El Lenguaje de marcado extensible de la voz (VXML) es una norma establecida por el World Wide Web Consortium (W3C). Se ha diseñado para crear diálogos audio que proporcionen habla sintetizada, reconocimiento de las palabras habladas, reconocimiento de dígitos DTMF y grabaciones del audio hablado. El servidor VXML y los clientes utilizan el conocido protocolo HTTP para intercambiar documentos VXML/páginas.
Cisco Voice Portal (CVP) ofrece aplicaciones de respuesta de voz (IVR) inteligentes e interactivas a las que se puede acceder a través del teléfono. Hay tres tipos de implementaciones de CVP:
Servicio independiente
Control de llamadas CVP
Cola de llamadas y transferencia
Las funciones de voz sintetizada y el reconocimiento de las palabras habladas / dígitos DTMF se proporcionan mediante Text-to-Speech (TTS) y Automatic Speech Recognition Servers (ASR). La puerta de enlace VXML IOS® se comunica con el servidor TTS/ASR a través del protocolo de control de recursos de medios (MRCP). Hay dos versiones de MRCP (RFC 4463), a saber, MRCPv1 (MRCP sobre RTSP) y MRCPv2 (MRCP sobre SIP).
Este documento describe el flujo de llamadas de una puerta de enlace XML de voz IOS a una llamada CVP en una implementación de servicio independiente que utiliza servidores MRCPv2 TTS / ASR. Se implementó una aplicación de farmacia de ejemplo en el servidor CVP VXML.
No hay requisitos específicos para este documento.
La información que contiene este documento se basa en las siguientes versiones de software y hardware.
Gateway VXML de IOS: Cisco AS5400XM, IOS 12.4(15)T1
Servidor VXML: CVP 4.0
Servidor ASR/TTS: Loquendo Speech Suite 7.0
The information in this document was created from the devices in a specific lab environment. All of the devices used in this document started with a cleared (default) configuration. If your network is live, make sure that you understand the potential impact of any command.
En esta sección encontrará la información para configurar las funciones descritas en este documento.
Nota: Utilice la herramienta Command Lookup (sólo para clientes registrados) para obtener más información sobre los comandos utilizados en esta sección.
En este documento, se utiliza esta configuración de red:
En este documento, se utilizan estas configuraciones:
Configuración de gateway VXML |
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!--- Define Hostname to IP Address !---- mapping for ASR and TTS servers ip host asr-en-us 172.18.110.76 ip host tts-en-us 172.18.110.76 !--- Define the Voice class URI to match !---- the SIP URI of ASR Server in the dial-peer voice class uri TTS sip pattern tts@172.18.110.76 !--- Define the Voice class URI to match !---- the SIP URI of TTS server in the dial-peer voice class uri ASR sip pattern asr@172.18.110.76 !--- Define the amount of maximum memory !---- to used for downloaded prompts ivr prompt memory 15000 !--- Define the SIP URI of ASR !---- and TTS Server ivr asr-server sip:asr@172.18.110.76 ivr tts-server sip:tts@172.18.110.76 !--- Configure an application service for !---- CVP VXML CVPSelfServiceBootstrap.vxml application service CVPSelfService flash: CVPSelfServiceBootstrap.vxml paramspace english language en paramspace english index 0 paramspace english location flash: paramspace english prefix en !--- Configure an application service for !---- CVP VXML CVPSelfService.tcl Script !--- CVPSelfService-app parameter specifies !---- the name of the VXML Application !--- CVPPrimary parameter specifies the !---- IP address of the VXML server service Pharmacy flash:CVPSelfService.tcl paramspace english index 0 paramspace english language en paramspace english location flash: param CVPSelfService-port 7000 param CVPSelfService-app GoodPrescriptionRefillApp7 paramspace english prefix en param CVPPrimaryVXMLServer 172.18.110.75 !--- Specifies the Gateway’s RTP !---- stream to the ASR / TTS to go around the !---- Content Service Switch !---- instead of through the CSS. mrcp client rtpsetup enable !--- Specify the maximum memory size !---- for the HTTP Client Cache http client cache memory pool 15000 !--- Specify the maximum number of file !---- that can be stored in the !---- HTTP Client Cache http client cache memory file 500 !--- Disable Persistent !---- HTTP Connections no http client connection persistent !--- Configure the T1 PRI controller T1 3/0 framing esf linecode b8zs pri-group timeslots 1-24 !--- Configure the ISDN switch !---- type and incoming-voice !---- under the D-channel interface interface Serial3/0:23 no ip address encapsulation hdlc isdn switch-type primary-net5 isdn incoming-voice modem no cdp enable ! --- Configure a POTS !---- dial-peer that will be used !---- as inbound dial-peer for calls coming ! --- in across the T1 PRI line. !---- The “pharmacy”service !---- is applied under this dial-peer. dial-peer voice 1 pots service pharmacy destination-pattern 5555 direct-inward-dial port 3/0:D forward-digits all !--- Configure a SIP Voip !---- dial-peer that will be used !---- as an outbound dial-peer when the !---Gateway initiates a MRCP overc SIP !---- session to the ASR server. !---- Codec = G711ulaw, DTMF-Relay !---- = RTP-NTE, No Vad dial-peer voice 5 voip session protocol sipv2 destination uri ASR dtmf-relay rtp-nte codec g711ulaw no vad !--- Configure a SIP Voip !---- dial-peer that will be used !---- as an outbound dial-peer when the !---Gateway initiates a MRCP !---- overc SIP session to the TTS server !--- Codec = G711ulaw, DTMF-Relay = RTP-NTE, !---- No Vad dial-peer voice 6 voip session protocol sipv2 destination uri TTS dtmf-relay rtp-nte codec g711ulaw no vad |
En esta sección se describe el flujo de llamada que resulta de este ejemplo de configuración.
Una llamada ISDN llega a la puerta de enlace PSTN/VXML a través de T1 PRI 3/0.
La puerta de enlace del IOS coincide con el dial-peer POTS 1 como el dial-peer entrante para esta llamada.
La puerta de enlace del IOS entrega el control de llamada al servicio Pharmacy que está asociado al dial-peer 1.
El script VXML / TCL de CVP asociado al servicio Farmacia envía una solicitud GET HTTP al servidor VXML.
El servidor VXML devuelve una respuesta de 200 OK. Esta respuesta contiene un documento VXML / página.
La puerta de enlace del IOS ejecuta el documento VXML.
Si el documento VXML especifica una dirección URL para un mensaje de audio, la puerta de enlace del IOS descarga el archivo de audio y reproduce el mensaje.
Si el documento VXML especifica un texto para un mensaje de audio, la puerta de enlace del IOS establece una sesión SIP con tts@172.18.110.76 (servidor TTS) utilizando dial-peer 5. Después de establecer la sesión SIP, abre una conexión TCP al servidor TTS utilizando el número de puerto TCP proporcionado en la respuesta SDP de 200 OK de SIP INVITE. Esta conexión TCP se utiliza para intercambiar mensajes MRCP como SPEAK, SPEAK-COMPLETE entre la gateway IOS y el servidor TTS.
El servidor TTS envía el flujo de audio RTP G.711ulaw a la dirección IP y al número de puerto UDP proporcionado por la puerta de enlace en el SDP de SIP INVITE.
Si el documento VXML especifica la puerta de enlace para reconocer dígitos DTMF y/o palabras habladas, la puerta de enlace del IOS establece una sesión SIP con asr@172.18.110.76 (servidor ASR) con dial-peer 6. Después de establecer la sesión SIP, abre una conexión TCP al servidor ASR utilizando el número de puerto TCP proporcionado en la respuesta SDP de 200 OK de SIP INVITE. Esta conexión TCP se utiliza para intercambiar mensajes MRCP tales como DEFINIR GRAMMAR, COMPLETE, RECONOCER y RECONOCIMIENTO COMPLETO entre la gateway IOS y el servidor ASR.
La gateway VXML del IOS envía el flujo de audio RTP G.711ulaw a la dirección IP y al número de puerto UDP proporcionado por el ASR en el SDP de la respuesta OK del SIP 200. La gateway VXML del IOS envía los dígitos introducidos por el usuario PSTN como eventos RTP-NTE al servidor ASR.
Después de la ejecución del documento VXML, el gateway envía una solicitud HTTP POST (con un conjunto de parámetros) como se especifica en la etiqueta <send> del documento VXML / página.
Los pasos 6 - 10 se producen para cada documento VXML enviado por el servidor.
Cuando la aplicación VXML finaliza el servicio proporcionado a la persona que llama, envía un documento VXML con sólo una etiqueta <exit/> dentro del elemento <form>.
La gateway IOS desconecta las sesiones MRCPv2 establecidas con los servidores TTS y ASR.
La puerta de enlace del IOS desconecta la llamada en el lado ISDN.
Use esta sección para confirmar que su configuración funciona correctamente.
La herramienta Output Interpreter Tool (clientes registrados solamente) (OIT) soporta ciertos comandos show. Utilice la OIT para ver un análisis del resultado del comando show.
Mostrar resumen de voz activa de llamada
11F8 : 160 333356110ms. 1 +10 pid:1 Answer 5555 active dur 00:00:54 tx:1740/300598 rx:364/85472 Tele 3/0:D (160) [3/0.1] tx:15145/15145/0ms None noise:-52 acom:6 i/0:-32/-64 dBm Telephony call-legs: 1 SIP call-legs: 0 H323 call-legs: 0 Call agent controlled call-legs: 0 SCCP call-legs: 0 Multicast call-legs: 0 Media call-legs: 0 Total call-legs: 1
Mostrar resumen de medios activos de llamada
11F8 : 163 333360880ms.1 +60 pid:6 Originate sip:tts@172.18.110.76:5060 active dur 00:00:44 tx:0/0 rx:2212/353545 IP 172.18.110.76:10000 SRTP: off rtt:0ms pl: 4485/0ms lost:0/1/0 delay:65/65/65ms g711ulaw TextRelay: off media inactive detected:n media contrl rcvd: n/a timestamp:n/a long duration call detected:n long duration call duration:n/a timestamp:n/a11F8 : 164 333360890ms.1 +20 pid:5 Originate sip:asr@172.18.110.76:5060 active dur 00:00:44 tx:1687/297152 rx:0/0 IP 172.18.110.76:10002 SRTP: off rtt:0ms pl:6550/30ms lost:0/2/0 delay:65/65/65ms g711ulaw TextRelay: off media inactive detected:n media contrl rcvd:n/a timestamp:n/a long duration call detected:n long duration call duration:n/a timestamp:n/a Telephony call-legs: 0 SIP call-legs: 0 H323 call-legs: 0 Call agent controlled call-legs: 0 SCCP call-legs: 0 Multicast call-legs: 0 Media call-legs: 2 Total call-legs: 2
Mostrar detalle activo de la sesión del cliente mrcp
No Of Active MRCP Sessions: 1 Call-ID: 0xA0 same: 0 -------------------------------------------- Resource Type: Synthesizer URL: sip:tts@172.18.110.76 Method In Progress: SPEAK State: S_SYNTH_SPEAKING Associated CallID: 0xA3 MRCP version: 2.0 Control Protocol: TCP Server IP Address: 172.18.110.76 Port: 51000 Data Protocol: RTP Server IP Address: 172.18.110.76 Port: 10000 Signalling URL: sip:tts@172.18.110.76:5060 Packets Transmitted: 0 (0 bytes) Packets Received: 2265 (361968 bytes) ReceiveDelay: 65 LostPackets: 0 -------------------------------------------- -------------------------------------------- Resource Type: Recognizer URL: sip:asr@172.18.110.76 Method In Progress: RECOGNIZE State: S_RECOG_RECOGNIZING Associated CallID: 0xA4 MRCP version: 2.0 Control Protocol: TCP Server IP Address: 172.18.110.76 Port: 51001 Data Protocol: RTP Server IP Address: 172.18.110.76 Port: 10002 Packets Transmitted: 1791 (313792 bytes) Packets Received: 0 (0 bytes) ReceiveDelay: 60 LostPackets: 0
Show voip rtp connections
VoIP RTP active connections : No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP 1 163 160 18964 10000 14.1.16.25 172.18.110.76 2 164 160 23072 10002 14.1.16.25 172.18.110.76 Found 2 active RTP connections
Mostrar caché del cliente http
HTTP Client cached information ============================== Maximum memory pool allowed for HTTP Client caching = 15000 K-bytes Maximum file size allowed for caching = 500 K-bytes Total memory used up for Cache = 410 Bytes Message response timeout = 10 secs Total cached entries = 1 Total non-cached entries = 0 Cached entries ============== entry 114, 1 entries Ref FreshTime Age Size context --- --------- --- ---- ------- 1 86400 48 1505 0 url: http://172.18.110.75/Welcome-1.wav
En esta sección encontrará información que puede utilizar para solucionar problemas de configuración.
Configure la puerta de enlace del IOS para registrar las depuraciones en su búfer de registro y desactivar la "consola de registro".
Nota: Consulte Información Importante sobre Comandos Debug antes de utilizar los comandos debug.
Nota: Estos son los comandos que se utilizan para configurar la puerta de enlace para almacenar las depuraciones en el búfer de registro de la puerta de enlace:
service timestamps debug datetime msec
secuencia de servicio
no logging console
logging buffered 5000000 debug
clear log
Los siguientes son los comandos debug utilizados para resolver problemas de la configuración:
debug isdn q931
debug voip ccapi inout
debug voip application vxml default
debug voip application vxml dump
debug ccsip message
debug mrcp detail
debug http client all
debug voip rtp session nte named-event
Esta sección proporciona resultados de depuración para este flujo de llamada de ejemplo:
La puerta de enlace coincide con el par de marcado 1 entrante.
La puerta de enlace inicia la ejecución del script CVPSelfServiceBootstrap.vxml VoiceXML.
La puerta de enlace envía una solicitud GET HTTP al servidor VXML.
El códec G711ulaw, la dirección IP y los números de puerto RTP para la secuencia de audio
El atributo de dirección de esta secuencia RTP: "recvonly"
El relé DTMF basado en RTP-NTE
El número de puerto TCP (51001) que utilizará la puerta de enlace para establecer una sesión MRCPv2 con el servidor ASR
La puerta de enlace recibe una respuesta COMPLETA de 200 para su solicitud DEFINE-GRAMMAR.
Códec G711ulaw, dirección IP y números de puerto RTP para la secuencia de audio
El atributo de dirección de esta secuencia RTP:"sendonly"
El relé DTMF basado en RTP-NTE
El número de puerto TCP (51000) que utilizará la puerta de enlace para establecer una sesión MRCPv2 con el servidor TTS
El servidor ASR envía una respuesta "EN CURSO" (para la solicitud RECONOCER) a la puerta de enlace.
El servidor TTS envía una respuesta "EN CURSO" a la solicitud SPEAK.
Gateway envía una solicitud MRCP "SPEAK" al servidor TTS para reproducir el mensaje "Menu" (Menú) (Ingrese 1 o Diga Refil / Ingrese 2 o diga farmacéutico). (No se muestran los resultados de la depuración.)
El servidor TTS envía un mensaje IN-PROGRESS, SPEAK-COMPLETE y finaliza la reproducción del mensaje. (No se muestran los resultados de la depuración.)
A continuación, el servidor VXML envía otro documento VXML que solicita a la persona que llama que introduzca aquí la receta. (No se muestran los resultados de la depuración.)
La puerta de enlace envía el mensaje MRCP a TTS para que hable las indicaciones. (No se muestran las salidas de depuración, pero son similares a los pasos 28-30.)
Gateway envía el mensaje MRCP al ASR para detectar el número de prescripción de 4 dígitos que habla el usuario. (No se muestran las salidas de depuración, pero son similares a los pasos 25-26.)
La puerta de enlace informa el número de prescripción al servidor VXML enviando una solicitud HTTP POST. (No se muestran las salidas de depuración, pero son similares al paso 35.)
El servidor VXML envía páginas VXML para recopilar el tiempo de recogida e informar a la persona que llama de que la receta estará lista para la recogida. La puerta de enlace ejecuta estas páginas mediante interacciones con el servidor TTS y ASR. (No se muestran los resultados de la depuración.)
La puerta de enlace desconecta la sesión SIP establecida con el servidor ASR.
La puerta de enlace desconecta la sesión SIP establecida con el servidor TTS.
*Jan 18 03:34:52.735: ISDN Se3/0:23 Q931: RX <- SETUP pd = 8 callref = 0x005A Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Called Party Number i = 0x81, '5555' Plan:ISDN, Type:Unknown *Jan 18 03:34:52.735: //-1/2AEE8C2A801C/ CCAPI/cc_api_display_ie_subfields: cc_api_call_setup_ind_common: cisco-username= ----- ccCallInfo IE subfields ----- cisco-ani= cisco-anitype=0 cisco-aniplan=0 cisco-anipi=0 cisco-anisi=0 dest=5555 cisco-desttype=0 cisco-destplan=1 cisco-rdie=FFFFFFFF cisco-rdn= cisco-rdntype=-1 cisco-rdnplan=-1 cisco-rdnpi=-1 cisco-rdnsi=-1 cisco-redirectreason=-1 fwd_final_type =0 final_redirectNumber = hunt_group_timeout =0
*Jan 18 03:34:52.735: //-1/2AEE8C2A801C/ CCAPI/cc_api_call_setup_ind_common: Interface=0x664B4BA4, Call Info( Calling Number=,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed), Called Number=5555(TON=Unknown, NPI=ISDN), Calling Translated=FALSE, Subscriber Type Str=RegularLine, FinalDestinationFlag=TRUE, Incoming Dial-peer=1, Progress Indication=NULL(0), Calling IE Present=FALSE, Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=-1
*Jan 18 03:34:52.739: //127/2AEE8C2A801C/CCAPI /cc_process_call_setup_ind: >>>>CCAPI handed cid 127 with tag 1 to app "_ManagedAppProcess_Pharmacy" *Jan 18 03:34:52.739: //127/2AEE8C2A801C/CCAPI/ccCallSetupAck: Call Id=127
*Jan 18 03:34:52.739: ISDN Se3/0:23 Q931: TX -> CONNECT pd = 8 callref = 0x805A *Jan 18 03:34:52.739: //127/2AEE8C2A801C/CCAPI/ccCallHandoff: Silent=FALSE, Application=0x663106C4, Conference Id=0xFFFFFFFF *Jan 18 03:34:52.743: //127//VXML:/Open_CallHandoff:
*Jan 18 03:34:52.755: //127/2AEE8C2A801C/VXML: /vxml_vxml_proc: <vxml> URI(abs):flash: CVPSelfServiceBootstrap.vxml scheme=flash path=CVPSelfServiceBootstrap.vxml base= URI(abs):flash: CVPSelfServiceBootstrap.vxml scheme=flash path=CVPSelfServiceBootstrap.vxml lang=none version=2.0 <script>: *Jan 18 03:34:52.799: //127/2AEE8C2A801C/VXML :/vxml_expr_eval: *Jan 18 03:34:52.863: //127/2AEE8C2A801C/VXML :/vxml_jse_global_switch: switch to scope(application) <var>: namep=handoffstring expr=session.handoff_string *Jan 18 03:34:52.867: //127/2AEE8C2A801C/VXML :/vxml_expr_eval: expr=(var handoffstring=session. handoff_string) <var>: namep=application expr=getValue('APP') *Jan 18 03:34:52.867: //127/2AEE8C2A801C/VXML :/vxml_expr_eval: expr=(var application=getValue('APP')) <var>: namep=port expr=getValue('PORT') *Jan 18 03:34:52.867: //127/2AEE8C2A801C/VXML :/vxml_expr_eval: expr=(var port=getValue('PORT')) <var>: namep=callid expr=getValue('CALLID') *Jan 18 03:34:52.867: //127/2AEE8C2A801C/VXML :/vxml_expr_eval: expr=(var callid=getValue('CALLID')) <var>: namep=servername expr=getValue('PRIMARY') *Jan 18 03:34:52.867: //127/2AEE8C2A801C/VXML :/vxml_expr_eval: expr=(var servername=getValue('PRIMARY')) <var>: namep=var1 expr=getValue('var1') *Jan 18 03:34:52.867: //127/2AEE8C2A801C/VXML :/vxml_expr_eval: expr=(var var1=getValue('var1')) <var>: namep=var2 expr=getValue('var2') *Jan 18 03:34:52.867: //127/2AEE8C2A801C/VXML :/vxml_expr_eval: expr=(var var2=getValue('var2')) <var>: namep=var3 expr=getValue('var3') *Jan 18 03:34:52.867: //127/2AEE8C2A801C/VXML :/vxml_expr_eval: expr=(var var3=getValue('var3')) <var>: namep=var4 expr=getValue('var4') *Jan 18 03:34:52.867: //127/2AEE8C2A801C/VXML :/vxml_expr_eval: expr=(var var4=getValue('var4')) <var>: namep=var5 expr=getValue('var5') *Jan 18 03:34:52.867: //127/2AEE8C2A801C/VXML :/vxml_expr_eval: expr=(var var5=getValue('var5')) <var>: namep=status expr=getValue('status') *Jan 18 03:34:52.867: //127/2AEE8C2A801C/VXML :/vxml_expr_eval: expr=(var status=getValue('status')) <var>: namep=prevapp expr=getValue('prevapp') *Jan 18 03:34:52.871: //127/2AEE8C2A801C/VXML :/vxml_expr_eval: expr=(var prevapp=getValue('prevapp')) <var>: namep=survive expr=getValue('survive') *Jan 18 03:34:52.871: //127/2AEE8C2A801C/VXML :/vxml_expr_eval: expr=(var survive=getValue('survive')) <var>: namep=handoffExit
*Jan 18 03:34:52.875: //127//HTTPC:/httpc_write_stream: Client write buffer fd(3): GET /CVP/Server?application= GoodPrescriptionRefillApp7&callid= 2AEE8C2A-0AFB11D6-801C0013- 803E8C8E&session.connection.remote.uri=555 5&session.connection.local.uri=5555 HTTP/1.1 Host: 172.18.110.75:7000 Content-Type: application/x-www-form-urlencoded Connection: close Accept: text/vxml, text/x-vxml, application/vxml, application/x-vxml, application/voicexml, application/x-voicexml, text/plain, tex t/html, audio/basic, audio/wav, multipart/form-data, application/octet-stream User-Agent: Cisco-IOS-C5400/12.4
El cuerpo del mensaje de esta respuesta contiene un documento VXML (1). El documento VXML indica a la puerta de enlace el archivo de medios de reproducción denominado Welcome-1.wav ubicado en un servidor de medios.
*Jan 18 03:34:52.883: processing server rsp msg: msg(67CA63A8) URL:http://172.18.110.75:7000/CVP/ Server?application=GoodPrescription RefillApp7&callid=2AEE8C2A-0AFB11D6-801C0013 -803E8C8E&session.connection. remote.uri=5555&session.connection.local. uri=5555, fd(3): *Jan 18 03:34:52.883: Request msg: GET /CVP/Server?application= GoodPrescriptionRefillApp7&callid= 2AEE8C2A-0AFB11D6-801C0013-803E8C8 E&session.connection.remote. uri=5555&session .connection.local.uri=5555 HTTP/1.1 *Jan 18 03:34:52.883: Message Response Code: 200 *Jan 18 03:34:52.883: Message Rsp Decoded Headers: *Jan 18 03:34:52.883: Date:Mon, 30 Apr 2007 16:58:39 GMT *Jan 18 03:34:52.883: Content-Type:text/xml; charset=ISO-8859-1 *Jan 18 03:34:52.883: Connection:close *Jan 18 03:34:52.883: Set-Cookie:JSESSIONID= BBCE0F948ADFDB720497F587A7997538; Path=/CVP *Jan 18 03:34:52.883: headers: *Jan 18 03:34:52.883: HTTP/1.1 200 OK Server: Apache-Coyote/1.1 Set-Cookie: JSESSIONID=BBCE0F948ADF DB720497F587A7997538; Path=/CVP Content-Type: text/xml;charset=ISO-8859-1 Date: Mon, 30 Apr 2007 16:58:39 GMT Connection: close *Jan 18 03:34:52.883: body: *Jan 18 03:34:52.883: <?xml version="1.0" encoding="UTF-8"?> <vxml version="2.0" application= "/CVP/Server?audium_root=true& calling_into=GoodPrescriptionRefillApp7" xml:lang="en-us"> <form id="audium_start_form"> <block> <assign name="audium_vxmlLog" expr="''" /> <assign name="audium_element _start_time_millisecs" expr="new Date().getTime()" /> <goto next="#start" /> </block> </form> <form id="start"> <block> <prompt bargein="true"> <audio src="http://172.18.110.75/ Welcome-1.wav" /> </prompt> <assign name="audium_vxmlLog" expr="audium_vxmlLog + '|||audio_group$$$' + 'initial_audio_group' + '^^^' + application.getEla psedTime(audium_element_start_time_millisecs)" /> <submit next="/CVP/Server" method="post" namelist=" audium_vxmlLog" /> </block> </form> </vxml>
GET /Welcome-1.wav HTTP/1.1 Host: 172.18.110.75 Content-Type: application/x-www-form-urlencoded Connection: close Accept: text/vxml, text/x-vxml, application/vxml, application/x-vxml, application/voicexml, application/x-voicexml, text/plain, tex t/html, audio/basic, audio/wav, multipart/form-data, application/octet-stream User-Agent: Cisco-IOS-C5400/12.4
*Jan 18 03:34:55.647: //127//HTTPC:/httpc_socket_read: *Jan 18 03:34:55.647: read data from the socket 3 : first 400 bytes of data: HTTP/1.1 200 OK Content-Length: 26450 Content-Type: audio/wav Last-Modified: Mon, 30 Apr 2007 15:36:51 GMT Accept-Ranges: bytes ETag: "e0c1445f3d8bc71:2d6" Server: Microsoft-IIS/6.0 Date: Mon, 30 Apr 2007 16:58:42 GMT Connection: close RIFFJg(Unprintable char...) 0057415645666D7420120001010401 F00401F00108000666163744000176700 64617461176700FFFFFF807 FFFFFFF80FFFFFF80F (other hex information not shown).
POST /CVP/Server HTTP/1.1 Host: 172.18.110.75:7000 Content-Length: 67 Content-Type: application/x-www-form-urlencoded Cookie: $Version=0; JSESSIONID=BBCE0F948 ADFDB720497F587A7997538; $Path=/CVP Connection: close Accept: text/vxml, text/x-vxml, application/vxml, application/x-vxml, application/voicexml, application/x-voicexml, text/plain, tex t/html, audio/basic, audio/wav, multipart/form-data, application/octet-stream User-Agent: Cisco-IOS-C5400/12.4
El cuerpo del mensaje contiene el documento VXML (2). El documento VXML indica a la puerta de enlace que reproduzca "Gracias por llamar a la farmacia audium". Tenga en cuenta que un servidor de texto a voz debe sintetizar este mensaje.
*Jan 18 03:34:55.651: processing server rsp msg: msg(67CA6960)URL: http://172.18.110.75: 7000/CVP/Server, fd(4): *Jan 18 03:34:55.651: Request msg: POST /CVP/Server HTTP/1.1 *Jan 18 03:34:55.651: Message Response Code: 200 *Jan 18 03:34:55.651: Message Rsp Decoded Headers: *Jan 18 03:34:55.651: Date:Mon, 30 Apr 2007 16:58:42 GMT *Jan 18 03:34:55.651: Content-Type:text/xml; charset=ISO-8859-1 *Jan 18 03:34:55.651: Connection:close *Jan 18 03:34:55.651: headers: *Jan 18 03:34:55.651: HTTP/1.1 200 OK Server: Apache-Coyote/1.1 Content-Type: text/xml;charset=ISO-8859-1 Date: Mon, 30 Apr 2007 16:58:42 GMT Connection: close *Jan 18 03:34:55.655: body: *Jan 18 03:34:55.655: <?xml version="1.0" encoding="UTF-8"?> <vxml version="2.0" application= "/CVP/Server?audium_root=true& calling_into=GoodPrescriptionRefillApp7" xml:lang="en-us"> <form id="audium_start_form"> <block> <assign name="audium_vxmlLog" expr="''" /> <assign name="audium_element _start_time_millisecs" expr="new Date().getTime()" /> <goto next="#start" /> </block> </form> <form id="start"> <block> <prompt bargein="true"> Thank you for calling Audium pharmacy. </prompt> <assign name="audium_vxmlLog" expr= "audium_vxmlLog + '|||audio_group$$$' + 'initial_audio_group' + '^^^' + application.getEla psedTime(audium_element_start_time_millisecs)" /> <submit next="/CVP/Server" method="post" namelist=" audium_vxmlLog" /> </block> </form> </vxml>
*Jan 18 03:34:55.667: //127//HTTPC:/httpc_write_stream: Client write buffer fd(4): POST /CVP/Server HTTP/1.1 Host: 172.18.110.75:7000 Content-Length: 67 Content-Type: application/x-www-form-urlencoded Cookie: $Version=0; JSESSIONID= BBCE0F948ADFDB720497F587A7997538; $Path=/CVP Connection: close Accept: text/vxml, text/x-vxml, application/vxml, application/x-vxml, application/voicexml, application/x-voicexml, text/plain, tex t/html, audio/basic, audio/wav, multipart/form-data, application/octet-stream User-Agent: Cisco-IOS-C5400/12.4
El cuerpo del mensaje contiene el documento VXML (3). Este documento VXML define un mensaje de menú que indica al autor de la llamada que introduzca 1 o diga Refill, o que introduzca 2 o diga farmacéutico. Los mensajes son sintetizados por un servidor de texto a voz. Las entradas (voz / DTMF) se reconocen con un Reconocidor de voz automático.
*Jan 18 03:34:57.499: processing server rsp msg: msg(67CA6B48)URL: http://172.18.110.75:7000/CVP/Server, fd(4): *Jan 18 03:34:57.499: Request msg: POST /CVP/Server HTTP/1.1 *Jan 18 03:34:57.499: Message Response Code: 200 *Jan 18 03:34:57.499: Message Rsp Decoded Headers: *Jan 18 03:34:57.499: Date:Mon, 30 Apr 2007 16:58:42 GMT *Jan 18 03:34:57.499: Content-Type:text/xml;charset=ISO-8859-1 *Jan 18 03:34:57.499: Connection:close *Jan 18 03:34:57.499: headers: *Jan 18 03:34:57.499: HTTP/1.1 200 OK Server: Apache-Coyote/1.1 Content-Type: text/xml;charset=ISO-8859-1 Date: Mon, 30 Apr 2007 16:58:42 GMT Connection: close *Jan 18 03:34:57.499: body: *Jan 18 03:34:57.499: ... Buffer too large - truncated to (4096) len. *Jan 18 03:34:57.499: <?xml version="1.0" encoding="UTF-8"?> <vxml version="2.0" application= "/CVP/Server?audium_root=true& calling_into=GoodPrescriptionRefillApp7" xml:lang="en-us"> <property name="timeout" value="60s" /> <property name="confidencelevel" value="0.40" /> <form id="audium_start_form"> <block> <assign name="audium_vxmlLog" expr="''" /> <assign name="audium_element _start_time_millisecs" expr="new Date().getTime()" /> <goto next="#start" /> </block> </form> <form id="start"> <block> <assign name="audium_vxmlLog" expr="audium_vxmlLog + '|||audio_group$$$' + 'initial_audio_group' + '^^^' + application.getElapsedTime (audium_element_start_time_millisecs)" /> <goto nextitem="choice_fld" /> </block> <field name="choice_fld" modal="false"> <property name="inputmodes" value="dtmf voice" /> <prompt bargein="true">Say refills or press 1. Or. Say pharmacist or press 2.</prompt> <catch event="nomatch"> <prompt bargein="true">Sorry. I did not understand that. Say refills or press 1. Say pharmacist or press 2.</prompt> <assign name="audium_vxmlLog" expr="audium_vxmlLog + '|||nomatch$$$' + '1' + '^^^' + application.getElapsedTime (audium_element_start_time_millisecs)" /> <assign name="audium_vxmlLog" expr="audium_vxmlLog + '|||audio_group$$$' + 'nomatch_audio_group' + '^^^' + application.getElapsedTime( audium_element_start_time_millisecs)" /> </catch> <catch event="nomatch" count="2"> <prompt bargein="true"> Sorry, I still did not get that. If you are using a speaker phone. Please use the phone keypad to make your selection. Press 1 for refills. Press 2 to speak to a pharmacist.</prompt> <assign name="audium_vxmlLog" expr="audium_vxmlLog + '|||nomatch$$$' + '2' + '^^^' + application.getElapsedTime (audium_element_start_time_millisecs)" /> <assign name="audium_vxmlLog" expr="audium_vxmlLog + '|||audio_group$$$' + 'nomatch_audio_group' + '^^^' + application.getElapsedTime (audium_element_start_time_millisecs)" /> </catch> <catch event="nomatch" count="3"> <prompt bargein="true">Gee. Looks like we are having some trouble.</prompt> <assign name="audium_vxmlLog" expr="audium_vxmlLog + '|||nomatch$$$' + '3' + '^^^' + application.getElapsedTime (audium_element_start_time_millisecs)" /> <assign name="audium_vxmlLog" expr="audium_vxmlLog + '|||audio_group$$$' + 'nomatch_audio_group' + '^^^' + application.getElapsedTime (audium_element_start_time_millisecs)" /> <var name="maxNoMatch" expr="'yes'" /> <submit next="/CVP/Server" method="post" namelist=" audium_vxmlLog maxNoMatch" /> </catch> <catch event="noinput"> <prompt bargein="true">Sorry. I did not hear that. Say refills or press 1. Say pharmacist or press 2.</prompt> <assign name="audium_vxmlLog" expr="audium_vxmlLog + '|||noinput$$$' + '1' + '^^^' + application.getElapsedTime (audium_element_start_time_millisecs)" /> <assign name="audium_vxmlLog" expr="audium_vxmlLog + '|||audio_group$$$' + 'noinput_audio_group' + '^^^' + application.getElapsedTime (audium_element_start_time_millisecs)" /> </catch> <catch event="noinput" count="2"> <prompt bargein="true">I am sorry. I still did not hear that. If you are using a speaker phone. Please use the phone keypad to make your selection. Press 1 for refills. Press 2 to speak to a pharmacist.</prompt> <assign name="audium_vxmlLog" expr="audium_vxmlLog + '|||noinput$$$' + '2' + '^^^' + application.getElapsedTime (audium_element_start_time_millisecs)" /> <assign name="audium_vxmlLog" expr="audium_vxmlLog + '|||audio_group$$$' + 'noinput_ audio_group' + '^^^' + application.getElapsedTime (audium_element_start_time_millisecs)" /> </catch> <catch event="noinput" count="3"> <prompt bargein="true">Gee. Looks like we are having some trouble.</prompt> <assign name="audium_vxmlLog" expr="audium_vxmlLog + '|||noinput$$$' + '3' + '^^^' + application.getElapsedTime (audium_element_start_time_millisecs)" /> <assign name="audium_vxmlLog" expr="audium_vxmlLog + '|||audio_group$$$' + 'noinput_ audio_group' + '^^^' + application.getElapsedTime (audium_element_start_time_millisecs)" /> <var name="maxNoInput" expr="'yes'" /> <submit next="/CVP/Server" method="post" namelist=" audium_vxmlLog maxNoInput" /> </catch> <option value="refills" dtmf="1"> prescription</option> <option value="refills">refills</option> <option value="refills"> prescription refills</option> <option value="refills"> refill my prescription</option> <option value="refills"> I want to refill my prescription</option> <option value="refills"> refills please</option> <option value="Pharmacist" dtmf="2">Pharmacist</option> <option value="Pharmacist"> I want to speak to a pharmacist</option> <option value="Pharmacist"> pharmacist please</option> <filled> <assign name="audium_vxmlLog" expr="audium_vxmlLog + '|||utterance$$$' + choice_fld$. utterance + '^^^' + application.getElapsedTime (audium_element_start_time_millisecs)" /> <assign name="audium_vxmlLog" expr="audium_vxmlLog + '|||inputmode$$$' + choice_fld$. inputmode + '^^^' + application.getElapsedTime (audium_element_start_time_millisecs)" /> <assign name="audium_vxmlLog" expr="audium_vxmlLog + '|||interpretation$$$' + choice_fld + '^^^' + application.getElapsedTim (audium_element_start_time_millisecs)" /> <assign name="audium_vxmlLog" expr="audium_vxmlLog + '|||confidence$$$' + choice_fld$. confidence + '^^^' + application.getElapsedTime (audium_element_start_time_millisecs)" /> <var name="confidence" expr="choice_fld$.confidence" /> <submit next="/CVP/Server" method="post" namelist=" audium_vxmlLog confidence choice_fld" /> </filled> </field> </form> </vxml>
Estos gramáticas se envían luego al servidor ASR una vez que la puerta de enlace establece una sesión con el servidor ASR.
*Jan 18 03:34:57.523: //127//AFW_:/vapp_asr_change_server: asr_server=sip:asr@172.18.110.76 *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: grammar_id=session:option485@field.grammar *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: xml_lang=en-us *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: encoding_name=UTF-8 *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: remoteupdate=0 *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: grammar=<?xml version="1.0" encoding="UTF-8"?> <grammar version="1.0" xm lns="http://www.w3.org/2001/06/grammar" xml:lang="en-us" root="root"><rule id="root" scope="public"> prescription</rule></grammar> *Jan 18 03:34:57.523: //-1//MRCP:/mrcp_get_ev: ****>Caller PC=0x61BE1F94, Count=339, Event=0x63ACCCF0 *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: grammar_id=session:option486@field.grammar *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: encoding_name=UTF-8 *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: remoteupdate=0 *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: grammar=<?xml version="1.0" encoding="UTF-8"?> <grammar version="1.0" xm lns="http://www.w3.org/2001/06/grammar" mode="dtmf" root= "root"><rule id="root" scope= "public">1</rule></grammar> *Jan 18 03:34:57.523: //-1//MRCP: /mrcp_get_ev: ****>Caller PC=0x61BE1F94, Count=340, Event=0x63ACCAE8 *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: grammar_id=session:option487@field.grammar *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: xml_lang=en-us *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: encoding_name=UTF-8 *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: remoteupdate=0 *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: grammar=<?xml version="1.0" encoding="UTF-8"?> <grammar version="1.0" xm lns="http://www.w3.org/2001/06/grammar" xml:lang="en-us" root="root"><rule id="root" scope="public"> refills</rule></grammar> *Jan 18 03:34:57.523: //-1//MRCP :/mrcp_get_ev: ****>Caller PC=0x61BE1F94, Count=341, Event=0x63ACBC88 *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: grammar_id=session:option488@field.grammar *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: xml_lang=en-us *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: encoding_name=UTF-8 *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: remoteupdate=0 *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: grammar=<?xml version="1.0" encoding="UTF-8"?> <grammar version="1.0" xm lns="http://www.w3.org/2001/06/grammar" xml:lang="en-us" root="root"><rule id="root" scope="public"> prescription refills</rule></grammar> *Jan 18 03:34:57.523: //-1//MRCP:/mrcp_get_ev: ****>Caller PC=0x61BE1F94, Count=342, Event=0x63ACBCB0 *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: grammar_id=session:option489@field.grammar *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: xml_lang=en-us *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: encoding_name=UTF-8 *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: remoteupdate=0 *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: grammar=<?xml version="1.0" encoding="UTF-8"?> <grammar version="1.0" xm lns="http://www.w3.org/2001/06/grammar" xml: lang="en-us" root="root"> <rule id="root" scope="public"> refill my prescription</rule>< /grammar> *Jan 18 03:34:57.523: //-1//MRCP:/mrcp_get_ev: ****>Caller PC=0x61BE1F94, Count=343, Event=0x63ACBCD8 *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: grammar_id=session:option490@field.grammar *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: xml_lang=en-us *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: encoding_name=UTF-8 *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: remoteupdate=0 *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: grammar=<?xml version="1.0" encoding="UTF-8"?> <grammar version="1.0" xm lns="http://www.w3.org/2001/06/grammar" xml:lang="en-us" root="root"> <rule id="root" scope="public"> I want to refill my prescription </rule></grammar> *Jan 18 03:34:57.523: //-1//MRCP:/mrcp_get_ev: ****>Caller PC=0x61BE1F94, Count=344, Event=0x63ACBD00 *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: grammar_id=session:option491@field.grammar *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: xml_lang=en-us *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: encoding_name=UTF-8 *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: remoteupdate=0 *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: grammar=<?xml version="1.0" encoding="UTF-8"?> <grammar version="1.0" xm lns="http://www.w3.org/2001/06/grammar" xml:lang="en-us" root="root"><rule id="root" scope="public"> refills please</rule></grammar > *Jan 18 03:34:57.523: //-1//MRCP:/mrcp_get_ev: ****>Caller PC=0x61BE1F94, Count=345, Event=0x63ACBD28 *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: grammar_id=session:option492@field.grammar *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: xml_lang=en-us *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: encoding_name=UTF-8 *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: remoteupdate=0 *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: grammar=<?xml version="1.0" encoding="UTF-8"?> <grammar version="1.0" xm lns="http://www.w3.org/2001/06/grammar" xml:lang="en-us" root="root"><rule id="root" scope="public"> Pharmacist </rule></grammar> *Jan 18 03:34:57.523: //-1//MRCP:/mrcp_get_ev: ****>Caller PC=0x61BE1F94, Count=346, Event=0x63ACBB20 *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: grammar_id=session:option493@field.grammar *Jan 18 03:34:57.523: //127//AFW_:/vapp_asr_define_grammar: encoding_name=UTF-8 *Jan 18 03:34:57.523: //127//AFW_:/vapp_asr_define_grammar: remoteupdate=0 *Jan 18 03:34:57.523: //127//AFW_:/vapp_asr_define_grammar: grammar=<?xml version="1.0" encoding="UTF-8"?> <grammar version="1.0" xm lns="http://www.w3.org/2001/06/grammar" mode="dtmf" root="root"> <rule id="root" scope= "public">2</rule></grammar> *Jan 18 03:34:57.523: //-1//MRCP:/mrcp_get_ev: ****>Caller PC=0x61BE1F94, Count=347, Event=0x63ACBD50 *Jan 18 03:34:57.523: //127//AFW_:/vapp_asr_define_grammar: *Jan 18 03:34:57.523: //127//AFW_:/vapp_asr_define_grammar: grammar_id=session: option494@field.grammar *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: xml_lang=en-us *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: encoding_name=UTF-8 *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: remoteupdate=0 *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: grammar=<?xml version="1.0" encoding="UTF-8"?> <grammar version="1.0" xm lns="http://www.w3.org/2001/06/grammar" xml:lang="en-us" root="root"><rule id="root" scope="public"> I want to speak to a pharmacist </rule></grammar> *Jan 18 03:34:57.523: //-1//MRCP:/mrcp_get_ev: ****>Caller PC=0x61BE1F94, Count=348, Event=0x63ACBFF8 *Jan 18 03:34:57.523: //127//AFW_ :/vapp_asr_define_grammar: *Jan 18 03:34:57.527: //127//AFW_ :/vapp_asr_define_grammar: grammar_id=session:option495@field.grammar *Jan 18 03:34:57.527: //127//AFW_ :/vapp_asr_define_grammar: xml_lang=en-us *Jan 18 03:34:57.527: //127//AFW_ :/vapp_asr_define_grammar: encoding_name=UTF-8 *Jan 18 03:34:57.527: //127//AFW_ :/vapp_asr_define_grammar: remoteupdate=0 *Jan 18 03:34:57.527: //127//AFW_ :/vapp_asr_define_grammar: grammar=<?xml version="1.0" encoding="UTF-8"?> <grammar version="1.0" xm lns="http://www.w3.org/2001/06/grammar" xml:lang="en-us" root="root"><rule id="root" scope="public"> pharmacist please </rule></grammar> *Jan 18 03:34:57.527: //-1//MRCP:/mrcp_get_ev: ****>Caller PC=0x61BE1F94, Count=349, Event=0x63ACC048 *Jan 18 03:34:57.527: //127//AFW_ :/vapp_asr_define_grammar: *Jan 18 03:34:57.527: //127//AFW_:/vapp_asr_define_grammar: grammar_id=session:link496@document.grammar *Jan 18 03:34:57.527: //127//AFW_:/vapp_asr_define_grammar: xml_lang=en-us *Jan 18 03:34:57.527: //127//AFW_:/vapp_asr_define_grammar: encoding_name=UTF-8 *Jan 18 03:34:57.527: //127//AFW_:/vapp_asr_define_grammar: remoteupdate=0 *Jan 18 03:34:57.527: //127//AFW_:/vapp_asr_define_grammar: grammar=<?xml version="1.0" encoding="UTF-8"?> <grammar xmlns="http://ww w.w3.org/2001/06/grammar" mode="voice" version="1.0" root="Hotlink_02_VOICE" xml:lang="en-us"> <rule id="Hotlink_02_VOICE" scope="public"> <one-of> <item>operator</item> <item>agent</item> <item>pharmacist</item> </one-of> </rule> </grammar> *Jan 18 03:34:57.527: //-1//MRCP:/mrcp_get_ev: ****>Caller PC=0x61BE1F94, Count=350, Event=0x63ACC098 *Jan 18 03:34:57.527: //127//AFW_:/vapp_asr_define_grammar: *Jan 18 03:34:57.527: //127//AFW_:/vapp_asr_define_grammar: grammar_id=session:link497@document.grammar *Jan 18 03:34:57.527: //127//AFW_:/vapp_asr_define_grammar: xml_lang=en-us *Jan 18 03:34:57.527: //127//AFW_:/vapp_asr_define_grammar: encoding_name=UTF-8 *Jan 18 03:34:57.527: //127//AFW_:/vapp_asr_define_grammar: remoteupdate=0 *Jan 18 03:34:57.527: //127//AFW_:/vapp_asr_define_grammar: grammar=<?xml version="1.0" encoding="UTF-8"?> <grammar xmlns="http://ww w.w3.org/2001/06/grammar" mode="voice" version="1.0" root="Hotlink_01_VOICE" xml:lang="en-us"> <rule id="Hotlink_01_VOICE" scope="public"> <one-of> <item>operator</item> <item>agent</item> <item>pharmacist</item> </one-of> </rule> </grammar> *Jan 18 03:34:57.527: //-1//MRCP:/mrcp_get_ev: ****>Caller PC=0x61BE1F94, Count=351, Event=0x63ACC0C0 *Jan 18 03:34:57.527: //127//AFW_:/vapp_asr_define_grammar: *Jan 18 03:34:57.527: //127//AFW_:/vapp_asr_define_grammar: grammar_id=session:help@grammar *Jan 18 03:34:57.527: //127//AFW_:/vapp_asr_define_grammar: xml_lang=en-us *Jan 18 03:34:57.527: //127//AFW_:/vapp_asr_define_grammar: encoding_name=UTF-8 *Jan 18 03:34:57.527: //127//AFW_:/vapp_asr_define_grammar: remoteupdate=1 *Jan 18 03:34:57.527: //127//AFW_:/vapp_asr_define_grammar: grammar=<?xml version="1.0" encoding="UTF-8"?> <grammar version="1.0" xm lns="http://www.w3.org/2001/06/grammar" xml:lang="en-us" root="root"><rule id="root" scope="public"> help</rule></grammar> *Jan 18 03:34:57.527: //-1//MRCP:/mrcp_get_ev: ****>Caller PC=0x61BE1F94, Count=352, Event=0x63ACBEE0 *Jan 18 03:34:57.527: //127//AFW_:/vapp_asr: grammar_id=session:option485@field.grammar grammar_id=session:option486@field.grammar grammar_id=session:option487@field.grammar grammar_id=session:option488@field.grammar grammar_id=session:option489@field.grammar grammar_id=session:option490@field.grammar grammar_id=session:option491@field.grammar grammar_id=session:option492@field.grammar grammar_id=session:option493@field.grammar grammar_id=session:option494@field.grammar grammar_id=session:option495@field.grammar grammar_id=session:link496@document.grammar grammar_id=session:link497@document.grammar grammar_id=session:help@grammar
El dial-peer 6 saliente coincide.
*Jan 18 03:34:57.527: //-1/xxxxxxxxxxxx/CCAPI/ccCallSetupRequest: Destination Pattern=, Called Number=sip:tts@172.18.110.76, Digit Strip=FALSE *Jan 18 03:34:57.527: //-1/xxxxxxxxxxxx/CCAPI/ccCallSetupRequest: Calling Number=5555(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed), Called Number=sip:tts@172.18.110.76(TON=Unknown, NPI=ISDN), Redirect Number=, Display Info= Account Number=, Final Destination Flag=TRUE, Guid=2AEE8C2A-0AFB-11D6-801C-0013803E8C8E, Outgoing Dial-peer=6 *Jan 18 03:34:57.531: //-1/xxxxxxxxxxxx/CCAPI/cc _api_display_ie_subfields: ccCallSetupRequest: cisco-username= ----- ccCallInfo IE subfields ----- cisco-ani=5555 cisco-anitype=0 cisco-aniplan=0 cisco-anipi=0 cisco-anisi=0 dest=sip:tts@172.18.110.76 cisco-desttype=0 cisco-destplan=1 cisco-rdie=FFFFFFFF cisco-rdn= cisco-rdntype=-1 cisco-rdnplan=-1 cisco-rdnpi=-1 cisco-rdnsi=-1 cisco-redirectreason=-1 fwd_final_type =0 final_redirectNumber = hunt_group_timeout =0 *Jan 18 03:34:57.531: //-1/xxxxxxxxxxxx/CCAPI/ ccIFCallSetupRequestPrivate: Interface=0x662CE538, Interface Type=3, Destination=, Mode=0x0, Call Params(Calling Number=5555, (Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed), Called Number=sip:tts@172.18.110.76 (TON=Unknown, NPI=ISDN), Calling Translated=FALSE, Subscriber Type Str=RegularLine, FinalDestinationFlag=TRUE, Outgoing Dial-peer=6, Call Count On=FALSE, Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)
El SDP del mensaje INVITE contiene información de medios para la secuencia Audio y la aplicación MRCPv2 (canal de sintetización de discursos).
*Jan 18 03:34:57.531: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: INVITE sip:tts@172.18.110.76:5060 SIP/2.0 Via: SIP/2.0/UDP 14.1.16.25: 5060;branch=z9hG4bK931F1D Remote-Party-ID: <sip:5555@14.1.16.25>; party=calling;screen=no;privacy=off From: <sip:5555@14.1.16.25> ;tag=E54D43C-1EC4 To: sip:tts@172.18.110.76 Date: Fri, 18 Jan 2002 03:34:57 GMT Call-ID: 2DCA5BEF-AFB11D6-80D3DC30 -3585E95A@14.1.16.25 Supported: 100rel,timer, resource-priority,replaces Min-SE: 1800 Cisco-Guid: 720276522-184226262 -2149318675-2151582862 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Max-Forwards: 70 Timestamp: 1011324897 Contact: <sip:5555@14.1.16.25:5060> Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 358 v=0 o=CiscoSystemsSIP-GW-UserAgent 6021 4611 IN IP4 14.1.16.25 s=SIP Call c=IN IP4 14.1.16.25 t=0 0 m=audio 16984 RTP/AVP 0 101 c=IN IP4 14.1.16.25 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=recvonly a=mid:1 m=application 9 TCP/MRCPv2 a=setup:active a=connection:new a=resource:speechsynth a=cmid:1
El dial-peer 5 saliente coincide.
*Jan 18 03:34:57.531: //-1/xxxxxxxxxxxx/CCAPI/ccCallSetupRequest: Destination Pattern=, Called Number=sip:asr@172.18.110.76, Digit Strip=FALSE *Jan 18 03:34:57.531: //-1/xxxxxxxxxxxx/CCAPI/ccCallSetupRequest: Calling Number=5555(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed), Called Number=sip:asr@172.18.110.76 (TON=Unknown, NPI=ISDN), Redirect Number=, Display Info= Account Number=, Final Destination Flag=TRUE, Guid=2AEE8C2A-0AFB-11D6-801C-0013803E8C8E, Outgoing Dial-peer=5 *Jan 18 03:34:57.531: //-1/xxxxxxxxxxxx/CCAPI/cc_api _display_ie_subfields: ccCallSetupRequest: cisco-username= ----- ccCallInfo IE subfields ----- cisco-ani=5555 cisco-anitype=0 cisco-aniplan=0 cisco-anipi=0 cisco-anisi=0 dest=sip:asr@172.18.110.76 cisco-desttype=0 cisco-destplan=1 cisco-rdie=FFFFFFFF cisco-rdn= cisco-rdntype=-1 cisco-rdnplan=-1 cisco-rdnpi=-1 cisco-rdnsi=-1 cisco-redirectreason=-1 fwd_final_type =0 final_redirectNumber = hunt_group_timeout =0 *Jan 18 03:34:57.535: //-1/xxxxxxxxxxxx/CCAPI /ccIFCallSetupRequestPrivate: Interface=0x662CE538, Interface Type=3, Destination=, Mode=0x0, Call Params(Calling Number=5555, (Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed), Called Number=sip:asr@172.18.110.76 (TON=Unknown, NPI=ISDN), Calling Translated=FALSE, Subscriber Type Str=RegularLine, FinalDestinationFlag=TRUE, Outgoing Dial-peer=5, Call Count On=FALSE, Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)
El SDP contiene la información de medios para la secuencia de audio, relé DTMF. y la aplicación MRCPv2 (canal de Speechrecg).
*Jan 18 03:34:57.535: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: INVITE sip:asr@172.18.110.76:5060 SIP/2.0 Via: SIP/2.0/UDP 14.1.16.25:5060;branch=z9hG4bK94C0B Remote-Party-ID: <sip:5555@14.1.16.25>; party=calling;screen=no;privacy=off From: <sip:5555@14.1.16.25>;tag=E54D440-1CDB To: sip:asr@172.18.110.76 Date: Fri, 18 Jan 2002 03:34:57 GMT Call-ID: 2DCAF817-AFB11D6 -80D5DC30-3585E95A@14.1.16.25 Supported: 100rel,timer, resource-priority,replaces Min-SE: 1800 Cisco-Guid: 720276522-184226262- 2149318675-2151582862 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Max-Forwards: 70 Timestamp: 1011324897 Contact: <sip:5555@14.1.16.25:5060> Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 358 v=0 o=CiscoSystemsSIP-GW-UserAgent 6805 2057 IN IP4 14.1.16.25 s=SIP Call c=IN IP4 14.1.16.25 t=0 0 m=audio 19994 RTP/AVP 0 101 c=IN IP4 14.1.16.25 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendonly a=mid:1 m=application 9 TCP/MRCPv2 a=setup:active a=connection:new a=resource:speechrecog a=cmid:1
Códec G711ulaw, dirección IP y números de puerto RTP para la secuencia de audio.
El atributo de dirección de esta secuencia RTP es "recvonly".
Retransmisión DTMF basada en RTP-NTE.
Número de puerto TCP (51001) que utilizará la puerta de enlace para establecer una sesión MRCPv2 con el servidor ASR.
*Jan 18 03:34:57.559: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 200 OK Via: SIP/2.0/UDP 14.1.16.25:5060; branch=z9hG4bK94C0B To: <sip:asr@172.18.110.76>;tag=a99d0500 From: <sip:5555@14.1.16.25>;tag=E54D440-1CDB Call-ID: 2DCAF817-AFB11D6-80D5DC30- 3585E95A@14.1.16.25 CSeq: 101 INVITE Contact: <sip:172.18.110.76:5060> Content-Type: application/sdp Content-Length: 342 v=0 o=MRCPv2Server 3386937590 3386937590 IN IP4 172.18.110.76 s=SIP Call c=IN IP4 172.18.110.76 t=3386937590 0 m=audio 10002 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=recvonly m=application 51001 TCP/MRCPv2 a=connection:new a=setup:passive a=model:besteffort a=channel:000023B846361276@speechrecog
La sesión SIP para el ASR se establece entre la puerta de enlace y el servidor ASR.
*Jan 18 03:34:57.563: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: ACK sip:172.18.110.76:5060 SIP/2.0 Via: SIP/2.0/UDP 14.1.16.25:5060;branch=z9hG4bK9520FA From: <sip:5555@14.1.16.25>;tag=E54D440-1CDB To: <sip:asr@172.18.110.76>;tag=a99d0500 Date: Fri, 18 Jan 2002 03:34:57 GMT Call-ID: 2DCAF817-AFB11D6-80D5DC30-3585E95A@14.1.16.25 Max-Forwards: 70 CSeq: 101 ACK Allow-Events: telephone-event Content-Length: 0
Aquí sólo se muestra una solicitud.
MRCP/2.0 446 DEFINE-GRAMMAR 1 Channel-Identifier: 000023B846361276@speechrecog : Speech-Language: en-us Content-Base: http://172.18.110.75:7000/CVP/ : Content-Type: application/srgs+xml Content-Id: option485@field.grammar Content-Length: 193 : <?xml version="1.0" encoding="UTF-8"?> <grammar version="1.0" mlns="http://www.w3.org/2001/06/grammar" xml:lang="en-us" root="root" ><rule id="root" scope="public"> prescription</rule></grammar>
*Jan 18 03:34:57.587: //-1//MRCP:/hash_get: Table=mrcpv2_socket_connect_table, Key=0: MRCP/2.0 80 1 200 COMPLETE Channel-Identifier: 000023B846361276@speechrecog
El SDP del mensaje SIP INVITE especifica lo siguiente:
Códec G711ulaw, dirección IP y números de puerto RTP para la secuencia de audio.
El atributo de dirección de esta secuencia RTP es "sólo envío".
Retransmisión DTMF basada en RTP-NTE
Número de puerto TCP (51000) que utilizará la puerta de enlace para establecer una sesión MRCPv2 con el servidor TTS.
*Jan 18 03:34:57.591: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 200 OK Via: SIP/2.0/UDP 14.1.16.25:5060; branch=z9hG4bK931F1D To: <sip:tts@172.18.110.76>;tag=c1160600 From: <sip:5555@14.1.16.25>;tag=E54D43C-1EC4 Call-ID: 2DCA5BEF-AFB11D6-80D3DC30- 3585E95A@14.1.16.25 CSeq: 101 INVITE Contact: <sip:172.18.110.76:5060> Content-Type: application/sdp Content-Length: 342 v=0 o=MRCPv2Server 3386937590 3386937590 IN IP4 172.18.110.76 s=SIP Call c=IN IP4 172.18.110.76 t=3386937590 0 m=audio 10000 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=sendonly m=application 51000 TCP/MRCPv2 a=connection:new a=setup:passive a=model:besteffort a=channel:000023EC46361276@speechsynth
La sesión SIP para el texto a voz se establece entre la puerta de enlace y el servidor TTS.
*Jan 18 03:34:57.595: //-1/xxxxxxxxxxxx/SIP/ Msg/ccsipDisplayMsg: Sent: ACK sip:172.18.110.76:5060 SIP/2.0 Via: SIP/2.0/UDP 14.1.16.25:5060; branch=z9hG4bK9626BC From: <sip:5555@14.1.16.25>;tag=E54D43C-1EC4 To: <sip:tts@172.18.110.76>;tag=c1160600 Date: Fri, 18 Jan 2002 03:34:57 GMT Call-ID: 2DCA5BEF-AFB11D6-80D3DC30 -3585E95A@14.1.16.25 Max-Forwards: 70 CSeq: 101 ACK Allow-Events: telephone-event Content-Length: 0
MRCP/2.0 987 RECOGNIZE 15 Channel-Identifier: 000023B846361276@speechrecog : Speech-Language: en-us Confidence-Threshold: 0.40 Sensitivity-Level: 0.50 Speed-Vs-Accuracy: 0.50 Cancel-If-Queue: false Dtmf-Interdigit-Timeout: 10000 Dtmf-Term-Timeout: 0 Dtmf-Term-Char: # No-Input-Timeout: 60000 N-Best-List-Length: 1 Logging-Tag: 127:127 Accept-Charset: charset: utf-8 Content-Base: http://172.18.110.75:7000/CVP/ Media-Type: audio/basic Start-Input-Timers: false : Content-Type: text/uri-list Content-Length: 453 : session:option485@field.grammar session:option486@field.grammar session:option487@field.grammar session:option488@field.grammar session:option489@field.grammar session:option490@field.grammar session:option491@field.grammar session:option492@field.grammar session:option493@field.grammar session:option494@field.grammar session:option495@field.grammar session:link496@document.grammar session:link497@document.grammar session:help@grammar
MRCP/2.0 84 15 200 IN-PROGRESS Channel-Identifier: 000023B846361276@speechrecog
Lo almacena en la caché y reproduce el mensaje a la persona que llama.
*Jan 18 03:35:04.335: //127//HTTPC:/httpc_is_cached: HTTPC_FILE_IS_CACHED *Jan 18 03:35:04.335: //-1//HTTPC: /httpc_set_cache_revoke_cb: Registering revoke_callback(0x61CDD948) +pcontext(0x63A7AAA8) for cach ep(0x68734930) *Jan 18 03:35:04.335: //127//AFW_:/vapp_driver: evtID: 146 vapp record state: 0 *Jan 18 03:35:04.335: //127//AFW_:/vapp_play_done: evID=146 reason=17, protocol=5, status_code=0, dur=3291, rate=0 *Jan 18 03:35:04.335: //127/2AEE8C2A801C/VXML: /vxml_media_done:
MRCP/2.0 376 SPEAK 1 Channel-Identifier: 000023EC46361276@speechsynth : Kill-On-Barge-In: true Speech-Language: en-us Logging-Tag: 127:127 Content-Base: http://172.18.110.75:7000/CVP/ : Content-Type: application/ssml+xml Content-Length: 123 : <?xml version="1.0" encoding="UTF-8"?> <speak version="1.0" xml:lang="en-us"> Thank you for calling Audium pharmacy.</speak>
MRCP/2.0 83 1 200 IN-PROGRESS Channel-Identifier: 000023EC46361276@speechsynth
MRCP/2.0 141 SPEAK-COMPLETE 1 COMPLETE Channel-Identifier: 000023EC46361276@speechsynth Completion-Cause: 000 normal Speech-Marker: ""
El gateway envía este dígito como un evento RTP-NTE al servidor ASR.
*Jan 18 03:35:12.583: s=DSP d=VoIP payload 0x65 ssrc 0x15 sequence 0x1E9B timestamp 0x2FADCC60 *Jan 18 03:35:12.583: Pt:101 Evt:1 Pkt:03 00 00 <Snd>>> *Jan 18 03:35:12.587: s=DSP d=VoIP payload 0x65 ssrc 0x15 sequence 0x1E9C timestamp 0x2FADCC60 *Jan 18 03:35:12.587: Pt:101 Evt:1 Pkt:03 00 00 <Snd>>> *Jan 18 03:35:12.631: s=DSP d=VoIP payload 0x65 ssrc 0x15 sequence 0x1E9E timestamp 0x2FADCC60 *Jan 18 03:35:12.631: Pt:101 Evt:1 Pkt:03 01 90 <Snd>>> *Jan 18 03:35:12.683: s=DSP d=VoIP payload 0x65 ssrc 0x15 sequence 0x1E9F timestamp 0x2FADCC60 *Jan 18 03:35:12.683: Pt:101 Evt:1 Pkt:03 03 20 <Snd>>> *Jan 18 03:35:12.703: s=DSP d=VoIP payload 0x65 ssrc 0x15 sequence 0x1EA0 timestamp 0x2FADCC60 *Jan 18 03:35:12.703: Pt:101 Evt:1 Pkt:83 03 38 <Snd>>> *Jan 18 03:35:12.707: s=DSP d=VoIP payload 0x65 ssrc 0x15 sequence 0x1EA1 timestamp 0x2FADCC60 *Jan 18 03:35:12.707: Pt:101 Evt:1 Pkt:83 03 38 <Snd>>> *Jan 18 03:35:12.711: s=DSP d=VoIP payload 0x65 ssrc 0x15 sequence 0x1EA2 timestamp 0x2FADCC60 *Jan 18 03:35:12.711: Pt:101 Evt:1 Pkt:83 03 38 <Snd>>>
Esto notifica al gateway que ha reconocido uno de los eventos solicitados (en este caso dígito 1).
MRCP/2.0 513 RECOGNITION-COMPLETE 15 COMPLETE Channel-Identifier: 000023B846361276@speechrecog Proxy-Sync-Id: 0B82553000000027 Completion-Cause: 000 success Content-Type: application/nlsml+xml Content-Length: 292 <?xml version="1.0" encoding="UTF-8"?> <result grammar="session:option486@field.grammar"> <interpretation grammar= "session:option486@field.grammar" confidence="0.000000"> <instance> 1 </instance> <input mode="dtmf" confidence="1.000000"> 1 </input> </interpretation> </result>
Después de recibir esta notificación, la puerta de enlace VXML envía una solicitud HTTP POST como se especifica en la etiqueta ENVIAR del documento VXML (3). Esta solicitud POST informa al servidor VXML que el autor de la llamada PSTN ingresó el dígito 1.
*Jan 18 03:35:12.863: //127/2AEE8C2A801C/VXML:/vxml_vapp_bgpost: url http://172.18.110.75:7000/CVP/Server cachable 1 timeout 0 body audium_vxmlLog=%7C%7C%7Caudio _group$$$initial_audio_group%5E% 5E%5E4%7C%7C%7Cutterance$$$1%5E%5E%5E153 40%7C%7C%7Cinputmode $$$dtmf%5E%5E%5E15344%7C%7C%7C interpretation$$$refills%5E%5E%5E15344%7C %7C%7Cconfidence$$$0%5E%5E%5E15344&confidence= 0&choice_fld=refills len 258maxage -1 maxstale -1 *Jan 18 03:35:12.863: //127//AFW_:/vapp_bgpost: url=http://172.18.110.75:7000/CVP/Server; mime_type=application/x-www-form-urlencod ed; len=258; iov_base=audium_vxmlLog=%7C%7C%7Caudio_ group$$$initial_audio_group %5E%5E%5E4%7C%7C%7Cutterance $$$1%5E%5E%5E15340%7C%7C %7Cinputmode$$$dtmf%5E%5E%5E15344% 7C%7C%7Cinterpretation$$$refills %5E%5E%5E15344%7C%7C%7Cconfidence$$$0 %5E%5E%5E15344&confidence=0& choice_fld=refills *Jan 18 03:35:12.931: about to send data to the socket 3 : first 400 bytes of data: POST /CVP/Server HTTP/1.1 Host: 172.18.110.75:7000 Content-Length: 258 Content-Type: application/x-www-form-urlencoded Cookie: $Version=0; JSESSIONID= BBCE0F948ADFDB720497F587A7997538; $Path=/CVP Connection: close Accept: text/vxml, text/x-vxml, application/vxml, application/x-vxml, application/voicexml, application/x-voicexml, text/plain, tex t/html, audio/basic, audio/wav, multipart/form-dat
El ASR envía un mensaje MRCP COMPLETO DE RECONOCIMIENTO a la gateway VXML del IOS.
MRCP/2.0 533 RECOGNITION-COMPLETE 21 COMPLETE Channel-Identifier: 000023B846361276@speechrecog Proxy-Sync-Id: 0B82553000000028 Completion-Cause: 000 success Content-Type: application/nlsml+xml Content-Length: 312 <?xml version="1.0" encoding="UTF-8"?> <result grammar= "session:field498@field.grammar"> <interpretation grammar= "session:field498@field.grammar" confidence="0.738968"> <instance> 1234 </instance> <input mode="speech" confidence="0.752155"> one two three four </input> </interpretation> </result> The final VXML document sent by the VXML server contains just the <exit\> tag in the <form> This tells the Gateway to terminate the VXML session
Esto indica a la puerta de enlace que finalice la sesión VXML
*Jan 18 03:36:07.159: processing server rsp msg: msg(67CA85F8)URL: http://172.18.110.75:7000/CVP/Server, fd(3): *Jan 18 03:36:07.159: Request msg: POST /CVP/Server HTTP/1.1 *Jan 18 03:36:07.159: Message Response Code: 200 *Jan 18 03:36:07.159: Message Rsp Decoded Headers: *Jan 18 03:36:07.159: D ate:Mon, 30 Apr 2007 16:59:53 GMT *Jan 18 03:36:07.159: Content-Type:text/xml;charset=ISO-8859-1 *Jan 18 03:36:07.159: Connection:close *Jan 18 03:36:07.159: Set-Cookie: JSESSIONID=NULL; Expires=Thu, 01-Jan-1970 00:00:10 GMT; Path=/CVP *Jan 18 03:36:07.159: headers: *Jan 18 03:36:07.159: HTTP/1.1 200 OK Server: Apache-Coyote/1.1 Set-Cookie: JSESSIONID=NULL; Expires=Thu, 01-Jan-1970 00:00:10 GMT; Path=/CVP Content-Type: text/xml;charset=ISO-8859-1 Date: Mon, 30 Apr 2007 16:59:53 GMT Connection: close *Jan 18 03:36:07.159: body: *Jan 18 03:36:07.159: <?xml version="1.0" encoding="UTF-8"?> <vxml version="2.0" xml:lang="en-us"> <catch event="vxml.session.error"> <exit /> </catch> <catch event="telephone.disconnect.hangup"> <exit /> </catch> <catch event="telephone.disconnect"> <exit /> </catch> <catch event="error.unsupported.object"> <exit /> </catch> <catch event="error.unsupported.language"> <exit /> </catch> <catch event="error.unsupported.format"> <exit /> </catch> <catch event="error.unsupported.element"> <exit /> </catch> <catch event="error.unsupported.builtin"> <exit /> </catch> <catch event="error.unsupported"> <exit /> </catch> <catch event="error.semantic"> <exit /> </catch> <catch event="error.noresource"> <exit /> </catch> <catch event="error.noauthorization"> <exit /> </catch> <catch event="error.eventhandler.notfound"> <exit /> </catch> <catch event="error.connection.noroute"> <exit /> </catch> <catch event="error.connection.noresource"> <exit /> </catch> <catch event="error.connection.nolicense"> <exit /> </catch> <catch event="error.connection.noauthorization"> <exit /> </catch> <catch event="error.connection.baddestination"> <exit /> </catch> <catch event="error.condition.baddestination"> <exit /> </catch> <catch event="error.com.cisco. media.resource.unavailable"> <exit /> </catch> <catch event= "error.com.cisco.handoff.failure"> <exit /> </catch> <catch event= "error.com.cisco.callhandoff.failure"> <exit /> </catch> <catch event= "error.com.cisco.aaa.authorize.failure"> <exit /> </catch> <catch event= "error.com.cisco.aaa.authenticate.failure"> <exit /> </catch> <catch event="error.badfetch.https"> <exit /> </catch> <catch event="error.badfetch.http"> <exit /> </catch> <catch event="error.badfetch"> <exit /> </catch> <catch event="error"> <exit /> </catch> <catch event="disconnect.com.cisco.handoff"> <exit /> </catch> <catch event="connection.disconnect.hangup"> <exit /> </catch> <catch event="connection.disconnect"> <exit /> </catch> <form> <block> <exit /> </block> </form> </vxml>
*Jan 18 03:36:14.155: //127/2AEE8C2A801C/VXML:/vxml_vapp_terminate: vapp_status=0 ref_count 0 *Jan 18 03:36:14.155: //127//AFW_:/vapp_terminate: *Jan 18 03:36:14.155: //127//AFW_ :/vapp_session_exit_event_name: Exit Event vxml.session.complete *Jan 18 03:36:14.155: //127//AFW_:/AFW_M_VxmlModule_Terminate: *Jan 18 03:36:14.155: //131/2AEE8C2A801C/CCAPI/ccCallDisconnect: Cause Value=16, Tag=0x0, Call Entry (Previous Disconnect Cause=0, Disconnect Cause=0) *Jan 18 03:36:14.155: //131/2AEE8C2A801C/CCAPI/ccCallDisconnect: Cause Value=16, Call Entry(Responsed=TRUE, Cause Value=16)
*Jan 18 03:36:14.159: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: BYE sip:172.18.110.76:5060 SIP/2.0 Via: SIP/2.0/UDP 14.1.16.25: 5060;branch=z9hG4bK971131 From: <sip:5555@14.1.16.25>;tag=E54D440-1CDB To: <sip:asr@172.18.110.76>;tag=a99d0500 Date: Fri, 18 Jan 2002 03:34:57 GMT Call-ID: 2DCAF817-AFB11D6-80D5DC30- 3585E95A@14.1.16.25 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 70 Timestamp: 1011324974 CSeq: 102 BYE Reason: Q.850;cause=16 Content-Length: 0 *Jan 18 03:36:14.607: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 200 OK Via: SIP/2.0/UDP 14.1.16.25: 5060;branch=z9hG4bK971131 To: <sip:asr@172.18.110.76>;tag=a99d0500 From: <sip:5555@14.1.16.25>;tag=E54D440-1CDB Call-ID: 2DCAF817-AFB11D6-80D5DC30- 3585E95A@14.1.16.25 CSeq: 102 BYE Contact: <sip:172.18.110.76:5060> Content-Length: 0
*Jan 18 03:36:14.159: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: BYE sip:172.18.110.76:5060 SIP/2.0 Via: SIP/2.0/UDP 14.1.16.25:5060;branch=z9hG4bK981487 From: <sip:5555@14.1.16.25>;tag=E54D43C-1EC4 To: <sip:tts@172.18.110.76>;tag=c1160600 Date: Fri, 18 Jan 2002 03:34:57 GMT Call-ID: 2DCA5BEF-AFB11D6- 80D3DC30-3585E95A@14.1.16.25 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 70 Timestamp: 1011324974 CSeq: 102 BYE Reason: Q.850;cause=16 Content-Length: 0 *Jan 18 03:36:14.215: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 200 OK Via: SIP/2.0/UDP 14.1.16.25:5060;branch=z9hG4bK981487 To: <sip:tts@172.18.110.76>;tag=c1160600 From: <sip:5555@14.1.16.25>;tag=E54D43C-1EC4 Call-ID: 2DCA5BEF-AFB11D6-80D3DC30-3585E95A@14.1.16.25 CSeq: 102 BYE Contact: <sip:172.18.110.76:5060> Content-Length: 0
*Jan 18 03:36:14.611: ISDN Se3/0:23 Q931: TX -> DISCONNECT pd = 8 callref = 0x805A Cause i = 0x8090 - Normal call clearing *Jan 18 03:36:14.623: ISDN Se3/0:23 Q931: RX <- RELEASE pd = 8 callref = 0x005A *Jan 18 03:36:14.623: ISDN Se3/0:23 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x805A