on 01-24-2014 04:02 PM
We are running into this same issue. DTMF shows up in the debugs as the only inputmode. The ASR leg comes up and uses a license for the duration of the call while in the app. We are trying to move from MRCPv1 to MRCPv2. We are using Nuance and 3945 gateways running 15.5(3)M3. We have many ivr apps and all will use ASR licenses if we can't find a solution. We have 1 app that needs ASR. TAC is telling us that they believe this is working as intended. I was hoping this thread found a solution and can give us a heads up.
Thanks,
Steve
Where you don't need ASR, set the adapter type to Cisco DTMF.
Remove the ivr asr-server setting from the gateway.
Where you do need ASR, specify it in your apps using the com.cisco.asr-server VoiceXML property.
Someone to talk too about this! Thanks for the reply.
We need ASR on 1 app. This likely means we need to keep ivr asr-server in the gateway. With that said, we use “Cisco Unified CVP 4.1/7.0/8.0/8.5/9.0 with Cisco DTMF" in the ivr app. The script nodes all use "D". The two things I just mentioned are a little outside my wheel house. I'm the CUCM/VXML Router person. In the debugs I see <property name="inputmodes" value="dtmf" throughout the entire debug. We still get an ASR session up on all calls. Removing the ivr asr-server and the above configuration works but for that 1 app that needs ASR, it won't work with ivr asr-server removed correct? or are you saying that ivr asr-server can be removed from the gateway and the application can call is via something set in com.cisco.asr-server VoiceXML property within the app?
Thanks a ton for the info.
Remove ivr asr-server from the gateway; if it's there, the gateway will establish sessions. In your CVP Call Studio app that needs to use ASR, add the VoiceXML property com.cisco.asr-server and set it to your recogniser SIP URI. You can do this in the application properties Root Doc Settings section. Redeploy the app.
Paul
This worked great Paul! I now get to make that call back to TAC to explain what was done to resolve. The only piece we don't have working yet is the SIP URI. We must use IP Address rather than the name that's configured as a ip host entry on the router. This makes things a tad more difficult but manageable unlike our previous issue. Thanks again!
No reason why host name ahouldn't work for you. If you use a hostname in the VoiceXML property and have a matching pattern it will work, provided of course you then resolve it OK statically or with DNS as normal.
For example, in the lab I currently have this set up:
voice class uri tts sip
pattern tts@nuance.rmlab.cisco.com
and com.cisco.tts-server set to sip:tts@nuance.rmlab.cisco.com
That is exactly right and worked like a champ! I had the ip host entry but still had the URI pointing to an ip address. Thanks Paul.
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