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Hola,
tengo instalado elastix 1.6 configurado para una troncal SIP de EPM en Bogota, Colombia. la troncal
tiene 1 CID y 20 DID se supone que en este momento tengo todo bien configurado para la entrada de
llamadas pero al momento de sacar las llamadas esta reportado el error "lo sentimos todas las lineas
estan ocupadas por favor intente mas tarde". Ayuda por favor
Andres
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Hay algunos proveedor SIP que si pones un CID correcto, no te dejan hacer la llamada. As mismo se
puede dar el caso de que ests enviando de formaincorrecta los dgitos, es decir, puede que tu
operador est esperando un prefijo y no se lo ests enviando.
Te recomiendo que cuando hagas una llamada, tengas en otr ventana ejecutando lo siguiente:
tail -f /var/log/asterisk/full | grep response
Y as veas el mensaje SIP de error, y en conjunto con tu proveedor vean cul es el problema.
============================================================================
Re:error
jgutierrez wrote:
Hay algunos proveedor SIP que si pones un CID correcto, no te dejan hacer la llamada. As mismo se
puede dar el caso de que ests enviando de formaincorrecta los dgitos, es decir, puede que tu
operador est esperando un prefijo y no se lo ests enviando.
Te recomiendo que cuando hagas una llamada, tengas en otr ventana ejecutando lo siguiente:
tail -f /var/log/asterisk/full | grep response
Y as veas el mensaje SIP de error, y en conjunto con tu proveedor vean cul es el problema.
Hola Jorge
Mi problema es similar a este y lo puse en otro hilo pero no obtuve solucin, si yo marco un n errneo o
que no existe, la centralita me dice que todas las lneas estn ocupadas, cuando el error debera ser
que el nmero marcado no existe o es incorrecto. Hay alguna manera de controlar esos errores para
que la locucin sea la correcta? Es que causa confusin porque la gente intenta una y otra vez llamar a
esos nmeros pensando en que efectivamente las lneas estn ocupadas, cuando no es as.
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Saludos
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Re: Re:error
Yo tambien tengo el mismo problema pero es con una troncal SIP de EMCALI en colombia.
Quedo en espera de una pronta solucion.
Saludos
JF
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Deben pegar la salida del CLI cuando se marca a un nmero equivocado, y a un nmero ocupado
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http://www.elastix.org/images/fbfiles/files/Numero_equivocado_o_no_existente.txt
http://www.elastix.org/images/fbfiles/files/Numero_equivocado_o_no_existente.txt
http://www.elastix.org/images/fbfiles/files/Numero_equivocado_o_no_existente.txt Buenas Noche, a
continuacion estos son los CLI solicitados:
NUMERO EQUIVOCADO
sterisk 1.4.36, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for detail
s.
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 1.4.36 currently running on elastix (pid = 2996)
Verbosity is at least 3
elastix*CLI>
elastix*CLI> clear
No such command 'clear' (type 'help clear' for other possible commands)
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elastix*CLI> cl
No such command 'cl' (type 'help cl' for other possible commands)
elastix*CLI>
Disconnected from Asterisk server
# clear
# asterisk -r
Asterisk 1.4.36, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer
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 1.4.36 currently running on elastix (pid = 2996)
Verbosity is at least 3
-- Executing Macro("SIP/599-00000f47", "user-callerid|SKIPTTL|") in new stack
-- Executing Set("SIP/599-00000f47", "AMPUSER=599") in new stack
-- Executing GotoIf("SIP/599-00000f47", "0?report") in new stack
-- Executing ExecIf("SIP/599-00000f47", "1|Set|REALCALLERIDNUM=599") in new stack
-- Executing Set("SIP/599-00000f47", "AMPUSER=599") in new stack
-- Executing Set("SIP/599-00000f47", "AMPUSERCIDNAME=599") in new stack
-- Executing GotoIf("SIP/599-00000f47", "0?report") in new stack
-- Executing Set("SIP/599-00000f47", "AMPUSERCID=599") in new stack
-- Executing Set("SIP/599-00000f47", "CALLERID(all)="599" ") in new stack
-- Executing ExecIf("SIP/599-00000f47", "0|Set|CHANNEL(language)=") in new stack
-- Executing GotoIf("SIP/599-00000f47", "1?continue") in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing NoOp("SIP/599-00000f47", "Using CallerID "599" ") in new stack
-- Executing NoOp("SIP/599-00000f47", "Calling Out Route: 6411000") in new stack
-- Executing Set("SIP/599-00000f47", "MOHCLASS=default") in new stack
-- Executing Set("SIP/599-00000f47", "_NODEST=") in new stack
-- Executing Macro("SIP/599-00000f47", "record-enable|599|OUT|") in new stack
-- Executing GotoIf("SIP/599-00000f47", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI("SIP/599-00000f47", "recordingcheck|20110420-002558|1303277158.3936") in new
stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20110420-002558|1303277158.3936: Outbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing MacroExit("SIP/599-00000f47", "") in new stack
-- Executing Macro("SIP/599-00000f47", "dialout-trunk|2|2356659|") in new stack
-- Executing Set("SIP/599-00000f47", "DIAL_TRUNK=2") in new stack
-- Executing GosubIf("SIP/599-00000f47", "0?sub-pincheck|s|1") in new stack
-- Executing GotoIf("SIP/599-00000f47", "0?disabletrunk|1") in new stack
-- Executing Set("SIP/599-00000f47", "DIAL_NUMBER=2356659") in new stack
-- Executing Set("SIP/599-00000f47", "DIAL_TRUNK_OPTIONS=trWw") in new stack
-- Executing Set("SIP/599-00000f47", "OUTBOUND_GROUP=OUT_2") in new stack
-- Executing GotoIf("SIP/599-00000f47", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing GotoIf("SIP/599-00000f47", "0?skipoutcid") in new stack
-- Executing Set("SIP/599-00000f47", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing Macro("SIP/599-00000f47", "outbound-callerid|2") in new stack
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_____________________________________
Segn veo, s hay forma de cambiar el mensaje, ahora has una llamada cuando la troncal no est
disponible, para vre si el hangupcause es diferente al que se setea cuadno el nmero es equivocado.
============================================================================
_____________________________________
Este es el CLI cuando la troncal no esta disponible tal cual como me solicitasta que hiciera.
LLAMADA CUANDO LA TRONCAL NO ES DISPONIBLE
# asterisk -r
Asterisk 1.4.36, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer
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 1.4.36 currently running on elastix (pid = 2996)
Verbosity is at least 3
-- Executing Macro("SIP/599-00001499", "user-callerid|SKIPTTL|") in new stack
-- Executing Set("SIP/599-00001499", "AMPUSER=599") in new stack
-- Executing GotoIf("SIP/599-00001499", "0?report") in new stack
-- Executing ExecIf("SIP/599-00001499", "1|Set|REALCALLERIDNUM=599") in new stack
-- Executing Set("SIP/599-00001499", "AMPUSER=599") in new stack
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Ok, lo probare y te estare comentando. otra cosa tu no sabes como cancelar el retorno de eco de
sangoma U100 FXO-USB? me puedes regalar tu msn?.
Saludos,
JF
============================================================================
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Hola, coloque el codigo que me indicaste, y marque a un numero que se encuentra daado y sigue
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Ahora se me presenta otro problema adicional, instale la version del elastis 2.2.0.rc1.
A veces se me presenta el problema en la salida de llamadas ( situacin que ya paso el lunes y el dia
de ayer en la tarde y nuevamente hoy) puesto que cuando se va a llamar o suena ocupado, o el
mensaje de todas las lneas estn ocupadas, se revisa las llamadas realizndolas de una lnea anloga
o de un celular externo y funciona ok.
Esto se soluciona realizando el servidor y los ATA FXS Linksys.
Saludos,
JF
============================================================================
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