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Re: audio files voice not coming

Posted: Thu Sep 01, 2016 1:41 pm
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

Reg: Call Transfer functionality implement using SIP

Posted: Fri Sep 02, 2016 2:56 am
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,

Re: audio files voice not coming

Posted: Fri Sep 02, 2016 11:15 am
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

Reg: Call Transfer functionality using SIP

Posted: Tue Sep 06, 2016 3:27 pm
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,

Re: audio files voice not coming

Posted: Tue Sep 06, 2016 3:53 pm
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

Reg: SIP based call Transfer

Posted: Wed Sep 07, 2016 2:07 am
by bmreddyrs
removed

Reg: SIP based call Transfer

Posted: Wed Sep 07, 2016 4:37 am
by bmreddyrs
removed

Re: Reg: SIP based call Transfer

Posted: Thu Sep 08, 2016 4:40 am
by bmreddyrs
removed

Re: audio files voice not coming

Posted: Thu Sep 08, 2016 5:17 pm
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

Reg: SIP using Call Transfer

Posted: Fri Sep 09, 2016 1:34 am
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,

Reg: Call Transfer functionality using SIP

Posted: Fri Sep 09, 2016 5:36 am
by bmreddyrs
removed

Re: audio files voice not coming

Posted: Fri Sep 09, 2016 4:10 pm
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

Re: audio files voice not coming

Posted: Mon Sep 12, 2016 12:37 am
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,

Re: audio files voice not coming

Posted: Mon Sep 12, 2016 9:42 am
by support
Hi,

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

Regards,
Plum Support

Reg: Call Transfer functionality using SIP

Posted: Wed Sep 14, 2016 1:29 am
by bmreddyrs
removed