PROBLEMAS INCOMING ROUTERS

presidio

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#1
problerma con Incoming Route no puedo recibir llamadas desde la PSTN. lo tengo todo configurado como dise el manual elastix_without_tears la vercind e elastix, estoy usando es 1.2.1-4

TENGO UNA TERJETA FXO DE UN SOLO PUERTO Y UN SIP TRUNK.

EN LA CONFIGURACION DE ADD (INCOMING ROUTERS) DEJO TODO EN BLANCO SOLO SELECCIONO LA EXT (Ejm 101) QUE QUEIRO QUE ME RESIVA LA LLAMADA Y NO ENTRAN (ANY DID/ANY CID)LAS LLAMADAS,PERO SI PUEDO REALIZAR LLAMADAS DESDE LA PBX EXLATIX.

CON EL SIP TRUNK EN(INCOMING ROUTERS)LE PONGO EN DID NUMBREY LA EXTQUE QUIERO QUE ME RESIVA LA LLAMADA(EXT 101)Y TAMPOCO RECIVE LA LLAMADA PERO SI PUEDO REALIZAR LLAMADAS

SI ALGUIN ME PUEDE AYUDAR CON ESTO POR FAVOR
 

nachogomez

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#2
Try this,

Open the Asterisk CLI (as root on console type "asterisk -rcvvvvvvvvv"), call to your system analog line first (from a cell phone or another line) and copy and paste the output here to see what's happening when a call comes into your system.
 

presidio

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#3
-- Executing [98095801111@from-internal:1] Macro("SIP/3000-09d6ce38", "user-callerid|SKIPTTL|") in new stack
-- Executing [s@macro-user-callerid:1] NoOp("SIP/3000-09d6ce38", "user-callerid: device 3000") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/3000-09d6ce38", "AMPUSER=3000") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/3000-09d6ce38", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("SIP/3000-09d6ce38", "1|Set|REALCALLERIDNUM=3000") in new stack
-- Executing [s@macro-user-callerid:5] NoOp("SIP/3000-09d6ce38", "REALCALLERIDNUM is 3000") in new stack
-- Executing [s@macro-user-callerid:6] Set("SIP/3000-09d6ce38", "AMPUSER=3000") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/3000-09d6ce38", "AMPUSERCIDNAME=Pablo A. Luciano") in new stack
-- Executing [s@macro-user-callerid:8] GotoIf("SIP/3000-09d6ce38", "0?report") in new stack
-- Executing [s@macro-user-callerid:9] Set("SIP/3000-09d6ce38", "AMPUSERCID=3000") in new stack
-- Executing [s@macro-user-callerid:10] Set("SIP/3000-09d6ce38", "CALLERID(all)="Pablo A. Luciano" <3000>") in new stack
-- Executing [s@macro-user-callerid:11] Set("SIP/3000-09d6ce38", "REALCALLERIDNUM=3000") in new stack
-- Executing [s@macro-user-callerid:12] ExecIf("SIP/3000-09d6ce38", "0|Set|CHANNEL(language)=") in new stack
-- Executing [s@macro-user-callerid:13] NoOp("SIP/3000-09d6ce38", "TTL: ARG1: SKIPTTL") in new stack
-- Executing [s@macro-user-callerid:14] GotoIf("SIP/3000-09d6ce38", "1?continue") in new stack
-- Goto (macro-user-callerid,s,23)
-- Executing [s@macro-user-callerid:23] NoOp("SIP/3000-09d6ce38", "Using CallerID "Pablo A. Luciano" <3000>") in new stack
-- Executing [98095801111@from-internal:2] Set("SIP/3000-09d6ce38", "_NODEST=") in new stack
-- Executing [98095801111@from-internal:3] Macro("SIP/3000-09d6ce38", "record-enable|3000|OUT|") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/3000-09d6ce38", "0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] AGI("SIP/3000-09d6ce38", "recordingcheck|20081105-092731|1225895251.1") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20081105-092731|1225895251.1: Outbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing [s@macro-record-enable:5] NoOp("SIP/3000-09d6ce38", "No recording needed") in new stack
-- Executing [98095801111@from-internal:4] Macro("SIP/3000-09d6ce38", "dialout-trunk|1|8095801111||") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/3000-09d6ce38", "DIAL_TRUNK=1") in new stack
-- Executing [s@macro-dialout-trunk:2] ExecIf("SIP/3000-09d6ce38", "0|Authenticate|") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/3000-09d6ce38", "0?disabletrunk|1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("SIP/3000-09d6ce38", "DIAL_NUMBER=8095801111") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/3000-09d6ce38", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/3000-09d6ce38", "GROUP()=OUT_1") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/3000-09d6ce38", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/3000-09d6ce38", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:10] Set("SIP/3000-09d6ce38", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/3000-09d6ce38", "outbound-callerid|1") in new stack
-- Executing [s@macro-outbound-callerid:1] GotoIf("SIP/3000-09d6ce38", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing [s@macro-outbound-callerid:3] NoOp("SIP/3000-09d6ce38", "REALCALLERIDNUM is 3000") in new stack
-- Executing [s@macro-outbound-callerid:4] GotoIf("SIP/3000-09d6ce38", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
-- Executing [s@macro-outbound-callerid:9] Set("SIP/3000-09d6ce38", "USEROUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:10] Set("SIP/3000-09d6ce38", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:11] Set("SIP/3000-09d6ce38", "TRUNKOUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:12] GotoIf("SIP/3000-09d6ce38", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing [s@macro-outbound-callerid:16] GotoIf("SIP/3000-09d6ce38", "1?usercid") in new stack
-- Goto (macro-outbound-callerid,s,18)
-- Executing [s@macro-outbound-callerid:18] GotoIf("SIP/3000-09d6ce38", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing [s@macro-outbound-callerid:22] NoOp("SIP/3000-09d6ce38", "CallerID set to "Pablo A. Luciano" <3000>") in new stack
-- Executing [s@macro-dialout-trunk:12] AGI("SIP/3000-09d6ce38", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
-- AGI Script fixlocalprefix completed, returning 0
-- Executing [s@macro-dialout-trunk:13] Set("SIP/3000-09d6ce38", "OUTNUM=8095801111") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/3000-09d6ce38", "custom=ZAP/g0") in new stack
-- Executing [s@macro-dialout-trunk:15] GotoIf("SIP/3000-09d6ce38", "1?gocall") in new stack
-- Goto (macro-dialout-trunk,s,17)
-- Executing [s@macro-dialout-trunk:17] Macro("SIP/3000-09d6ce38", "dialout-trunk-predial-hook|") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/3000-09d6ce38", "0?bypass|1") in new stack
-- Executing [s@macro-dialout-trunk:19] GotoIf("SIP/3000-09d6ce38", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:20] Dial("SIP/3000-09d6ce38", "ZAP/g0/8095801111|300|") in new stack
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [s@macro-dialout-trunk:21] Goto("SIP/3000-09d6ce38", "s-CONGESTION|1") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing [s-CONGESTION@macro-dialout-trunk:1] GotoIf("SIP/3000-09d6ce38", "1?noreport") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,3)
-- Executing [s-CONGESTION@macro-dialout-trunk:3] NoOp("SIP/3000-09d6ce38", "TRUNK Dial failed due to CONGESTION - failing through to other trunks") in new stack
-- Executing [98095801111@from-internal:5] Macro("SIP/3000-09d6ce38", "outisbusy|") in new stack
-- Executing [s@macro-outisbusy:1] Playback("SIP/3000-09d6ce38", "all-circuits-busy-now|noanswer") in new stack
-- <SIP/3000-09d6ce38> Playing 'all-circuits-busy-now' (language 'en')
-- Executing [s@macro-outisbusy:2] Playback("SIP/3000-09d6ce38", "pls-try-call-later|noanswer") in new stack
-- <SIP/3000-09d6ce38> Playing 'pls-try-call-later' (language 'en')
== Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/3000-09d6ce38' in macro 'outisbusy'
== Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/3000-09d6ce38'


eso es realizando una llamada Via el zaptel Osea una tarjeta A100P FXO PCI CARD

no sale n las llamadas ni entran. esto paso despuesd eaverla actualizado a la vercion 1.3-2
 

nachogomez

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#4
Veo que estas realizando la llamada desde un teléfono SIP interno utilizando tu trunk zaptel como ruta de salida.

Que ocurre si configuras tu ruta de salida por el trunk SIP???

Me ayudaría mucho saber como es tu configuración de estas rutas de salida.

Revisa tambien si despues de la actualización Elastix te detecto la tarjeta TDM (en la opción de detección de hardware o "Hardware Detection" ).

Si puedes manda por aquí algunos screenshots de la configuración de tu lineas SIP, Zaptel y de las rutas de salida (trata de que no tengan mucha resolución pero que se puedan leer las letras) a ver si encuentro algo.

Todavía tienes problemas para recibir las llamadas???


Nacho
 

presidio

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#5
ya lo resolvi muchas gracias por la ayuda Prestada, el problema lo tenia mi provedor

gracias
 

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