¿me falla extension o telefono?

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sae...@comyc.com

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Nov 27, 2010, 8:48:05 AM11/27/10
to aster...@googlegroups.com
Hola a todos , tengo una extensi�n , la 200 , que me lanza un mensaje
siempre, que dice:
Lo siento ha ocurrido un error, y no se por donde meterle mano. Todo el
resto de extensiones bien y la que falla si puede llamar a las dem�s.
Adjunto el log de asterisk -r , llamando a esa extensi�n y a otra que si
funciona .
El problema empieza aqu�:
-- Called 200
-- Got SIP response 486 "Busy Here" back from 192.168.1.122
-- SIP/200-00000031 is busy
== Everyone is busy/congested at this time (1:1/0/0)
�me podeis orientar?�es asterisk o el telefono quien dice "Busy Here"?

Gracias
Saegen


1� log de extension que falla "Mensaje, lo siento ha ocurrido un error"
*********************************************************************************************************************


Connected to Asterisk 1.4.32 currently running on com (pid = 3172)
Verbosity is at least 3
-- Executing [200@from-internal:1] Macro("SIP/220-00000030",
"exten-vm|200|200") in new stack
-- Executing [s@macro-exten-vm:1] Macro("SIP/220-00000030",
"user-callerid|") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/220-00000030",
"AMPUSER=220") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/220-00000030",
"0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/220-00000030",
"1|Set|REALCALLERIDNUM=220") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/220-00000030",
"AMPUSER=220") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/220-00000030",
"AMPUSERCIDNAME=220 ADMINISTRACION") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/220-00000030",
"0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/220-00000030",
"AMPUSERCID=220") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/220-00000030",
"CALLERID(all)="220 ADMINISTRACION" <220>") in new stack
-- Executing [s@macro-user-callerid:9] ExecIf("SIP/220-00000030",
"1|Set|CHANNEL(language)=es") in new stack
-- Executing [s@macro-user-callerid:10] GotoIf("SIP/220-00000030",
"0?continue") in new stack
-- Executing [s@macro-user-callerid:11] Set("SIP/220-00000030",
"__TTL=64") in new stack
-- Executing [s@macro-user-callerid:12] GotoIf("SIP/220-00000030",
"1?continue") in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing [s@macro-user-callerid:19] NoOp("SIP/220-00000030",
"Using CallerID "220 ADMINISTRACION" <220>") in new stack
-- Executing [s@macro-exten-vm:2] Set("SIP/220-00000030",
"RingGroupMethod=none") in new stack
-- Executing [s@macro-exten-vm:3] Set("SIP/220-00000030",
"VMBOX=200") in new stack
-- Executing [s@macro-exten-vm:4] Set("SIP/220-00000030",
"__EXTTOCALL=200") in new stack
-- Executing [s@macro-exten-vm:5] Set("SIP/220-00000030",
"CFUEXT=") in new stack
-- Executing [s@macro-exten-vm:6] Set("SIP/220-00000030",
"CFBEXT=") in new stack
-- Executing [s@macro-exten-vm:7] Set("SIP/220-00000030", "RT=15")
in new stack
-- Executing [s@macro-exten-vm:8] Macro("SIP/220-00000030",
"record-enable|200|IN") in new stack
-- Executing [s@macro-record-enable:1]
MacroExit("SIP/220-00000030", "") in new stack
-- Executing [s@macro-exten-vm:9] Macro("SIP/220-00000030",
"dial|15|tr|200") in new stack
-- Executing [s@macro-dial:1] GotoIf("SIP/220-00000030", "1?dial")
in new stack
-- Goto (macro-dial,s,3)
-- Executing [s@macro-dial:3] AGI("SIP/220-00000030",
"dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
dialparties.agi: Caller ID name is '220 ADMINISTRACION' number is '220'
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 200 to extension map
-- dialparties.agi: Extension 200 cf is disabled
-- dialparties.agi: Extension 200 do not disturb is disabled
dialparties.agi: ExtensionState: 0
dialparties.agi: Extension 200 has ExtensionState: 0
-- dialparties.agi: Checking CW and CFB status for extension 200
-- dialparties.agi: dbset CALLTRACE/200 to 220
-- dialparties.agi: Filtered ARG3: 200
== Manager 'admin' logged off from 127.0.0.1
-- AGI Script dialparties.agi completed, returning 0
-- Executing [s@macro-dial:7] Dial("SIP/220-00000030",
"SIP/200|15|tr") in new stack
-- Called 200
-- Got SIP response 486 "Busy Here" back from 192.168.1.122
-- SIP/200-00000031 is busy
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing [s@macro-dial:8] Set("SIP/220-00000030",
"DIALSTATUS=BUSY") in new stack
-- Executing [s@macro-dial:9] GosubIf("SIP/220-00000030",
"0?BUSY|1") in new stack
-- Executing [s@macro-exten-vm:10] GotoIf("SIP/220-00000030",
"0?exit|return") in new stack
-- Executing [s@macro-exten-vm:11] Set("SIP/220-00000030",
"SV_DIALSTATUS=BUSY") in new stack
-- Executing [s@macro-exten-vm:12] GosubIf("SIP/220-00000030",
"0?docfu|1") in new stack
-- Executing [s@macro-exten-vm:13] GosubIf("SIP/220-00000030",
"0?docfb|1") in new stack
-- Executing [s@macro-exten-vm:14] Set("SIP/220-00000030",
"DIALSTATUS=BUSY") in new stack
-- Executing [s@macro-exten-vm:15] NoOp("SIP/220-00000030",
"Voicemail is 200") in new stack
-- Executing [s@macro-exten-vm:16] GotoIf("SIP/220-00000030",
"0?s-BUSY|1") in new stack
-- Executing [s@macro-exten-vm:17] NoOp("SIP/220-00000030",
"Sending to Voicemail box 200") in new stack
-- Executing [s@macro-exten-vm:18] Macro("SIP/220-00000030",
"vm|200|BUSY|") in new stack
-- Executing [s@macro-vm:1] Macro("SIP/220-00000030",
"user-callerid|SKIPTTL") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/220-00000030",
"AMPUSER=220") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/220-00000030",
"0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/220-00000030",
"0|Set|REALCALLERIDNUM=220") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/220-00000030",
"AMPUSER=220") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/220-00000030",
"AMPUSERCIDNAME=220 ADMINISTRACION") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/220-00000030",
"0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/220-00000030",
"AMPUSERCID=220") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/220-00000030",
"CALLERID(all)="220 ADMINISTRACION" <220>") in new stack
-- Executing [s@macro-user-callerid:9] ExecIf("SIP/220-00000030",
"1|Set|CHANNEL(language)=es") in new stack
-- Executing [s@macro-user-callerid:10] GotoIf("SIP/220-00000030",
"1?continue") in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing [s@macro-user-callerid:19] NoOp("SIP/220-00000030",
"Using CallerID "220 ADMINISTRACION" <220>") in new stack
-- Executing [s@macro-vm:2] Set("SIP/220-00000030", "VMGAIN=""") in
new stack
-- Executing [s@macro-vm:3] GotoIf("SIP/220-00000030", "1?vmx|1")
in new stack
-- Goto (macro-vm,vmx,1)
-- Executing [vmx@macro-vm:1] Set("SIP/220-00000030", "MEXTEN=200")
in new stack
-- Executing [vmx@macro-vm:2] Set("SIP/220-00000030", "MMODE=BUSY")
in new stack
-- Executing [vmx@macro-vm:3] Set("SIP/220-00000030", "RETVM=") in
new stack
-- Executing [vmx@macro-vm:4] Set("SIP/220-00000030", "MODE=busy")
in new stack
-- Executing [vmx@macro-vm:5] GotoIf("SIP/220-00000030",
"1?chknomsg") in new stack
-- Goto (macro-vm,vmx,7)
-- Executing [vmx@macro-vm:7] GotoIf("SIP/220-00000030",
"0?s-BUSY|1") in new stack
-- Executing [vmx@macro-vm:8] GotoIf("SIP/220-00000030",
"1?notdirect") in new stack
-- Goto (macro-vm,vmx,10)
-- Executing [vmx@macro-vm:10] NoOp("SIP/220-00000030", "Checking
if ext 200 is enabled: ") in new stack
-- Executing [vmx@macro-vm:11] GotoIf("SIP/220-00000030",
"1?s-BUSY|1") in new stack
-- Goto (macro-vm,s-BUSY,1)
-- Executing [s-BUSY@macro-vm:1] NoOp("SIP/220-00000030", "BUSY
voicemail") in new stack
-- Executing [s-BUSY@macro-vm:2] Macro("SIP/220-00000030",
"get-vmcontext|200") in new stack
-- Executing [s@macro-get-vmcontext:1] Set("SIP/220-00000030",
"VMCONTEXT=general") in new stack
-- Executing [s@macro-get-vmcontext:2] GotoIf("SIP/220-00000030",
"0?200:300") in new stack
-- Goto (macro-get-vmcontext,s,300)
-- Executing [s@macro-get-vmcontext:300] NoOp("SIP/220-00000030",
"") in new stack
-- Executing [s-BUSY@macro-vm:3] VoiceMail("SIP/220-00000030",
"200@general|b") in new stack
-- Executing [s-BUSY@macro-vm:4] Goto("SIP/220-00000030",
"exit-FAILED|1") in new stack
-- Goto (macro-vm,exit-FAILED,1)
-- Executing [exit-FAILED@macro-vm:1] Playback("SIP/220-00000030",
"im-sorry&an-error-has-occured") in new stack
-- <SIP/220-00000030> Playing 'im-sorry' (language 'es')
-- <SIP/220-00000030> Playing 'an-error-has-occured' (language 'es')
-- Executing [exit-FAILED@macro-vm:2] GotoIf("SIP/220-00000030",
"0?exit-RETURN|1") in new stack
-- Executing [exit-FAILED@macro-vm:3] Hangup("SIP/220-00000030",
"") in new stack
== Spawn extension (macro-vm, exit-FAILED, 3) exited non-zero on
'SIP/220-00000030' in macro 'vm'
== Spawn extension (macro-exten-vm, s, 18) exited non-zero on
'SIP/220-00000030' in macro 'exten-vm'
== Spawn extension (from-internal, 200, 1) exited non-zero on
'SIP/220-00000030'
-- Executing [h@from-internal:1] Macro("SIP/220-00000030",
"hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/220-00000030",
"1?skiprg") in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4] GotoIf("SIP/220-00000030",
"1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf("SIP/220-00000030",
"1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] Hangup("SIP/220-00000030", "")
in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on
'SIP/220-00000030' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on
'SIP/220-00000030'

2� log de extension que no falla
*********************************************************************************************************************


Connected to Asterisk 1.4.32 currently running on com (pid = 3172)
Verbosity is at least 3
-- Executing [230@from-internal:1] Macro("SIP/220-00000032",
"exten-vm|novm|230") in new stack
-- Executing [s@macro-exten-vm:1] Macro("SIP/220-00000032",
"user-callerid|") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/220-00000032",
"AMPUSER=220") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/220-00000032",
"0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/220-00000032",
"1|Set|REALCALLERIDNUM=220") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/220-00000032",
"AMPUSER=220") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/220-00000032",
"AMPUSERCIDNAME=220 ADMINISTRACION") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/220-00000032",
"0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/220-00000032",
"AMPUSERCID=220") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/220-00000032",
"CALLERID(all)="220 ADMINISTRACION" <220>") in new stack
-- Executing [s@macro-user-callerid:9] ExecIf("SIP/220-00000032",
"1|Set|CHANNEL(language)=es") in new stack
-- Executing [s@macro-user-callerid:10] GotoIf("SIP/220-00000032",
"0?continue") in new stack
-- Executing [s@macro-user-callerid:11] Set("SIP/220-00000032",
"__TTL=64") in new stack
-- Executing [s@macro-user-callerid:12] GotoIf("SIP/220-00000032",
"1?continue") in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing [s@macro-user-callerid:19] NoOp("SIP/220-00000032",
"Using CallerID "220 ADMINISTRACION" <220>") in new stack
-- Executing [s@macro-exten-vm:2] Set("SIP/220-00000032",
"RingGroupMethod=none") in new stack
-- Executing [s@macro-exten-vm:3] Set("SIP/220-00000032",
"VMBOX=novm") in new stack
-- Executing [s@macro-exten-vm:4] Set("SIP/220-00000032",
"__EXTTOCALL=230") in new stack
-- Executing [s@macro-exten-vm:5] Set("SIP/220-00000032",
"CFUEXT=") in new stack
-- Executing [s@macro-exten-vm:6] Set("SIP/220-00000032",
"CFBEXT=") in new stack
-- Executing [s@macro-exten-vm:7] Set("SIP/220-00000032", "RT=""")
in new stack
-- Executing [s@macro-exten-vm:8] Macro("SIP/220-00000032",
"record-enable|230|IN") in new stack
-- Executing [s@macro-record-enable:1]
MacroExit("SIP/220-00000032", "") in new stack
-- Executing [s@macro-exten-vm:9] Macro("SIP/220-00000032",
"dial||tr|230") in new stack
-- Executing [s@macro-dial:1] GotoIf("SIP/220-00000032", "1?dial")
in new stack
-- Goto (macro-dial,s,3)
-- Executing [s@macro-dial:3] AGI("SIP/220-00000032",
"dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
dialparties.agi: Caller ID name is '220 ADMINISTRACION' number is '220'
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 230 to extension map
-- dialparties.agi: Extension 230 cf is disabled
-- dialparties.agi: Extension 230 do not disturb is disabled
dialparties.agi: ExtensionState: 0
dialparties.agi: Extension 230 has ExtensionState: 0
-- dialparties.agi: Checking CW and CFB status for extension 230
-- dialparties.agi: dbset CALLTRACE/230 to 220
-- dialparties.agi: Filtered ARG3: 230
== Manager 'admin' logged off from 127.0.0.1
-- AGI Script dialparties.agi completed, returning 0
-- Executing [s@macro-dial:7] Dial("SIP/220-00000032",
"SIP/230||tr") in new stack
-- Called 230
-- SIP/230-00000033 is ringing

Elio Rojano

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Nov 28, 2010, 8:35:26 AM11/28/10
to aster...@googlegroups.com
2010/11/27 sae...@comyc.com <sae...@comyc.com>
>
> Hola a todos , tengo una extensión , la 200 , que me lanza un mensaje siempre, que dice:
> Lo siento ha ocurrido un error, y no se por donde meterle mano. Todo el resto de extensiones bien y la que falla si puede llamar a las demás.
> Adjunto el log de asterisk -r , llamando a esa extensión y a otra que si funciona .
> El problema empieza aquí:

>    -- Called 200
>    -- Got SIP response 486 "Busy Here" back from 192.168.1.122
>    -- SIP/200-00000031 is busy
>  == Everyone is busy/congested at this time (1:1/0/0)
> ¿me podeis orientar?¿es asterisk o el telefono quien dice "Busy Here"?
>
> Gracias
> Saegen
>
>
>
>
>
>
> 1º log de extension que falla "Mensaje, lo siento ha ocurrido un error"
> 2º log de extension que no falla


El teléfono dice: "Lo siento ha ocurrido un error, y no se por donde
meterle mano."
Estos terminales, cada vez más elocuentes...

Ahora en serio... a pesar que haya gente que le encante cómo trabajan
los interfaces web, el primer problema del "newbie" que utiliza este
interfaz es que piensa que todo es más fácil, pero luego se encuentran
con algo que no cuadra y ahora adivina entre las 80 líneas de código
del dialplan qué ha hecho, porqué no hace lo que quieres y cómo
solucionarlo.

En Asterisk "a pelo" sin interfaces generalistas, llamar a otro
usuario es tan sencillo como:

exten => _2XX,1,Dial(SIP/${EXTEN}) ; (si, solo esto)

Si al ejecutar esa única línea, el sistema devuelve algo como: "User
is busy or congested" ya te puedes hacer una idea de por dónde van los
tiros.

En fín, siento no poder ayudarte a solucionar tu problema... yo
prefiero ver los mensajes de error antes de aventurarme a dar
elucubraciones.


--
http://www.sinologic.net/

Iñaki Baz Castillo

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Nov 28, 2010, 8:57:15 AM11/28/10
to aster...@googlegroups.com
2010/11/27 sae...@comyc.com <sae...@comyc.com>:
> Adjunto el log de asterisk -r , llamando a esa extensión y a otra que si
> funciona .

El log que muestras es el de algún enlatado Asterisk+Web+PHP+MySQL.
Imposible ayudar así.

Los interfaces web son útiles cuando no los necesitas.

--
Iñaki Baz Castillo
<i...@aliax.net>

Paco Gil

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Nov 28, 2010, 11:45:36 AM11/28/10
to aster...@googlegroups.com
seguramente no tienes registrado al 200...

¿qué te sale un "sip show peers"?

On Sat, Nov 27, 2010 at 14:48, sae...@comyc.com <sae...@comyc.com> wrote:
Hola a todos , tengo una extensión , la 200 , que me lanza un mensaje siempre, que dice:
Lo siento ha ocurrido un error, y no se por donde meterle mano. Todo el resto de extensiones bien y la que falla si puede llamar a las demás.
Adjunto el log de asterisk -r , llamando a esa extensión y a otra que si funciona .
El problema empieza aquí:

   -- Called 200
   -- Got SIP response 486 "Busy Here" back from 192.168.1.122
   -- SIP/200-00000031 is busy
 == Everyone is busy/congested at this time (1:1/0/0)
¿me podeis orientar?¿es asterisk o el telefono quien dice "Busy Here"?

Gracias
Saegen






1º log de extension que falla "Mensaje, lo siento ha ocurrido un error"
2º log de extension que no falla
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Entra ahora en el canal de irc de Asterisk-ES para charlar en directo sobre VoIP y
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Raúl Alexis Betancor Santana

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Nov 29, 2010, 5:04:57 AM11/29/10
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On Domingo 28 Noviembre 2010 16:45:36 Paco Gil escribió:
> seguramente no tienes registrado al 200...
>
> ¿qué te sale un "sip show peers"?

Idependientemente de que el pobre desgraciado esté usando un enlatado... y que
va sufir una lapidación pública, si vuelve a pegar semejante tipo de traza ...
es relativamente fácil ver que el 486 lo devuelve el teléfono. ¿Porqué? ...
eso ya le toca a él depurarlo, con ngrep, como dios manda.

Saludos
--
Raúl Alexis Betancor Santana
Dimensión Virtual

Fernando Villares

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Nov 29, 2010, 9:13:43 AM11/29/10
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no tendra el boton de DND habilitado?????

Martin Rodriguez

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Nov 29, 2010, 10:18:08 AM11/29/10
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Sumandome a la causa de Raul. No se olviden que esto es Telefonia IP y el trafico que viaja por al red tienen codigos de respuesta que con una miradita de RFC entendemos el idioma.

Ya me paso de postear que revisen el DND del Telefono, y muchos respondieron "no creo que sea eso por SIP me llego un ocupado" jejejej.

Saludos
Martin Rodriguez
VoIP Engineer Globant

El 29 de noviembre de 2010 11:13, Fernando Villares <fvil...@gmail.com> escribió:
no tendra el boton de DND habilitado?????

Fernando Villares

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Nov 29, 2010, 10:39:05 AM11/29/10
to aster...@googlegroups.com
acabo de confirmar con grandstream que en efecto el DND habilitado devuelve el sip code  486 correspondiente a Busy here...
o sea que es recontra valido que un dnd de ese error
saludos

Iñaki Baz Castillo

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Nov 29, 2010, 10:51:31 AM11/29/10
to aster...@googlegroups.com
El día 29 de noviembre de 2010 16:39, Fernando Villares
<fvil...@gmail.com> escribió:

> acabo de confirmar con grandstream que en efecto el DND habilitado devuelve
> el sip code  486 correspondiente a Busy here...
> o sea que es recontra valido que un dnd de ese error

Un error común es suponer que DND significa algo concreto y exacto,
cuando en realidad simplemente significa "Don't Disturb" y cada
teléfono lo implementa como quiere (algunos devuelven 480 "Not
Available", otros 486 "Busy Now", otros cualquier otro código de
rechazo como 403 "Forbidden" o 603 "Decline", etc).

Martin Rodriguez

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Nov 29, 2010, 11:18:33 AM11/29/10
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Si eso que dice Iñaki es verdad. Hay varios telefonos y dispositivos sip que te permiten cambiar el codigo de respuesta desde la GUI de gestion o lo traen implementado de otra manera.

Por eso es valido lo del DND y es valido que sea otra respuesta del Telefono. Igualmente como dice el LOG se ve claro quien manda el Busy. -- Got SIP response 486 "Busy Here" back from 192.168.1.122

Saludos
Martin Rodriguez
VoIP Engineer Globant
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