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#43317 Config Timeout

Posted by SupportTeam on 18 January 2018 - 05:18 PM in General

Could you please post the VG.INI file and the vgEngine trace that captures the latest VoiceGuide service restart.




#43302 How To Separate Lines For Testing?

Posted by SupportTeam on 09 January 2018 - 09:07 PM in General

You can set what script is to be ran on each of the analog ports, or E1/T1 channels. This is done in VoiceGuide's Config.xml file.

 

Also, a common approach is to jump to a different script based on certain CallerID. Please see the "Branching to other scripts and calling subscrips" section in this Help file page: http://www.voiceguid.../html/paths.htm

 

You can use an Evaluate Expression module to evaluate this expression:

 

"$RV_CIDNUMBER"

 

and then have a path that matches the test caller's CallerID point to a new script.

 

eg. if you would like calls from number 5551234 to run the Test script then this path can be used in this Evaluate Expression module:

 

on {5551234} goto [myTestCallflow.vgs|myStartModule]

 

and these paths can be used to forward all other calls to the 'production' part of the callflow:

 

on {Success} goto [nextModule]

on {Fail} goto [nextModule]

 

The "On {Success}" path would be match all other CallerIDs (other then 5551234 that has a specific path for it already), and the "On {Fail}" path would match cases where no CallerID was supplied.




#43289 Pause On Answer Than Play Dtmf Code

Posted by SupportTeam on 06 January 2018 - 08:23 PM in General

If you do not want to set the tone as a global DISCONNECT type tone then you will need to check for presence of tone in each module individually.

 

There is no way to set a global 'on this tone goto this module regardless of where the callflow is up to currently'. Only 'DISCONNECT' type tones do that, and system will hangup  when a DISCONNECT tone is heard.

 

You can have a "on {DISCONNECT_MyToneName} goto [MyModule]" type paths. If the DISCONNECT tone occurs in module that has that path specified then that path will be taken instead.




#43285 Hmp Ip Media Server No Voice Response.

Posted by SupportTeam on 06 January 2018 - 03:27 PM in General

Instead of trying to test this system with soft-phone from another country, it may be just faster to connect this VoiceGuide server to the lines that it will be using.

 

Do you have connection details of the switch/service that will be used to route calls to this system? Will you have a direct IP routes SIP trunk, or will you need to SIP REGISTER the system with the switch/service ?




#43279 Hmp Ip Media Server No Voice Response.

Posted by SupportTeam on 05 January 2018 - 10:49 PM in General

Trace on 183.82.98.134 should show the SIP packets, as we see those are exchanged. Does 183.82.98.134 trace show SIP ?

 

Does 183.82.98.134 trace show RTP being sent out?

 

If its not showing either of the the above then does the 183.82.98.134 machine have multiple network interfaces? Perhaps the WireShark tracing was done on wrong interface?




#43277 Hmp Ip Media Server No Voice Response.

Posted by SupportTeam on 05 January 2018 - 09:44 PM in General

Attached WireShark trace was done on the VoiceGuide machine, right? (Win2012, 10.1.1.5).

 

Attached trace shows that 10.1.1.5 is sending RTP to caller (183.82.98.134), but 183.82.98.134 is not sending any RTP to 10.1.1.5 

(also the RTP sent is just silence...)

 

Did you do the WireShark trace on the 183.82.98.134 machine? What did that show? Can you post it here?

 

Can you try using MicroSIP softphonefor testing instead ( https://www.microsip.org/ )

 

Also, can you, for initial testing, have the softphone machine on the same network (ie, on  10.1.1.X network)? It looks like right now you are most likely having some firewall issues that prevent RTP from 183.82.98.134 reaching 10.1.1.5

 

Is 10.1.1.5 a separate physical machine, or a VM? If a VM then what VM is it running on?




#43274 Linphone On Virtual Vm\os

Posted by SupportTeam on 05 January 2018 - 07:10 PM in General

Screenshot shows the IP address of the Windows machine running VoiceGuide is: 10.0.2.15

 

Your Ubuntu ipconfig is reporting IP 192.168.0.3

Is 192.168.0.3 the machine that you are running the Linphone on?

 

Is the 10.0.2.15 VM running on the 192.168.0.3 system?

 

Can you ping that IP address from the machine that is running Linphone?

If the Windows machine can be pinged then all that you need to do is to enter 10.0.2.15 in the Linphone text box where it says "SIP Address or Phone Number"

 

 

 

We'd recommend setting up HMP+VoiceGuide on its own physical system, or a VM running on top of VmWare ESXi

 

Running HMP in the Oracle VirtualBox is not supported. It might work, or it might not.

The fact that it is not supported suggests that there will be problems running on it, but you may find that those problems only appear when running a larger number of simultaneous calls.




#43265 Service Stop After 1 Hour Run

Posted by SupportTeam on 03 January 2018 - 03:31 PM in General

ktTel trace shows that there is still a "serverResponseDataMismatch" error ( IPEC_REG_FAIL_serverResponseDataMismatch ) during the SIP Registration :

210 130954.703  4716           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=[]

WireShark shows the sent out Contact header was:

 

<sip:1004@192.168.111.235>

 

and the Yeastar responded with this Contact header in the response message:

 

<sip:1004@192.168.111.235:5060>;expires=59

 

 

please set the <LocalAlias> field in VoiceGuide's Config.xml to:

 

1004@192.168.111.235:5060

 

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

 

 

Please start WireShark before starting the VoiceGuide service, to capture the Registration messages, and post both ktTel and WireShark traces if still encountering issues.




#43263 Service Stop After 1 Hour Run

Posted by SupportTeam on 03 January 2018 - 03:01 PM in General

Please post the WireShark traces and the ktTel traces, as before. Then we can see what is happening on this system and comment.




#43261 Service Stop After 1 Hour Run

Posted by SupportTeam on 03 January 2018 - 08:59 AM in General

In VoiceGuide's Config.xml, in section <VoIP-Registration> please set the <LocalAlias> field to:

 

1004@192.168.111.235

 

the LocalAlias should be the local address, not a remote address.

 

The reason why the SIP Register did not work as far as HMP was concerned was because of the Contact: field mismatch between the Register request and response.

SIP Registrars usually just use the Contact: setting provided in the original Register request, but the Yeastar PBX decided to use a different setting for Contact: in its reply.

In situations where the SIP Registrar changes the Contact:  details we must change the LocalAlias to match.

 

(the actual Contact: field returned by the Yeastar PBX is 1004@192.168.111.235:5060 but just using 1004@192.168.111.235 should be enough)

 

 

Also please add this entry to the <VoIP-Registration> section:

 

<Expires>60</Expires>

 

This will allow to quickly see if the REGISTER requests are being resent, without waiting for an hour.

 

After changing the Config.xml please stop both the VoiceGuide service and the Dialogic HMP service, and after the Dialogic HMP service has fully stopped the please start the Dialogic service and then start the VoiceGuide service.

 

Starting WireShark before starting the VoiceGuide service will let you monitor the REGISTER messages and confirm if they are being regularly sent to keep VoiceGuide registered.

 

 

Another option is to set up a direct IP trunk in the PBX. With direct IP trunk there is no need to do any registration.

PBX will just send calls direct to VoiceGuide IP address and VoiceGuide will answer all calls arriving at its IP address.

 

 

.

177 232531.237  7876     fn    VoIPProvider_Register(protocol=SIP, reg_server=192.168.111.4, reg_client=1004@192.168.111.4, local_alias=1004@192.168.111.4:5060, sH323SupportedPrefixes=)210 232531.299  2912     ev    GCEV_SERVICERESP (board device)

211 232531.303  2912           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=[]
  • ws2.png



#43260 Service Stop After 1 Hour Run

Posted by SupportTeam on 03 January 2018 - 08:06 AM in General

Could you please .ZIP up and post all the files in this directory:

 

C:\Program Files (x86)\Dialogic\HMP\log




#43258 Service Stop After 1 Hour Run

Posted by SupportTeam on 02 January 2018 - 10:55 PM in General

Attached WireShark trace does not capture any startup REGISTER messages, and does not capture any calls. It just captures a series of OPTIONS/OK messages.

 

Please start the WireShark capture before starting the VoiceGuide service. This way any REGISTER messages that are sent out by VoiceGuide to register itself with the PBX will be captured. And please make a call into the system as well.

 

We can then see how the communications between the PBX and VoiceGuide is working.

 

No need to do capture for an hour, ending capture 5 minutes after end of the call into system is sufficient.

 

After finishing the trace please post the VoiceGuide ktTel trace as well (.ZIP up first also). We can then see both the SIP comms and VoiceGuide trace together,

 

Most likely the Registration between the VoiceGuide and this Yeastar S100 PBX is not fully set up. Traces will let us confirm and advise.

  • ws1.png



#43255 Hmp Ip Media Server No Voice Response.

Posted by SupportTeam on 02 January 2018 - 09:13 PM in General

Can you please do a WireShark capture of the SIP and RTP traffic.

 

Are you perhaps able to run the WireShark capture on both systems? ie. both the Win2012 server and Win10 soft-phone machine?

When we have both captures we can the see what packets are are seen at both ends.

 

Could it be some firewall issue?

 

You need to open ports 49100 and above (to say 50000) on both systems to allow RTP traffic (the sound data). see: http://www.voiceguid...dialogichmp.htm

 

Are you able to temporarily just disable the firewalls on both systems and do a test call?

 

When doing a WireShark capture you can use this in the Display Filter Expression text box (at top of WireShark):

 

sip || rtp

 

To save WireShark trace use the File->Export Specified Packets and make sure the 'Displayed' option is selected.




#43252 Hmp Ip Media Server No Voice Response.

Posted by SupportTeam on 02 January 2018 - 07:27 PM in General

Does this server have multiple network interfaces?

 

If yes then please stop the Dialogic HMP service (using DCM), select the card, go to Config -> Default IP, and select the 'first' IP interface. Then start the Dialogic HMP service again (using DCM) and try running the IP Media Server test again.




#43251 Hmp Ip Media Server No Voice Response.

Posted by SupportTeam on 02 January 2018 - 07:21 PM in General

Customer wrote:

 

We are using Windows server 2012 64bit operating system with Bria xlite softphone.

Please note we have not yet registered with sip service provider because it is still in the testing mode.

1.server 2012 IVR setup done which is located @ datacenter.

2.softphone is setup on my local machine windows 10 which is connected to public ip.

3.when I dial the with softphone can see this message on server side:-  IP Media server no voice response.

4. server 2012 operating system does not have sound card drivers setup is it ok ? OR IVR Needs sound card ?

  • ipmediaserver_error.jpg



#43244 Pause On Answer Than Play Dtmf Code

Posted by SupportTeam on 01 January 2018 - 09:02 AM in General

If you want to consider a certain tone to be a sign of a disconnect then it is best to define that tone as a 'DISCONNECT' type tone. Then the system will react to it and hangup automatically when that tone is heard.

 

Why do you need to turn off the DISCONNECT tone detection on a particular call?




#43243 Licence Is Opening As Evaluation

Posted by SupportTeam on 01 January 2018 - 08:58 AM in General

When I first boot up the PC the evaluation statement appears in Line Manager even though I have brought the Proffesional + Dialler.

 

Please contact sales@voiceguide.com regarding this, and forward VoiceGuide traces capturing VoiceGuide startup.

 

 

My other issue is after booting up I am getting no dail tone from the simple y spliiter to handset + Dialogic card setup.

 

You need to ensure that the Dial tone is present on line. if Dial tone is present on line, but not when the splitter is attached then it is likely that the splitter is faulty or the system that supplies the line does not support splitters. Some telephone PBXs/Switches will expect to drive only one telephone device on the line, not two.

 

 

But after a round of calls my handset dialtone come back, I dont understand how my dial tone is lost after booting the PC then comes back after some calls. If I disconnect from the PC the RJ45 cable after the initial boot my dial tone comes back.

 

You should speak to whatever company is supplying the line regarding this. Most likely you are only allowed to connect one device to the telephone line at a time.




#43240 Pause On Answer Than Play Dtmf Code

Posted by SupportTeam on 31 December 2017 - 09:30 AM in General

The campaign name is accessible within the script using the $RV:

 

$RV[OutDial_CampaignName]

 

( other often used OutDial family $RVs are: $RV[OutDial_RetriesLeft] $RV[OutDial_ID] $RV[OutDial_GUID] )

 

The path statements do not allow for Boolean expressions with in them, but to switch based on Campaign you can use an Evaluate Expression module etc. to check value of  $RV[OutDial_CampaignName] and switch accordingly.




#43238 Pause On Answer Than Play Dtmf Code

Posted by SupportTeam on 30 December 2017 - 12:27 PM in General

The guard time between calls can be set in VG.INI, section [VGDialer] entry: "AfterIdleWaitBeforeDial"

 

eg: to make time after end of last call and stat of new call 10 seconds set:

 

AfterIdleWaitBeforeDial=10

 

VoiceGuide service will need to be restarted to read in the new VG.INI settings.

 

 

The tone detection settings are set ConfigLine.xml file in VoiceGuide's \conf\ subdirectory. (file used can actually be set per lie by editing Config.xml)

 

VoiceGuide service will need to be restarted to read in the new tone settings.

 

it is not possible to turn off the tone detection on a per call basis, but if you rename the tone to say MYTONE_USER_1 then VoiceGuide will no longer regard it as a 'Disconnect' type tone and will not hang up the call automatically. You can have paths in script like this:

 

on {MYTONE_USER_1} goto [my hangup module]

 

to act on the tone being present.

 

and on calls on which you do not want to react to the MYTONE_USER_1 tone just use a script that does not have those paths.




#43237 Outgoing Call Gets Disconnected With New Sip

Posted by SupportTeam on 30 December 2017 - 12:21 PM in General

OK, Thanks for letting us know this is now resolved,




#43233 Pause On Answer Than Play Dtmf Code

Posted by SupportTeam on 30 December 2017 - 09:02 AM in General

I am using a Y Splitter on the line with a simple phone, when the Dialogic D/4PCI card is on and IVR is not dialing but in the start state, their is no tone on my simple phone, is that normal and why don't I get a dial tone.

 

There should be a dial tone. Not sure why you would not be hearing it.

 

 

  <OnAnswerMachine>DISABLE</OnAnswerMachine> Is this the correct line for the XML file to turn off calls to answer machines or should I just leave out?

 

<OnAnswerMachine>DISABLE</OnAnswerMachine> will stop VoiceGuide from trying to detect whether the answered call was answered by a live person or by answering machine. The 'Live Answer' script will be used in both cases.

 

Is it better to drop 1 Outdial xml file with 2 ports coded into it or 2 XMLs files with one file per port, I want to multi dial off 2 ports at the same time, is their a preferred method?

 

One outdial file with multiple dial entries in it. Can be more then 2. The calls will get queued and made as lines become available. If you only have 2 lines attached to a 4 port card then you can disable outbound dialing on the other 2 ports in the Config.xml

 

Is their a silence detection mode on outbound calls using the dialler? if so can I turn it off, I just want to hang up the call on timeout/timer, nothing else needed to complicate things.

 

Before call is connected there is no 'silence detection'.

 

For outbound calls is their a log file that you can see what terminated the call, for instance in the script I have a hangup module, can I tell from a log file if that triggered or was it something else, did the call make the duration I wanted it to run for, or did something else end the call.

 

There is a 'CallEvents' log that shows the path through the script that every call took. The CallEvents log is in VoiceGuide's \log\ subdirectory. A new CallEvents log is created every day and  holds all the calls made that day.

 

Each script also has log files - in 4 different formats:  .XML, .JSON, .CSV and .VGL

Those log files will also show you what path each call took though the script.

The .VGL log can be opened using the vgLogViewer.exe application that comes with VoiceGuide. All calls handled by the script are stored in those log files. Each log file stores the same information, just in different format.

 

There are also CDR logs in VoiceGuide's \cdr\ subdirectory. The CDR logs will show overall call length.

 

Please see:

 

http://www.voiceguid..._scriptlogs.htm

 

http://www.voiceguid...ml/log_cdrs.htm

 

How can I increase the time between auto dialled calls so I can check the tone signal, can this be done inside the Outdial file?

 

Do you mean the time between end of one call and start of next call on same line? or something else?

Could you please describe in more detail what you mean by "so I can check the tone signal," How would you check this?




#43217 Pause On Answer Than Play Dtmf Code

Posted by SupportTeam on 28 December 2017 - 04:42 PM in General

Would you like to not play anything after call answer and only play a DTMF 2 10 seconds after the script has started?

 

Please see attached script. It just plays a DTMF tone 10 seconds after the script is started.

 

This is done by specifying first module to play:

 

none

 

and then using a 'Timeout 10' path to move to the next module that plays a DTMF 2.

 

Also, does the called system play some message that will allow Dialogic to detect when call has been answered? When using analog systems you need some message playing to let system know call is answered. Please see: http://www.voiceguid...tcallanswer.htm




#43216 Windows 7 - Installation Problems

Posted by SupportTeam on 28 December 2017 - 04:25 PM in General

The Dialogic System Release should be listed in the "Programs and Features" list. Its entry is titled:

 

Dialogic® System Release 6.0 PCI

 

VoiceGuide does not prompt to install/uninstall Dialogic drivers. Sounds like you are installing Dialogic drivers.

 

If you are given the option to uninstall then this should be done. You need to uninstall Dialogic System Release drivers before installing HMP.

 

Is this a clean install of Win7? If not then are you able to re-format Hard Disk and install just Windows7 and the Dialogic HMP afterwards?




#43213 Windows 7 - Installation Problems

Posted by SupportTeam on 28 December 2017 - 09:28 AM in General

VoiceGuide can use VoIP, and handle all calls using SIP.

 

Just uninstall the Dialogic System Release drivers and uninstall VoiceGuide and then install Dialogic HMP drivers and then install VoiceGuide again (selecting 'VoIP' option at install time).

 

But the "doesn't recognize the script and load numbers to be dialed out" does not seem like an issue that is related to what platform is used. Can you describe in more detail what is the problem that you are experiencing?

 

Please note that the evaluation version will allow only a small number of outgoing calls before the VoiceGuide IVR service needs to be restarted. (after which you can make more outgoing calls).




#43212 Campaign Schedule Stalled

Posted by SupportTeam on 28 December 2017 - 08:35 AM in General

Did the license you purchase include the Dialer?

 

From http://www.voiceguid...em-features.htm :

 

Dialer Add-on is required whenever the system needs to make an outgoing call on a new line.