Llamadas entrantes

**navaismo** · 05-12-2013, 12:29 PM

Si esa ultima grabación que comentas te la pone la compañia Telefónica hay un problema fuera del pbx. Por otro lado muestranos lo que arroja el cli de asterisk con:

Sip show peers
Sip show registry
Y cuando haces una llamada entrante

**Marco Zink** · 05-13-2013, 10:06 AM

Hola Javier,

Si no me equivoco Alestra te pide mandar las llamadas a 12 dígitos para llamadas locales y 13 para celulares, es decir:

(52) ZXXXXXXXXX para llamadas a landlines.
(521) ZXXXXXXXXX para llamadas a celulares.

¿Cómo estás enviando las llamadas al carrier?

Saludos

**javierbueno** · 05-19-2013, 10:44 PM

Gracias, solo era problema del context, que ha quedado solucionado, pero ahora tengo otro problema, que me entran dos llamadas o y hago llamadas, y el PBX esta fuera de línea, es decir que me dice "TODAS LAS LINEAS ESTAN OCUPADAS" y cuando una persona llama, se escucha ocupado, y no se que hacer, marque a Alestra, pero me comentaron que tenía que pagar por la póliza y ellos no se hacen cargo

**navaismo** · 05-20-2013, 12:23 AM

Necesitas saber:

-¿Cuantos canales tiene tu linea SIP?
-¿Con que servicios cuenta?
Y mostrarnos la configuración de tu pbx o el debug de las llamadas fallidas para poder ayudarte.

**hector** · 05-20-2013, 12:29 PM

Asegurate tambien que tienes conectividad de red entre tu Asterisk y Alestra, y que los paquetes SIP están por lo menos llegando a su destino y te están contestando.

**javierbueno** · 05-20-2013, 10:44 PM

Saludos a ambos, gracias por atender mis dudas, el los canales de mi línea SIP son 5, que son los que Alestra te dan al contratar la línea, y cuenta con los servicios de llamada local y a celuar y lista para montarse en asterisk, si hay una red entre mi asterisk y alestra, el problema de aquí es que a cada rato tengo que estar reiniciando mi conmutador para que pueda funcionar, y el otro problema es que no se si sea en el context porque puse el de from pstn, saludos y espero sus respuestas.

**navaismo** · 05-21-2013, 12:43 AM

Necesitamos ver el debug del cli cuando pasan estos problemas para poderte ayudar.

**javierbueno** · 05-21-2013, 05:36 PM

Hola, buenas tardes, me puedes decir como lo hago, ya que a algunas cosas de asterisk no le entiendo, si me puedes decir por favor te lo agradecería mil. Gracias.

**navaismo** · 05-21-2013, 06:49 PM

Conectate via ssh a tu server ejecuta el comando:

Código:

asterisk -rvvvvvdd

una vez dentro del cli de asterisk ejecuta:

Código:

sip set debug on

Y copia y pega aquí TODO lo que salga cuando fallan las llamadas.

**javierbueno** · 05-22-2013, 08:52 AM

Hola Buenas tardes, aquí esta todo lo que me salio, saludos.

[root@localhost ~]# asterisk -rvvvvvdd
Asterisk 1.8.11.0, Copyright (C) 1999 - 2012 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
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.
================================================== =======================
== Parsing '/etc/asterisk/asterisk.conf': Parsing /etc/asterisk/asterisk.conf
== Found
Seeding global EID '08:00:27:da:0b:ab' from 'eth0' using 'siocgifhwaddr'
== Parsing '/etc/asterisk/extconfig.conf': Parsing /etc/asterisk/extconfig.conf
== Found
Connected to Asterisk 1.8.11.0 currently running on localhost (pid = 2566)
Verbosity is at least 5
Core debug is at least 2
localhost*CLI> sip set debug on
SIP Debugging re-enabled
-- Remote UNIX connection
-- Remote UNIX connection disconnected
localhost*CLI>

**hector** · 05-22-2013, 09:39 AM

ok cuando tengas eso en la pantalla:

localhost*CLI> sip set debug on
SIP Debugging re-enabled
-- Remote UNIX connection
-- Remote UNIX connection disconnected
localhost*CLI>

Haz la llamada... y te van a salir otras cosas.... copialas y pegalas por aqui

**javierbueno** · 05-22-2013, 11:54 AM

<------------->
Really destroying SIP dialog 'NDZjNjAxZWU5YTU5NDM3MmM4YzIxMTYzMzA0NDVhNGM.' Method: REGISTER

<--- SIP read from UDP:192.168.1.67:62102 --->

<------------->
Reliably Transmitting (NAT) to 192.168.1.67:62102:
OPTIONS sip:1000@192.168.1.67:62102;rinstance=5347eadfc3c8 0891 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.68:5060;branch=z9hG4bK1e203aad;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.68>;tag=as692e178a
To: <sip:1000@192.168.1.67:62102;rinstance=5347eadfc3c 80891>
Contact: <sip:Unknown@192.168.1.68:5060>
Call-ID: 4f21f9f13ffcf08f6f30e5bf1330a37b@192.168.1.68:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.8.1(1.8.11.0)
Date: Wed, 22 May 2013 17:52:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

Esto fue lo que ahora me salio
---

<--- SIP read from UDP:192.168.1.67:62102 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.68:5060;branch=z9hG4bK1e203aad;rport=506 0
Contact: <sip:192.168.1.67:62102>
To: <sip:1000@192.168.1.67:62102;rinstance=5347eadfc3c 80891>;tag=aa1db15c
From: "Unknown"<sip:Unknown@192.168.1.68>;tag=as692e 178a
Call-ID: 4f21f9f13ffcf08f6f30e5bf1330a37b@192.168.1.68:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Supported: replaces
Allow-Events: presence, message-summary, tunnel-info
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Really destroying SIP dialog '4f21f9f13ffcf08f6f30e5bf1330a37b@192.168.1.68:506 0' Method: OPTIONS

<--- SIP read from UDP:192.168.1.67:62102 --->

<------------->
localhost*CLI>

**javierbueno** · 05-22-2013, 11:59 AM

y con el debug off me sale esto

-- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/1000-00000009", "0?Set(REALCALLERIDNUM=1000)") in new stack
-- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/1000-00000009", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,6)
-- Executing [s@macro-outbound-callerid:6] Set("SIP/1000-00000009", "USEROUTCID=1000") in new stack
-- Executing [s@macro-outbound-callerid:7] Set("SIP/1000-00000009", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:8] Set("SIP/1000-00000009", "TRUNKOUTCID=528180006573") in new stack
-- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/1000-00000009", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,12)
-- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/1000-00000009", "1?Set(CALLERID(all)=528180006573)") in new stack
-- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/1000-00000009", "1?Set(CALLERID(all)=1000)") in new stack
-- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/1000-00000009", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/1000-00000009", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
-- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/1000-00000009", "0?sub-flp-2,s,1") in new stack
-- Executing [s@macro-dialout-trunk:13] Set("SIP/1000-00000009", "OUTNUM=528182440085") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/1000-00000009", "custom=SIP/ALESTRA") in new stack
-- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/1000-00000009", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)) ") in new stack
-- Executing [s@macro-dialout-trunk:16] Macro("SIP/1000-00000009", "dialout-trunk-predial-hook,") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/1000-00000009", "") in new stack
-- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/1000-00000009", "0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/1000-00000009", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:19] Dial("SIP/1000-00000009", "SIP/ALESTRA/528182440085,300,") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/ALESTRA/528182440085
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [s@macro-dialout-trunk:20] NoOp("SIP/1000-00000009", "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 34") in new stack
-- Executing [s@macro-dialout-trunk:21] Goto("SIP/1000-00000009", "s-CONGESTION,1") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing [s-CONGESTION@macro-dialout-trunk:1] Set("SIP/1000-00000009", "RC=34") in new stack
-- Executing [s-CONGESTION@macro-dialout-trunk:2] Goto("SIP/1000-00000009", "34,1") in new stack
-- Goto (macro-dialout-trunk,34,1)
-- Executing [34@macro-dialout-trunk:1] Goto("SIP/1000-00000009", "continue,1") in new stack
-- Goto (macro-dialout-trunk,continue,1)
-- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/1000-00000009", "1?noreport") in new stack
-- Goto (macro-dialout-trunk,continue,3)
-- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/1000-00000009", "TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34 - failing through to other trunks") in new stack
-- Executing [continue@macro-dialout-trunk:4] Set("SIP/1000-00000009", "CALLERID(number)=1000") in new stack
-- Executing [82440085@from-internal:7] Macro("SIP/1000-00000009", "outisbusy,") in new stack
-- Executing [s@macro-outisbusy:1] Progress("SIP/1000-00000009", "") in new stack
-- Executing [s@macro-outisbusy:2] GotoIf("SIP/1000-00000009", "0?emergency,1") in new stack
-- Executing [s@macro-outisbusy:3] GotoIf("SIP/1000-00000009", "0?intracompany,1") in new stack
-- Executing [s@macro-outisbusy:4] Playback("SIP/1000-00000009", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
-- <SIP/1000-00000009> Playing 'all-circuits-busy-now.gsm' (language 'en')
-- <SIP/1000-00000009> Playing 'pls-try-call-later.gsm' (language 'en')
-- Executing [s@macro-outisbusy:5] Congestion("SIP/1000-00000009", "20") in new stack
== Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'SIP/1000-00000009' in macro 'outisbusy'
== Spawn extension (from-internal, 82440085, 7) exited non-zero on 'SIP/1000-00000009'
-- Executing [h@from-internal:1] Macro("SIP/1000-00000009", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/1000-00000009", "1?endmixmoncheck") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] NoOp("SIP/1000-00000009", "End of MIXMON check") in new stack
-- Executing [s@macro-hangupcall:10] GotoIf("SIP/1000-00000009", "1?nomeetmemon") in new stack
-- Goto (macro-hangupcall,s,15)
-- Executing [s@macro-hangupcall:15] NoOp("SIP/1000-00000009", "MEETME_RECORDINGFILE=") in new stack
-- Executing [s@macro-hangupcall:16] GotoIf("SIP/1000-00000009", "1?noautomon") in new stack
-- Goto (macro-hangupcall,s,18)
-- Executing [s@macro-hangupcall:18] NoOp("SIP/1000-00000009", "TOUCH_MONITOR_OUTPUT=") in new stack
-- Executing [s@macro-hangupcall:19] GotoIf("SIP/1000-00000009", "1?noautomon2") in new stack
-- Goto (macro-hangupcall,s,25)
-- Executing [s@macro-hangupcall:25] NoOp("SIP/1000-00000009", "MONITOR_FILENAME=") in new stack
-- Executing [s@macro-hangupcall:26] GotoIf("SIP/1000-00000009", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,29)
-- Executing [s@macro-hangupcall:29] GotoIf("SIP/1000-00000009", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,32)
-- Executing [s@macro-hangupcall:32] GotoIf("SIP/1000-00000009", "1?theend") in new stack
-- Goto (macro-hangupcall,s,34)
-- Executing [s@macro-hangupcall:34] Hangup("SIP/1000-00000009", "") in new stack
== Spawn extension (macro-hangupcall, s, 34) exited non-zero on 'SIP/1000-00000009' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/1000-00000009'
-- Remote UNIX connection
-- Remote UNIX connection disconnected
localhost*CLI>

**hector** · 05-22-2013, 12:08 PM

en el debug que pones no se muestra una llamada... yo creo que lo estas perdiendo con el scroll... arranca el CLI de asterisk de la siguiente manera: asterisk -rvvvvvdd | tee logfile.txt y cuando capture una llamada que falla te sales y mandas el archivo logfile.txt

Llamadas entrantes

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