record the conversation between the customer and a customer

[gmanigopal]

Hi,

I am new to VoiceXML IVR development. currenlty i'm developing ivr application using asp.net. I tried some samples of record option using record tag and saved as wav file. But my criteria
Is it possible to record the conversation between the customer and a customer support executive once the call is be transferred to customer care?

I used to following code to transfer the call

<?xml version="1.0" encoding="utf-8" ?>
<vxml version="2.0">
<form>
<property name="recordcall" value="true"/>
<transfer name="outcome" dest="tel:+1xxxxxxxxxxx" connecttimeout="18s">
</transfer>
</form>
</vxml>

[support]

Hi gmanigopal,

Yes, it is possible for you to record the conversation between the customer and a customer support executive once the call is transferred to customer care. You can do this by using the "recordcall" property.

This previous forum post, http://support.plumvoice.com/viewtopic.php?f=2&t=39633, contains a code example of how you would be able to set up your application to use the "recordcall" property and then submit the recording for post-call processing.

Regards,
Plum Support

[gmanigopal]

Hi,

Thanks for your reply. Now i'm getting idea how to save the conversation but i'm facing problem in transferring the call. I just tried to transfer the call to particular no. In the log shows makecall failed. The following things was showed in log details

Log details
--------------------------------------------------------------------------------------------------------------------------------
VXI::queue_prompts()
Bridge Transfer: tel=+1447586062438
Makecall Failed: General Failure - 0x506.
bargein set to true
INPUTMODES set to "DTMF VOICE"
Audio segment added to prompt queue from TTS application/synthesis+ssml for:
---------
<?xml version='1.0'?><speak>
You will hear this message in your recording.
</speak>
---------
starting playback: bargein=true, inputmodes="dtmf speech"
Newly queued prompts are now being played

and also shows
HTTP/1.1 500 Internal Server Error - http://xxxxxxxxx/recordmessage.aspx
DocumentParser::FetchBuffer - could not open URL: recordmessage.aspx
DocumentParser::FetchDocument - exiting with error result 2
errno: 203 uri recordmessage.aspx
------------------------------------------------------------------------------------------------------------------------------------------


call transfer coding
-----------------------------------------------------------------------------------------------------------------------------------------
<?xml version="1.0" ?>
<vxml version="2.0">
<form>
<property name="recordcall" value="true"/>
<transfer name="mycall" dest="tel:+1447586062438"/>
</form>
</vxml>
------------------------------------------------------------------------------------------------------------------------------------------

but its shows recordmessage.aspx file exiting with error result 2.
But that recordmessage.aspx file was working fine. i tried with url (http://xxxx/recordmessage.aspx).

Regards.,
Manikandan

[support]

Hi,

From the log details, it appears that you are attempting to transfer to an international number. Please note that for international transfers from the US, you would need to enter the phone number in the following format:

{US exit code} + {country code} + {phone number}

Regarding the HTTP 500 error you are seeing within the logs, this response is typically returned when there are issues with your web server. You should check with your network team to find out why HTTP 500 errors are being returned by your web server.

Regards,
Plum Support

[gmanigopal]

Hi,

Sorry to again. I tried call transferring as you told {US exit code} + {country code} + {phone number}. But i unable to get call.

<transfer name="mycall" dest="tel:+1011447586062438" connecttimeout="180s"/>

while i'm checking log it shows as follows
-----------------------------------------------------------------------------------------------------------------------------------------------
Makecall Failed: Timeout of 30s reached
received event: connection.disconnect.hangup:
VXI::exit_element()
Call End Event
Ending session
Ending Session On Channel 1
------------------------------------------------------------------------------------------------------------------------------------------------
one more question is, i changed the call transfer no and i deployed files in server. I validated the remote url using plum voice. I'm getting "This document is valid VoiceXML" and i saw my changed no. But in log it shows old no.

I dont know why the call no was not changed in log. Is it keep trying the old no or new no?.

Regards.,
Manikandan

[support]

Hi,

Looking through your code, you should remove the '+1' from the number that you are attempting to transfer to so that the number inside your "dest" attribute is just "011447586062438".

Regards,
Plum Support

[gmanigopal]

Hi,

If i tried without '+1' i'm getting no one answering the call message. This is my code

-----------------------------------------------------------------------------------------------------------------------------------------------
<?xml version="1.0"?>
<vxml version="2.0">
<form>
<property name="recordcall" value="true"/>
<block>
<prompt> preparing to dialing
</prompt>
</block>
<transfer name="MyCall" dest="tel:+1011447586062438" connecttimeout="180s">
<filled>
<if cond="MyCall == 'busy'">
<prompt>
The lines are busy. Please try again later.
</prompt>
<exit/>

<elseif cond="MyCall == 'noanswer'"/>
<prompt>
No one is answering the line.
</prompt>

<elseif cond="MyCall == 'far_end_disconnect'"/>
<prompt>
The called party has hung up.
</prompt>

<elseif cond="MyCall == 'near_end_disconnect'"/>
<prompt>
You have pressed 1 to disconnect the transfer.
</prompt>

<elseif cond="MyCall == 'maxtime.disconnect'"/>
<prompt>
Your call ran over the maxtime of sixty seconds,
and the called party has been disconnected.
</prompt>

<elseif cond="MyCall == 'unknown'"/>
<prompt>
The call failed for an unknown reason.
</prompt>

</if>
</filled>
</transfer>


</form>
</vxml>

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

I can receive the call from US for above no (01447586062438) what i used. I tested with my friend. But In plumvoice i can't receive any call.


can you help this.

[support]

Hi,

If you are using a UK hosted number (you can check this by seeing if the "Location" for your phone number says "UK - London"), then you would set up the line of code for your transfer to be the following:

<transfer name="MyCall" dest="tel:+447586062438" connecttimeout="180s">

Regards,
Plum Support

[gmanigopal]

Hi,

I have two doubt in this post.
1> Is any cache details are need to clear in plumvoice during call?. I'm getting Max Disconnect Count Exceeded and variable not defined in line error in log details. what is Max disconnect count exceeded and how to solve this? and "my_transfer" is not defined error shows. I'm not using that variable in my program. But log show as follows...
--------------------------------------------------------------------------------------------------------------------------------------------
Tue 02 Jul 2013 11:42:16 AM EDT:
ReferenceError: my_transfer is not defined line 1
received event: error.semantic.ecmascript:
bargein set to true
INPUTMODES set to "DTMF VOICE"
Audio segment added to prompt queue from TTS application/synthesis+ssml for:
---------
<?xml version='1.0'?><speak>
A serious error of type </speak>
---------
bargein set to true
INPUTMODES set to "DTMF VOICE"
Audio segment added to prompt queue from TTS application/synthesis+ssml for:
---------
<?xml version='1.0'?><speak>error.semantic.ecmascript</speak>
---------
bargein set to true
INPUTMODES set to "DTMF VOICE"
Audio segment added to prompt queue from TTS application/synthesis+ssml for:
---------
<?xml version='1.0'?><speak> has occurred. Exiting. </speak>
---------
VXI::exit_element()
starting playback: bargein=true, inputmodes="dtmf speech"
Newly queued prompts are now being played

Tue 02 Jul 2013 11:42:22 AM EDT:
Line disconnect detected. Abandoning queued audio data.
received event: connection.disconnect.hangup:
Can not queue audio -- line disconnected
received event: connection.disconnect.hangup:
Max Disconnect Count Exceeded
Call End Event
Ending session
Ending Session On Channel 16
------------------------------------------------------------------------------------------------------------------------------

2> you guys gave a link viewtopic.php?f=2&t=39633, contains a code example of how you would be able to set up your application to use the "recordcall" property and then submit the recording for post-call processing for record the caller and callee conversation. In this article namelist="callrecording" was used in submit tag. But it was not declared in that program.
then how the data (conversation details )was passed ( by post) to another page?

[support]

Hi gmanigopal,

1) Please note that if you are using a static file for your application script, you may need to flush out the cache by using the "maxage" and "maxstale" properties.

You can view the following documentation regarding caching: http://www.plumvoice.com/docs/dev/devel ... ce:caching

2) "callrecording" is a shadow variable that stores your call recording when using the "recordcall" property. Please see our documentation here, http://www.plumvoice.com/docs/dev/voice ... lrecording, for more information.

Regards,
Plum Support

[gmanigopal]

Thanks a lot. It very useful to me.