Tecnicas monitoreo reportes con Asterisk

Técnicas para
monitoreo y reportes
con Asterisk
Nicolás Gudiño
asternic@gmail.com

4K Conference 2011
Buenos Aires

Nicolás Gudiño - http://www.asternic.biz
Tuesday, December 6, 2011

Técnicas para monitoreo y reportes con Asterisk 4K Conference 2011

Agenda
Archivos de registro / logs

Call Detail Records / CDR

Registro actvidad de colas / queue_log

Channel Event Logging / CEL

Asterisk Manager Interface / AMI



Archivos de registro - Logs
Ideales para depuración de problemas

/etc/asterisk/logger.conf

;console => notice,warning,error
;console => notice,warning,error,debug
;messages => notice,warning,error
full => notice,warning,error,debug,verbose



/var/log/asterisk/full
[2011-11-18 15:42:36] VERBOSE[13927] pbx.c: -- Executing [s@macro-dial-one:37] Dial
("Local/605@from-queue-8891;2", "SIP/605,25,trM(auto-blkvm)") in new
stack
[2011-11-18 15:42:36] VERBOSE[13927] netsock2.c: == Using SIP RTP TOS bits 184
[2011-11-18 15:42:36] VERBOSE[13927] netsock2.c: == Using SIP RTP CoS mark 5
[2011-11-18 15:42:36] VERBOSE[13927] app_dial.c: -- Called SIP/605
[2011-11-18 15:42:36] VERBOSE[13887] app_queue.c: -- Local/605@from-queue-8891;1 is
ringing
[2011-11-18 15:42:36] VERBOSE[13928] app_mixmonitor.c: == Begin MixMonitor Recording
Local/605@from-queue-8891;2
[2011-11-18 15:42:36] VERBOSE[13927] app_dial.c: -- SIP/605-00000094 is ringing
[2011-11-18 15:42:36] VERBOSE[13887] app_queue.c: -- Local/605@from-queue-8891;1 is
ringing
[2011-11-18 15:42:37] VERBOSE[13927] app_dial.c: -- SIP/605-00000094 is ringing

Fecha Nivel Thread Módulo Detalle


Revisando últimas líneas de log
#> tail -n 1000 /var/log/asterisk/full

Filtrando logs para depuración
#> grep 13928 /var/log/asterisk/full

[2011-11-18 15:42:36] VERBOSE[13928] app_mixmonitor.c: == Begin MixMonitor Recording ...
[2011-11-18 15:43:14] VERBOSE[13928] app_mixmonitor.c: == MixMonitor close filestream
[2011-11-18 15:43:14] VERBOSE[13928] app_mixmonitor.c: == Executing [/var/lib/asterisk/...
[2011-11-18 15:43:14] VERBOSE[13928] app_mixmonitor.c: == End MixMonitor Recording ...



Comandos de consola
Recargar configuración
#> asterisk -rx “logger reload”

Rotación de logs
#> asterisk -rx “logger rotate”

Ajuste de nivel de verbosidad
#> asterisk -rx “core set verbose 10”

Ajuste de nivel de depuración
#> asterisk -rx “core set debug 3”



Call Detail Records - CDR
Registro histórico de llamados
Tarifación / facturación
Análisis de volumen de llamados
Estadísticas de uso
Auditoría



Diferentes back-ends
csv
manager
radius
sqlite
odbc, etc
Back-end mysql disponible en addons



/etc/asterisk/cdr.conf
[general]
unanswered=no
endbeforehexten=no

[csv]
usegmtime = yes ; log date/time in GMT
loguniqueid = yes ; log uniqueid.
loguserfield = yes ; log user field.
accountlogs = yes ; create separate log file for
each account code. Default is "yes"



Campo de Usuario (userﬁeld)
Set(CDR(userfield)=Value)
Puede usarse para guardar información extra en un
registro CDR como ser:
Tecla pulsada en IVR, nombre de grabación
Cualquier variable de canal

/etc/asterisk/cdr_mysql.conf
[global]
userfield=1



/var/log/asterisk/cdr-csv/Master.csv
"","19524742","609","from-did-direct","""19524742""
<19524742>","SIP/zapuser-00000179","Local/FMPR-609@from-
internal-b2c7;1","Dial","Local/FMPR-609@from-internal&Local/
FMGL-606@from-internal,22,trM(auto-blkvm)","2011-11-24
09:18:10","2011-11-24 09:18:26","2011-11-24 09:18:44",
34,18,"ANSWERED","DOCUMENTATION","1322126290.637",""

"","19524742","605","from-internal-xfer","""19524742""
<19524742>","SIP/609-0000017a","SIP/605-0000017c","Dial","SIP/
605,25,trM(auto-blkvm)","2011-11-24 09:18:44",,"2011-11-24
09:18:59",15,0,"NO ANSWER","DOCUMENTATION","1322126301.642",""



Información util pero a veces incompleta

Hay inconsistencias cuando se
transfieren llamadas

No es util para determinar el flujo de un
llamado



Asternic CDR Reports



Asternic CDR Reports

Módulo de FreePBX - GPL
http://www.asternic.biz/downloads/asternic-cdr-reports



Actividad de colas - queue_log
Registro detallado de actividad de colas
Estadísticas de call center
Eﬁciencia de agentes
Nivel de servicio
Auditoría



/var/log/asterisk/queue_log
1321903401|1321903368.260|101|NONE|ENTERQUEUE||asterisk|1
1321903417|1321903368.260|101|Soporte Paula|RINGNOANSWER|15000
1321903449|1321903368.260|101|Soporte Paula|CONNECT|48|1321903443.267|5
1321904957|1321903368.260|101|Soporte Paula|COMPLETEAGENT|48|1508|1
1321945326|NONE|NONE|NONE|CONFIGRELOAD|
1321964669|1321964645.270|100|NONE|ENTERQUEUE||44438722|1
1321964738|1321964645.270|100|Soporte Paula|CONNECT|69|1321964727.277|10
1321964795|1321964645.270|100|Soporte Paula|COMPLETECALLER|69|57|1

timestamp cola info adicional
uniqueid agente evento


Asternic Call Center Stats Lite



Asternic Call Center Stats Lite - GPL3

http://www.asternic.biz/downloads/lite



Channel Event Logging - CEL
Registro detallado de
llamados

Híbrido entre queue_log
y CDR

CDR con esteroides


/etc/asterisk/cel.conf
[general]
enable=yes
apps=all
events=ALL
dateformat=%F %T

Lo mínimo para empezar a usar CEL es
habilitarlo con enable=yes y enumerar
eventos con events=xxx.



Eventos disponibles en CEL

CHAN_START , CHAN_END, ANSWER, HANGUP,
CONF_ENTER, CONF_EXIT, CONF_START, CONF_END,
APP_START, APP_END, PARK_START, PARK_END,
BRIDGE_START, BRIDGE_END, 3WAY_START ,
3WAY_END, BLINDTRANSFER, ATTENDEDTRANSFER,
TRANSFER, HOOKFLASH, USER_EVENT

¡UN MONTONAZO!



/etc/asterisk/cel_custom.conf
[mappings]
Master.csv => ${CSV_QUOTE(${eventtype})},${CSV_QUOTE($
{eventtime})},${CSV_QUOTE(${CALLERID(name)})},${CSV_QUOTE
(${CALLERID(num)})},${CSV_QUOTE(${CALLERID(ANI)})},$
{CSV_QUOTE(${CALLERID(RDNIS)})},${CSV_QUOTE(${CALLERID
(DNID)})},${CSV_QUOTE(${CHANNEL(exten)})},${CSV_QUOTE($
{CHANNEL(context)})},${CSV_QUOTE(${CHANNEL(channame)})},$
{CSV_QUOTE(${CHANNEL(appname)})},${CSV_QUOTE(${CHANNEL
(appdata)})},${CSV_QUOTE(${CHANNEL(amaflags)})},$
{CSV_QUOTE(${CHANNEL(accountcode)})},${CSV_QUOTE(${CHANNEL
(uniqueid)})},${CSV_QUOTE(${CHANNEL(linkedid)})},$
{CSV_QUOTE(${CHANNEL(peer)})},${CSV_QUOTE(${CHANNEL
(userfield)})},${CSV_QUOTE(${eventextra})}



/var/log/asterisk/cel_custom/Master.csv
"CHAN_START" "1322126290.417117" "" "from-trunk-sip-zapuser" "" "HANGUP" "1322126306.752231" "609" "macro-dial-one" "AppDial"
"ANSWER" "1322126290.433885" "49524742" "ivr-2" "Answer" "CHAN_END" "1322126306.752260" "609" "macro-dial-one" "AppDial"
"APP_START" "1322126301.270181" "49524742" "macro-dial" "Dial" "APP_END" "1322126306.752321" "49524742" "macro-dial-one" "Dial"
"CHAN_START" "1322126301.270458" "" "from-internal" "" "HANGUP" "1322126306.752498" "49524742" "from-internal" ""
"CHAN_START" "1322126301.270551" "" "from-internal" "" "CHAN_END" "1322126306.752534" "49524742" "from-internal" ""
"CHAN_START" "1322126301.271182" "" "from-internal" "" "BLINDTRANSFER" "1322126324.863833" "" "macro-dial" "AppDial"
"CHAN_START" "1322126301.271246" "" "from-internal" "" "BRIDGE_END" "1322126324.934059" "49524742" "from-internal-xfer" "Dial"
"APP_START" "1322126301.286019" "49524742" "macro-dial-one" "Dial" "HANGUP" "1322126324.934099" "" "from-internal-xfer" "AppDial"
"CHAN_START" "1322126301.307455" "" "from-internal" "" "CHAN_END" "1322126324.937976" "" "from-internal-xfer" "AppDial"
"APP_START" "1322126303.485284" "49524742" "macro-dial" "Dial" "APP_END" "1322126324.938097" "49524742" "from-internal-xfer" "Dial"
"CHAN_START" "1322126303.485900" "" "from-internal" "" "APP_START" "1322126324.943018" "49524742" "macro-dial-one" "Dial"
"ANSWER" "1322126306.698194" "609" "from-internal" "AppDial" "CHAN_START" "1322126324.943507" "" "from-internal" ""
"ANSWER" "1322126306.713192" "49524742" "macro-dial-one" "Dial" "HANGUP" "1322126339.351786" "605" "from-internal" "AppDial"
"BRIDGE_START" "1322126306.713233" "49524742" "macro-dial-one" "Dial" "CHAN_END" "1322126339.352030" "605" "from-internal" "AppDial"
"ANSWER" "1322126306.713394" "" "from-internal" "AppDial" "APP_END" "1322126339.352151" "49524742" "macro-dial-one" "Dial"
"HANGUP" "1322126306.713480" "" "from-internal" "AppDial" "HANGUP" "1322126339.353329" "49524742" "from-internal-xfer" ""
"CHAN_END" "1322126306.713512" "" "from-internal" "AppDial" "CHAN_END" "1322126339.354762" "49524742" "from-internal-xfer" ""
"HANGUP" "1322126306.714272" "606" "from-internal" "AppDial" "LINKEDID_END" "1322126339.354858" "49524742" "from-internal-xfer" ""
"CHAN_END" "1322126306.714476" "606" "from-internal" "AppDial"
"APP_END" "1322126306.714525" "49524742" "macro-dial" "Dial"
"HANGUP" "1322126306.715063" "49524742" "from-internal" ""

events=all
"CHAN_END" "1322126306.724464" "49524742" "from-internal" ""
"BRIDGE_START" "1322126306.730180" "49524742" "macro-dial" "Dial"
"BRIDGE_UPDATE" "1322126306.746361" "" "macro-dial" "AppDial"

¡UN SOLO LLAMADO!


¡¿Y el screenshot de la
aplicación ?!
Estoy trabajando
en eso. Por lo
pronto es una
idea.


Asterisk Manager Interface - AMI
Interfaz de gestión por socket tcp/5038
Permite controlar asterisk desde una
aplicación externa.
Recepción de eventos
Envío de comandos
Orientado a desarrolladores y para ser
accedida por aplicaciones.



Es un protocolo simple de texto plano
Desde Asterisk 1.6.2 soporta cifrado TLS
Este tipo de interfaz no es privativa de
Asterisk. Freeswitch dispone de una
interfaz similar llamada Event Socket.



/etc/asterisk/manager.conf
[general]
enabled = yes

[admin]
secret = claveSuperSecreta
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
read = all
write = all
eventfilter=!Event: RTCPSent
eventfilter=!Event: RTCPReceived



Eventos (recepción)

Event: Hangup
Privilege: call,all
Channel: DAHDI/i2/555502341-2864
Uniqueid: 1313184096.179300
CallerIDNum: 555502341
CallerIDName: <unknown>
ConnectedLineNum: 555502341
ConnectedLineName: <unknown>
Cause: 16
Cause-txt: Normal Clearing



Comandos (envío)

Action: MailboxStatus
Mailbox: 602@default

Action: DBGet
Family: CF
Key: 606
ActionID: getvar!custom!CF/606



¿Que tipo de aplicación podemos
desarrollar usando AMI?



Asterisk Manager Interface - FOP2 - www.fop2.com



Desafío: FOP2 + Voz
Adobe Flash soporta transmisión de datos
continua via xmlsockets

Adobe Flash soporta transporte de audio y
video en tiempo real usando RTMP (Real Time
Messaging Protocol)

FOP2 utiliza xmlsockets de Flash para
mostrar y controlar estado en tiempo real

Entonces, ¿por qué no integrar audio en
FOP2 usando RTMP?



Arquitectura FOP2 - RTMP

*
SIP RTMP

UDP 5060 TCP 1935
Servidor Cliente/Navegador
Proxy RTMP
Asterisk Softphone Web

Asterisk puro y duro Mediador / Proxy Softphone WEB

Dispositivos SIP Libfreeswitch Javascript

Sesion RTMP = Registro SIP Adobe Flash

Transcodiﬁcación Speex
Wideband - Codec Asterisk.



FOP2 - De consola a teléfono

Reconversión de la interfaz de
usuario, hay que superar la
consola e incorporar la
funcionalidad de teléfono.

Agregado de Dialpad y DTMF

Agregado de Acciones:
Answer, Hold, Reject

Reubicación de botones de
control de llamados dentro del
llamado activo.



FOP2: de consola a teléfono

Dialpad / DTMF

Módulo de softphone
Selección de entrada de audio



FOP2: de consola a teléfono

Reubicación de
botones
de control



¿ Preguntas ?

Nicolás Gudiño
asternic@gmail.com


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