Configuracion de una E1/PRI Digium TE420P

linuxitojr

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#1
basicamente lo que nesesitamos con Pedro es lo siguiente:


yo estaba con asterisk corriendo en Debian pero me dececcione ya que no pude instalar FreePBX pues nesesitaba tener instalado PHP4 y para la version de Debian Lenny ya no sale PHP4, maas sin embargo configure la E1/PRI sin problemas, todo eso lo hice a puro scrip, luego lei sobre el poder de Elastix y decidi montarlo, hasta ahora ya tengo configuradas las extensiones SIP pero no se como crear los contextos que tenia para llamadas salientes, entrantes y no encuentro el zapata.conf, ni el zaptel.conf, en esta nueva instalacion de elastix, quiero mencionar que en la interface grafica me aparece que me reconocio los 4 Span entonces ahora solo nesesito portar esta config que me habia funcionado en el debian.

extension.conf

[DID_span_1]

exten => 5061010,1,Goto(internal,1010,1)
exten => 5061011,1,Goto(internal,1011,1)
exten => 5061012,1,Goto(internal,1012,1)

[internal]

exten => _9XXXXXXXX,1,Dial(ZAP/g1/${EXTEN:1},50,Tt)
exten => _9XXXXXXXX,n,Hangup()

exten => _9XXXXXXX,1,Dial(Zap/g1/${EXTEN:1},50,Tt) ;
exten => _9XXXXXXX,n,Hangup()


exten => 1010,1,Dial(SIP/1010,40)
exten => 1011,1,Dial(SIP/1011,40)
exten => 1012,1,Dial(SIP/1012,40)

/etc/asterisk/zaptel.conf
#########################
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16



zapata.conf.conf


[channels]
language=en
overlapdial=yes
rxwink=300
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
immediate=no
pridialplan=local
prilocaldialplan=local
switchtype=euroisdn
signalling = pri_cpe
context=DID_span_1
group=1
channel => 1-15,17-31
 

asepulveda

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#2
Si estas utilizando la version 1.5 de elastix no encuentras el zapata.con por que ya no existe en las nuevas veriosnes de Asterisk, fue sustituido por dahdi , busca los archivos de configuración en /etc/asterisk , en zaptel lo encuentras en /etc.

El dadhi se configura exactamente igual que el zapata solo fue un cambio de nombre por cuestiones comerciales y de registro de marcas,

para configurar tus contextos y no sean sobreescritos por el free pbx debes de agregarlos en el extension_custom.conf

espero esto te sea de utilidad

saludos
 

linuxitojr

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#3
bueno hice lo que mencionastes pero no me anda.

me recomiendan trabajar con una version anterior a la que tengo instalada ?

en caso que si me la recomiendan que version puedo descargar y de donde ?

:unsure:
 

jcastellanos

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#4
es que seria lo mismo, aqui tu problema es de configuracion, como conectyas tus lineas a tu E1? el medio es RJ45 o tienes un Ballum?
 

linuxitojr

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#5
Jorge

no es un problema asociado a conexion fisica pues como comente anteriormente la tarjeta ya la tenia funcionando, lastimosamente no pude cargar Debian Etch en ese server por que no me reconoce la tarjeta de red, entonces hice la prueba con Debian Lenny y su me andaba la tarjeta de red y ahi si pude lavantar asterisk y configurar la tarjeta Digium, ahora la configuracion que postie anteriormete de los file sip.conf, entension.conf, zapata.conf, zaptel.conf yo sacaba llamadas hacia la PSTN y viceversa, lastimosamente es muy complejo trabajar a puro scrip y documentandome lei de la gran capacidad de Elastix siendo asi que decidi reinstalar el server, ahora el problema es que no puedo definir la configuracion que en su momento monte en el asterisk con debian, ya que configure con esos parametros los ficheros extension_custom.conf, dahdi-channels.conf,chan_dahdi.conf y no me funciona:S .



:S :S :S :S :S :S
 

jcastellanos

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#6
ok, ahora te entiendo mejor, me imagino que la tarjeta si la detecta son problemas el elastix?
 

asepulveda

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#7
Intenta hacer tus rutas de salida desde el free pbx y ver si asi es posible sacar llamadas , para descartar problemas de configuración de la tarjeta ya que comentas que si ves los span y los canales.
 

linuxitojr

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#8
en efecto la reconoce aqui esta el fichero dahdi-channels.conf


; Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" (MASTER) HDB3/CCS/CRC4 RECOVERINGC
lockSource
group=0,11
context=from-pstn
switchtype = euroisdn
signalling = pri_cpe
channel => 1-15,17-31
context = default
group = 63

; Span 2: TE4/0/2 "T4XXP (PCI) Card 0 Span 2" HDB3/CCS/CRC4 RED
group=0,12
context=from-pstn
switchtype = euroisdn
signalling = pri_cpe
channel => 32-46,48-62
context = default
group = 63

; Span 3: TE4/0/3 "T4XXP (PCI) Card 0 Span 3" HDB3/CCS/CRC4 RED
group=0,13
context=from-pstn
switchtype = euroisdn
signalling = pri_cpe
channel => 63-77,79-93
context = default
group = 63

; Span 4: TE4/0/4 "T4XXP (PCI) Card 0 Span 4" HDB3/CCS/CRC4 RED
group=0,14
context=from-pstn
switchtype = euroisdn
signalling = pri_cpe
channel => 94-108,110-124
context = default
group = 63



ahora el chan_dahdi.conf



[trunkgroups]

[channels]
context=from-pstn
signalling=fxs_ks <--- he cambiando manualmente la senalizacion pero no anda :S
rxwink=300 ; Atlas seems to use long (250ms) winks
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
faxdetect=incoming
echotraining=800
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1

;Uncomment these lines if you have problems with the disconection of your analog lines
;busydetect=yes
;busycount=3


immediate=no

#include dahdi-channels.conf
#include chan_dahdi_additional.conf
 

linuxitojr

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#9
Pregunta

lis ficheros dahdi-channels.conf y chan_dahdi.conf lo puedo manipular desde los Scrip agregando los parametros que ya tenia configurado en es scrip de zapata.con y zaptel.conf :dry:
 

juanelop

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#10
Hola:

Espero que me puedan ayudar, mi problema o conflicto que tengo es que quiero hacer llamadas por medio de mi E1 via softphone? se puede hacer esto... ya tengo instalada mi E1 en mi elastix , yo ya he hecho llamamadas sip... por ip claro.. pero lo que quiero ahora es por la ruta de la E1 salir para hacer llamadas..

Espero que me puedan orientar ...


Saludos!!.
 

asepulveda

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#11
De hecho la señalización esta correcta , para módulos FXO debe de ser fxs_ks.

Me extraña que no seas capas de sacar llamadas, es R2 modificado tu enlace? que de muestra cuando le pones dadhi shoe channels en el cli d asterisk?
 

security

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#12
en la parte de atras de la tarjeta el bombillo deben de estar de color verde cuando conectas los primarios, si es asi deja ver el archivo de configuracion de dahdi a ver, o prueba quitando cr4
 

victorhugo76

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#13
hola que tal yo hice una pregunta creo que es parecida a la que esta en esta es la siguiente tengo elstix puse la A101 sangoma tambien configure troncal zap ruta saliente y extension sip en los archivos de dahdi he cambiado infinidad de veces ya que he encontrado en otros foros diferentes configuraciones ya no se nipor donde les envio mis archivos

dahdi-channels.conf
; Span 2: WRTDM/0 "wrtdm Board 1"
;;; line="32 WRTDM/0/0 FXSKS (In use) (EC: MG2)"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 32
callerid=
group=
context=default

;;; line="33 WRTDM/0/1 FXSKS (In use) (EC: MG2)"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 33
callerid=
group=
context=default

;;; line="34 WRTDM/0/2 FXSKS (In use) (EC: MG2)"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 34
callerid=
group=
context=default

;;; line="35 WRTDM/0/3 FXSKS (In use) (EC: MG2)"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 35
callerid=
group=
context=default
[/quote]

chan_dahdi.conf

callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no

;Sangoma A101 port 1 [slot:6 bus:1 span:1] <wanpipe1>
context=from-trunk
group=0
echocancel=yes
signalling=fxo_ls
channel => 1-31

;Sangoma AFT-A200 [slot:5 bus:1 span:2] <wanpipe2>
context=from-zaptel
group=0
echocancel=yes
signalling = fxs_ks
channel => 32

context=from-zaptel
group=0
echocancel=yes
signalling = fxs_ks
channel => 33

context=from-zaptel
group=0
echocancel=yes
signalling = fxs_ks
channel => 34

context=from-zaptel
group=0
echocancel=yes
signalling = fxs_ks
channel => 35
system.conf

# This file is parsed by the Dahdi Configurator, dahdi_cfg
#
# Span 1: WPE1/0 "wanpipe1 card 0" (MASTER) HDB3/CCS
span=1,1,0,ccs,hdb3,crc4
# termtype: te
bchan=1-15,17-31
dchan=16
echocanceller=oslec,1-15,17-31

# Span 2: WRTDM/0 "wrtdm Board 1"
fxsks=32
echocanceller=oslec,32
fxsks=33
echocanceller=oslec,33
fxsks=34
echocanceller=oslec,34
fxsks=35
echocanceller=oslec,35
# channel 36, WRTDM/0/4, no module.
# channel 37, WRTDM/0/5, no module.
# channel 38, WRTDM/0/6, no module.
# channel 39, WRTDM/0/7, no module.
# channel 40, WRTDM/0/8, no module.
# channel 41, WRTDM/0/9, no module.
# channel 42, WRTDM/0/10, no module.
# channel 43, WRTDM/0/11, no module.
# channel 44, WRTDM/0/12, no module.
# channel 45, WRTDM/0/13, no module.
# channel 46, WRTDM/0/14, no module.
# channel 47, WRTDM/0/15, no module.
# channel 48, WRTDM/0/16, no module.
# channel 49, WRTDM/0/17, no module.
# channel 50, WRTDM/0/18, no module.
# channel 51, WRTDM/0/19, no module.
# channel 52, WRTDM/0/20, no module.
# channel 53, WRTDM/0/21, no module.
# channel 54, WRTDM/0/22, no module.
# channel 55, WRTDM/0/23, no module.

# Global data

loadzone = us
defaultzone = us
extensions.conf
include => ext-local
exten => s,1,Playback(vm-goodbye)
exten => s,2,Macro(hangupcall)
disculpen tan larga la pregunta pero son los archivos que he editado les pido encaresidamente me puedan apoyar

gracias
victor
victor.lopez@mail.vvangard.com.mx
puebla puebla (mexico)
 

asepulveda

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#14
linuxitojr said:
en efecto la reconoce aqui esta el fichero dahdi-channels.conf


; Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" (MASTER) HDB3/CCS/CRC4 RECOVERINGC
lockSource
context=from-pstn
switchtype = euroisdn
signalling = pri_cpe
channel => 1-15,17-31
context = default
group = 63
linuxitojr said:
[internal]

exten => _9XXXXXXXX,1,Dial(ZAP/g1/${EXTEN:1},50,Tt)
exten => _9XXXXXXXX,n,Hangup()

exten => _9XXXXXXX,1,Dial(Zap/g1/${EXTEN:1},50,Tt) ;
exten => _9XXXXXXX,n,Hangup()
te das cuenta que tus canales estan en el group=63?? pero tu dial plans esta configurados para el grupo 1?? cambia los grupos de cualqueuiera de los dos y se debe de solucionar tu problema

saludos
 

victorhugo76

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#15
gracias ..
lo de los canales ya le cambie el grupo =1 igual al dial plans
tambien en la troncal le puse en lugar de la default g0 le puse g1 y en la saliente reconocio zap/g1.

tambien le puse [internal]

exten => _9XXXXXXXX,1,Dial(ZAP/g1/${EXTEN:1},50,Tt)
exten => _9XXXXXXXX,n,Hangup()

exten => _9XXXXXXX,1,Dial(Zap/g1/${EXTEN:1},50,Tt) ;
exten => _9XXXXXXX,n,Hangup()

en el archivo de extensions.conf y despues quise marcar con el x-lite a un numero local de 7 digitos y nada marque con 9 +7 digitos y nada, me podrian seguir ayudando?
 

asepulveda

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#16
Trata quitando el 9 en tu dial plan y marcando los 7 dígitos , cuando marcas que te aparece en consola de asterisk?? si lo pegas aqui podemos tener mas idea de que error te da

prueba asi

exten => _XXXXXXX,1,Dial(ZAP/g1/${EXTEN},50,Tt)
exten => _XXXXXXX,n,Hangup()

exten => _XXXXXXX,1,Dial(Zap/g1/${EXTEN},50,Tt) ;
exten => _XXXXXXX,n,Hangup()

Ahora por que no pruebas haciendo tu dial plan en las outbounds de freepbx??
 

victorhugo76

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#17
ok probare y te envio el resultado
 

victorhugo76

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#18
esto es lo que sale en el log despues de haber quitado el 9 en el dial plan

-- Goto (macro-user-callerid,s,20)
-- Executing [s@macro-user-callerid:20] NoOp("SIP/1100-09f4f120", "Using Cal lerID "1100" <1100>") in new stack
-- Executing [5946400@from-internal:2] Set("SIP/1100-09f4f120", "_NODEST=") in new stack
-- Executing [5946400@from-internal:3] Macro("SIP/1100-09f4f120", "record-en able|1100|OUT|") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/1100-09f4f120", "1?check" ) in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] AGI("SIP/1100-09f4f120", "recordingch eck|20090520-161844|1242854324.2") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20090520-161844|1242854324.2: Outbound recording enabled.
recordingcheck|20090520-161844|1242854324.2: CALLFILENAME=OUT-1100-1242854324. 2
-- AGI Script recordingcheck completed, returning 0
-- Executing [s@macro-record-enable:999] MixMonitor("SIP/1100-09f4f120", "OU T-1100-1242854324.2.wav||") in new stack
-- Executing [5946400@from-internal:4] Macro("SIP/1100-09f4f120", "dialout-t runk|2|5946400||") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/1100-09f4f120", "DIAL_TRUNK= 2") in new stack
-- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/1100-09f4f120", "0?sub-p incheck|s|1") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/1100-09f4f120", "1?disabl etrunk|1") in new stack
-- Goto (macro-dialout-trunk,disabletrunk,1)
-- Executing [disabletrunk@macro-dialout-trunk:1] NoOp("SIP/1100-09f4f120", "TRUNK: SIP/FCOM-01 DISABLED - falling through to next trunk") in new stack
-- Executing [5946400@from-internal:5] Macro("SIP/1100-09f4f120", "outisbusy |") in new stack
-- Executing [s@macro-outisbusy:1] Playback("SIP/1100-09f4f120", "all-circui ts-busy-now|noanswer") in new stack
-- <SIP/1100-09f4f120> Playing 'all-circuits-busy-now' (language 'en')
== Begin MixMonitor Recording SIP/1100-09f4f120
-- Executing [s@macro-outisbusy:2] Playback("SIP/1100-09f4f120", "pls-try-ca ll-later|noanswer") in new stack
-- <SIP/1100-09f4f120> Playing 'pls-try-call-later' (language 'en')
-- Executing [s@macro-outisbusy:3] Macro("SIP/1100-09f4f120", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("SIP/1100-09f4f120", "w") in ne w stack
-- Executing [s@macro-hangupcall:2] NoCDR("SIP/1100-09f4f120", "") in new st ack
-- Executing [s@macro-hangupcall:3] GotoIf("SIP/1100-09f4f120", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] GotoIf("SIP/1100-09f4f120", "1?skipblkvm ") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] GotoIf("SIP/1100-09f4f120", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] Hangup("SIP/1100-09f4f120", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/1100-09f4 f120' in macro 'hangupcall'
== Spawn extension (macro-outisbusy, s, 3) exited non-zero on 'SIP/1100-09f4f1 20' in macro 'outisbusy'
== Spawn extension (from-internal, 5946400, 5) exited non-zero on 'SIP/1100-09 f4f120'
-- Executing [h@from-internal:1] Macro("SIP/1100-09f4f120", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("SIP/1100-09f4f120", "w") in ne w stack
-- Executing [s@macro-hangupcall:2] NoCDR("SIP/1100-09f4f120", "") in new st ack
-- Executing [s@macro-hangupcall:3] GotoIf("SIP/1100-09f4f120", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] GotoIf("SIP/1100-09f4f120", "1?skipblkvm ") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] GotoIf("SIP/1100-09f4f120", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] Hangup("SIP/1100-09f4f120", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/1100-09f4 f120' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/1100-09f4f120 '
== End MixMonitor Recording SIP/1100-09f4f120
-- Executing [5946400@from-internal:1] Macro("SIP/1100-09f58e88", "user-call erid|SKIPTTL|") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/1100-09f58e88", "AMPUSER=110 0") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/1100-09f58e88", "0?report ") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/1100-09f58e88", "1|Set|RE ALCALLERIDNUM=1100") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/1100-09f58e88", "AMPUSER=110 0") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/1100-09f58e88", "AMPUSERCIDNAME=1100") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/1100-09f58e88", "0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/1100-09f58e88", "AMPUSERCID=1100") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/1100-09f58e88", "CALLERID(all)="1100" <1100>") in new stack
-- Executing [s@macro-user-callerid:9] Set("SIP/1100-09f58e88", "REALCALLERIDNUM=1100") in new stack
-- Executing [s@macro-user-callerid:10] ExecIf("SIP/1100-09f58e88", "0|Set|CHANNEL(language)=") in new stack
-- Executing [s@macro-user-callerid:11] GotoIf("SIP/1100-09f58e88", "1?continue") in new stack
-- Goto (macro-user-callerid,s,20)
-- Executing [s@macro-user-callerid:20] NoOp("SIP/1100-09f58e88", "Using CallerID "1100" <1100>") in new stack
-- Executing [5946400@from-internal:2] Set("SIP/1100-09f58e88", "_NODEST=") in new stack
-- Executing [5946400@from-internal:3] Macro("SIP/1100-09f58e88", "record-enable|1100|OUT|") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/1100-09f58e88", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] AGI("SIP/1100-09f58e88", "recordingcheck|20090520-161855|1242854334.3") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20090520-161855|1242854334.3: Outbound recording enabled.
recordingcheck|20090520-161855|1242854334.3: CALLFILENAME=OUT-1100-1242854334.3
-- AGI Script recordingcheck completed, returning 0
-- Executing [s@macro-record-enable:999] MixMonitor("SIP/1100-09f58e88", "OUT-1100-1242854334.3.wav||") in new stack
-- Executing [5946400@from-internal:4] Macro("SIP/1100-09f58e88", "dialout-trunk|2|5946400||") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/1100-09f58e88", "DIAL_TRUNK=2") in new stack
-- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/1100-09f58e88", "0?sub-pincheck|s|1") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/1100-09f58e88", "1?disabletrunk|1") in new stack
-- Goto (macro-dialout-trunk,disabletrunk,1)
-- Executing [disabletrunk@macro-dialout-trunk:1] NoOp("SIP/1100-09f58e88", "TRUNK: SIP/FCOM-01 DISABLED - falling through to next trunk") in new stack
-- Executing [5946400@from-internal:5] Macro("SIP/1100-09f58e88", "outisbusy|") in new stack
-- Executing [s@macro-outisbusy:1] Playback("SIP/1100-09f58e88", "all-circuits-busy-now|noanswer") in new stack
-- <SIP/1100-09f58e88> Playing 'all-circuits-busy-now' (language 'en')
== Begin MixMonitor Recording SIP/1100-09f58e88
-- Executing [s@macro-outisbusy:2] Playback("SIP/1100-09f58e88", "pls-try-call-later|noanswer") in new stack
-- <SIP/1100-09f58e88> Playing 'pls-try-call-later' (language 'en')
-- Executing [s@macro-outisbusy:3] Macro("SIP/1100-09f58e88", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("SIP/1100-09f58e88", "w") in new stack
-- Executing [s@macro-hangupcall:2] NoCDR("SIP/1100-09f58e88", "") in new stack
-- Executing [s@macro-hangupcall:3] GotoIf("SIP/1100-09f58e88", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] GotoIf("SIP/1100-09f58e88", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] GotoIf("SIP/1100-09f58e88", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] Hangup("SIP/1100-09f58e88", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/1100-09f58e88' in macro 'hangupcall'
== Spawn extension (macro-outisbusy, s, 3) exited non-zero on 'SIP/1100-09f58e88' in macro 'outisbusy'
== Spawn extension (from-internal, 5946400, 5) exited non-zero on 'SIP/1100-09f58e88'
-- Executing [h@from-internal:1] Macro("SIP/1100-09f58e88", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("SIP/1100-09f58e88", "w") in new stack
-- Executing [s@macro-hangupcall:2] NoCDR("SIP/1100-09f58e88", "") in new stack
-- Executing [s@macro-hangupcall:3] GotoIf("SIP/1100-09f58e88", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] GotoIf("SIP/1100-09f58e88", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] GotoIf("SIP/1100-09f58e88", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] Hangup("SIP/1100-09f58e88", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/1100-09f58e88' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/1100-09f58e88'
== End MixMonitor Recording SIP/1100-09f58e88
elasix*CLI>
Disconnected from Asterisk server
[root@elasix ~]# asterisk -r
Asterisk 1.4.25-rc1, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 1.4.25-rc1 currently running on elasix (pid = 3150)
Verbosity is at least 3
-- Executing [5946400@from-internal:1] Macro("SIP/1100-09f5d1e8", "user-callerid|SKIPTTL|") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/1100-09f5d1e8", "AMPUSER=1100") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/1100-09f5d1e8", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/1100-09f5d1e8", "1|Set|REALCALLERIDNUM=1100") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/1100-09f5d1e8", "AMPUSER=1100") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/1100-09f5d1e8", "AMPUSERCIDNAME=1100") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/1100-09f5d1e8", "0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/1100-09f5d1e8", "AMPUSERCID=1100") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/1100-09f5d1e8", "CALLERID(all)="1100" <1100>") in new stack
-- Executing [s@macro-user-callerid:9] Set("SIP/1100-09f5d1e8", "REALCALLERIDNUM=1100") in new stack
-- Executing [s@macro-user-callerid:10] ExecIf("SIP/1100-09f5d1e8", "0|Set|CHANNEL(language)=") in new stack
-- Executing [s@macro-user-callerid:11] GotoIf("SIP/1100-09f5d1e8", "1?continue") in new stack
-- Goto (macro-user-callerid,s,20)
-- Executing [s@macro-user-callerid:20] NoOp("SIP/1100-09f5d1e8", "Using CallerID "1100" <1100>") in new stack
-- Executing [5946400@from-internal:2] Set("SIP/1100-09f5d1e8", "_NODEST=") in new stack
-- Executing [5946400@from-internal:3] Macro("SIP/1100-09f5d1e8", "record-enable|1100|OUT|") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/1100-09f5d1e8", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] AGI("SIP/1100-09f5d1e8", "recordingcheck|20090520-161910|1242854350.4") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20090520-161910|1242854350.4: Outbound recording enabled.
recordingcheck|20090520-161910|1242854350.4: CALLFILENAME=OUT-1100-1242854350.4
-- AGI Script recordingcheck completed, returning 0
-- Executing [s@macro-record-enable:999] MixMonitor("SIP/1100-09f5d1e8", "OUT-1100-1242854350.4.wav||") in new stack
-- Executing [5946400@from-internal:4] Macro("SIP/1100-09f5d1e8", "dialout-trunk|2|5946400||") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/1100-09f5d1e8", "DIAL_TRUNK=2") in new stack
-- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/1100-09f5d1e8", "0?sub-pincheck|s|1") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/1100-09f5d1e8", "1?disabletrunk|1") in new stack
-- Goto (macro-dialout-trunk,disabletrunk,1)
-- Executing [disabletrunk@macro-dialout-trunk:1] NoOp("SIP/1100-09f5d1e8", "TRUNK: SIP/FCOM-01 DISABLED - falling through to next trunk") in new stack
-- Executing [5946400@from-internal:5] Macro("SIP/1100-09f5d1e8", "outisbusy|") in new stack
-- Executing [s@macro-outisbusy:1] Playback("SIP/1100-09f5d1e8", "all-circuits-busy-now|noanswer") in new stack
== Begin MixMonitor Recording SIP/1100-09f5d1e8
-- <SIP/1100-09f5d1e8> Playing 'all-circuits-busy-now' (language 'en')
-- Executing [s@macro-outisbusy:2] Playback("SIP/1100-09f5d1e8", "pls-try-call-later|noanswer") in new stack
-- <SIP/1100-09f5d1e8> Playing 'pls-try-call-later' (language 'en')
== Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/1100-09f5d1e8' in macro 'outisbusy'
== Spawn extension (from-internal, 5946400, 5) exited non-zero on 'SIP/1100-09f5d1e8'
-- Executing [h@from-internal:1] Macro("SIP/1100-09f5d1e8", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("SIP/1100-09f5d1e8", "w") in new stack
-- Executing [s@macro-hangupcall:2] NoCDR("SIP/1100-09f5d1e8", "") in new stack
-- Executing [s@macro-hangupcall:3] GotoIf("SIP/1100-09f5d1e8", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] GotoIf("SIP/1100-09f5d1e8", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] GotoIf("SIP/1100-09f5d1e8", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] Hangup("SIP/1100-09f5d1e8", "") in new stack
[04:59:17 p.m.] jpablo: wget -c
de verdad agradesco tu ayuda.
cres que pueda contactarte por msg o skype? o a tu tel yo vivo en puebla mexico
 

linuxitojr

Joined
May 15, 2009
Messages
57
Likes
0
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#19
Bueno Sres les comento que yo sigo con mi problema, en el log que me muestra en el CLI me dice "all-circuits-busy-now| y si hago un lsdahdi

en algunos casos tengo llamadas entrantes pero salientens si que no se pueden

### Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" (MASTER) HDB3/CCS/CRC4 RECOVERINGClockSource
1 PRI Clear (In use) (EC: OSLEC) RED
2 PRI Clear (In use) (EC: OSLEC) RED
3 PRI Clear (In use) (EC: OSLEC) RED
4 PRI Clear (In use) (EC: OSLEC) RED
5 PRI Clear (In use) (EC: OSLEC) RED
6 PRI Clear (In use) (EC: OSLEC) RED
7 PRI Clear (In use) (EC: OSLEC) RED
8 PRI Clear (In use) (EC: OSLEC) RED
9 PRI Clear (In use) (EC: OSLEC) RED
10 PRI Clear (In use) (EC: OSLEC) RED
11 PRI Clear (In use) (EC: OSLEC) RED
12 PRI Clear (In use) (EC: OSLEC) RED
13 PRI Clear (In use) (EC: OSLEC) RED
14 PRI Clear (In use) (EC: OSLEC) RED
15 PRI Clear (In use) (EC: OSLEC) RED
16 PRI HDLCFCS (In use) RED
17 PRI Clear (In use) (EC: OSLEC) RED
18 PRI Clear (In use) (EC: OSLEC) RED
19 PRI Clear (In use) (EC: OSLEC) RED
20 PRI Clear (In use) (EC: OSLEC) RED
21 PRI Clear (In use) (EC: OSLEC) RED
22 PRI Clear (In use) (EC: OSLEC) RED
23 PRI Clear (In use) (EC: OSLEC) RED
24 PRI Clear (In use) (EC: OSLEC) RED
25 PRI Clear (In use) (EC: OSLEC) RED
26 PRI Clear (In use) (EC: OSLEC) RED
27 PRI Clear (In use) (EC: OSLEC) RED
28 PRI Clear (In use) (EC: OSLEC) RED
29 PRI Clear (In use) (EC: OSLEC) RED
30 PRI Clear (In use) (EC: OSLEC) RED
31 PRI Clear (In use) (EC: OSLEC) RED


asterisk-*CLI> zap show channels
Chan Extension Context Language MOH Interpret
pseudo default default
1 from-pstn default
2 from-pstn default
3 from-pstn default
4 from-pstn default
5 from-pstn default
6 from-pstn default
7 from-pstn default
8 from-pstn default
9 from-pstn default
10 from-pstn default
11 from-pstn default
12 from-pstn default
13 from-pstn default
14 from-pstn default
15 from-pstn default
17 from-pstn default
18 from-pstn default
19 from-pstn default
20 from-pstn default
21 from-pstn default
22 from-pstn default
23 from-pstn default
24 from-pstn default
25 from-pstn default
26 from-pstn default
27 from-pstn default
28 from-pstn default
29 from-pstn default
30 from-pstn default
31 from-pstn default
 

jcastellanos

Joined
Feb 10, 2009
Messages
2,404
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#20
no sera cosa de tu medio?
 

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