Llamadas exten remotas se cortan en 20segundos

**navaismo** · 12-18-2013, 10:48 AM

El pastebin que proporcionas no tiene información útil, deshabilita el debug del cli que solo genera datos innecesarios. Lo que necesitamos es el sip debug de una llamada completa no solo el final.

**Allan Porras** · 12-18-2013, 11:36 AM

El sip debug de una llamada completa?? No sé cómo generarlo. He logrado obtener esto justo en el momento de la llamada:

Executing [83226044@from-internal:1] Macro("SIP/500-00000091", "user-callerid,SKIPTTL,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/500-00000091", "AMPUSER=500") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/500-00000091", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/500-00000091", "1?Set(REALCALLERIDNUM=500)") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/500-00000091", "AMPUSER=500") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/500-00000091", "AMPUSERCIDNAME=500") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/500-00000091", "0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/500-00000091", "AMPUSERCID=500") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/500-00000091", "CALLERID(all)="500" <500>") in new stack
-- Executing [s@macro-user-callerid:9] ExecIf("SIP/500-00000091", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [s@macro-user-callerid:10] GotoIf("SIP/500-00000091", "1?continue") in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing [s@macro-user-callerid:19] Set("SIP/500-00000091", "CALLERID(number)=500") in new stack
-- Executing [s@macro-user-callerid:20] Set("SIP/500-00000091", "CALLERID(name)=500") in new stack
-- Executing [s@macro-user-callerid:21] NoOp("SIP/500-00000091", "Using CallerID "500" <500>") in new stack
-- Executing [83226044@from-internal:2] NoOp("SIP/500-00000091", "Calling Out Route: nacional") in new stack
-- Executing [83226044@from-internal:3] Set("SIP/500-00000091", "MOHCLASS=default") in new stack
-- Executing [83226044@from-internal:4] Set("SIP/500-00000091", "_NODEST=") in new stack
-- Executing [83226044@from-internal:5] Macro("SIP/500-00000091", "record-enable,500,OUT,") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/500-00000091", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] ExecIf("SIP/500-00000091", "0?MacroExit()") in new stack
-- Executing [s@macro-record-enable:5] GotoIf("SIP/500-00000091", "0?Group:OUT") in new stack
-- Goto (macro-record-enable,s,15)
-- Executing [s@macro-record-enable:15] GotoIf("SIP/500-00000091", "0?IN") in new stack
-- Executing [s@macro-record-enable:16] ExecIf("SIP/500-00000091", "1?MacroExit()") in new stack
-- Executing [83226044@from-internal:6] Macro("SIP/500-00000091", "dialout-trunk,3,83226044,") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/500-00000091", "DIAL_TRUNK=3") in new stack
-- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/500-00000091", "0?sub-pincheck,s,1") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/500-00000091", "0?disabletrunk,1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("SIP/500-00000091", "DIAL_NUMBER=83226044") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/500-00000091", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/500-00000091", "OUTBOUND_GROUP=OUT_3") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/500-00000091", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/500-00000091", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:10] Set("SIP/500-00000091", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/500-00000091", "outbound-callerid,3") in new stack
-- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/500-00000091", "0?Set(CALLERPRES()=)") in new stack
-- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/500-00000091", "0?Set(REALCALLERIDNUM=500)") in new stack
-- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/500-00000091", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,6)
-- Executing [s@macro-outbound-callerid:6] Set("SIP/500-00000091", "USEROUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:7] Set("SIP/500-00000091", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:8] Set("SIP/500-00000091", "TRUNKOUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/500-00000091", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,12)
-- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/500-00000091", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/500-00000091", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/500-00000091", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/500-00000091", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
-- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/500-00000091", "0?sub-flp-3,s,1") in new stack
-- Executing [s@macro-dialout-trunk:13] Set("SIP/500-00000091", "OUTNUM=83226044") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/500-00000091", "custom=SIP/gateway") in new stack
-- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/500-00000091", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)) ") in new stack
-- Executing [s@macro-dialout-trunk:16] Macro("SIP/500-00000091", "dialout-trunk-predial-hook,") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/500-00000091", "") in new stack
-- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/500-00000091", "0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/500-00000091", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:19] Dial("SIP/500-00000091", "SIP/gateway/83226044,300,") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/gateway/83226044
ippbx*CLI>
-- SIP/gateway-00000092 is making progress passing it to SIP/500-00000091
-- SIP/gateway-00000092 answered SIP/500-00000091
-- Locally bridging SIP/500-00000091 and SIP/gateway-00000092
-- Executing [h@macro-dialout-trunk:1] Macro("SIP/500-00000091", "hangupcall,") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/500-00000091", "1?endmixmoncheck") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] NoOp("SIP/500-00000091", "End of MIXMON check") in new stack
-- Executing [s@macro-hangupcall:10] GotoIf("SIP/500-00000091", "1?nomeetmemon") in new stack
-- Goto (macro-hangupcall,s,28)
-- Executing [s@macro-hangupcall:28] NoOp("SIP/500-00000091", "End of MEETME check") in new stack
-- Executing [s@macro-hangupcall:29] GotoIf("SIP/500-00000091", "1?noautomon") in new stack
-- Goto (macro-hangupcall,s,34)
-- Executing [s@macro-hangupcall:34] NoOp("SIP/500-00000091", "TOUCH_MONITOR_OUTPUT=") in new stack
-- Executing [s@macro-hangupcall:35] GotoIf("SIP/500-00000091", "1?noautomon2") in new stack
-- Goto (macro-hangupcall,s,41)
-- Executing [s@macro-hangupcall:41] NoOp("SIP/500-00000091", "MONITOR_FILENAME=") in new stack
-- Executing [s@macro-hangupcall:42] GotoIf("SIP/500-00000091", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,45)
-- Executing [s@macro-hangupcall:45] GotoIf("SIP/500-00000091", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,48)
-- Executing [s@macro-hangupcall:48] GotoIf("SIP/500-00000091", "1?theend") in new stack
-- Goto (macro-hangupcall,s,50)
-- Executing [s@macro-hangupcall:50] AGI("SIP/500-00000091", "hangup.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.agi
-- <SIP/500-00000091>AGI Script hangup.agi completed, returning 0
-- Executing [s@macro-hangupcall:51] Hangup("SIP/500-00000091", "") in new stack
== Spawn extension (macro-hangupcall, s, 51) exited non-zero on 'SIP/500-00000091' in macro 'hangupcall'
== Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/500-00000091'
== Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/500-00000091' in macro 'dialout-trunk'
== Spawn extension (from-internal, 83226044, 6) exited non-zero on 'SIP/500-00000091'

**navaismo** · 12-18-2013, 11:44 AM

Con el comando: sip set debug on

**Allan Porras** · 12-18-2013, 12:23 PM

Ahora si. Aquí tengo toda la traza completa de la llamada. Tuve que pegarlo en pastebin nuevamente: http://pastebin.com/gh34E69f

**navaismo** · 12-18-2013, 12:35 PM

Tu dispositivo en la dirección 190.113.109.189 no esta repondiendo a las peticiones de Asterisk, las retransmisiones comienzan en la línea 246 y terminan en la línea 413 del pastebin, esto generalmente es problema de NAT o de Ruteo, el sip debug se ve sano con las direcciones públicas correctas etc, por lo que podrías verifican en ambos routers que no existan opciones del tipo SIP ALG o SPI activadas.

**Allan Porras** · 12-18-2013, 01:01 PM

Gracias por responder.

En la sección ALG Service de mi router ubicado en 190.113.109.189 he DESACTIVADO: FTP, TFTP, IPSec, PPTP y L2TP. Además que DESACTIVADO las configuraciones UPnP, porque había leído en un foro que es recomendable quitarlo.

Tengo el mismo problema. No logré la solución con esto.

**navaismo** · 12-18-2013, 02:06 PM

¿Y del lado del server no tienes esas opciones activadas? Por cierto debes buscar alguna otra opción en el router remoto con respecto a SIP. ¿Que modelo de router usas?

**Allan Porras** · 12-18-2013, 02:29 PM

Del lado del server tengo un modem/router marca Pirelli 6332AP y no hay ninguna opción como las que usted me indica. Del lado remoto (donde estoy ahora), tengo un router Nexxt.

Me estoy volviendo loco y no encuentro la solución. En el firewall tengo abierto todo el tráfico entre el lado server y el lado remoto.

**navaismo** · 12-18-2013, 04:48 PM

Realiza un trazado del lado remoto con wireshark para ver si están llegando esas retransmisiones de asterisk y a la par obtén el sip debug desde asterisk, así podrás/podremos ver que pasa en realidad con esos paquetes.

**Allan Porras** · 12-18-2013, 08:28 PM

He colocado canreinvite=very en las extensiones remotas, y me funciona perfecto. Ya no se corta. Solución válida o perjudica seguridad?

**navaismo** · 12-18-2013, 09:11 PM

No me suena ese valor Para canreinvite/directmedia, no será insecure? Canreinvite es usado para dirigir el rtp y los invites: http://www.voip-info.org/wiki/view/A...ip+canreinvite

insecure=very sirve para permitir a los hosts no reautenticarse como lo hacen en un register o en un invite.

Llamadas exten remotas se cortan en 20segundos

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