Problemas con llamadas salientes

Discussion in 'Elastix 2.x' started by jaystb, Sep 16, 2009.

  1. jaystb

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    Saludos a toda la comunidad.

    Tengo un problema con las llamadas salientes. Me dice que están las líneas ocupadas y que lo intente más tarde.

    Con las llamadas entrantes no tengo ningún problema.

    Las troncales son SIP.

    En la ruta saliente he probado la siguiente configuración de "Dial Patterns":
    X
    XXXXXXXXX
    .

    En la troncal de salida, la configuración es la siguiente:
    Dial rules: Está en blanco
    Outgoing settings:
    context=from-pstn
    canreinvite=no
    host=serversip.com
    insecure=invite
    port=5060
    qualify=yes
    secret=*******
    type=friend
    username=********


    Os agradezco nuevamente vuestras sugerencias y colaboración.
     
  2. telecomtechnician

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    Hola

    El problema parece ser que el troncal sip esta mal configurado.

    1) De donde vienen o van las lineas SIP? (vienen de un proveedor de telefonía IP, o quieres enlazar un gateway o puerta de enlace a elastix)

    2) Configuraste correctamente la ruta de llamadas salientes y el troncal?

    Espero tus comentarios

    David Medina
     
  3. jaystb

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    La línea es de un proveedor de telefonía IP.

    En la configuración de la ruta de llamadas salientes y la troncal probé lo que ya indiqué anteriormente, pero no funciona así.

    La configuración no tiene problemas para llamadas entrantes, pero parece estar mal para las salientes.
     
  4. jaystb

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    He probado varias configuraciones, pero no hay forma.

    ¿Alguien sabe cómo solucionarlo?

    Recibo todo OK, pero no consigo salir. Siempre dice que lo intente más tarde!!!
     
  5. ramoncio

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    Con
    Code:
    asterisk -rx "sip show peers" 
    
    desde la consola de linux aparece el peer registrado?

    Prueba con esto desde la consola de linux (ojo, todo en una línea):
    Code:
    asterisk -rx "core set verbose 99" && asterisk -rx sip set debug peer nombredetupeer" && tail -f /var/log/asterisk/full > sipdebug.txt
    
    sustituyendo nombredetupeer por el nombre que te aparece con sip show peers, e intenta una llamada saliente.

    Cuando acabes, puedes salir con CTRL+C, y echale un vistazo al debug en sipdebug.txt
    Ahi deberías encontrar el problema.

    Luego recuerda desactivar el debug para que no se te llene el disco duro (también en una línea):
    Code:
    asterisk -rx "sip set debug off" && asterisk -rx "core set verbose 3"
    
     
  6. jaystb

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    He hecho lo que me comentas.

    El peer me aparece registrado. Cómo puedo acceder al sipdebug.txt para ver el problema???
     
  7. ramoncio

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    En el directorio donde ejecutaste el script se ha debido generar un archivo sipdebug.txt
    Puedes abrirlo con cualquier editor de texto: nano, pico, vi, etc., si usas linux o notepad ++, pspad o incluso el bloc de notas (si usas linux acostumbrate a no usarlo JAMAS, rompe todo lo que abre), si usas Windows.
     
  8. jaystb

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    Imagino que este archivo debe estar en la equipo de Elastix, ¿verdad?

    Cómo puedo acceder a él, con samba o a través de la consola??
     
  9. netsfk

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    creo que esto esta no tan complejo perdón si escribo cosas muy básicas en las cuales digas eso ya lo realice y no queda no es con el afán de molestar.

    1.- una vez que has creado tu troncal ya asignaste una ruta de salida para usarla???

    2.- tu ruta de salida que datos has llenado ej.

    name route. SIP
    route password
    Dial Patterns 9|. ; esto es que al descolgar marcas 9 mas lo que quieras 1234455....
    Trunk Sequence <-------------en esta parte tienes q seleccionar la troncal creaste


    ahora bien si ya has hecho esto puedes hacer un pequeño test en desde la interface web ve a tools y donde dice Asterisk-Cli pon lo siguiente

    sip show peers

    esto de dara la información de todas tus troncales sip o bien puedes poner

    iax2 show peer {nombre de tu troncal} ; va sin los corchetes


    si en lo que te arroje dichos comandos ves todo en OK siginifica que has registrado correctamente las troncales ahora intenta sacar la llamada con ese dial plan 9|.


    si no funciona tendremos que ver que esta pasando para esto te recomiendo uses putty es un cliente ssh para windows (aunq yo prefiero linux) ahi teclearas la dir ip de tu elastix 192.168.X.X te pedira te logues das root y tu password que pusiste durante la instalacion, una vez dentro das asterisk -vvvvvvvvvvvvvvr el promp cambiara por algo similar a esto

    elastix*CLI>


    ahora intenta nuevamente hacer la llamada 9|. veras que en tu consola aparecera algo similar a esto

    elastix*CLI> Macro("SIP/11-082cbf30", "exten-vm|novm|333") in new stack
    -- Executing [s@macro-exten-vm:1] Macro("SIP/11-082cbf30", "user-callerid") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/11-082cbf30", "AMPUSER=11") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/11-082cbf30", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/11-082cbf30", "1|Set|REALCALLERIDNUM=11") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/11-082cbf30", "AMPUSER=11") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/11-082cbf30", "AMPUSERCIDNAME=Soporte Tecnico1") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/11-082cbf30", "0?report") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/11-082cbf30", "AMPUSERCID=11") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/11-082cbf30", "CALLERID(all)="Soporte Tecnico1" <11>") in new stack
    -- Executing [s@macro-user-callerid:9] Set("SIP/11-082cbf30", "REALCALLERIDNUM=11") in new stack
    -- Executing [s@macro-user-callerid:10] ExecIf("SIP/11-082cbf30", "1|Set|CHANNEL(language)=es") in new stack
    -- Executing [s@macro-user-callerid:11] GotoIf("SIP/11-082cbf30", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:12] Set("SIP/11-082cbf30", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:13] GotoIf("SIP/11-082cbf30", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,20)
    -- Executing [s@macro-user-callerid:20] NoOp("SIP/11-082cbf30", "Using CallerID "Soporte Tecnico1" <11>") in new stack
    -- Executing [s@macro-exten-vm:2] Set("SIP/11-082cbf30", "RingGroupMethod=none") in new stack
    -- Executing [s@macro-exten-vm:3] Set("SIP/11-082cbf30", "VMBOX=novm") in new stack
    -- Executing [s@macro-exten-vm:4] Set("SIP/11-082cbf30", "EXTTOCALL=333") in new stack
    -- Executing [s@macro-exten-vm:5] Set("SIP/11-082cbf30", "CFUEXT=") in new stack
    -- Executing [s@macro-exten-vm:6] Set("SIP/11-082cbf30", "CFBEXT=") in new stack
    -- Executing [s@macro-exten-vm:7] Set("SIP/11-082cbf30", "RT=""") in new stack
    -- Executing [s@macro-exten-vm:8] Macro("SIP/11-082cbf30", "record-enable|333|IN") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/11-082cbf30", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] AGI("SIP/11-082cbf30", "recordingcheck|20090923-163524|1253741724.0") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
    recordingcheck|20090923-163524|1253741724.0: Inbound recording not enabled
    -- AGI Script recordingcheck completed, returning 0
    -- Executing [s@macro-record-enable:5] MacroExit("SIP/11-082cbf30", "") in new stack
    -- Executing [s@macro-exten-vm:9] Macro("SIP/11-082cbf30", "dial||Ttr|333") in new stack
    -- Executing [s@macro-dial:1] GotoIf("SIP/11-082cbf30", "1?dial") in new stack
    -- Goto (macro-dial,s,3)
    -- Executing [s@macro-dial:3] AGI("SIP/11-082cbf30", "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 'Soporte Tecnico1' number is '11'
    dialparties.agi: USE_CONFIRMATION: 'FALSE'
    dialparties.agi: RINGGROUP_INDEX: ''
    dialparties.agi: Methodology of ring is 'none'
    -- dialparties.agi: Added extension 333 to extension map
    -- dialparties.agi: Extension 333 cf is disabled
    -- dialparties.agi: Extension 333 do not disturb is disabled
    > dialparties.agi: extnum 333 has: cw: 0; hascfb: 0 [] hascfu: 0 []
    > dialparties.agi: ExtensionState: 4
    dialparties.agi: Extension 333 has ExtensionState: 4
    -- dialparties.agi: Checking CW and CFB status for extension 333
    -- dialparties.agi: dbset CALLTRACE/333 to 11
    -- dialparties.agi: Filtered ARG3: 333
    == Manager 'admin' logged off from 127.0.0.1
    -- AGI Script dialparties.agi completed, returning 0
    -- Executing [s@macro-dial:7] Dial("SIP/11-082cbf30", "SIP/333||Ttr") in new stack
    == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [s@macro-dial:8] Set("SIP/11-082cbf30", "DIALSTATUS=CHANUNAVAIL") in new stack
    -- Executing [s@macro-dial:9] GosubIf("SIP/11-082cbf30", "0?CHANUNAVAIL|1") in new stack
    -- Executing [s@macro-exten-vm:10] GotoIf("SIP/11-082cbf30", "0?exit|return") in new stack
    -- Executing [s@macro-exten-vm:11] Set("SIP/11-082cbf30", "SV_DIALSTATUS=CHANUNAVAIL") in new stack
    -- Executing [s@macro-exten-vm:12] GosubIf("SIP/11-082cbf30", "0?docfu|1") in new stack
    -- Executing [s@macro-exten-vm:13] GosubIf("SIP/11-082cbf30", "0?docfb|1") in new stack
    -- Executing [s@macro-exten-vm:14] Set("SIP/11-082cbf30", "DIALSTATUS=CHANUNAVAIL") in new stack
    -- Executing [s@macro-exten-vm:15] NoOp("SIP/11-082cbf30", "Voicemail is novm") in new stack
    -- Executing [s@macro-exten-vm:16] GotoIf("SIP/11-082cbf30", "1?s-CHANUNAVAIL|1") in new stack
    -- Goto (macro-exten-vm,s-CHANUNAVAIL,1)
    -- Executing [s-CHANUNAVAIL@macro-exten-vm:1] NoOp("SIP/11-082cbf30", "IVR_RETVM: IVR_CONTEXT: ") in new stack
    -- Executing [s-CHANUNAVAIL@macro-exten-vm:2] GotoIf("SIP/11-082cbf30", "0?exit|1") in new stack
    -- Executing [s-CHANUNAVAIL@macro-exten-vm:3] PlayTones("SIP/11-082cbf30", "congestion") in new stack
    -- Executing [s-CHANUNAVAIL@macro-exten-vm:4] Congestion("SIP/11-082cbf30", "10") in new stack
    == Spawn extension (macro-exten-vm, s-CHANUNAVAIL, 4) exited non-zero on 'SIP/11-082cbf30' in macro 'exten-vm'
    == Spawn extension (from-internal, 333, 1) exited non-zero on 'SIP/11-082cbf30'
    -- Executing [h@from-internal:1] Macro("SIP/11-082cbf30", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] ResetCDR("SIP/11-082cbf30", "w") in new stack
    -- Executing [s@macro-hangupcall:2] NoCDR("SIP/11-082cbf30", "") in new stack
    -- Executing [s@macro-hangupcall:3] GotoIf("SIP/11-082cbf30", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,6)
    -- Executing [s@macro-hangupcall:6] GotoIf("SIP/11-082cbf30", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] GotoIf("SIP/11-082cbf30", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,11)
    -- Executing [s@macro-hangupcall:11] Hangup("SIP/11-082cbf30", "") in new stack
    == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/11-082cbf30' in macro 'hangupcall'
    == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/11-082cbf30'



    esta es la info que necesitamos para ver que es lo q pasa y como poder ayudarte
     
  10. jaystb

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    Esto es lo que me aparece con el sip show peers:

    Name/username Host Dyn Nat ACL Port Status
    TSP/968970776 86.109.97.3 5060 OK (81 ms)
    TSP2/8780028792 193.22.119.20 5060 OK (95 ms)
    GH/8780029768 193.22.119.20 5060 OK (96 ms)
    F&B/8780029296 193.22.119.20 5060 OK (96 ms)
    103/103 192.168.2.101 D N 6942 OK (54 ms)
    100/100 192.168.2.100 D N 60254 OK (104 ms)
    6 sip peers [Monitored: 6 online, 0 offline Unmonitored: 0 online, 0 offline]

    == Parsing '/etc/asterisk/asterisk.conf': Found
    == Parsing '/etc/asterisk/extconfig.conf': Found
    Connected to Asterisk 1.4.24 currently running on tusecrepersonal (pid = 2857)
    Verbosity was 3 and is now 15
    -- Executing [0918838422@from-internal:1] Macro("SIP/103-0833c890", "user-ca llerid|SKIPTTL|") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/103-0833c890", "AMPUSER=103" ) in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/103-0833c890", "0?report" ) in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/103-0833c890", "1|Set|REA LCALLERIDNUM=103") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/103-0833c890", "AMPUSER=103" ) in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/103-0833c890", "AMPUSERCIDNA ME=Test") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/103-0833c890", "0?report" ) in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/103-0833c890", "AMPUSERCID=1 03") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/103-0833c890", "CALLERID(all )="Test" <103>") in new stack
    -- Executing [s@macro-user-callerid:9] Set("SIP/103-0833c890", "REALCALLERID NUM=103") in new stack
    -- Executing [s@macro-user-callerid:10] ExecIf("SIP/103-0833c890", "0|Set|CH ANNEL(language)=") in new stack
    -- Executing [s@macro-user-callerid:11] GotoIf("SIP/103-0833c890", "1?contin ue") in new stack
    -- Goto (macro-user-callerid,s,20)
    -- Executing [s@macro-user-callerid:20] NoOp("SIP/103-0833c890", "Using Call erID "Test" <103>") in new stack
    -- Executing [0918838422@from-internal:2] Set("SIP/103-0833c890", "_NODEST=" ) in new stack
    -- Executing [0918838422@from-internal:3] Macro("SIP/103-0833c890", "record- enable|103|OUT|") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/103-0833c890", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] AGI("SIP/103-0833c890", "recordingche ck|20090924-101225|1253779945.45") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
    recordingcheck|20090924-101225|1253779945.45: Outbound recording not enabled
    -- AGI Script recordingcheck completed, returning 0
    -- Executing [s@macro-record-enable:5] MacroExit("SIP/103-0833c890", "") in new stack
    -- Executing [0918838422@from-internal:4] Macro("SIP/103-0833c890", "dialout -trunk|5|968838422||") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/103-0833c890", "DIAL_TRUNK=5 ") in new stack
    -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/103-0833c890", "0?sub-pi ncheck|s|1") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/103-0833c890", "0?disable trunk|1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/103-0833c890", "DIAL_NUMBER= 968838422") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/103-0833c890", "DIAL_TRUNK_O PTIONS=trTW") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/103-0833c890", "OUTBOUND_GRO UP=OUT_5") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/103-0833c890", "1?nomax") in new stack
    -- Goto (macro-dialout-trunk,s,9)
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/103-0833c890", "0?skipout cid") in new stack
    -- Executing [s@macro-dialout-trunk:10] Set("SIP/103-0833c890", "DIAL_TRUNK_ OPTIONS=tTuV") in new stack
    -- Executing [s@macro-dialout-trunk:11] Macro("SIP/103-0833c890", "outbound- callerid|5") in new stack
    -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/103-0833c890", "0|Set CallerPres|") in new stack
    -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/103-0833c890", "0|Set |REALCALLERIDNUM=103") in new stack
    -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/103-0833c890", "1?nor mcid") in new stack
    -- Goto (macro-outbound-callerid,s,6)
    -- Executing [s@macro-outbound-callerid:6] Set("SIP/103-0833c890", "USEROUTC ID=") in new stack
    -- Executing [s@macro-outbound-callerid:7] Set("SIP/103-0833c890", "EMERGENC YCID=") in new stack
    -- Executing [s@macro-outbound-callerid:8] Set("SIP/103-0833c890", "TRUNKOUT CID="GH" <916261582>") in new stack
    -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/103-0833c890", "1?tru nkcid") in new stack
    -- Goto (macro-outbound-callerid,s,12)
    -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/103-0833c890", "1|Se t|CALLERID(all)=GH <916261582>") in new stack
    -- Executing [s@macro-outbound-callerid:13] GotoIf("SIP/103-0833c890", "1?ex it") in new stack
    -- Goto (macro-outbound-callerid,s,11)
    -- Executing [s@macro-outbound-callerid:11] MacroExit("SIP/103-0833c890", "" ) in new stack
    -- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/103-0833c890", "1|AGI|fi xlocalprefix") 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/103-0833c890", "OUTNUM=9688 38422") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("SIP/103-0833c890", "custom=SIP/ HotelGH") in new stack
    -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/103-0833c890", "0|Set|DI AL_TRUNK_OPTIONS=M(setmusic^)tTuV") in new stack
    -- Executing [s@macro-dialout-trunk:16] Macro("SIP/103-0833c890", "dialout-t runk-predial-hook|") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/103-0833c 890", "") in new stack
    -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/103-0833c890", "0?bypass |1") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/103-0833c890", "0?custom trunk") in new stack
    -- Executing [s@macro-dialout-trunk:19] Dial("SIP/103-0833c890", "SIP/HotelG H/968838422|300|tTuV") in new stack
    -- Called GH/918838422
    -- SIP/GH-08303d18 is circuit-busy
    == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [s@macro-dialout-trunk:20] Goto("SIP/103-0833c890", "s-CONGESTI ON|1") in new stack
    -- Goto (macro-dialout-trunk,s-CONGESTION,1)
    -- Executing [s-CONGESTION@macro-dialout-trunk:1] GotoIf("SIP/103-0833c890", "1?noreport") in new stack
    -- Goto (macro-dialout-trunk,s-CONGESTION,3)
    -- Executing [s-CONGESTION@macro-dialout-trunk:3] NoOp("SIP/103-0833c890", " TRUNK Dial failed due to CONGESTION - failing through to other trunks") in new s tack
    -- Executing [0918838422@from-internal:5] Macro("SIP/103-0833c890", "outisbu sy|") in new stack
    -- Executing [s@macro-outisbusy:1] Playback("SIP/103-0833c890", "all-circuit s-busy-now|noanswer") in new stack
    -- <SIP/103-0833c890> Playing 'all-circuits-busy-now' (language 'en')
    -- Executing [s@macro-outisbusy:2] Playback("SIP/103-0833c890", "pls-try-cal l-later|noanswer") in new stack
    -- <SIP/103-0833c890> Playing 'pls-try-call-later' (language 'en')
    -- Executing [s@macro-outisbusy:3] Macro("SIP/103-0833c890", "hangupcall") i n new stack
    -- Executing [s@macro-hangupcall:1] ResetCDR("SIP/103-0833c890", "w") in new stack
    -- Executing [s@macro-hangupcall:2] NoCDR("SIP/103-0833c890", "") in new sta ck
    -- Executing [s@macro-hangupcall:3] GotoIf("SIP/103-0833c890", "1?skiprg") i n new stack
    -- Goto (macro-hangupcall,s,6)
    -- Executing [s@macro-hangupcall:6] GotoIf("SIP/103-0833c890", "1?skipblkvm" ) in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] GotoIf("SIP/103-0833c890", "1?theend") i n new stack
    -- Goto (macro-hangupcall,s,11)
    -- Executing [s@macro-hangupcall:11] Hangup("SIP/103-0833c890", "") in new s tack
    == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/103-0833c 890' in macro 'hangupcall'
    == Spawn extension (macro-outisbusy, s, 3) exited non-zero on 'SIP/103-0833c89 0' in macro 'outisbusy'
    == Spawn extension (from-internal, 0918838422, 5) exited non-zero on 'SIP/103- 0833c890'
    -- Executing [h@from-internal:1] Macro("SIP/103-0833c890", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] ResetCDR("SIP/103-0833c890", "w") in new stack
    -- Executing [s@macro-hangupcall:2] NoCDR("SIP/103-0833c890", "") in new sta ck
    -- Executing [s@macro-hangupcall:3] GotoIf("SIP/103-0833c890", "1?skiprg") i n new stack
    -- Goto (macro-hangupcall,s,6)
    -- Executing [s@macro-hangupcall:6] GotoIf("SIP/103-0833c890", "1?skipblkvm" ) in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] GotoIf("SIP/103-0833c890", "1?theend") i n new stack
    -- Goto (macro-hangupcall,s,11)
    -- Executing [s@macro-hangupcall:11] Hangup("SIP/103-0833c890", "") in new s tack
    == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/103-0833c 890' in macro 'hangupcall'
    == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/103-0833c890'
    tusecrepersonal*CLI> exit
     
  11. ElasMex

    Joined:
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    Hola jaystb

    Te recomiendo que coloques mejor un archivo de texto.

    Vamos a revisar los que dices:

    1. ¿Puedes decir con que proveedor de IP tienes contratadas las líneas?

    2. ¿Qué versión de Elastix tienes ?

    3. ¿Hay forma de que coloques una imagen de la configuración de "rutas de salida"?

    4. ¿Qué avances tienes ?

    Saludos
     
  12. jaystb

    Joined:
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    Hola ElasMex,

    1. Mi proveedor ip para las líneas es neogen (www.neogen.es)

    2. La versión de Elastix es la 1.5.2 de 31 de marzo de 2009.

    3. Esta es la configuración de la ruta de salida
    [​IMG]


    4. No sé qué avances tengo.


    Te agradezco tu colaboración. Si necesitas más información pídemela, a ver si me ayudáis a solucionar esto porque no entiendo nada de nada.
     
  13. ElasMex

    Joined:
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    jaystb

    ¿En la configuración de la troncal tiene este dato?

    host=i2next.com.mx
     
  14. jaystb

    Joined:
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    Si, en la configuración de la troncal, en los peers details, tengo lo siguiente:
    context=from-pstn
    canreinvite=no
    host=serversip.com
    insecure=invite
    port=5060
    qualify=yes
    secret=********
    type=friend
    username=********
     
  15. jaystb

    Joined:
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    Estoy probando con otro proveedor de línea IP (www.telsome.es)

    Con este proveedor si puedo realizar llamadas, pero no con el anterior. He puesto la misma configuración en ambas troncales.

    Con el proveedor con el que puedo hacer llamadas:
    username=*********
    type=friend
    secret=*********
    qualify=yes
    port=5060
    insecure=invite
    host=voip3.telsome.com
    context=from-pstn
    canreinvite=no

    Con el que no puedo hacer llamadas:
    username=*********
    type=friend
    secret=*********
    qualify=yes
    port=5060
    insecure=invite
    host=serversip.com
    context=from-pstn
    canreinvite=no
     
  16. jaystb

    Joined:
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    Cómo el problema lo tengo con un proveedor, pero no con el otro, he estado comprobando cosas para que estén igual, pero no consigo nada. No sé si debo cambiar la configuración en la troncal o en la configuración general, en la que en la opción "Asterisk Outbound Dial command options:" la he dejado en blanco.

    Cada vez que intento una llamada me dice que las líneas están ocupadas.
     
  17. jaystb

    Joined:
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    En el FreePBX Notices he visto el siguiente error:

    /var/lib/asterisk/bin/module_admin listonline (255)
    (cron_manager.EXECFAIL)

    No sé si tendrá algo que ver con mi problema.
     
  18. zeoneo

    Joined:
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    mira, he visto en algunos casos que cuando el proveedor SIP ocupa servidores como trixbox, estas tonteras probocan un conflicto simple solo por el orden de estos comenados...

    context=from-pstn
    canreinvite=no
    host=serversip.com
    insecure=invite
    port=5060
    qualify=yes
    secret=********
    type=friend
    username=********


    Yo te recomiendo ordenarlos de la siguente manera....


    context=from-pstn
    host=serversip.com
    username=********
    secret=********
    port=5060
    type=friend
    canreinvite=no
    insecure=yes
    qualify=yes


    Cuentame como te va...


    Nos vemos
     
  19. jaystb

    Joined:
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    Gracias Renato,

    Yo también pienso que podría ser lo que tú dices. He probado ordenándolo así, pero me sigue diciendo que todas las líneas están ocupadas. Todavía no salen llamadas.
     
  20. jpablodeleon

    Joined:
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    Hola soy Juan Pablo, tengo un grandstream ht503 y me funciona muy bien en mi elastix.

    elastix 2.3.0 5

    Eh adquirido otro grandstream gxw4108 y no me entran llamadas y cuando quiero sacar una llamada me dice "todas las líneas están ocupadas".

    Me podrías ayudar?.

    Tengo Iusacell Enlace.
     

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