problema con las conferencias al ingresar el PIN

problema con las conferencias al ingresar el PIN

Posted Marzo 10th, 2010 by dataver86

Estoy trabajando con asterisk 1.6 y freepbx 2.5 y he tenido problemas al crear una conferencia pues cuando se marca el número de la conferencia ésta pide el PIN para ingresar pero al digitarlo se corta la conferencia o la llamada... anexo el log que genera asterisk para que alguien me pueda decir cual es el posible error:

[1;35;40mUsing CallerID "Servidor VoIP" <300>") in new stack
[Mar 17 03:57:16] VERBOSE[29115] pbx.c: -- Executing [1234@from-internal:2] Set("SIP/300-00000001", "MEETME_ROOMNUM=1234") in new stack
[Mar 17 03:57:16] VERBOSE[29115] pbx.c: -- Executing [1234@from-internal:3] Set("SIP/300-00000001", "MEETME_RECORDINGFILE=/var/spool/asterisk/monitor/meetme-conf-rec-1234-1268816236.1") in new stack
[Mar 17 03:57:16] VERBOSE[29115] pbx.c: -- Executing [1234@from-internal:4] GotoIf("SIP/300-00000001", "0?READPIN") in new stack
[Mar 17 03:57:16] VERBOSE[29115] pbx.c: -- Executing [1234@from-internal:5] Answer("SIP/300-00000001", "") in new stack
[Mar 17 03:57:16] VERBOSE[29115] pbx.c: -- Executing [1234@from-internal:6] Wait("SIP/300-00000001", "1") in new stack
[Mar 17 03:57:17] VERBOSE[29115] pbx.c: -- Executing [1234@from-internal:7] Set("SIP/300-00000001", "PINCOUNT=0") in new stack
[Mar 17 03:57:17] VERBOSE[29115] pbx.c: -- Executing [1234@from-internal:8] Read("SIP/300-00000001", "PIN,enter-conf-pin-number,,,,") in new stack
[Mar 17 03:57:17] VERBOSE[29115] file.c: -- Playing 'enter-conf-pin-number.gsm' (language 'es')
[Mar 17 03:57:33] VERBOSE[29115] app_read.c: -- User entered '12345'
[Mar 17 03:57:33] VERBOSE[29115] pbx.c: -- Executing [1234@from-internal:9] GotoIf("SIP/300-00000001", "1?USER") in new stack
[Mar 17 03:57:33] VERBOSE[29115] pbx.c: -- Goto (from-internal,1234,17)
[Mar 17 03:57:33] VERBOSE[29115] pbx.c: -- Executing [1234@from-internal:17] Set("SIP/300-00000001", "MEETME_OPTS=cIMsr") in new stack
[Mar 17 03:57:33] VERBOSE[29115] pbx.c: -- Executing [1234@from-internal:18] Goto("SIP/300-00000001", "STARTMEETME,1") in new stack
[Mar 17 03:57:33] VERBOSE[29115] pbx.c: -- Goto (from-internal,STARTMEETME,1)
[Mar 17 03:57:33] VERBOSE[29115] pbx.c: -- Executing [STARTMEETME@from-internal:1] MeetMe("SIP/300-00000001", "1234,cIMsr,12345") in new stack
[Mar 17 03:57:33] VERBOSE[29115] config.c: == Parsing '/etc/asterisk/meetme.conf': [Mar 17 03:57:33] VERBOSE[29115] config.c: == Found
[Mar 17 03:57:33] VERBOSE[29115] config.c: == Parsing '/etc/asterisk/meetme_additional.conf': [Mar 17 03:57:33] VERBOSE[29115] config.c: == Found
[Mar 17 03:57:33] WARNING[29115] app_meetme.c: Unable to open pseudo device
[Mar 17 03:57:33] VERBOSE[29115] pbx.c: == Spawn extension (from-internal, STARTMEETME, 1) exited non-zero on 'SIP/300-00000001'
[Mar 17 03:57:33] VERBOSE[29115] pbx.c: -- Executing [h@from-internal:1] Macro("SIP/300-00000001", "hangupcall") in new stack
[Mar 17 03:57:33] VERBOSE[29115] pbx.c: -- Executing [s@macro-hangupcall:1] GotoIf("SIP/300-00000001", "1?skiprg") in new stack
[Mar 17 03:57:33] VERBOSE[29115] pbx.c: -- Goto (macro-hangupcall,s,4)
[Mar 17 03:57:33] VERBOSE[29115] pbx.c: -- Executing [s@macro-hangupcall:4] GotoIf("SIP/300-00000001", "1?skipblkvm") in new stack
[Mar 17 03:57:33] VERBOSE[29115] pbx.c: -- Goto (macro-hangupcall,s,7)
[Mar 17 03:57:33] VERBOSE[29115] pbx.c: -- Executing [s@macro-hangupcall:7] GotoIf("SIP/300-00000001", "1?theend") in new stack
[Mar 17 03:57:33] VERBOSE[29115] pbx.c: -- Goto (macro-hangupcall,s,9)
[Mar 17 03:57:33] VERBOSE[29115] pbx.c: -- Executing [s@macro-hangupcall:9] Hangup("SIP/300-00000001", "") in new stack
[Mar 17 03:57:33] VERBOSE[29115] app_macro.c: == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/300-00000001' in macro 'hangupcall'
[Mar 17 03:57:33] VERBOSE[29115] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/300-00000001'

Solución...

Problemas con el Dahdi, por lo que fue necesito reinstalarlo y listo. (dede estar bien instalado para que sincronice el hardware...)

Posted by dataver86 on Jue, 2010-03-18 21:35
sigue el problema con las conferencias

creo las conferencias normalmente a través del freepbx, pero sin embargo al ejecutar el comando meetme en el CLI o entrar al asterisk info figura que no tiene conferencias activas (No active MeetMe conferences.) cúal será el problema, por qué no las crea?

Posted by dataver86 on Lun, 2010-03-15 15:51
Conferencias

Hola,
¿Viste en el log que sale a la hora de digitar el PIN?
Podría ser que no reconozca los digitos y corte, o alguna otra cosa.

El freePBx lo que hace es "automatizar" y por eso pide que no cambies los archivos. Debes tener unos archivos "custom" que puedes cambiar.

Saludos

Posted by Nestor on Mié, 2010-03-10 17:32