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Entered Number Not Work after the 4th test

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Hi Team Support.

 

The situation is...

 

I installed VG7 with SIP support, and HMP software. We evaluate the system and purchase the licenses to 12 HMP ports and the license fot VG7, but always in the 4th or 5th test the Get Entered Number does not work, only replay the wav file but not collect the numbers only when restart the service begins to collect the data.

 

I believed that this problem was related to the evaluation license but the problem persist.

 

The log files are attached to this message.

 

Please help me as quickly as possible because we try to implement this IVR the first days in January.

 

 

Thanks.

1227_1415_vgEngine.txt

1227_ktTel.txt

1227_ktTts.txt

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Attached trace shows that on first 4 calls the system is registering the DTMF tones being received, but on subsequent calls there are no reports of any DTMF tones being pressed by the caller.

 

Easiest way to test if the DTMF tones are being sent/received is to start the script with a Record module. This will let you confirm if the DTMF tones are being received by listening to the recording afterwards.

 

Another source of the problem could be the mismatch between the HMP and the VoiceGuide licenses. You mention that the HMP is licensed for 12 channels - but the trace shows that only a 4 line license has been applied on this system, so another test would be to change to using the evaluation 4 line license from Dialogic and see if having the matching licenses line count helps anything here.

 

The Evaluation version of VoiceGuide does not restrict VG from receiving DTMF tones.

 

 

 

 

141627.608 4 3 state [Menu Principal] Playing wav (wav\gracias_por_llamar.wav)

...

141641.123 9 3 ev dtmf 5 (3,53,0)

 

 

141715.951 4 7 state [Menu Principal] Playing wav (wav\gracias_por_llamar.wav)

...

141725.128 9 7 ev dtmf 5 (7,53,0)

 

 

141807.613 4 10 state [Menu Principal] Playing wav (wav\gracias_por_llamar.wav)

...

141812.535 9 10 ev dtmf 5 (10,53,0)

 

 

141856.613 4 13 state [Menu Principal] Playing wav (wav\gracias_por_llamar.wav)

...

141903.910 9 13 ev dtmf 5 (13,53,0)

 

 

141945.285 4 3 state [Menu Principal] Playing wav (wav\gracias_por_llamar.wav)

...

142001.144 4 3 state [Menu Principal] Playing wav (wav\gracias_por_llamar.wav)

..

142003.003 9 3 ev Dialogic 2086,GCEV_DISCONNECTED,2086,0,0,,,

 

 

 

142006.941 4 7 state [Menu Principal] Playing wav (wav\gracias_por_llamar.wav)

...

142020.331 9 7 ev Dialogic 2086,GCEV_DISCONNECTED,2086,0,0,,,

 

 

142038.878 4 10 state [Menu Principal] Playing wav (wav\gracias_por_llamar.wav)

...

142049.191 9 10 ev Dialogic 2086,GCEV_DISCONNECTED,2086,0,0,,,

 

 

142311.456 4 13 state [Menu Principal] Playing wav (wav\gracias_por_llamar.wav)

...

142319.753 9 13 ev Dialogic 2086,GCEV_DISCONNECTED,2086,0,0,,,

 

 

 

 

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The first time I used the evaluation license HMP 4-ports and the problem happens exactly after the 4th call.

I test with both licenses (4 and 12 ports) and the problem persist.

 

I make a test with the Answer and Record Script sample, in the first 4 calls I can hear the tones, but the next calls no.

 

I don't know what happens, in our network we use SIP all the time, we have 3 asterisk servers so we don't restrict anything.

 

 

Another suggestion?

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I make a test with the Answer and Record Script sample, in the first 4 calls I can hear the tones, but the next calls no.

Can you hear what the caller says in the recording?

 

ie. On calls where the DTMF is not heard, is it just the DTMF tones that cannot be heard, or is it the entire receive voice channel that is silent? I take it that the caller can still hear the gracias_por_llamar.wav being played, yes?

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The caller can hear the wav file, but the VG not detect the DTMF tone.

 

I test another time with "Answer and Record Script sample", in every test I say a few words and press 3 digits. The results are the following:

 

Test Hear Voice Hear DTMF

*** ********* *********

1 yes yes

2 yes yes

3 yes yes

4 yes yes

5 yes no

6 yes no

7 yes no

8 yes no

 

Why only the first 4 times work very well?

 

VG7 is installed under Windows XP SP2, .Net 2 core, .Net 3 core, HMP.

The platform is Intel Pentium 4, 3.2GHz, 1Gb RAM, 120Gb HD, Linksys 10/100/1000 network card.

 

How I can detect the DTMF setting that is used in the call??

 

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Looks like for some reason something along the line is 'clamping' the DTMF tones after the 4th call...

 

How are you placing the calls into the system? Does the call go through any other switches/PBXs ?

 

I'd next try placing calls directly into VoiceGuide using SJPhone. Set SJPhone to work in "Direct SIP" mode and dial into the system by specifying the IP of the HMP/VG machine.

 

Repeat the DTMF test as before to see if DTMF can be heard.

 

Can you do a WireShark capture of one of the calls when the DTMF cannot be heard?

 

 

 

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I tested the SJPhone in Direct SIP mode and the system always work very well, but in SIP Proxy mode after the 4th call the system can't receive the DTMF tones, if I return to Direct SIP the IVR always receive the DTMF tones.

 

 

We have two SIP PBX: 3Com NBX V3000 and Asterisk.

We use 2 kinds of Sip Phones: the 3com(pcXset) and Free Softphones(X-Lite, PortSIP, etc), in both sip phones we use SIP Proxy settings.

We have one E1 line card in the 3com PBX and through this we are goin to transfer the calls to the IVR(Asterisk Extension).

In Asterisk PBX we have one Analog Line card and we use this to make the IVR test too.

VoiceGuide use one Asterisk Extension for register and receive the calls.

 

In this scenario, please tell me

 

Why in SIP Proxy the IVR can't receive the DTMF tones?

How I can integrate my PBXs whith the IVR?

o You need additional information about our SIP PBXs?

 

Thanks for help me

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It looks like the switch through which you are sending the calls is 'clamping' the DTMF tones. As you can see when recording the incoming sound stream, the DTMF is just not passed on. VoiceGuide itself or HMP does not do any DTMF tone clamping.

 

You should be speaking with the Switch supplier about why is the Switch clamping the DTMFs.

 

I assume that you are using the Asterisk as the SIP proxy? Have you tried using 3Com instead?

 

Does everything work ok when calls are routed out the Asterisk's analog port?

 

How many IVR channels will you ultimately require?

 

 

 

 

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Dear Support Team,

 

We use Asterisk as the SIP proxy because the configuration of the VG use parameters like Asterisk. In 3com its very hard to obtain support for this, really we are disappointed with this provider. I believe that in SIP mode the IVR is related to one extension, and every incoming call from 3com or asterisk I go to transfer to this extension (ivr). That's all. I am wrong or not???

 

I researched about the DTMFs and asterisk can use rfc2833, inband and sip_info, these are supported in VG. I use rfc2833.

 

I make new tests installing Wireshark in the IVR server and the DTMF tones REACH THIS SERVER (I filtered by RTP protocol), something happens in VG or HMP software that after the 4th call not receive this DTMFs.

 

RTP EVENT Payload type=RTP Event, DTMF Four 4

RTP EVENT Payload type=RTP Event, DTMF Four 5

 

I know that if I test with SJphone in SIP Direct mode the IVR works fine all the time, but in SIP Proxy only the first 4 incomming calls. Exactly the numbers of channels/ports in the VG, it seams like the channels stay unavailable after the first use of every one.

 

Another anomaly is that I intentionally modified the Config.xml changing the protocol to SIP, normally is IP

 

<Channel>

<Device_Voice>dxxxB1C1</Device_Voice>

<Device_Network>iptB1T1</Device_Network>

<Device_Media>ipmB1C1</Device_Media>

<Protocol>SIP</Protocol> <------ This line

<script>C:\VG\PRUEBAS\SaldoCredipycca.vgs</Script>

<AllowDialOut>1</AllowDialOut>

</Channel>

 

I reset the service and the system works fine the first 4 times, but in the next loop the incoming call can't be transfer to the Script, it's like the firts time the VG use another config and bypass this error, but the second time stay in error because SIP is not a valid parameter.

 

Another test was that I leave only 3 channels in the Config.xml, and after the 3th call the system not detect the DTMF tones.

 

 

I need Your help, we bought 12 HMP and 4 VG licenses to begin, but we want to expand this channels.

To buy HMP licenses is a littlet bit difficult than VG licenses because they don't work with credit card payment so we bougth 12 once.

 

 

All the tests were made with evaluation licenses (4 ports).

 

Thanks for help.

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Could you please setup the system in a 3-channel mode (only setup 3 channels in config.xml) and restart the VG service and make 10 calls into the system. Then post all the VoiceGuide traces and the WireShark trace. Please include everything in the WireShark trace (ie. do not filter).

 

I believe that in SIP mode the IVR is related to one extension, and every incoming call from 3com or asterisk I go to transfer to this extension (ivr). That's all. I am wrong or not???

On HMP the 'next' channel answers the incoming VoIP call - it's not the same channel answering over and over.

 

I researched about the DTMFs and asterisk can use rfc2833, inband and sip_info, these are supported in VG. I use rfc2833.

Please try setting Asterisk to use Inband.

 

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I take traces in the IVR Server with 3 channels and I make 10 calls.

 

The VoiceGuideScript play a sound file and wait for press 4, after this execute the GetNumber and I press 5.

Finally I hung up

Edited by SupportTeam
removed attachment

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I try with inband in Asterisk and nothing.

In the 3com NBX I set one SIP TRUSTED INTERFACE with the IVR IP address with the port 5060, the result is the same, after the 4th call the IVR can't read the DTMF tones.

In 3com the SIP Trusted Interface are used to comunicate 3com with other components, like Asterisk and the Attendant that work with 3com (IP Messaging).

 

 

But with wireshark I can view the DTMT tones, so... the problem is in HMP or VG.

 

I take the follow steps to enable the IVR:

 

1. Install .Net Framework 2.0 and 3.0

2. Install HMP driver 3.0

3. With DCM Assign the 4LinesLicenseEval 4r4v4e4c4s4f4i4m_host_eva.lic

4. VoiceGuide v7.0.4

5a. Set the Config.xml with VOIP_Registration to Asterisk Extension

5b. Set the Config.xml without VOIP_Registration

 

When I use 5a I make a call from the extension XXXX to extension YYYY, when XXXX is a softphone and YYYY is the extension related to the IVR.

 

When I use 5b I use the SIP TRUSTED Interface in 3COM.

 

In both cases after the 4th call the IVR can't receive the DTMF tones.

 

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Lets just explore the Inband setting on Asterisk a bit more.

 

When you set the Asterisk to use Inband, and setup the VG script to just do a recording of what VoiceGuide/HMP can hear when the incoming call is answered (ie. just have a single record module in the script) do the recordings show that the DTMF sound is being played on the line on all calls?

 

As per previous testing - the calls made direct to VG from are all fine - with the HMP hearing the inband tones OK. What looks like is happening on this system is that the Asterisk is clamping the inband tones and sending a rfc2833 instead. We'd need to have a close look at the traces and setup more tests to see why this rfc2833 message is not recognized - but the quickest way to get this working is to ensure that the Asterisk is allowing the inband signal to passthrough and be heard by HMP.

 

If you can setup Asterisk to let the DTMF sound passthrough and VG/HMP can hear the DTMF then things will work as well as when direct test calls from softphone are made.

 

NB. do you have any softphone which can send just rfc2833 without sending Inband?

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Another possibility is that the HMP itself is clamping the tone and not sending the inband tone onto the internal voice device which does the recording. Only way to really see what is actaully being sent on the line is to extract the data portion of the RTP packets into a sound file and look at the sound file, but we do not have any tools which can do that right now.

 

We have passed the provided traces to the development team who will look into why the system is not reporting rfc2833 messages which can be seen in the WireShark trace.

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Which version of Asterisk are you using?

 

Can you please post the changes you made to Asterisk configuration file and the VoiceGuide's Config.xml.

 

We will try to reproduce your setup here in our lab.

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Please uninstall the current version of VG on your system (backup you scripts and config files) and the download and install the version which has a fix for the rfc2833 from here:

[old link removed]


--------


We had a look at the RTP streams in the provided WireShark trace and extracted the sound files from all 10 calls (attached).

It looks like the DTMF tones on the first 3 calls were not sent via Inband after all. The DTMF on all 10 calls was sent via rfc2833.

So the problem was not that 'inband dtmf transmission went away', but that rfc2833 detection was only working on first call.

Attached version now sets rfc2833 detection support for every call. The enabling of rfc2833 support for every call in the previous version was not documented in the Help file. Having both the inband and rfc2833 DTMF detection enabled on every VoIP call is probably a better approach anyway.

RTP_streams.zip

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Hello,

 

I installed the VG7 fix, but not work.

I uninstalled VG7 and HMP. After this I install HMP an VG7 without make modifications in Config.Xml, but do not work.

 

I think that this fix has some problem with HMP because in the log directory only create the vgEngine.txt file and there are line with errors.

 

 

load: C:\Archivos de programa\VoiceGuide\ktTelDialogicHMP30.dll, (ktTelDialogicHMP30, Version=7.0.2925.31076, Culture=neutral, PublicKeyToken=d1aea68d146fa1e6)

ERROR v7.0.2925.31193 (Fri 04/01/2008 17:19:46.29) Load_Layer_ktTel_DynamicPlugins Assembly.Load : No se puede encontrar el módulo especificado. (Excepción de HRESULT: 0x8007007E)

en System.Reflection.Assembly.nLoadFile(String path, Evidence evidence)

en System.Reflection.Assembly.LoadFile(String path)

en ..()

ERROR v7.0.2925.31193 (Fri 04/01/2008 17:19:46.29) mainLoad ktTel_ApiMode : Referencia a objeto no establecida como instancia de un objeto.

en ..(String sAppPath, String sLayer_ktTel_ToLoad)

timer init

VGINI_Filename=C:\Archivos de programa\VoiceGuide\VG.INI

kttel call Initialize(C:\Archivos de programa\VoiceGuide\, config_ktTel=C:\Archivos de programa\VoiceGuide\Config\Config.xml, config_ktTts=)

ERROR v7.0.2925.31193 (Fri 04/01/2008 17:19:46.29) mainLoad A : Referencia a objeto no establecida como instancia de un objeto.

en ..(String sAppPath, String sLayer_ktTel_ToLoad)

 

 

The complete log file is attached.

 

0104_1240_vgEngine.txt

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Looks like the install build made especially for you was supplied with the wrong ktTelDialogicHMP30 version, so the service is not starting up at all. We will look into this and should have a new version available for download for you within a couple of hours.

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We have been installing the same install package on our test machines and it is installing fine. And we have been advised that the ktTelDialogicHMP30 version is correct.

 

Can you please check if the following two files have also been installed in VG's main directory: libsofia_sip_ua.dll and pthreadVC2.dll

 

Is it possible for us to have a look at this system using logmein.com service or VNC/PCAnywhere or similar ?

 

If yes then could you please forward us the connection details via email to support@voiceguide.com and include in the email a link to this forum thread.

 

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I use HMP 3.0

 

The original installation was made with VG7, in this version the Line Status shown the 4 lines idle.

When I uninstall the VG7 and install the VG7Fix I can't view the 4 lines.

When I uninstall the VG7Fix and reinstall the VG7 I can view the 4 lines.

When I came back to VG7Fix I can't view the 4 lines in Line Status option.

 

When I can view the 4 lines I make test to IVR and it's ok only the first four times.

But, when I can't view the 4 lines I can't test the IVR.

 

The error happens when tray to use something subroutine in the ktTelDialogicHMP30.dll, in the log files there are error messages.

 

I review the files in the VG directory and I have the libsofia_sip_ua.dll and pthreadVC2.dll when is installed the Fix.

 

I compare the files between versions, the first line is original VG7 and de second line is VG7 fix:

 

11/05/2007 04:20 AM 40,448kb ktTelCommon.dll

12/11/2007 08:30 PM 40,448kb ktTelCommon.dll

 

11/15/2007 01:17 AM 933,888kb ktTelDialogicHMP30.dll

01/04/2008 01:15 AM 1,036,288kb ktTelDialogicHMP30.dll

 

11/15/2007 01:17 AM 593,920kb ktTelDialogicSR60.dll

01/04/2008 01:20 AM 659,456kb ktTelDialogicSR60.dll

 

11/15/2007 01:17 AM 585,728kb ktTelDonjinNADK181.dll

01/04/2008 01:20 AM 647,168kb ktTelDonjinNADK181.dll

 

11/14/2007 05:29 PM 61,440kb ktTtsSapi.dll

01/03/2008 10:22 PM 61,440kb ktTtsSapi.dll

 

11/15/2007 01:16 AM 1,146,368kb vgEngine.dll

01/04/2008 01:55 AM 1,152,512kb vgEngine.dll

 

01/04/2008 07:26 PM 112kb VoiceGuide.url

01/04/2008 07:28 PM 51kb VoiceGuide.url

 

libsofia_sip_ua.dll and pthreadVC2.dll are new files in the VG7Fix.

 

I can't allow the remote access without authorization, today is friday, in monday I can give authorization for this.

But, I'm goin to work the weekend, I appreciate that You can help me before monday.

 

 

Please help me, I am desperate.

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About a preview question, in the last test I use only rfc2833.

 

--------------------------------------

 

 

The error in the VG7 Fix is....

 

Load_Layer_ktTel_DynamicPlugins Assembly.Load

The specified module could not be found. (Exception from HRESULT: 0x8007007E)

at System.Reflection.Assembly.nLoadFile(String path, Evidence evidence)

at System.Reflection.Assembly.LoadFile(String path)

mainLoad ktTel_ApiMode : Object reference not set to an instance of an object.

 

 

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I use HMP 3.0

HMP3.0 drivers are updated every few weeks or so. Which "Service Update" of the drivers is installed on the system?

Common recent version is SU157, and SU171 was just released. see: http://www.dialogic.com/support/helpweb/dx...p30/default.htm

 

Either the SU157 or SU171 is fine. VG has been tested with both.

 

Another thing to check would be to use the depends.exe app from here: http://www.dependencywalker.com/ and use it to view the ktTelDialogicHMP30.dll (from the new version).

This should indicate which DLLs that it requires are not present on the system. Please post a screenshot of dependencywalker's reported dll dependency list of the ktTelDialogicHMP30.dll.

 

Also, which version of Windows are you using?

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After user upgraded to latest HMP and latest version of VoiceGuide now available we have received this email note:

 

Thanks, that was the solution for the DTMF tones detection, now I can make calls and always the IVR receive the DTMF tones.

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