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Nii

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hi support , 

may i know how to locate the config.xml and the VG.INI, files,   I am using the trial version,   we are  using win server 2012.

 

 

tnx

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VG.INI  is in the main VoiceGuide directory – the directory into which VoiceGuide was installed.

Config.xml is in VoiceGuide’s \conf\ subdirectory.

From https://www.voiceguide.com/vghelp/source/html/install_v7_dialogichmp.htm and https://www.voiceguide.com/vghelp/source/html/install_v7_dialogicsr60.htm :

 

Configuring VoiceGuide

VoiceGuide configuration is set using the following files: VG.INI and Config.xml
VG.INI file is located in VoiceGuide's main directory. Usually: C:\Program Files (x86)\VoiceGuide\
Config.xml file is located in VoiceGuide's \conf\ subdirectory.

Config.xml is used to set what lines/channels are opened by VoiceGuide and what script is to be used when answering incoming calls on each of those lines/channels.

Sample Config files are provided showing how to set up the Config.xml file for larger systems.

If not running as Administrator then some permission changes may need to be made to allow editing of Config.xml, VG.INI, etc. files.

In Windows Explorer:

  1. Right click on the VoiceGuide directory
  2. Select Properties
  3. Go to the Security tab
  4. Make the necessary changes

Alternatively, the text editor application used to edit the Config.xml, VG.INI, etc. files would need to be started using the 'Run as administrator' option. Then Windows will not block the saving of new versions of Config.xml, VG.INI, etc. files.
Notepad++ https://notepad-plus-plus.org/ is a good text editor.

 

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hi Support,

 

1, test calls completing well from the dealer,  however,  we are using the evaluation version,  how do we get a sound file to play  when call arty picks up ....  we tried the normal process , but sound does not play.

 

2, apart from the 1 ports , what are the other limitations of the evalatuion software ,  this will aid  clearer testing please, realise we can only make one good call.

 

 

thank you .

 

n

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Are incoming calls working OK, and you are now trying to place outgoing calls?

For outgoing calls please see: https://www.voiceguide.com/vghelp/source/html/detectcallanswer.htm

Sounds like you are using VoIP. For outgoing VoIP calls it's usually best to just disable the answering machine detection by setting the "Answering Machine .wav / script" to DISABLE.

see below screenshot:

image.png.632194d3393b0fda4aadff8058f7fedf.png

 

The evaluation version of Dialogic HMP only supports one channel.

The evaluation version of VoiceGuide needs restarting every 20 minutes and is limited in the number of outgoing calls that it can make.

Please post the trace files from your system if you are still encountering problems. The trace files can be found in VoiceGuide's \log\ subdirectory.

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thanks ,

 

ys we can see call in , on the line status, and can make call out, alright ,  but can't play the sound file . 

 

please attached, call event log . hope that helps, else which specific log , should i send ...

 

 

Nii

 

0728_CallEvents.txt

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thanks ,

 

ys we can see call in , on the line status, and can make call out, alright ,  but can't play the sound file . 

 

please attached, call event log . hope that helps, else which specific log , should i send ...

 

Nii

 

we have  DISABLE,  the Answering machine answer :   still getting same,  can hear any sound ......

 

thanks.

 

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Please .ZIP up and post the ktTel trace file from same day as the CallEvents trace file. The filename is: 0728_ktTel.txt

We can see many calls getting disconnected straight after dialing. What device is at IP 192.168.11.35 ?

Can you please make a WireShark trace of the SIP messages from/to the VoiceGuide system and post the .pcapng file here.

From https://www.voiceguide.com/vghelp/source/html/install_v7_dialogichmp.htm :

Tracing SIP Messages

WireShark : https://www.wireshark.org can be used to confirm that SIP packets are arriving at VoiceGuide/HMP system.

To isolate SIP traffic the following should be specified in WireShark's Filter text box:

sip

image.png.888c3cdc9d3bf3197c253c1855b027e4.png

To isolate SIP and RTP traffic the following should be specified in WireShark's Filter text box:

sip || rtp

image.png.b6460b10c3f972cb58bd370f05fd25a5.png

To save the isolated traces go to WireShark's File->Export Specified Packets menu and save the displayed packets as .pcapng format.

 

Please ensure all posted files are .ZIPed up before posting.

 

{ "call": {
  "crn": "8000001",
  "port": "1",
  "direction": "out",
  "callcounter": "3",
  "time_start": "2023-07-28 08:44:00",
  "callflow": "C:\Program Files (x86)\VoiceGuide\Scripts\Credit Card Payment\PayGetCc.wav",
  "events": "
    084400.640|command|MakeCall_Completed|0244891860@192.168.11.35|10013@192.168.11.35||<result>ok</result><crn>134217729</crn><crnx>8000001</crnx>|0|0|0|0
    084400.643|callstate|GCEV_DIALING|||||0|16|16|0
    084400.644|event|GCEV_DIALING|||||16|0|16|0
    084400.672|event|GCEV_ALERTING|||||2081|0|0|0
    084401.152|callstate|GCEV_DISCONNECTED|||||0|16384|64|0
    084401.155|event|GCEV_DISCONNECTED|||||16384|0|64|0
    084401.156|command|cmd_DropCall|||||0|0|0|0
    084401.160|callstate|GCEV_DROPCALL|||||0|1|32|0
    084401.171|event|GCEV_DROPCALL|||||1|0|32|0
    084401.177|callstate|GCEV_RELEASECALL|GCST_NULL||||2137|0|0|0
  ",
  "length": "0.537"
}},

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thanks,

 

The device on 192.168.11.35 is an IPPBX .    indeed also realise most calls gets discounted at once, and  or,  while you have discount the call from the handset, the call

on the line status remains active.

 

attached the Zipped log files.  

 logs2.zip     logs.zip.  

 

Nii.

 

 

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Please post the WireShark trace as advised before. Also post the ktTel trace file for the day when the WireShark trace was captured.

Also, it looks like the SIP REGISTER is not successfully registering extension 10013.

Looks like PBX is responding with a "403 - FORBIDDEN" when SIP REGISTER is attempted, so you will need to check with PBX why that is the case. Not being able to Register your extension with PBX will most likely prevent you from placing outgoing calls from that extension though the PBX to the outside number. WireShark trace would show exactly what is happening between VoiceGuide/HMP and the PBX.

From ktTel trace file:

153 110223.350  4936         fn    VoIPProvider_Register(protocol=Sip, reg_server=192.168.11.35, reg_client=10013@192.168.11.35, local_alias=10013@192.168.11.37, sH323SupportedPrefixes=)
...
191 110223.477  2884               GCEV_SERVICERESP ResultInfo: gcValue=1286(0x506|GCRV_CCLIBSPECIFIC|event caused by cclib specific failure) gcMsg=[Event caused by call control library specific failure] ccLibId=8 ccLibName=[GC_H3R_LIB] ccValue=[0x151b||] ccMsg=[IPEC_SIPReasonStatus403Forbidden] additionalinfo=[]
192 110223.477  2884         ERROR ccValue!=IPERR_OK (VoIPProvider_AuthenticationSet)
193 110223.477  2884         WARN  **********************************************
194 110223.477  2884         WARN  SIP_REGISTER Failed
195 110223.477  2884         WARN  **********************************************

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hi,

 please see attached. however, sound file played well yesterday, i tried this morning , and did not play, nothing changed.  

could it be, due to the trial version... ?

thanks

 NII.pcapng

logs2.zip

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The last call in the 0730_ktTel trace file shows the all was connected OK, but that the .WAV file did not play as there was something wrong with the .WAV file itself.

See below trace extract from 0730_ktTel.txt:

549 082938.683  4748   3   1 fn    PlayStart file=E:\._20230730_032921843.wav, options=<option>trim_silence_at_front</option>
..
557 082938.685  4348         WARN  play  E:\._20230730_032921843.wav is not a RIFF file.
558 082938.685  4348   3   1 ERROR  ReadDataFromWavPcmIntoMemBuffer returned -203 (bytes_to_play_out=0) when reading file E:\._20230730_032921843.wav, skipping file.
559 082938.685  4348   3   1       play  PlayStart : no sound files queued to play.
560 082938.685  4348   3   1 r     PlayEnd 2011118477

ktTel trace shows 3 previous calls tried to play that file and all failed in in same way.

An even earlier call played a different sound file:

C:\Users\Administrator\Desktop\20230730_032921843.wav

and that sound file played OK:

264 073421.438  2904   3   1 fn    PlayStart file=C:\Users\Administrator\Desktop\20230730_032921843.wav, options=<option>trim_silence_at_front</option>
...
272 073421.498  2204   3   1       play  wav_hdr: format=1, channels=1, hz=8000, byte_ratec=16000, block_align=2, bits=16, data_block_size=186880  C:\Users\Administrator\Desktop\20230730_032921843.wav
...
288 073421.514  2204   3   1       play xpb: wDataFormat=8, wFileFormat=1, nSamplesPerSec=0x40, wBitsPerSample=8
289 073421.515  2204   3   1       chapter_1 play_iott(2, iott=0xfce8708 (len=93440, buff=0xfd96748), xpb=0xfbe46d8) call
290 073421.516  2204   3   1       play(2, 0xfce8708) => 0, hli=0FBE1FD0

 

so looks like there is something wrong with the E:\._20230730_032921843.wav sound file itself.

Recommend you keep testing with with verified valid sound files. You can use sound files that came with VoiceGuide. See VoiceGuide's \System\voice\ subdirectory, or the sound files that came with the sample VoiceGuide scripts.

Does E:\._20230730_032921843.wav play ok when using Audacity? (see https://www.audacityteam.org/ )

If E:\._20230730_032921843.wav plays using Audacity then please post the .WAV file here.

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hi support , 

"so looks like there is something wrong with the E:\._20230730_032921843.wav sound file itself."  yes, wasnt  saved in the proper format.

 how can we/I , play 2 sound file at the same time ?

what do i need to check , need  to keep restarting device or the PBX, for   VG  device to register on the PBX   after some , 

thanks

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hi support,

kindly, show  with image how that is done,  yes i followed through the link,  still some errors, is adding file  ,  and adding the second  file, 

thanks

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in any module in VoiceGuide script it's be done like this:

image.png.fd37ffede7a8adb4e3370f57c60b296b.png

 

and similar in the outbound call loader:

image.png.c39cdb8f2434a15d34b29d95db73aa5e.png

 

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thanks support,

b,ut , guess am not doing some thing right ,    i followed through still i can't add, 

 

am testing from  the trial version 

? does file locations count 

thanks

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Please post traces capturing the system startup and the call. We can then see what has happened during the call and advise.

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please,  

 the module do not allow me to add another sound as done above simply ,,,, any time i add the second , the earlier si taken off... 

we don't get to the point placing the call.

thanks 

 

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Are you adding files using the "Select" button? That will only select one file and populate the 'sound file' text box with that file.

To specify multiple files to be played you need to type the paths/filenames into the text box.

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thanks support,

 i hope you are working on using the "select button"  to achieve same ,,,,,, soon.

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On 8/4/2023 at 7:13 AM, SupportTeam said:

Are you adding files using the "Select" button? That will only select one file and populate the 'sound file' text box with that file.

To specify multiple files to be played you need to type the paths/filenames into the text box.

 hi support,

am getting "valid file name "     per the above. i.e. typing the file name or path....  

what do we do next ...

thanks 

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Can you please post some screenshots or a screen video capture so that we can see how you are using the software.

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20 hours ago, SupportTeam said:

Can you please post some screenshots or a screen video capture so that we can see how you are using the software.

HI, 

please see attached .

VG.jpg

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Are you trying to play over the local speakers of headphones the list of comma delimited sound files specified as the sound files to be used by that 'Play' module?

What button did you press immediately before the "Invalid Filename" error box popped up?

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2 minutes ago, SupportTeam said:

Are you trying to play over the local speakers of headphones the list of comma delimited sound files specified as the sound files to be used by that 'Play' module?

What button did you press immediately before the "Invalid Filename" error box popped up?

playing of local speakers , and  i pressed the play button on the play module...   

 

thanks 

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That play button will not play back a list of sound files. The "Record", "Play" and "Select" buttons are meant to be used for easy selection and recording or verification of contents of single files only.

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1 minute ago, SupportTeam said:

That play button will not play back a list of sound files. The "Record", "Play" and "Select" buttons are meant to be used for easy selection and recording or verification of contents of single files only.

so,  how do we  playing multiple sound files ... please 

 

thanks

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There is no way to play back concatenated sound files in the script editor. Best way to test playback in that case is to just make a call into the system. That way you will hear the sound files played in exactly same way as the IVR user will.

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9 minutes ago, SupportTeam said:

There is no way to play back concatenated sound files in the script editor. Best way to test playback in that case is to just make a call into the system. That way you will hear the sound files played in exactly same way as the IVR user will.

oh ok, so what does it mean,  when the system says you can play multiple sound file, or to  add" files by typing"  as mentioned earlier ..  indeed i place a call from the system to my self , but did not hear any voice or sound,   but when when its a single sound file, i do hear the sound or voice,,,,,

 

thanks

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Quote

 i place a call from the system to my self , but did not hear any voice or sound,  

Please post VoiceGuide traces capturing the call (vgEngine and ktTel traces). We can then see what has happened during the call and advise.

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  hello, 

please, 

1, how can I make or get an IVR to play for an incoming call ..       achiveing that, do i need to make the VG an extension in the PBX ? 

2,how do i use call transfer to ,  to bringss calls from a sip server through the VG to  the teleco sip trunk.

thanks

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8 hours ago, SupportTeam said:

Please post VoiceGuide traces capturing the call (vgEngine and ktTel traces). We can then see what has happened during the call and advise.

Sure will test that again and post , please.      

 but my main question, is how to get multiple sound to play .....  using the method you shown above ....

 

thanks 

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Quote

1, how can I make or get an IVR to play for an incoming call ..       achiveing that, do i need to make the VG an extension in the PBX ? 

Easiest way is to just call the VoiceGude's IP address direct from an IP phone.

from https://www.voiceguide.com/vghelp/source/html/install_v7_dialogichmp.htm :

 

 

Testing VoiceGuide

To test the VoiceGuide installation you can place a call into the system by dialing the IP address of the system from SIP VoIP phone, or from a SIP softphone on another system.
Best if for initial testing the dialing phone is on same network segment as the VoiceGuide system, with no other routers etc. in between.

When a SIP call arrives at the VoiceGuide system, VoiceGuide should answer the call and start the demonstration 'Credit Card Payment' script, which asks caller to enter some details (card number, amount etc.) and the plays the entered information back to the caller.

The demonstration script is located in VoiceGuide's \Scripts\Credit Card Payment\ subdirectory.

After you hear the demonstration script answer the call you are now ready to start creating your own scripts, make outgoing calls etc.
The Script Designer and other VoiceGuide applications can be started by right-clicking on the IVR tray icon and selecting from the context menu.

If not running as a user with Administrator privileges then you will not be able to stop and start the VoiceGuide service using the VoiceGuide Service Monitor applet from the icon tray. You will need to go to Control Panel -> System and Security -> Administrative tools -> Services and stop and start the VoiceGuide service from there.

 

 

Softphones

NOTE: The testing softphone MUST NOT be installed on the same system as Dialogic HMP.
It must be installed on some other system.
There can be no other SIP software that uses SIP 5060 port installed on same system as Dialogic HMP.
Only one software can use the SIP 5060 port at a time.

 

One softphone that can place calls to an IP address is MicroSIP : https://www.microsip.org

MicroSIP allows dialing addresses in format:

number@ip_addres

eg:

1234@192.168.0.87

Some versions of MicroSIP also allow dialing an IP address without the number@ before it.

eg:

192.168.0.87

 

Another softphone that can place calls to an IP address is Linphone : http://www.linphone.org

With Linphone the SIP address to be dialed must be prefixed with "sip:"

eg:

sip:1234@192.168.0.87

or:

sip:192.168.0.87

Also please select Linphone's Options->Preferences menu, and set the following:
Network Settings tab : Media Encryption is set to 'None'
Codecs tab: only enable the G.711 ULaw and/or the G.711 ALaw codecs (PCMU and PCMA codecs), and disable the rest.

 

Another older softphone that can also dial the IP address directly is SJPhone.
With SJPhone the number@ip_addres address or just the IP address by itself can be used.

 

 

Quote

2,how do i use call transfer to ,  to bringss calls from a sip server through the VG to  the teleco sip trunk.

You would use the Transfer Call module.

Please see: https://www.voiceguide.com/vghelp/source/html/modxfer.htm

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how to get multiple sound to play .....

Just specify the multiple sound files one after another with a comma between each filename. When you call into the system the files will play. They will not play in the script editor, but will play one after another during the actual call.

 

in any module in VoiceGuide script it's be done like this:

image.png.fd37ffede7a8adb4e3370f57c60b296b.png

 

and similar in the outbound call loader:

image.png.c39cdb8f2434a15d34b29d95db73aa5e.png

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hi support , 

on the call transfer , i.e. SIP refer transfer ,     can you show an example how his done, after going the  he pointed resource,  my thoughts--->   Sip Server,  send calls to VG (via Hookflash blind)..  ? where do i input the IP/url of the destination sip trunck that receives the call.

thanks

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