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audio files voice not coming

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support
Posts: 3632
Joined: Mon Jun 02, 2003 3:47 pm
Location: Boston, MA
Contact:

Re: audio files voice not coming

Post by support »

Hi,

We have tested your code with some adjustments and successfully got your expected result:

Code: Select all

<?xml version="1.0"?>
<vxml version="2.0">
	<form>
		<var name="CallSid" expr="'a9e72239-f524-401f-a803-acba55a04aff'"/>
		<var name="autoDigits" expr="1"/>
		<block>
			<prompt bargein="false">
				<audio src="https://wcdevint2.wavecrest.in:444/static/ivr/Afligo/en/wc-A21-1.wav" fetchtimeout="120s"/>
				<voice xml:lang="en_us" gender="female" name="crystal"><say-as>0 dollars. 0 cents</say-as></voice>
				<break time="4s" />
			</prompt>
		</block>
		<property name="termtimeout" value="12s"/>
		<property name="termchar" value="#"/>

		<field name="Digits" type="digits?length=1"><grammar type="application/srgs+xml" root="Root" mode="dtmf">
			<rule id="Root" scope="public"><one-of><item repeat="0-255"/></one-of></rule></grammar>

			<prompt bargein="true">
				<audio src="https://wcdevint2.wavecrest.in:444/static/ivr/Afligo/en/wc-A12-suspendCard.wav" fetchtimeout="120s" />
				<voice xml:lang="en_us" gender="female" name="crystal"><say-as>1</say-as></voice>
				<break time="4s" />
				<audio src="https://wcdevint2.wavecrest.in:444/static/ivr/Afligo/en/wc-A12-transactionHistory.wav" fetchtimeout="120s" />
				<voice xml:lang="en_us" gender="female" name="crystal"><say-as>2</say-as></voice>
				<break time="4s" />
				<audio src="https://wcdevint2.wavecrest.in:444/static/ivr/Afligo/en/wc-A12-managePIN.wav" fetchtimeout="120s"/>
				<voice xml:lang="en_us" gender="female" name="crystal"><say-as>3</say-as></voice>
				<break time="4s" />
				<audio src="https://wcdevint2.wavecrest.in:444/static/ivr/Afligo/en/wc-A12-blockCard.wav" fetchtimeout="120s" />
				<voice xml:lang="en_us" gender="female" name="crystal"><say-as>4</say-as></voice>
				<break time="4s" />
				<audio src="https://wcdevint2.wavecrest.in:444/static/ivr/Afligo/en/wc-A12-customerService.wav" fetchtimeout="120s" />
				<voice xml:lang="en_us" gender="female" name="crystal"><say-as>9</say-as></voice>
				<break time="4s" />
			</prompt>
			<nomatch count="1">
				<reprompt/>
			</nomatch>
			<nomatch count="2">
				<reprompt/>
			</nomatch>
			<nomatch count="3">
				<reprompt/>
			</nomatch>
			<nomatch count="4">
				<exit/>
			</nomatch>
			<noinput count="1">
				<reprompt/>
			</noinput>
			<noinput count="2">
				<reprompt/>
			</noinput>
			<noinput count="3">
				<reprompt/>
			</noinput>
			<noinput count="4">
				<exit/>
			</noinput>
			<filled>
				<submit next="https://wcuser:wcuser@dev2.api.wavecrest.in:444/v1/services/ivr/xmlivrprocessor/" namelist="Digits CallSid" method="post"/>
			</filled>
		</field>
	</form>
</vxml>
You may have been coming across those issues by accidentally setting off a "no match" event. If this was done after "transactionHistory.wav", it would reprompt the user starting with "suspendCard.wav" again. We recommend editing your code to alert the user of a nomatch or noinput event before reprompting. For example:

Code: Select all

<nomatch count="1">
        <prompt>
                I'm sorry, I didn't understand you.
        </prompt>
	<reprompt/>
</nomatch>
Regards,
Plum Support

bmreddyrs
Posts: 44
Joined: Wed May 25, 2016 4:03 am

Reg: Call Transfer functionality implement using SIP

Post by bmreddyrs »

Hi Team,

we are going the implement call transfer functionality using SIP aai.



Please let me know, Plum environment dest="sip:9160111490@xxxxx.xxxx.xxx.com - this value.

Thanks,

support
Posts: 3632
Joined: Mon Jun 02, 2003 3:47 pm
Location: Boston, MA
Contact:

Re: audio files voice not coming

Post by support »

Hi,

Can you please clarify if you trying to transfer a SIP call out of our environment or in to it? That will help us determine what information you may need from us.

Regards,
Plum Support

bmreddyrs
Posts: 44
Joined: Wed May 25, 2016 4:03 am

Reg: Call Transfer functionality using SIP

Post by bmreddyrs »

support wrote:Hi,

Can you please clarify if you trying to transfer a SIP call out of our environment or in to it? That will help us determine what information you may need from us.

Regards,
Plum Support
Hi Team,

In the IVR application, we can able to connect to the Live agent application in call transfer functionality using in SIP.

Please provide example of the SIP based call transfer functionality

Thanks,

support
Posts: 3632
Joined: Mon Jun 02, 2003 3:47 pm
Location: Boston, MA
Contact:

Re: audio files voice not coming

Post by support »

Hi,

You should be able to place a SIP transfer by using a sip address for the "dest" attribute for the <transfer> tag:

Code: Select all

<transfer dest="sip:9999@123.456.789.2">
Additionally, the <transfer> tag also has a "aaiexpr" attribute with which you can send aai data. At this time, only custom SIP headers beginning with "X-" are supported. Below is an example of how you can use the transfer tag to send aai data:

Code: Select all

<?xml version="1.0" encoding="UTF-8"?>
<vxml version="2.0">
  <form>
    <!-- NOTE: ALL CUSTOM SIP HEADERS WILL NEED TO BE PREFIXED WITH "x-" -->
    <script>
      var aai = new Object();
      aai.headers = new Object();
      aai.headers['x-CustomHeader1'] = 'predefined text string';
    </script>

    <field name="id" type="digits">
      <prompt>Please enter your account ID.</prompt>
      <filled>
        <script>
          aai.headers['x-CustomHeader2'] = 'accountID='+id;
        </script>
      </filled>
    </field>

    <!-- NOTE: ENTER DESIRED EXTENSION AND IP ADDRESS FOR SIP CALL -->
    <transfer dest="sip:9999@123.456.789.2" aaiexpr="JSON.stringify(aai)">
      <prompt>Please hold while you are transferred.</prompt>
    </transfer>
  </form>
</vxml>
We hope that this helps.

Regards,
Plum Support

bmreddyrs
Posts: 44
Joined: Wed May 25, 2016 4:03 am

Reg: SIP based call Transfer

Post by bmreddyrs »

removed
Last edited by bmreddyrs on Thu Mar 09, 2017 3:17 am, edited 1 time in total.

bmreddyrs
Posts: 44
Joined: Wed May 25, 2016 4:03 am

Reg: SIP based call Transfer

Post by bmreddyrs »

removed
Last edited by bmreddyrs on Thu Mar 09, 2017 3:15 am, edited 1 time in total.

bmreddyrs
Posts: 44
Joined: Wed May 25, 2016 4:03 am

Re: Reg: SIP based call Transfer

Post by bmreddyrs »

removed
Last edited by bmreddyrs on Thu Mar 09, 2017 3:16 am, edited 1 time in total.

support
Posts: 3632
Joined: Mon Jun 02, 2003 3:47 pm
Location: Boston, MA
Contact:

Re: audio files voice not coming

Post by support »

Hi,

We've been looking into this and we wanted a bit more information. Are you attempting to send a transfer call to an Indian number?

Regards,
Plum Support

bmreddyrs
Posts: 44
Joined: Wed May 25, 2016 4:03 am

Reg: SIP using Call Transfer

Post by bmreddyrs »

support wrote:Hi,

We've been looking into this and we wanted a bit more information. Are you attempting to send a transfer call to an Indian number?

Regards,
Plum Support
Hi Team,

Yes, I am trying to call Indian Number - (+919160111490).

Thanks,

bmreddyrs
Posts: 44
Joined: Wed May 25, 2016 4:03 am

Reg: Call Transfer functionality using SIP

Post by bmreddyrs »

removed
Last edited by bmreddyrs on Thu Mar 09, 2017 3:17 am, edited 1 time in total.

support
Posts: 3632
Joined: Mon Jun 02, 2003 3:47 pm
Location: Boston, MA
Contact:

Re: audio files voice not coming

Post by support »

Hi,

We have done some transfer test calls to the numbers you provided at sip-twilio.ubiquitygs.com on a test IVR where we could run packet captures while we placed these calls. We have found that your system returns a SIP response 403 Forbidden for both numbers after we have attempted a SIP INVITE. This is why the calls are failing during the transfer attempt. Please look into why your system is returning a 403 Forbidden.

Regards,
Plum Support

bmreddyrs
Posts: 44
Joined: Wed May 25, 2016 4:03 am

Re: audio files voice not coming

Post by bmreddyrs »

support wrote:Hi,

We have done some transfer test calls to the numbers you provided at sip-twilio.ubiquitygs.com on a test IVR where we could run packet captures while we placed these calls. We have found that your system returns a SIP response 403 Forbidden for both numbers after we have attempted a SIP INVITE. This is why the calls are failing during the transfer attempt. Please look into why your system is returning a 403 Forbidden.

Regards,
Plum Support

Hi Team,

Thank you for your response. Please provide Your Plum Dev environment server configuration details (IP address), these details we will configure at sip-twilio.ubiquitygs.com server environment.

Thanks,

support
Posts: 3632
Joined: Mon Jun 02, 2003 3:47 pm
Location: Boston, MA
Contact:

Re: audio files voice not coming

Post by support »

Hi,

These transfer calls can come from a variety of different IP addresses.

Regards,
Plum Support

bmreddyrs
Posts: 44
Joined: Wed May 25, 2016 4:03 am

Reg: Call Transfer functionality using SIP

Post by bmreddyrs »

removed
Last edited by bmreddyrs on Thu Mar 09, 2017 3:18 am, edited 1 time in total.

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