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Some Questions On The Scripting

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I have a couple questions on my voiceguide script.

 

1. I have it set up so that it plays an initial greeting, and when that times out it goes to the options menu. This is so that when I press # on the options menu to repeat the options, the system doesn't have to replay the initial greeting. But that means my options don't work when the initial greeting is played. Is there any way to make this work other than copying the paths over?

 

2. When I press # anywhere in my script to return to the main menu, I get disconnected instead of being returned to the menu. How do I fix that? I attached my script.

 

3. I have it so that from the main menu, if I press a 4-digit extension, it will transfer me to [Voicemail Box $RV[menumain]]. But it takes a really long time (like 5-7 seconds) before I'm transferred and hear the voicemail greeting. That's way too long - how can I fix this?

 

Thanks.

vivobrain.zip

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In each module you need to define the paths that you want to be valid in that module.

 

We will need to see the vgEngine traces to see why the system is hanging up.

 

You should use "*" to return the main menu. "*" is the accepted standard for returning to the top level menu. "#" is used to end input in a Get Numbers module.

 

 

3. I have it so that from the main menu, if I press a 4-digit extension, it will transfer me to [Voicemail Box $RV[menumain]]. But it takes a really long time (like 5-7 seconds) before I'm transferred and hear the voicemail greeting. That's way too long - how can I fix this?

You must have the Get Number module set to accept a variable length number. Set it to expect a number of maximum of 4 digits and the Success path will then be taken as soon as the 4th digit is pressed.

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In each module you need to define the paths that you want to be valid in that module.

 

We will need to see the vgEngine traces to see why the system is hanging up.

 

You should use "*" to return the main menu. "*" is the accepted standard for returning to the top level menu. "#" is used to end input in a Get Numbers module.

 

 

3. I have it so that from the main menu, if I press a 4-digit extension, it will transfer me to [Voicemail Box $RV[menumain]]. But it takes a really long time (like 5-7 seconds) before I'm transferred and hear the voicemail greeting. That's way too long - how can I fix this?

You must have the Get Number module set to accept a variable length number. Set it to expect a number of maximum of 4 digits and the Success path will then be taken as soon as the 4th digit is pressed.

 

Ok. I will send vgEngine.

 

To continue back to our question on PBX options, will Panasonic KX-TA624 or KX-TA824 work with Vonage trunk lines? Will they work with Voice Guide? You have mentioned KX-TDA100 to me in our previous postings.

 

 

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To continue back to our question on PBX options, will Panasonic KX-TA624 or KX-TA824 work with Vonage trunk lines? Will they work with Voice Guide?

VoiceGuide can be attached to any analog (or T1/E1 or VoIP) line/extension.

 

You will be able to connect VoiceGuide to the analog extensions of the PBXs you mention and any other PBX.

 

The PBXs you mention can accept analog lines as trunk lines, so you will not have any problems plugging the Vonage lines into it (after they are converted to analog using the Vonage supplied media gateway box)

 

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In each module you need to define the paths that you want to be valid in that module.

 

We will need to see the vgEngine traces to see why the system is hanging up.

 

You should use "*" to return the main menu. "*" is the accepted standard for returning to the top level menu. "#" is used to end input in a Get Numbers module.

 

 

3. I have it so that from the main menu, if I press a 4-digit extension, it will transfer me to [Voicemail Box $RV[menumain]]. But it takes a really long time (like 5-7 seconds) before I'm transferred and hear the voicemail greeting. That's way too long - how can I fix this?

You must have the Get Number module set to accept a variable length number. Set it to expect a number of maximum of 4 digits and the Success path will then be taken as soon as the 4th digit is pressed.

 

Ok. I will send vgEngine.

 

To continue back to our question on PBX options, will Panasonic KX-TA624 or KX-TA824 work with Vonage trunk lines? Will they work with Voice Guide? You have mentioned KX-TDA100 to me in our previous postings.

 

Also, if I get a PBX with a caller id card built in, is that useless since voiceguide already has a Caller ID function? Please clarify this.

 

And if I have a voicemail menu for employee Vm retrieval, how should I set it so that customers cannot accidentally stumble upon that menu?

 

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1. Do you know if Panasonic KX-TA624 and KX-TA824 support inband signaling? I checked the manuals and it seems like they support DTMF integration/signaling. Please let me know.

 

3. How can voiceguide display the caller id passed from the PBX while the user is still in the menu system (before transferring to a station)?

 

2. And if I have a voicemail menu for employee Vm retrieval, how should I set it so that customers cannot accidentally stumble upon that menu?

 

 

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1. Do you know if Panasonic KX-TA624 and KX-TA824 support inband signaling? I checked the manuals and it seems like they support DTMF integration/signaling. Please let me know.

Don’t thinks they do. Best to check with PBX supplier.

 

3. How can voiceguide display the caller id passed from the PBX while the user is still in the menu system (before transferring to a station)?

Where do you need the CallerID displayed?

 

2. And if I have a voicemail menu for employee Vm retrieval, how should I set it so that customers cannot accidentally stumble upon that menu?

That's why you have a PINs for the mailboxes, so even if caller accidentally selects a voicemail box, they will be unable to listen to the messages within it.

 

Here attached are the vgEngine logs to diagnose why the system hangs up when I press #.

The log files are quire long. Which call should we be looking at? Please give time of call and we can then see what happened.

Can you reproduce this? Can you describe when do you press the "#"? Is it possible that you are pressing it in a Get Numbers module before entering any other data and the "Fail" path in that module is not defined?

 

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1. Do you know if Panasonic KX-TA624 and KX-TA824 support inband signaling? I checked the manuals and it seems like they support DTMF integration/signaling. Please let me know.

Don’t thinks they do. Best to check with PBX supplier.

 

3. How can voiceguide display the caller id passed from the PBX while the user is still in the menu system (before transferring to a station)?

Where do you need the CallerID displayed?

 

2. And if I have a voicemail menu for employee Vm retrieval, how should I set it so that customers cannot accidentally stumble upon that menu?

That's why you have a PINs for the mailboxes, so even if caller accidentally selects a voicemail box, they will be unable to listen to the messages within it.

 

Here attached are the vgEngine logs to diagnose why the system hangs up when I press #.

The log files are quire long. Which call should we be looking at? Please give time of call and we can then see what happened.

Can you reproduce this? Can you describe when do you press the "#"? Is it possible that you are pressing it in a Get Numbers module before entering any other data and the "Fail" path in that module is not defined?

 

 

1. For the caller ID, I'm not sure how I want it to be displayed. How is it usually done? I just want some way of seeing the caller ID while they are still in the menu options.

 

2. One more thing - can I configure the voicemail so that a voicemail left to a specific employee is emailed to that specific employee's email address? From the voicemail manager's email tab it seems like it's a global email where all voicemails are sent to.

 

 

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1. For the caller ID, I'm not sure how I want it to be displayed. How is it usually done?

It's usually not done. Why would you want it displayed?

 

I just want some way of seeing the caller ID while they are still in the menu options

In v6 you can use a VBScript to display a MessageBox - and you can specify for the MessageBox to contain the CallerID.

 

2. One more thing - can I configure the voicemail so that a voicemail left to a specific employee is emailed to that specific employee's email address? From the voicemail manager's email tab it seems like it's a global email where all voicemails are sent to.

The email address (and forwarding phone number etc) are on a per-voicemail box basis. You can set a different email address, phone number etc for each voicemail box.

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1. Okay. I'm having some trouble setting the Pager Tel/Msg and Forward Tel to notify a person that they have a voicemail. In both cases, the system calls my specified number but when I pick up, I get silence. Nothing is said. When I put a message in the Pager Msg folder, the system doesn't read that. Just silence.

 

2. In the Voicemail Retrieve Menu, how do I make it so that before or after the messages are played back, the time/date the message was left is read out? If that cannot be done, is there any other way that an employee can know the date/time of a voicemail left to them?

 

Thanks.

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1. Okay. I'm having some trouble setting the Pager Tel/Msg and Forward Tel to notify a person that they have a voicemail. In both cases, the system calls my specified number but when I pick up, I get silence. Nothing is said. When I put a message in the Pager Msg folder, the system doesn't read that. Just silence.

Please post a vgEngine trace showing the outgoing call.

 

2. In the Voicemail Retrieve Menu, how do I make it so that before or after the messages are played back, the time/date the message was left is read out?

You can just modify the voicemail scripts directly and insert a module that plays the time/date before the module hat plays the message.

 

is there any other way that an employee can know the date/time of a voicemail left to them?

See: http://www.voiceguide.com/vghelp/source/ht...stemmanager.htm, Retrieve Messages Menu, option 7.

 

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1. Okay. I'm having some trouble setting the Pager Tel/Msg and Forward Tel to notify a person that they have a voicemail. In both cases, the system calls my specified number but when I pick up, I get silence. Nothing is said. When I put a message in the Pager Msg folder, the system doesn't read that. Just silence.

Please post a vgEngine trace showing the outgoing call.

 

2. In the Voicemail Retrieve Menu, how do I make it so that before or after the messages are played back, the time/date the message was left is read out?

You can just modify the voicemail scripts directly and insert a module that plays the time/date before the module hat plays the message.

 

is there any other way that an employee can know the date/time of a voicemail left to them?

See: http://www.voiceguide.com/vghelp/source/ht...stemmanager.htm, Retrieve Messages Menu, option 7.

 

I've posted the vgEngine traces.

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The last set of traces posted contains many calls. Which of the calls are we supposed to be looking at? Please advice the date and time of the call and we can then have a look at it.

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Here I've attached a new vgEngine log with just one call showing the problem with Pager tel & msg.

 

Please let me know if you have a chance to trace the problem. I am waiting to program further upon receipt of your response, so I'd appreciate if you can take a look. Thanks.

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Trace shows that the phone number to forward the voicemail recording to was not specified.

 

Trace also shows that the paging server that answer the call did not play any message after the call was answered. Does the paging service that you are using do not play any pre-recorded message prompting caller to enter the pager message? Which pager message service did you use in this test?

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Trace shows that the phone number to forward the voicemail recording to was not specified.

 

Trace also shows that the paging server that answer the call did not play any message after the call was answered. Does the paging service that you are using do not play any pre-recorded message prompting caller to enter the pager message? Which pager message service did you use in this test?

 

1. I want to have it so that if an employee gets a voicemail, their cellphone number is called and when they pick up, they will hear a message saying "You have new voicemail." What is the best way to do this?

 

2. I just put a regular cellphone number in the pager tel field. It's a cellphone, not a pager. What is the pager tel/msg for then (since nobody has pagers anymore)?

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Another question: here I've attached some logs showing that when I put a number into Forward Tel. field in the Voicemail manager, it calls that number but when I pick up there's just silence. Please tell me how to fix htis.

 

Thanks.

log.zip

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1. I want to have it so that if an employee gets a voicemail, their cellphone number is called and when they pick up, they will hear a message saying "You have new voicemail." What is the best way to do this?

To get that precise announcement you would need to modify the voicemail scripts. Right now the default voicemail delivery script will announce how many new messages are in the voicemail box.

 

2. I just put a regular cellphone number in the pager tel field. It's a cellphone, not a pager. What is the pager tel/msg for then (since nobody has pagers anymore)?

The numeric pager message is still included for legacy users. Sounds like this is not what you require, so you can just ignore that option.

 

I put a number into Forward Tel. field in the Voicemail manager, it calls that number but when I pick up there's just silence.

Please download the latest VG v7 which is now available from our WWW, and post traces from that version if problem still persists for you.

 

When answering a call please make sure to say "Hello" or something similar. The Dialogic analog cards need to hear "Hello"/etc to know that the call has been answered.

 

This is the main reason why it’s better to use ISDN for outgoing calls. A T1 ISDN will be able to detect immediately when the recipient of the call picks up the handset to answer the call.

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1. I want to have it so that if an employee gets a voicemail, their cellphone number is called and when they pick up, they will hear a message saying "You have new voicemail." What is the best way to do this?

To get that precise announcement you would need to modify the voicemail scripts. Right now the default voicemail delivery script will announce how many new messages are in the voicemail box.

 

2. I just put a regular cellphone number in the pager tel field. It's a cellphone, not a pager. What is the pager tel/msg for then (since nobody has pagers anymore)?

The numeric pager message is still included for legacy users. Sounds like this is not what you require, so you can just ignore that option.

 

I put a number into Forward Tel. field in the Voicemail manager, it calls that number but when I pick up there's just silence.

Please download the latest VG v7 which is now available from our WWW, and post traces from that version if problem still persists for you.

 

When answering a call please make sure to say "Hello" or something similar. The Dialogic analog cards need to hear "Hello"/etc to know that the call has been answered.

 

This is the main reason why it’s better to use ISDN for outgoing calls. A T1 ISDN will be able to detect immediately when the recipient of the call picks up the handset to answer the call.

 

Ah, the key is to say hello. Thanks, it works now.

 

I have two more questions:

 

1.

For incoming fax, how do I set it up so that each employee can receive faxes to their individual boxes?

 

2.

I set it up so that voicemail will be automatically emailed to the employee. The email is sent, but the email subject is "Voicemail received from , at 21:14 on MMMM 2/25/2008 9:14:38 PM" and the message is

 

"Voicemail received from: ,

on 2/25/2008 9:14:38 PM MMMM at 21:14:38

 

Voicemail box: 8465"

 

Why is there no "from" information and what is that MMMM after the PM?

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Ah, the key is to say hello. Thanks, it works now.

See: http://www.voiceguide.com/vghelp/source/ht...tcallanswer.htm

 

For incoming fax, how do I set it up so that each employee can receive faxes to their individual boxes?

Fax is not supported in the downloadable version of VoiceGuide. If you require fax support please contact sales@voiceguide.com Fax is now supported in v7. Please see: http://www.voiceguide.com/vghelp/source/html/fax_intro.htm

 

Why is there no "from" information and what is that MMMM after the PM?

Looks like there was no CallerID on this call.

 

You may want to read: http://www.voiceguide.com/vghelp/source/ht...stemmanager.htm , see the section on "Email Forwarding".

Edited by SupportTeam

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But what is the MMMM after the PM?

In the version you have now there is a bug with $RV_MONTHNAME variable.

 

Easiest resolution is to change the subject and body templates as described in Help file section we referenced.

 

The $RV_MONTHNAME variable will be fixed in next release.

 

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But what is the MMMM after the PM?

In the version you have now there is a bug with $RV_MONTHNAME variable.

 

Easiest resolution is to change the subject and body templates as described in Help file section we referenced.

 

The $RV_MONTHNAME variable will be fixed in next release.

 

Ok. We will give it a try.

 

Also, does Voice Guide support "Music on hold" while caller is put on hold for the next available person to answer the call? How was it implemented?

 

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Also, does Voice Guide support "Music on hold" while caller is put on hold for the next available person to answer the call? How was it implemented?

You can just use a Play module to play a sound file before transferring the call to the extension/number.

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Your solution for playing music on hold is a bit weird - doesn't that just play a piece of music *before* transferring them, not while they are actually waiting? In the Transfer Call Module, under the Dial and Conference tab, there is this field to specify the On-Hold Sound File. What is that for then, if not for on-hold music?

 

Thanks.

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The "On-Hold" sound file is played only when "Dial and Conference" transfers are used.

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Hi,

 

My PBX, a Panasonic KX-TA824 arrived, and I'm attaching a proprietary phone (Panasonic KX-T7731) to one of the extensions. I hooked up my Dialogic card to one of the extensions.

 

When I use a call transfer module inside VG to transfer my call to the station, I can hear some tones as if the system is dialing, then it disconnects me. Please help me solve this. All the log files are attached.

log.zip

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Thanks, that works. I'm running into another problem.

 

A diagram is attached of the current setup I have. I'm using VONAGE for my two CO lines that go into the PBX. Vonage's Call Hunting feature is turned on so that when a customer tries to dial into a busy line, they get automatically transferred to another Vonage line.

 

When a customer calls in from Vonage line 1, which goes through CO1 and Station 1 on the PBX, Voiceguide picks up fine. But when they call in from Vonage line 2 (or are transferred there), it goes through CO2 and Station 2 on the PBX, but Voiceguide never picks up.

 

Please help, and you can refer to the diagram attached for more detail of my current setup.

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Thanks, that works. I'm running into another problem.

 

A diagram is attached of the current setup I have. I'm using VONAGE for my two CO lines that go into the PBX. Vonage's Call Hunting feature is turned on so that when a customer tries to dial into a busy line, they get automatically transferred to another Vonage line.

 

When a customer calls in from Vonage line 1, which goes through CO1 and Station 1 on the PBX, Voiceguide picks up fine. But when they call in from Vonage line 2 (or are transferred there), it goes through CO2 and Station 2 on the PBX, but Voiceguide never picks up.

 

Please help, and you can refer to the diagram attached for more detail of my current setup.

 

Here is the attachment.

post-2066-1204504268_thumb.jpg

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But when they call in from Vonage line 2 (or are transferred there), it goes through CO2 and Station 2 on the PBX, but Voiceguide never picks up.

Does the Ext2 ring when the second call arrives? Try plugging the Ext 2 line into an analog handset instead of the Dialogic card and see if the analog handset rings.

 

Also try changing the lines attached to the PBX around to make sure the problem is not with the Dialogic card ports themselves.

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But when they call in from Vonage line 2 (or are transferred there), it goes through CO2 and Station 2 on the PBX, but Voiceguide never picks up.

Does the Ext2 ring when the second call arrives? Try plugging the Ext 2 line into an analog handset instead of the Dialogic card and see if the analog handset rings.

 

Also try changing the lines attached to the PBX around to make sure the problem is not with the Dialogic card ports themselves.

 

Yes, Ext 2 rings when I try plugging in an analog handset. I tried using PBXpert but ran into a few problems - I've attached images of my process here.

 

Please view the images in order, from 1.gif to 6.gif - it shows my PBXpert errors.

post-2066-1204507166_thumb.jpg

post-2066-1204507177_thumb.jpg

post-2066-1204507181_thumb.jpg

post-2066-1204507184_thumb.jpg

post-2066-1204507189_thumb.jpg

post-2066-1204507194_thumb.jpg

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Yes, Ext 2 rings when I try plugging in an analog handset.

Have you tried changing the lines attached to the PBX around to make sure the problem is not with the Dialogic card ports themselves.

 

The PBXpert shows that it's having problems controlling the second port on the card. Suggest you try another port and stop using port 2 on that card.

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A couple more questions:

 

1. When customers call in, they should be able to dial the extension of the party they are looking for. Then they should be transferred to the party and if it's not picked up, go to voicemail. The employee should be able to set it so that when their extension is dialed, the call goes to voicemail right away, or goes to an external cellphone right away. Please let me know how to implement these features.

 

2. I'm using Dialogic D4PCIUF with the PBX. Can I set it up so that when a certain extension is called, it will dial out to a certain cellphone number instead of a station?

 

3. For my PBX, the default extension numbers are 101 for Jack 1, 102 for Jack 2, etc. How do I make it so that they are 4-digit and start with 8 (e.g. 8765 for Jack 1...)

 

Thanks.

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1. When customers call in, they should be able to dial the extension of the party they are looking for. Then they should be transferred to the party and if it's not picked up, go to voicemail. The employee should be able to set it so that when their extension is dialed, the call goes to voicemail right away, or goes to an external cellphone right away. Please let me know how to implement these features.

Same question was asked here already: http://voiceguide.com/forums/index.php?showtopic=5445

 

2. I'm using Dialogic D4PCIUF with the PBX. Can I set it up so that when a certain extension is called, it will dial out to a certain cellphone number instead of a station?

This is something that the PBX would do - it if supports that feature. If the PBX does not support that feature then you will need to use a D/41JCT card and do a “Dial and Conference” transfer direct to the external number.

 

3. For my PBX, the default extension numbers are 101 for Jack 1, 102 for Jack 2, etc. How do I make it so that they are 4-digit and start with 8 (e.g. 8765 for Jack 1...)

Change the extension numbers in the PBX - if the PBX allows it... some PBX allow you to set the extension numbers, and some do not. Most lower end PBXs do not let you change extension numbers.

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2. I'm using Dialogic D4PCIUF with the PBX. Can I set it up so that when a certain extension is called, it will dial out to a certain cellphone number instead of a station?

This is something that the PBX would do - it if supports that feature. If the PBX does not support that feature then you will need to use a D/41JCT card and do a “Dial and Conference” transfer direct to the external number.

 

 

 

3. For my PBX, the default extension numbers are 101 for Jack 1, 102 for Jack 2, etc. How do I make it so that they are 4-digit and start with 8 (e.g. 8765 for Jack 1...)

Change the extension numbers in the PBX - if the PBX allows it... some PBX allow you to set the extension numbers, and some do not. Most lower end PBXs do not let you change extension numbers.

 

 

2. What feature is this called so I can look in the manuals of the PBX? Or Can you tell me where I should be looking in the manuals?

 

3. I am not sure if KX-TA824 can or cannot allow change of extension numbering. Do you know that?

Assuming it cannot, can I program it so that when a user calls "8465" (an extension we announce to customer), Voice Guide will convert it to the PBX's internal extension number (8645 goes to "103") and transfer the call to station 103?

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2. What feature is this called so I can look in the manuals of the PBX?

It may be called "Forwarding to external number" or something line that. Most low end PBXs will not have this feature. You need to speak with PBX suppliers to see which ones do.

 

3. I am not sure if KX-TA824 can or cannot allow change of extension numbering. Do you know that?

Dontt know. You need to speak with PBX supplier.

 

Assuming it cannot, can I program it so that when a user calls "8465" (an extension we announce to customer), Voice Guide will convert it to the PBX's internal extension number (8645 goes to "103")

Yes, that's possible. You'd need to setup some sort of translation mechanism within the script. Either a database lookup or a translation table encoded in a VBScript etc.

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Thanks, that works. I'm running into another problem.

 

A diagram is attached of the current setup I have. I'm using VONAGE for my two CO lines that go into the PBX. Vonage's Call Hunting feature is turned on so that when a customer tries to dial into a busy line, they get automatically transferred to another Vonage line.

 

When a customer calls in from Vonage line 1, which goes through CO1 and Station 1 on the PBX, Voiceguide picks up fine. But when they call in from Vonage line 2 (or are transferred there), it goes through CO2 and Station 2 on the PBX, but Voiceguide never picks up.

 

Please help, and you can refer to the diagram attached for more detail of my current setup.

 

Here is the attachment.

 

I'm still running into this problem. I changed Dialogic cards to a D/120JCT and line 2 still is not picked up by Voiceguide. I programmed the PBX to transfer CO2 immediately to Station2, just like CO1 transfers immediately to Station1.

 

It's already proven that the Dialogic card is not the problem, since a new one doesn't fix the problem.

 

I've attached the diagram again of my current setup - can you confirm that this is right or tell me how to change it?

 

Thanks.

post-2066-1204614602_thumb.jpg

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Could you please post a copy of VoiceGuide's Trace Logs which captures the system startup and the incoming call, this will allow us to see what happened.

 

When posting traces/scripts please .ZIP them up and post them as attachments.

 

The trace would show whether the Dialogic card is reporting Ringing on that line or not.

 

Running a series of tests swapping the lines/ports between each test should let you deduce whether the problem is in the PBX or Wiring or Dialogic or VoiceGuide.

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Could you please post a copy of VoiceGuide's Trace Logs which captures the system startup and the incoming call, this will allow us to see what happened.

 

When posting traces/scripts please .ZIP them up and post them as attachments.

 

The trace would show whether the Dialogic card is reporting Ringing on that line or not.

 

Running a series of tests swapping the lines/ports between each test should let you deduce whether the problem is in the PBX or Wiring or Dialogic or VoiceGuide.

 

I've attached the files. Please help me diagnose this.

 

Also, I have a Dialogic D/120JCT card and outbound dialing to a cellphone does not work - this is in the log files as well. Please take a look, thanks.

log.zip

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Another question: how do I increase the delay before the system thinks that the user won't type in anymore numbers?

 

For example, I have an extension that is 103. But pressing 0 goes to voicemail. So if a user types in 103 with hesitations, the system will ignore the 1, recognize the zero, and go to voicemail.

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I have a Dialogic D/120JCT card and outbound dialing to a cellphone does not work

Trace shows that the outgoing leg of the Dial and Conference call was made on port 2, and that number 4254588464 was dialed. I suspect you need to place a "0," before the dialed number to make the call go out on external line, just like you would dial on an external line when placing a call from the telephone handset attached to line.

 

234648.816 4 1 dial found free dev=2 (LineId=2)

234648.822 4 2 ktTel_MakeCall([4254588464],CONNECT_IMMEDIATELY, 60,0,0,,<calltype>DialAndConf</calltype>) call

234648.838 4 2 ktTel_MakeCall => []

234648.839 4 2 state dialing 4254588464 (for conference with dxxxB1C1)

 

Alternatively you should try specifying that line 3 be used for the outgoing calls.

 

The 0303_0014_vgEngine trace shows calls arriving on lines 1 and 3 but the Dialogic card does not report any Rings for any incoming calls on line 2.

 

Another question: how do I increase the delay before the system thinks that the user won't type in anymore numbers?

Use a timeout path. See: http://www.voiceguide.com/vghelp/source/ht...odgetnumber.htm

 

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1. For dialing out, I'll try adding a 0, but I'm not sure if that will work because when I'm dialing an external number from a telephone handset attached to the PBX, I don't need to put a 0 in. Is there any other way?

 

2. For the other problem of Line 2 not picking up, are you saying that port 2 of the dialogic card is not ringing? Can you clarify this and how to fix this?

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Neither of the two traces supplied showed the Dialogic card reporting hearing any Ringing on line 2. Only lines 1 and 3.

 

The outgoing call placed to 4254588464 was placed on line 2 (as it was the next available line).

 

I suspect what is going on here is that you have not wired up the card properly and the two lines you have are wired up to channels 1 and 3 on the card. Refer to the wiring diagram on the installation notes for the D/120JCT.

 

I'd try specifying that lines 3 be used for outgoing calls in the transfer module and see if that line can place the outgoing calls OK. Type this in the Line Selection box in the transfer module:

 

,3,

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Okay, I think that explains why Dialout is not working. But this doesn't explain why the Dialogic card/VG is not picking up on Jack 2 of the PBX, right?

 

Because even if my phone line from Jack 2 of the PBX is plugged into port 3 of the Dialogic card, Voiceguide should still pick up. Please let me know what's going on, thanks.

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The trace showed that VoiceGude was answering calls on lines/ports 1 and 3, and playing sound files after answering.

 

From 0304_0014_vgEngine.txt:

 

Call on line 1:

 

001426.157 9 1 callstate: OFFERING crn=0,ev=2,LINECALLSTATE_OFFERING,0,0

001426.157 9 1 vgEngine v7.0.2985.21127 (Tue 04/03/2008 11:44:14.64)

001426.157 9 1 FireEvent_RemotingIf_VgEngine OFFERING 0

001426.168 9 1 AnswerTheCallIfAllowed iIvrDev=1, strDlgcDevName_Network=dxxxB1C1

001426.169 9 1 rings=0, min rings before answer=0 (iCallerIdHasArrived=0)

001426.174 9 1 ev ring 1, hCall=1 vgEngine v7.0.2985.21127 (Tue 04/03/2008 11:44:14.64)

001426.174 9 1 FireEvent_RemotingIf_VgEngine RING 1

001426.174 9 1 time since last ring : 0

001426.174 9 1 state Ring 1

...

 

Call on line 3:

 

001443.834 9 3 callstate: OFFERING crn=0,ev=2,LINECALLSTATE_OFFERING,0,0

001443.834 9 3 vgEngine v7.0.2985.21127 (Tue 04/03/2008 11:44:14.64)

001443.834 9 3 FireEvent_RemotingIf_VgEngine OFFERING 0

001443.834 9 3 AnswerTheCallIfAllowed iIvrDev=3, strDlgcDevName_Network=dxxxB1C3

001443.834 9 3 rings=0, min rings before answer=0 (iCallerIdHasArrived=0)

001443.834 9 3 ev ring 1, hCall=3 vgEngine v7.0.2985.21127 (Tue 04/03/2008 11:44:14.64)

001443.835 9 3 FireEvent_RemotingIf_VgEngine RING 1

001443.835 9 3 time since last ring : 0

001443.835 9 3 state Ring 1

001443.835 9 3 AnswerTheCallIfAllowed iIvrDev=3, strDlgcDevName_Network=dxxxB1C3

001443.835 9 3 rings=1, min rings before answer=0 (iCallerIdHasArrived=0)

001443.835 9 3 should answer as number of rings reached

001443.835 9 3 rv clear all

001443.835 9 3 LineState ResetAtStartOfNewCall

001443.835 9 3 Set_hCall crn=3, called from AnserThisCallIfAllowed

001443.835 9 3 start script after going offhook

001443.836 9 3 answering the call (picking up the line)

001443.836 9 3 timer clear (force=True)

001443.838 9 3 ktTel_LineAnswerCall =>

001443.838 9 3 ev Dialogic 134,TDX_CST, crn=0x0 (0), 0,0,0,DE_RINGS,ET_RON,

001444.417 9 3 ev Dialogic 135,TDX_SETHOOK, crn=0x0 (0), 0,0,0,DX_OFFHOOK,CALL_INBOUND,

001444.417 9 3 ev CallState sEvent=LINECALLSTATE_CONNECTED, hDevice=0, iEvent=2 ,256,0,,,]

001444.418 9 3 callstate: CONNECTED crn=0,ev=2,LINECALLSTATE_CONNECTED,1,0

001444.418 9 3 FireEvent_RemotingIf_VgEngine CONNECTED 0

001444.418 9 3 Set_hCall crn=1, called from LINECALLSTATE_CONNECTED

001444.418 9 3 Inband detection not enabled

001444.418 9 3 StartLoadedVgs at 2008-03-04 00:14:44 7.0.2985.21127 (Tue 04/03/2008 11:44:14.64)

001444.419 9 3 StartLoadedVgs LineState.iVgsIdx=1

001444.419 9 3 rvns add [scriptPath]{C:\Users\Joyce\Documents\ivr\}

001444.419 9 3 rvns add [scriptsPath]{C:\Users\Joyce\Documents\ivr\}

001444.419 9 3 set sScriptToRunOnHangup=[] in StartLoadedVgs

001444.419 9 3 rv add [$RV_STARTTIME]{2008-03-04 00:14:44}

001444.419 9 3 rv add [$RV_DEVICEID]{3}

001444.420 9 3 rv add [DlgcVoice]{dxxxB1C3}

001444.420 9 3 rv add [DlgcNetwork]{dxxxB1C3}

001444.423 9 3 rvns add [PathApp]{C:\Program Files\VoiceGuide\}

001444.423 9 3 rvns add [scriptsPath]{C:\Users\Joyce\Documents\ivr\}

001444.423 9 3 timer clear (force=False)

001444.423 9 3 RunModule start [Get Numbers,[greeting],13]

001444.423 9 3 state [greeting] Number Input

001444.424 9 3 PrepareAndPlayWavAndTts start iModuleIdx=13, iPlayFileIdx=1, bMakeTtsFile=True, sSoundFileIs=

001444.424 9 3 PrepareAndPlayWavAndTTS sPlayFile=recordings\greeting.wav

001444.424 9 3 state [greeting] Playing wav (recordings\greeting.wav)

001444.424 9 3 timer clear (force=False)

001444.433 9 3 play start ok, playId=650122

...

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The trace showed that VoiceGude was answering calls on lines/ports 1 and 3, and playing sound files after answering.

 

From 0304_0014_vgEngine.txt:

 

Call on line 1:

 

001426.157 9 1 callstate: OFFERING crn=0,ev=2,LINECALLSTATE_OFFERING,0,0

001426.157 9 1 vgEngine v7.0.2985.21127 (Tue 04/03/2008 11:44:14.64)

001426.157 9 1 FireEvent_RemotingIf_VgEngine OFFERING 0

001426.168 9 1 AnswerTheCallIfAllowed iIvrDev=1, strDlgcDevName_Network=dxxxB1C1

001426.169 9 1 rings=0, min rings before answer=0 (iCallerIdHasArrived=0)

001426.174 9 1 ev ring 1, hCall=1 vgEngine v7.0.2985.21127 (Tue 04/03/2008 11:44:14.64)

001426.174 9 1 FireEvent_RemotingIf_VgEngine RING 1

001426.174 9 1 time since last ring : 0

001426.174 9 1 state Ring 1

...

 

Call on line 3:

 

001443.834 9 3 callstate: OFFERING crn=0,ev=2,LINECALLSTATE_OFFERING,0,0

001443.834 9 3 vgEngine v7.0.2985.21127 (Tue 04/03/2008 11:44:14.64)

001443.834 9 3 FireEvent_RemotingIf_VgEngine OFFERING 0

001443.834 9 3 AnswerTheCallIfAllowed iIvrDev=3, strDlgcDevName_Network=dxxxB1C3

001443.834 9 3 rings=0, min rings before answer=0 (iCallerIdHasArrived=0)

001443.834 9 3 ev ring 1, hCall=3 vgEngine v7.0.2985.21127 (Tue 04/03/2008 11:44:14.64)

001443.835 9 3 FireEvent_RemotingIf_VgEngine RING 1

001443.835 9 3 time since last ring : 0

001443.835 9 3 state Ring 1

001443.835 9 3 AnswerTheCallIfAllowed iIvrDev=3, strDlgcDevName_Network=dxxxB1C3

001443.835 9 3 rings=1, min rings before answer=0 (iCallerIdHasArrived=0)

001443.835 9 3 should answer as number of rings reached

001443.835 9 3 rv clear all

001443.835 9 3 LineState ResetAtStartOfNewCall

001443.835 9 3 Set_hCall crn=3, called from AnserThisCallIfAllowed

001443.835 9 3 start script after going offhook

001443.836 9 3 answering the call (picking up the line)

001443.836 9 3 timer clear (force=True)

001443.838 9 3 ktTel_LineAnswerCall =>

001443.838 9 3 ev Dialogic 134,TDX_CST, crn=0x0 (0), 0,0,0,DE_RINGS,ET_RON,

001444.417 9 3 ev Dialogic 135,TDX_SETHOOK, crn=0x0 (0), 0,0,0,DX_OFFHOOK,CALL_INBOUND,

001444.417 9 3 ev CallState sEvent=LINECALLSTATE_CONNECTED, hDevice=0, iEvent=2 ,256,0,,,]

001444.418 9 3 callstate: CONNECTED crn=0,ev=2,LINECALLSTATE_CONNECTED,1,0

001444.418 9 3 FireEvent_RemotingIf_VgEngine CONNECTED 0

001444.418 9 3 Set_hCall crn=1, called from LINECALLSTATE_CONNECTED

001444.418 9 3 Inband detection not enabled

001444.418 9 3 StartLoadedVgs at 2008-03-04 00:14:44 7.0.2985.21127 (Tue 04/03/2008 11:44:14.64)

001444.419 9 3 StartLoadedVgs LineState.iVgsIdx=1

001444.419 9 3 rvns add [scriptPath]{C:\Users\Joyce\Documents\ivr\}

001444.419 9 3 rvns add [scriptsPath]{C:\Users\Joyce\Documents\ivr\}

001444.419 9 3 set sScriptToRunOnHangup=[] in StartLoadedVgs

001444.419 9 3 rv add [$RV_STARTTIME]{2008-03-04 00:14:44}

001444.419 9 3 rv add [$RV_DEVICEID]{3}

001444.420 9 3 rv add [DlgcVoice]{dxxxB1C3}

001444.420 9 3 rv add [DlgcNetwork]{dxxxB1C3}

001444.423 9 3 rvns add [PathApp]{C:\Program Files\VoiceGuide\}

001444.423 9 3 rvns add [scriptsPath]{C:\Users\Joyce\Documents\ivr\}

001444.423 9 3 timer clear (force=False)

001444.423 9 3 RunModule start [Get Numbers,[greeting],13]

001444.423 9 3 state [greeting] Number Input

001444.424 9 3 PrepareAndPlayWavAndTts start iModuleIdx=13, iPlayFileIdx=1, bMakeTtsFile=True, sSoundFileIs=

001444.424 9 3 PrepareAndPlayWavAndTTS sPlayFile=recordings\greeting.wav

001444.424 9 3 state [greeting] Playing wav (recordings\greeting.wav)

001444.424 9 3 timer clear (force=False)

001444.433 9 3 play start ok, playId=650122

...

 

Still having the same problems - outbound dialing is not working, and line 2 does not pick up. For outbound dialing, I tried putting ,3, in the transfer module but it doesn't work.

 

Here are my logs attached.

 

Interestingly, when I first call the second line after restarting VG IVR service, it works, but any subsequent calls are not picked up. Please help. Thanks.

log.zip

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Trace shows that the outbound cal was made on line 3, but about 2 seconds after the transfer was attempted the Disconnect tone was detected on line 1 (DISCONNECT_TAPI1).

 

You should probably change the tone detection settings to make the disconnect tone detection less likely to falsely report disconnect tones. See: http://www.voiceguide.com/vghelp/source/ht...ctiondetect.htm

 

About 15 seconds after end of 1st call trace shows that there was a call arriving on line 2 and that VoiceGuide answered it and started playing a sound file.

 

You should look at the traces yourself to see if there are any differences between first call to line 2 and subsequent calls to line 2.

 

 

 

205620.781 16 1 state [111_transfer] Blind Dial and Connect to 4254588464

205620.792 16 1 dial sCidSentThisCall= (pre RV replace)

205620.793 16 1 outbound call announce message not defined. vgm=15 [111_transfer]

205620.793 16 1 setting CONNECT_IMMEDIATELY as iXferType = XT_DC_BLIND

205620.793 16 1 calling FindFreeLine

205620.799 16 Search for free lines in list [,3,] in LineID range [0-4] src=XferDialAndConf

205620.800 16 1 LineIf_ActiveCall_Present TRUE crn=1

205620.800 16 1 dial busy: active call

205620.800 16 2 LineIf_ActiveCall_Present FALSE crn=0

205620.800 16 2 dial device [dxxxB1C2,dxxxB1C2] not in selection list [,,3,,]

205620.801 16 3 LineIf_ActiveCall_Present FALSE crn=0

205620.801 16 3 dial reserving: found in comma delimited id list [,,3,,]

205620.801 16 1 dial found free dev=3 (LineId=3)

205620.806 16 3 ktTel_MakeCall([4254588464],CONNECT_IMMEDIATELY, 60,0,0,,<calltype>DialAndConf</calltype>) call

205620.822 16 3 ktTel_MakeCall => []

205620.823 16 3 state dialing 4254588464 (for conference with dxxxB1C1)

205620.827 16 1 on-hold sound file defined. Play on-hold file while dialing.

205620.830 16 1 timer clear (force=False)

205620.836 16 1 play start ok, playId=489514

205620.837 16 1 timer set 3600 EV_TIMEOUT_HANGUP

205620.837 16 3 timer set 3600 EV_TIMEOUT_HANGUP

205620.837 16 3 waiting till outcome of outgoing call leg known, as CallProgressOption=CONNECT_IMMEDIATELY. Play OnHold file if present.

205621.354 9 3 ev Dialogic 135,TDX_SETHOOK, crn=0x0 (0), 0,0,0,DX_OFFHOOK,CALL_OUTBOUND,

205621.361 9 3 LsXferWaitOutcomeOutgoingCall_OutLeg : 135,[TDX_SETHOOK],0,0,0,DX_OFFHOOK,CALL_OUTBOUND,

205621.374 9 1 LsXfer_6_dc_WaitOutcomeOutgoingCall_InLeg : [DC_LEG2_TDX_SETHOOK], 135(EV_UNKNOWN_135),0,0,0,CALL_OUTBOUND,,), xferType=31, leg2=3

205621.374 9 1 path {DC_LEG2_TDX_SETHOOK} not found

205621.378 9 3 ev Dialogic 134,TDX_CST, crn=0x0 (0), 2820,0,0,DE_LCON,,

205621.379 9 3 LsXferWaitOutcomeOutgoingCall_OutLeg : 134,[TDX_CST],2820,0,0,DE_LCON,,

205621.379 9 1 LsXfer_6_dc_WaitOutcomeOutgoingCall_InLeg : [DC_LEG2_TDX_CST], 134(EV_UNKNOWN_134),2820,0,0,,,), xferType=31, leg2=3

205621.379 9 1 path {DC_LEG2_TDX_CST} not found

205622.840 9 3 ev Dialogic 132,TDX_DIAL, crn=0x0 (0), 0,0,0,,,

205622.841 9 3 LsXferWaitOutcomeOutgoingCall_OutLeg : 132,[TDX_DIAL],0,0,0,,,

205622.841 9 1 LsXfer_6_dc_WaitOutcomeOutgoingCall_InLeg : [DC_LEG2_TDX_DIAL], 132(EV_UNKNOWN_132),0,0,0,,,), xferType=31, leg2=3

205622.841 9 1 path {DC_LEG2_TDX_DIAL} not found

205622.841 9 1 LsXfer_6_dc_WaitOutcomeOutgoingCall_InLeg : [XFER_ANN_ACCEPTED], 132(EV_UNKNOWN_132),0,0,0,,,), xferType=31, leg2=3

205622.847 9 3 TwoCalls_Bridge 3<=>1 (dxxxB1C3<=>dxxxB1C1)

205622.848 9 3 play PlaySoundStop ok

205622.859 9 1 play PlaySoundStop ok

205622.859 9 3 rvns add [TsRoutedTo]{1}

205622.859 9 1 rvns add [TsRoutedTo]{3}

205622.859 9 1 rvns add [Conf_DevName_1]{dxxxB1C3}

205622.859 9 1 rvns add [Conf_LineId_1]{3}

205622.859 9 1 rvns add [Conf_LineNbr_1]{3}

205622.860 9 1 rv add [Conf_DevName_2]{dxxxB1C1}

205622.860 9 1 rv add [Conf_LineId_2]{1}

205622.860 9 1 rv add [Conf_LineNbr_2]{1}

205622.860 9 3 rvns add [Conf_DevName_1]{dxxxB1C3}

205622.860 9 3 rvns add [Conf_LineId_1]{3}

205622.860 9 3 rvns add [Conf_LineNbr_1]{3}

205622.860 9 3 rv add [Conf_DevName_2]{dxxxB1C1}

205622.861 9 3 rv add [Conf_LineId_2]{1}

205622.861 9 3 rv add [Conf_LineNbr_2]{1}

205622.895 9 3 tsroute dxxxB1C3<=>dxxxB1C1 SC_FULLDUP:

205622.896 9 1 state [111_transfer] bridged with dxxxB1C1 (incoming leg)

205622.896 9 3 state [111_transfer] bridged with dxxxB1C3 (outgoing leg)

205622.898 9 3 ev CallState sEvent=LINECALLSTATE_CONNECTED, hDevice=0, iEvent=2 ,256,0,,,]

205622.898 9 3 callstate: CONNECTED crn=0,ev=2,LINECALLSTATE_CONNECTED,2,0

205622.908 9 1 LsXfer_9_WaitEndCall_InLeg : 0(EV_UNKNOWN_0),DC_LEG2_CONNECTED,256,2,0,,, (iModuleIdx=15)

205622.908 9 1 LsXfer_9_WaitEndCall_InLeg DC_LEG2_ type event

205622.909 9 3 FireEvent_RemotingIf_VgEngine CONNECTED 0

205622.909 9 3 Set_hCall crn=1, called from LINECALLSTATE_CONNECTED

205622.913 9 3 strWorkingModeScript=

205622.913 9 3 connected event no action as: outgoing leg of call now awaits end of call (state=waitendcall)

205622.913 9 3 Set_hCall crn=1, called from LineEvCallState_Connected - LS_XFER_9_WAITENDCALL_DialingSide

205622.913 9 1 ev PlayEnd 1 489514

205622.913 9 1 play end id=489514, (current ID)

205622.913 9 1 LsXfer_9_WaitEndCall_InLeg : 8001(EV_PLAY_FINISHED),EV_PLAY_FINISHED,0,0,0,,, (iModuleIdx=15)

205622.914 9 1 LsXfer_9_WaitEndCall_InLeg try match event by name

205622.914 9 1 path {EV_PLAY_FINISHED} not found

205624.639 9 1 ev Dialogic 134,TDX_CST, crn=0x0 (0), 192,0,0,DE_TONEON,DISCONNECT_TAPI1,

205624.639 9 1 LsXfer_9_WaitEndCall_InLeg : 134(EV_UNKNOWN_134),DISCONNECT,192,0,0,DE_TONEON,DISCONNECT_TAPI1, (iModuleIdx=15)

205624.639 9 1 LsXfer_9_WaitEndCall_InLeg local disconnet handler

205624.640 9 3 LsXfer_9_WaitEndCall_OutLeg : 134(), [DC_LEG2_DISCONNECT], 192,0,0,DE_TONEON,DISCONNECT_TAPI1,

205624.640 9 3 hanging up 2nd leg of call (1st leg cause)

205624.641 9 3 HangupCall ('and any associated'=False), source=xfer_9_leg2 remote

205624.641 9 3 state Hanging up call... [xfer_9_leg2 remote]

 

 

incoming call:

 

205638.660 9 3 ev CallState sEvent=LINECALLSTATE_OFFERING, hDevice=0, iEvent=2 ,2,0,,,]

205638.661 9 3 callstate: OFFERING crn=0,ev=2,LINECALLSTATE_OFFERING,0,0

205638.661 9 3 vgEngine v7.0.2985.21127 (Tue 04/03/2008 11:44:14.64)

205638.661 9 3 FireEvent_RemotingIf_VgEngine OFFERING 0

205638.661 9 3 AnswerTheCallIfAllowed iIvrDev=3, strDlgcDevName_Network=dxxxB1C3

205638.661 9 3 rings=0, min rings before answer=0 (iCallerIdHasArrived=0)

205638.661 9 3 ev ring 1, hCall=3 vgEngine v7.0.2985.21127 (Tue 04/03/2008 11:44:14.64)

205638.662 9 3 FireEvent_RemotingIf_VgEngine RING 1

205638.662 9 3 time since last ring : 0

205638.662 9 3 state Ring 1

205638.662 9 3 AnswerTheCallIfAllowed iIvrDev=3, strDlgcDevName_Network=dxxxB1C3

205638.662 9 3 rings=1, min rings before answer=0 (iCallerIdHasArrived=0)

205638.662 9 3 should answer as number of rings reached

205638.662 9 3 rv clear all

205638.662 9 3 LineState ResetAtStartOfNewCall

205638.663 9 3 Set_hCall crn=3, called from AnserThisCallIfAllowed

205638.663 9 3 start script after going offhook

205638.663 9 3 answering the call (picking up the line)

205638.663 9 3 timer clear (force=True)

205638.665 9 3 ktTel_LineAnswerCall =>

205638.666 9 3 ev Dialogic 134,TDX_CST, crn=0x0 (0), 0,0,0,DE_RINGS,ET_RON,

205639.250 9 3 ev Dialogic 135,TDX_SETHOOK, crn=0x0 (0), 0,0,0,DX_OFFHOOK,CALL_INBOUND,

205639.251 9 3 ev CallState sEvent=LINECALLSTATE_CONNECTED, hDevice=0, iEvent=2 ,256,0,,,]

205639.251 9 3 callstate: CONNECTED crn=0,ev=2,LINECALLSTATE_CONNECTED,1,0

205639.251 9 3 FireEvent_RemotingIf_VgEngine CONNECTED 0

205639.251 9 3 Set_hCall crn=1, called from LINECALLSTATE_CONNECTED

205639.251 9 3 Inband detection not enabled

205639.251 9 3 StartLoadedVgs at 2008-03-04 20:56:39 7.0.2985.21127 (Tue 04/03/2008 11:44:14.64)

205639.252 9 3 StartLoadedVgs LineState.iVgsIdx=1

205639.252 9 3 rvns add [scriptPath]{C:\Users\Joyce\Documents\ivr\}

205639.252 9 3 rvns add [scriptsPath]{C:\Users\Joyce\Documents\ivr\}

205639.252 9 3 set sScriptToRunOnHangup=[] in StartLoadedVgs

205639.252 9 3 rv add [$RV_STARTTIME]{2008-03-04 20:56:39}

205639.252 9 3 rv add [$RV_DEVICEID]{3}

205639.252 9 3 rv add [DlgcVoice]{dxxxB1C3}

205639.253 9 3 rv add [DlgcNetwork]{dxxxB1C3}

205639.253 9 3 rv add [$RV_CIDNAME]{}

205639.253 9 3 rvns add [PathApp]{C:\Program Files\VoiceGuide\}

205639.253 9 3 rvns add [scriptsPath]{C:\Users\Joyce\Documents\ivr\}

205639.253 9 3 rv add [$RV_CIDNUMBER]{}

205639.253 9 3 timer clear (force=False)

205639.253 9 3 RunModule start [Get Numbers,[greeting],13]

205639.253 9 3 state [greeting] Number Input

205639.254 9 3 PrepareAndPlayWavAndTts start iModuleIdx=13, iPlayFileIdx=1, bMakeTtsFile=True, sSoundFileIs=

205639.254 9 3 PrepareAndPlayWavAndTTS sPlayFile=recordings\greeting.wav

205639.254 9 3 state [greeting] Playing wav (recordings\greeting.wav)

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I tried using PBXpert to configure my disconnect time, but I'm running into a channel selection error. I've attached a zip file of my PBXpert process - please look and tell me what's wrong.

 

Jack 1 (ext 101) of the PBX is plugged into port 1 of Dialogic, and Jack 2 (ext 102) of PBX is plugged into port 3 of Dialogic.

Pictures.zip

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Are you using Vista? You should really be using WinXP or Win2003, unless you are using special Vista version of the Dialogic Drivers.

 

Please contact Dialogic direct for issues related to their PBXpert program.

 

We'd recommend just recording the tones and analyzing and setting the tones manually as described in http://www.voiceguide.com/vghelp/source/ht...ctiondetect.htm

 

You should also rewire your connections so that you are actually using ports 1 and 2 on the card, not ports 1 and 3. Otherwise you’ll need to set “,1,3,” as outbound line selection in the various places where outbound calls are made.

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I installed Windows XP Pro (SP2) and I'm getting an error when installing the latest Dialogic release drivers from the VG website (SR 6.0 SU181 drivers at http://membersresource.dialogic.com/_relea...1/dev/dev.zip):

 

"Error encountered after attempting to launch:

 

Program: C:\Program Files\Dialogic\bin\DM3Config.exe

Parameters: -u

Return code: -1073741515

Timeout Code: 0"

 

Can you help me resolve this?

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Not sure why you would be getting this error. Does the Dialogic service itself detect the Dialogic card and starts ok?

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Not sure why you would be getting this error. Does the Dialogic service itself detect the Dialogic card and starts ok?

 

The Dialogic card I am using is D/120JCT-LS. Am I installing the correct software from VG website? I know you are not sure why I got this error, but can you think of some things I could try to resolve this problem? Thanks.

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SR 6.0 are the correct drivers.

 

Does the Dialogic service itself detect the Dialogic card and starts ok?

 

D/120JCT is not a DM3 type card, so the DM3Config.exe error could probably be just ignored in your case.

 

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Okay - I have no problem with the Dialogic SR6.0 drivers anymore. It's fine now, it recognizes the card, and the card starts.

 

1. Outbound dialing *still* is not working. I'm using Windows XP now, and I did some testing - I think the false disconnects etc that I'm getting are due to the card starting to dial out too quickly (before the dial tone). I'm trying to solve this by using PBX command Strings inside VG Script Designer. I'm putting more commas to make the system wait inside each entry under 'Call Conference.' Is this the right way to do it? Where exactly should I put the commas and how many should I put?

 

Logs are attached. Thanks.

log.zip

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Trace shows that the script is dialing out on the second line and connecting the two legs of the call immediately after dialing, and then disconnects about 35 seconds later as there was a Disconnect tone reported on the line.

 

What is the actual problem? Is the dialed number ringing? When you pick it up can the two people hear each other?

 

 

 

state [111_transfer] Blind Dial and Connect to 4254588464

...

004510.687 8 1 state [111_transfer] bridged with dxxxB1C1 (incoming leg)

004510.687 8 2 state [111_transfer] bridged with dxxxB1C2 (outgoing leg)

...

004544.718 8 1 ev Dialogic 134,TDX_CST, crn=0x0 (0), 192,0,0,DE_TONEON,DISCONNECT_TAPI1,

 

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Trace shows that the script is dialing out on the second line and connecting the two legs of the call immediately after dialing, and then disconnects about 35 seconds later as there was a Disconnect tone reported on the line.

 

What is the actual problem? Is the dialed number ringing? When you pick it up can the two people hear each other?

 

 

 

state [111_transfer] Blind Dial and Connect to 4254588464

...

004510.687 8 1 state [111_transfer] bridged with dxxxB1C1 (incoming leg)

004510.687 8 2 state [111_transfer] bridged with dxxxB1C2 (outgoing leg)

...

004544.718 8 1 ev Dialogic 134,TDX_CST, crn=0x0 (0), 192,0,0,DE_TONEON,DISCONNECT_TAPI1,

It is very strange. For example, out of 10 times, the dailed number sometimes rang, sometimes didn't ring; a few times, the call went to someone else (yes, it rang on some strange number; I used a phone attached to line 2 to overhear the call); At NO times were the two people able to hear each other. I am now very confused. Can you help shed some light to this mystery?

 

 

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Suggest removing the PBX from the loop for now and plugging the external lines direct into the Dialogic card and Performing some dial out tests like that.

 

I used a phone attached to line 2 to overhear the call

This would change he impedance of the line as seen by both the PBX and the Dialogic card which may in turn distort the DTMF signals used during dialing. You should not connect additional handsets to the line like that.

 

At NO times were the two people able to hear each other.

The traces show that the Dialogic API replied that it did connect the two calls together, so if neither of the callers could hear the other party then this would suggest there is a problem with the Dialogic card.

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1.

For the oubound dialing, I suspect that Voiceguide is picking up line 2 and dialing the phone number out right away, which causes problems and makes me disconnect.

 

I think it should pick up and wait a few seconds before dialing out. How do I introduce this delay so that Voiceguide picks up the outgoing line, waits, and then dials? Please tell me exactly how to do it. Is it through Edit > PBX Command Strings?

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I think it should pick up and wait a few seconds before dialing out. How do I introduce this delay so that Voiceguide picks up the outgoing line, waits, and then dials? Please tell me exactly how to do it. Is it through Edit > PBX Command Strings?

Just add a comma before the number to be dialed. If you want more delay then just add more commas. Each comma is about 2 seconds delay (by default).

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I think it should pick up and wait a few seconds before dialing out. How do I introduce this delay so that Voiceguide picks up the outgoing line, waits, and then dials? Please tell me exactly how to do it. Is it through Edit > PBX Command Strings?

Just add a comma before the number to be dialed. If you want more delay then just add more commas. Each comma is about 2 seconds delay (by default).

 

I tried to put in commas (varying numbers of them) before the phone number to be dialed out, and it still doesn't work. When I dial out, I can see from VG Line Status Monitor that line 2 is being dialed out, but then Line 1 disconnects and Line 2 disconnects shortly afterward.

 

For the call bridging method in the VG Script, I tried "Immediately after Dialing" and "On Voice Detect." Neither of them work. Which bridging method should I use?

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When I connect the Dialogic card directly to the phone lines, removing the PBX from the loop, calling out works fine. Please help me solve this with the PBX. thanks.

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When I connect the Dialogic card directly to the phone lines, removing the PBX from the loop, calling out works fine.

Looks like the PBX is faulty then...

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I want to confirm that Voiceguide is capable of picking up 2 or more simultaneous calls into the Dialogic card. Thanks.

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Are you asking whether VoiceGuide can handle multiple calls arriving on multiple lines at the same time? Yes, of course it can. It can handle hundreds of simultaneous calls if required.

 

Just place calls on the lines attached to the Dialogic card and you will see VoiceGuide will answer all the calls that arrive into the system.

 

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Are you asking whether VoiceGuide can handle multiple calls arriving on multiple lines at the same time? Yes, of course it can. It can handle hundreds of simultaneous calls if required.

 

Just place calls on the lines attached to the Dialogic card and you will see VoiceGuide will answer all the calls that arrive into the system.

 

I'm connecting two Vonage lines to CO1 and CO2, and Station1 and Station2 are connected to a Dialogic card that's hooked up to VG.

 

1. If I call in on CO1, then it's transferred to Jack 1 and VG picks it up. Then, when I try to call out from within VG, it calls out on the line connected to Jack 2 (ext 102) on the PBX. What I believe is happening is that Jack 2 tries to call outbound on CO1, which is in use. It then gives a busy tone and disconnects me. I'm having trouble making Jack 2 (ext 102) call out on CO2. I've configured the KX-TA Maintenance Console so that in PT personal settings, Jack 02's outgoing line preference is CO2. This doesn't work though - please help me fix this. I know perhaps you will say that I should talk to my PBX guy. I did and he confirm my setting is right. Other than that, he cannot be of more help.

 

2. Another problem is when I call into CO1 and my call is picked up by VG. Then if I use CO2 to call in, VG does not pick up. I think this is because the PBX is not transferring a call to CO2 correctly to Jack 2. But when I call CO1 and CO2 separately, they are both picked up. Any idea I can try to fix this problem?

 

Many thanks.

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These are both PBX related issues and you really need to speak with the PBX supplier to get them resolved. You should probably get someone who knows PBXs on site to get all these issues sorted out.

 

Sounds like you should test out the PBX first with just telephone handsets instead of the Dialogic card. Once you have it working as you want then attach the Dialogic/VoiceGuide instead of the handsets.

 

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I have a Dialogic D/120JCT card and outbound dialing to a cellphone does not work

Trace shows that the outgoing leg of the Dial and Conference call was made on port 2, and that number 4254588464 was dialed. I suspect you need to place a "0," before the dialed number to make the call go out on external line, just like you would dial on an external line when placing a call from the telephone handset attached to line.

 

234648.816 4 1 dial found free dev=2 (LineId=2)

234648.822 4 2 ktTel_MakeCall([4254588464],CONNECT_IMMEDIATELY, 60,0,0,,<calltype>DialAndConf</calltype>) call

234648.838 4 2 ktTel_MakeCall => []

234648.839 4 2 state dialing 4254588464 (for conference with dxxxB1C1)

 

Alternatively you should try specifying that line 3 be used for the outgoing calls.

 

The 0303_0014_vgEngine trace shows calls arriving on lines 1 and 3 but the Dialogic card does not report any Rings for any incoming calls on line 2.

 

Another question: how do I increase the delay before the system thinks that the user won't type in anymore numbers?

Use a timeout path. See: http://www.voiceguide.com/vghelp/source/ht...odgetnumber.htm

 

Another question on the timeout path:

 

In my current script, I have a get numbers module that plays a greeting, followed by a get numbers module that plays the main menu options. I need it to start playing the menu options right after the greeting ends. What's the right way of doing this automatic transfer?

 

I tried a timeout path at first, but that messes up the timeout between key presses. I just tried a fail path, but it gives a big delay before playing the menu options.

 

What's the right way to do this? thanks.

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I suspect you should be using Play modules instead of Get Numbers modules for both the introduction and menu modules.

 

If you still have problems please post your script here.

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I suspect you should be using Play modules instead of Get Numbers modules for both the introduction and menu modules.

 

If you still have problems please post your script here.

 

The play module doesn't seem to accept multi-digit input (like '123'), which I need. How would I do it?

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The Play module is for accepting single digit selections and use a Get Numbers module for multi-digit input.

 

If you know the length of the number to be typed in in a Get Numbers module then just set that to be the "Max Length", and a Success path will be taken immediately when that number of digits is entered.

 

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Hi,

 

That's not answering my problem. When a customer first calls in, this is what should happen:

* the call is answered by a get numbers module that plays a greeting.

* the script them passes the call on to *another* get numbers module that plays the menu options.

 

How do I automatically make the script pass the call onto the next get numbers module without using a timeout path (which screws up key press timeouts) and without introducing a delay? I want the transition to be smooth. I can't use a play module because I need multidigit input

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If you know the length of the first entered number then spcify that to be "maximum neterd number length", or tell the caller to press # after tehy have enterdthe number.

 

Otherwise a timeout will be used to determine that the caller has finished entering the digits.

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If you know the length of the first entered number then spcify that to be "maximum neterd number length", or tell the caller to press # after tehy have enterdthe number.

 

Otherwise a timeout will be used to determine that the caller has finished entering the digits.

 

 

This again is not addressing the problem I have.

Let me describe my problem more clearly.

Our company’s IVR script is like this:

 

Get Numbers module: “Greeting”    Welcome and thank you for calling our company. This automated system provides easy access to information about our company. 
Get Numbers module: “Menu options”    If you know your party’s extension number, you can dial it now. 

To inquire about or order our products, press 2.

To repeat the menu options, please press the * key.  (goes to beginning of menu options)

 

 

And the reason why we are separating “Greeting” and “Menu Options” is that if the caller presses *, then he/she doesn’t need to listen to the replay of “Greeting” and can listen to “menu options” immediately.

The problem I have now is that there is a 4 second “silence” between the end of “Greeting” and the beginning of “Menu options.” if I use a FAIL pathway to transfer to the menu options module; if I use a timeout pathway to play the greeting, then there’s no delay between greeting and menu options, but there’s one problem: when caller is pressing one of the menu options (say, 123) and if he is a tiny bit slow, then the key press timeout delay is too short (because of the timeout pathway).

 

I have also attached this reply in a word file (since there is a table in it). Please take a look.

 

 

VG_problem.doc

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Sounds like the easiest way for you to do what you want, you should just use a normal Play module for the "Greeting" module.

 

Just set the "Greeting" Play module to have 0 replays and just use a {Timeout 0} path going to the "Menu Options" module, to have the "Menu Options" module start immediately when the "Greeting" module's sound file finishes.

 

The caller would of course only be able to make a selection in the "Menu Options" module, and the greeting module doing just that - playing a Greeting only.

 

 

If you want to be able for the caller to enter extension numbers in the first module, but also immediately jump to another module if nothing was entered then do this for the "Greeting" module:

 

1. Use a Get Number module type

2. Do not use any Timeout paths

3. Set replays to 0

4. Use a Fail path to go to "Menu Options" module

5. In VG.INI, section [moduleGetNbrs] add/set this entry: Timeout_1stDigit=0

6. Set the "maximum entered number length" to be the length of the extension number entered.

 

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Hi,

 

Another question:

 

Right now, callers can press an extension number to be transferred to the appropriate desktop phone. However, when the employee at the desk picks up, there is a delay of about 4 seconds before the caller and the employee can hear each other. I'm using a PBX Hookflash Transfer - Monitored. Can you tell me how to set it so that they can hear each other immediately?

 

Thanks.

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Does the person answering the phone say "Hello" or something similar? on monitored transfers VoiceGuide will only connect the call after if hears the person answering the call say something (eg "Hello" etc)

 

Better option is to use a Blind Hooklfash Transfer, but the PBX must support Blind Hooklfash Transfers.

 

Another options is to use "Dial and Conference" transfers but that uses up another line for the outgoing leg of the call.

 

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If you want to be able for the caller to enter extension numbers in the first module, but also immediately jump to another module if nothing was entered then do this for the "Greeting" module:

 

1. Use a Get Number module type

2. Do not use any Timeout paths

3. Set replays to 0

4. Use a Fail path to go to "Menu Options" module

5. In VG.INI, section [moduleGetNbrs] add/set this entry: Timeout_1stDigit=0

6. Set the "maximum entered number length" to be the length of the extension number entered.

 

This is still not working - I tried it, and I still have a 3-4 second gap between Greeting and Menu Options. Logs are attached. Thanks.

log.zip

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Looks like the lowest value for Timeout_1stDigit is 1. Please try setting Timeout_1stDigit=1

 

We will change the software to accept Timeout_1stDigit=0 in future versions.

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Looks like the lowest value for Timeout_1stDigit is 1. Please try setting Timeout_1stDigit=1

 

We will change the software to accept Timeout_1stDigit=0 in future versions.

 

Great, this works. Thanks.

 

Does the person answering the phone say "Hello" or something similar? on monitored transfers VoiceGuide will only connect the call after if hears the person answering the call say something (eg "Hello" etc)

 

Better option is to use a Blind Hooklfash Transfer, but the PBX must support Blind Hooklfash Transfers.

 

Another options is to use "Dial and Conference" transfers but that uses up another line for the outgoing leg of the call.

 

This won't work, because the reason I'm using a monitored hookflash transfer is so that after a certain timeout period, the caller can be automatically transferred to the employee's voicemail. This won't happen with a blind hookflash transfer. Or is there any other way to do this?

 

Or with a Monitored hookflash transfer, can I make it so that the system can recognize when the employee picks up, rather than detecting the voice?

 

How do all the other companies do this without a delay? Is Voiceguide incapable of doing it?

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This won't work, because the reason I'm using a monitored hookflash transfer is so that after a certain timeout period, the caller can be automatically transferred to the employee's voicemail. This won't happen with a blind hookflash transfer. Or is there any other way to do this?

If you need to detect the actual time the handset has been picked up, rather then relying on the Dialogic card detecting speech after the call is answered, then you would need to have a CTI link to the PBX. I don't think your PBX supports any CTI integration which would allow external apps to be alerted when a telephone handset goes off-hook etc.

 

The way this is done usually is that the PBX is configured to transfer calls to another extension after a certain number of rings, and you just configure that extension to be the voicemail system's extension. That way you can just do a blind transfer out of the IVR and rely on the PBX to transfer a call to voicemail etc if that extension is not answered.

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So let me clarify how this works:

 

I need to set up the PBX so that after a certain number of rings (if the call is unanswered) then the call will be transferred to another extension. How is that extension linked back to VG voicemail?

 

Do I need another phone line coming out of that extension and going back into the Dialogic card? That seems like an awkward way of doing it. But if so, how do I link it to just the specific voicemail box for that employee?

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How is that extension linked back to VG voicemail?

You just set it to be the extension number of one of the lines plugged into the VoiceGuide’s Dialogic card. Or a hunt group which points to all of the lines plugged into the VoiceGuide's Dialogic card. Inband Signaling or CTI link must be used to distinguish these "transfer on no answer" calls from normal incoming calls, and also to receive the extension number. The script then can jump to the right voicemail box using the Inband Signaling (or CTI) data.

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