Answers marked with #### Missing audio files. IOS VB returned error.badfetch.http.404. VVB (pre patch ES81) returned error.badfetch.http. With ES81 returns no notification at all. ####  ES81 error not throw is as per compliance.     In VVB ES81 we followed the Specs as recommended in Conformance : https://www.w3.org/Voice/2004/vxml-ir/     375    [4.1.3]   Yes. Yes.       If the URI associated with src is unavailable, and the audio element doesn't contain content, no error is thrown.     376    [4.1.3]   Yes. Yes.       If the URI associated with expr is unavailable, and the audio element doesn't contain content, no error is thrown. Bad speech reco grammar. IOS VB returned error.noresource.  VVB returns error.badfetch #### it needs to be mapped to error.noresource interdigittimeout. IOS VB was 10s default. VVB seems to be 3s default. #### 3s was optimal value used. In future it should be configurable. Form element. DTMF Grammar builtin:dtmf/digits. IOS VB - it used to work at some point. VVB doesn't seem to work. #### Problem is that if we use form element the grammar is not explicitly specified by vxmlserver (compared to how it is received for digit element). So VVB doesn't know how many digits to expect. Example is below for what is received for form element.   <form id="start">^M     <property name="inputmodes" value="dtmf" />^M     <block>^M       <assign name="audium_vxmlLog" expr="audium_vxmlLog + '|||audio_group$$$' + 'initial_audio_group' + '^^^' + application.getElapsedTime(audium_element_start_time_millisecs)" />^M       <goto nextitem="foundation_fld" />^M     </block>^M     <field name="foundation_fld" modal="false">^M       <property name="inputmodes" value="dtmf" />^M       <property name="com.cisco.secureLogging" value="false" />^M       <prompt bargein="true">^M         <audio src="/CVP/audio/sample.wav" />^M       </prompt>^M ... View more

This always worked since 11.5 and nothing to do with ES81. But I believe alternative text for a prompt had an issue in ES80. Example: "com.cisco.tts-server" can be set to sip:tts@hostname Continuing with the above example: 1. Add TTS server (use hostname instead of IP) 2. Add dns entry for hostname should be added through CLI. Thanks, Anjum ... View more

Use ES53 which was released this week. That has another fix for multi-lang where MRCP Msg length was calculated incorrectly.  ... View more

Dilip, Not sure how IOS-VB will handle when one of the server is unreachable.  You can configure a wrong IP in one of the dial-peer and check how it behaves. I can comment on CVVB behavior in this scenario, Lets say you have these TTS servers configured: 10.10.10.80/81/90/73 If any of these is out-of-service or unreachable (marked as unreachable on appadmin), then that server will be skipped in the server selection logic. So as per your example, 80 will not be considered while selecting a TTS server. Thanks, Anjum ... View more

Handoff error can be because of the maximum number of handoffs (i.e.100) is reached. 100 is hardcoded in IOS-GW code. You can still manage 90m>EWT>60m. We have recently tested for around 80m and we didn't see handoff failures. But If we go close to 90m, we start seeing the issue. Thanks ... View more

MRCPv2 sessions are setup using SIP. So you could CUSP and have Nuance server in a server group. Simpler approach is to have dial-peers on IOS-GW load balance between the Nuance server. Here is an example for this approach where we load balance between 3 Nuance server and putting upper limit on max sessions for each server:- Step 1: create voice class uri for ASRs VBGW4(config)#voice class uri asr sip VBGW4(config-voice-uri-class)#pattern asr@10.10.10.80 VBGW4(config-voice-uri-class)#exi VBGW4(config)#exi VBGW4#sh conf | s asr ip host asr-en-us 10.10.10.80 voice class uri asr sip pattern asr@10.10.10.80 Step 2: Add Dial-Peers which define destination uri for 'asr' dial-peer voice 9990207 voip max-conn 94 session protocol sipv2 session target ipv4:10.10.10.81 destination uri asr dtmf-relay rtp-nte codec g711ulaw no vad dial-peer voice 9990201 voip max-conn 94 session protocol sipv2 session target ipv4:10.10.10.90 destination uri asr dtmf-relay rtp-nte codec g711ulaw no vad dial-peer voice 9990202 voip max-conn 94 session protocol sipv2 session target ipv4:10.10.10.73 destination uri asr dtmf-relay rtp-nte codec g711ulaw no vad ... View more

Exjun, If you are using 10.5 or later version, you could divide the elements across multiple subdialogs. These are like sub-routines and can make the application modular and then you o not have to worry of elements in a page. See attached screen shot as an example. You do not have limit on number of elements in  a call as such. Refer to " Element Specifications for Cisco Unified CVP VXML Server and Cisco Unified Call Studio" for more details. Thanks ... View more

Gerry, Thank you for the reply. I can take this suggestion to the team and see if this can be improved. As mentioned earlier this was intentionally done during the time subdialog was implemented for the studio. See the attached screen shot which shows the tab name mapping for each flow. If we use the app name for the main flow, it would be inconsistent when we have subdialogs. Thanks, Anjum ... View more

This is expected behavior of CallStudio 11.0. The main flow name is app.callflow. Thanks ... View more