows version for new deployments.

[Guest Illy]

I managed to install Dialogic HMP with new SP1 package. But now when I start DCM I do not see any device. If I add device HMP_Software manually still I can not start the service. 

On previus installation on my laptop and win 10 OS I didn t have this kind of issue, ie there was HMP Software in DCM, DM3 device.

[SupportTeam]

Please install VoiceGuide on this system, and post  here the vgEngine and ktTel trace that capture the VoiceGuide service startup attempt.

This will give us more information about this system setup and we can then advise how to proceed.

Trace files are saved in in VoiceGuide's \log\ subdirectory. Please .ZIP up traces before posting them.

[SupportTeam]

Quote

with new SP1 package.

It sounds like previously you were using just "2008 R2", without the SP1. Correct?

If you only just updated to SP1 then it sounds like you may not have all the available Windows Updates applied on this system. Recommend opening the "Windows Update" app (in Control Panel) and proceed with installing all the windows updates available. Once all available updates have been installed then try starting DCM again.

 

[Guest Illy]

Thanks, I managed to finish the installation and started both services. This server doesn t have internet connection that s why some updates were not done automatically and I installed it manually. Now I ll try configuration with PBX. 

[Guest Illy]

HI,

I m testing now Outbound Dial loader. I entered two phone no, one is our local other is mobile phone. I get local ringing but mobile phone no. 

When I answer to the local I dont hear script that I put in for answer. I put CreditCardPayment.vgs

Attached are logs.

 

0821_1140_vgEngine.txt

0821_ktTel.txt

 

Edited 09/18/2019 04:18 AM by Illy

[SupportTeam]

WireShark trace shows that VoiceGuide registered itself successfully as extension 339 with PBX/Switch at 192.168.10.61

At timestamp '42.5" VoiceGuide made an outgoing call to 063408691@192.168.10.61, but the PBX/Switch at 192.168.10.61 declined the call.

192.168.10.61 advised "Q.850 cause 16" as reason for declining the call, which is just a normal clearing code.

You should look into 192.168.10.61's logs (192.168.10.61 looks to be a Grandstream UCM6208 IP PBX Appliance) to determine why it returned this response.

Grandstream first advised that it is "Trying", and then about 100ms later advised that the call wasDeclined, so most likely it did try placing an outgoing leg of the call to 063408691, but that outgoing call attempt failed, and as a result the Grandstream had to return the "Decline" message to VoiceGuide.

 

The second call made was to extension 326, and it was successful, with Grandstream advising that call was connected/answered about 5 seconds later.

The second call had a script specified for both live answer and answering machine, so the HMP started listening to the line to determine what answered the call, but looks it could not determine what answered the call and the call recipient hung up after 10 seconds.

Are you able to make a test call which does not specify any script for the answering machine options (ie. leave the 'answering machined' field blank), or set that field to:

none  

and confirm if the main 'live' script starts immediately after the outgoing call is answered?

 

 

[Guest Illy]

HI, for outside no we solved it changing some settings on pbx and it is ok now. I put none on answering machine and now everything is ok. Script starts immediately. I ll do some more test and let you know if I find some other issues. 

 

Thanks

[SupportTeam]

Another note:

In VoiceGuide's Config.xml file, in section <VoIP_Registration> please set the <LocalAlias> field in VoiceGuide's Config.xml to:

339@192.168.10.172:5060

and then stop both VoiceGuide and Dialogic HMP and then start Dialogic HMP and then start VoiceGuide.

 

vgEngine trace currently shows:

190 114100.691  8036               GCEV_SERVICERESP ResultInfo: gcValue=1285(0x505|GCRV_INTERNAL|event caused internal failure) gcMsg=[Event caused internal failure] ccLibId=8 ccLibName=[GC_H3R_LIB] ccValue=[0x93||] ccMsg=[IPEC_REG_FAIL_serverResponseDataMismatch] additionalinfo=[]
191 114100.691  8036         WARN  IPEC_REG_FAIL_serverResponseDataMismatch : there was a mismatch between the internal IP CCLib data, and the data contained in the Registrar's response.
192 114100.691  8036         WARN  **********************************************
193 114100.691  8036         WARN  SIP REGISTER Failed
194 114100.691  8036         WARN  **********************************************
195 114100.691  8036         NOTE  Most likely the Contact: field in Registrar's response was different then the Contact: field in the Register request sent out by VoiceGuide.
196 114100.691  8036         NOTE  Some SIP Registrars ignore the Contact: field sent to them and set their own instead.
197 114100.691  8036         NOTE  Please use WireShark to confirm contents of the SIP Registraton messages exchanged.
198 114100.691  8036         NOTE  Please set the LocalAlias field in VoiceGuide's Config.xml to match the Contact: field in Registrar's responses.
199 114100.691  8036         NOTE **********************************************

 

and changing the  <LocalAlias> field should fix this. Otherwise you could encounter issue same as here:

 

 

 

 

[Guest Illy]

Hi, 

It was working just with Credit card Payment script. If I put some other wav file or script I designed I just here some noice.

Logs attached.

I m not sure if in this testing phase I can put other scripts in outcall loader or not?

BR

0821_1446_vgEngine.txt

0821_ktTel.txt

[SupportTeam]

Please .ZIP up and post your bulk_test_music.wav file.

We need to see what is the format of that sound file. It looks like an unsupported format sound file.

Supported sound file formats are listed here: https://www.voiceguide.com/vghelp/source/html/soundfiles.htm

 

For best results recommend using ULAW or ALAW encoded .WAV files - same as the voice data format used for the SIP connection. This way the bytes in the file will be the actual data transmitted over the SIP connection and there will be no trans-coding of any type done from the input WAV to output RTP stream.

See the sound files used by the demo Credit card Payment script and VoiceGuide's 'system' sound files. They will be either 'ALAW' or 'ULAW' (selectable at VoiceGuide install time)

For best results you should match RTP encoding type between VoiceGuide and your PBX to be the type used by PBX when connecting over it's external SIP trunks. Other It is actually ULAW that is used on those external SIP trunks, regardless of where the system is located geographically.

[SupportTeam]

In the case of the call where the HMP started listening to the line to determine what answered the call, but could not determine what answered the call and the call recipient hung up after 10 seconds -> it's possible that the "Hello" said at the beginning when the call was answered was just not loud enough to allow recognition to occur.

If there is something that can be set on the handset used as extension 326 to increase the loudness with which the sound from that handset is transmitted then recommend doing that, otherwise you can try just speaking louder into that handset when answering call.

To test how loud the sound can be heard by the VoiceGuide system you can set up a script that records a sound file and then immediately plays the recorded sound files back to the other party. You can then compare the volume of that playback with the volume of the sample sound files that are played by VoiceGuide in demo scripts.

[Guest Illy]

Hi,

 

Attaching the file. During the installation we selected the option for other countries then NAmerica as we are in Europe. 

bulk_test_music.rar

[SupportTeam]

The "bulk_test_music.wav" file is in format: PCM 22.050 kHz, 16 Bit, Stereo

This is an unsupported format. Please see here for list of supported formats: https://www.voiceguide.com/vghelp/source/html/soundfiles.htm

Please see attached files, which are converted to "PCM 8 kHz, 8 Bit, Mono" and "ALaw 8 kHz, 8 Bit, Mono"

You should be able to play these new files.

The original 22kHz file already had some 'hiss' in file, suggesting that it was not the original recording. Recommend that you convert direct to one of the supported formats from the original recording to reduce the amount of 'hiss' that is present in the played back file.

For best results you should ask for the recording studio to provide the ALaw and/or ULaw format files. This way the file you play will be free from any hiss that is added to the sound whenever the sound is re-encoded into different format.

 

Right now your VoiceGuide system is set up to prefer ALaw encoding, as that was the option selected at install time. Does the Grandstream appliance use ALaw encoding when delivering the calls over the external trunk to external numbers? Or does it use ULaw? (ULaw is commonly used on SIP connections worldwide).

 

bulk_test_music_ALaw_8khz8bitMono.rar

bulk_test_music_PCM_8khz8bitMono.rar

[Guest Illy]

Hi,

 

I was able to play second file you sent. We will pay attention on this when we start recording new files. This file is just for testing purposes, is not recorded professionally. Thanks for your help. Next week we will test on Cisco PBX and win server installation.

That one will be used in production. 

 

BR,

[Guest Illy]

Sorry not Cisco, we will use IP Centrex. I guess it is compatible with your sw.

 

BR,