Ayuda con problemas al configurar Trunk SIP por IP

Asterisk Asterisk Support
#1

Saludos

tengo inocnvenientes con un troncal SIP que autentica por IP de Claro Dominicana
el proveedor solo me da el numero de Telefono piloto o prueba
la direccion del HOST la direccion del Router y la direccion de mi central.

Nota: tengo una T1 analoga funcionando correctamente y el servidor posee 2 tarjetas de Red en la eth0 tengo la configuracion de la red local
eth1 tengo la configuracion que se conecta al router cisco

en mi central agregue una troncal SIP con la siguiente configuracion

type=peer
qualify=yes
nat=yes
insecure=invite,port
host= 172.19.38.90(No. del Host que me brindo el proveedor de servicio)
dtmfmode=rfc2833
context=default
canreinvite=no

agregue mi regla de marcado

cuando hago en el asterisk -r luego sip show peers
Claroout/192.172.0.2 172.19.38.90 N 5060 UNREACHABLE

y no me salen las llamadas ni me entran las llamadas…
no se logra conectar.

porfavor ayudenmeeeeeeeeeeeee, Gracias anticipadas

#2

[root@Elastix ~]# asterisk -r
Asterisk 1.6.2.13, Copyright © 1999 - 2010 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.

Connected to Asterisk 1.6.2.13 currently running on Elastix (pid = 23482)
Verbosity is at least 3
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Executing [88092270399@from-internal:1] Macro(“SIP/2345-00000090”, “user-callerid,SKIPTTL,”) in new stack
– Executing [s@macro-user-callerid:1] Set(“SIP/2345-00000090”, “AMPUSER=2345”) in new stack
– Executing [s@macro-user-callerid:2] GotoIf(“SIP/2345-00000090”, “0?report”) in new stack
– Executing [s@macro-user-callerid:3] ExecIf(“SIP/2345-00000090”, “1?Set(REALCALLERIDNUM=2345)”) in new stack
– Executing [s@macro-user-callerid:4] Set(“SIP/2345-00000090”, “AMPUSER=2345”) in new stack
– Executing [s@macro-user-callerid:5] Set(“SIP/2345-00000090”, “AMPUSERCIDNAME=Jonathan Martinez”) in new stack
– Executing [s@macro-user-callerid:6] GotoIf(“SIP/2345-00000090”, “0?report”) in new stack
– Executing [s@macro-user-callerid:7] Set(“SIP/2345-00000090”, “AMPUSERCID=2345”) in new stack
– Executing [s@macro-user-callerid:8] Set(“SIP/2345-00000090”, “CALLERID(all)=“Jonathan Martinez” <2345>”) in new stack
– Executing [s@macro-user-callerid:9] ExecIf(“SIP/2345-00000090”, “0?Set(CHANNEL(language)=)”) in new stack
– Executing [s@macro-user-callerid:10] GotoIf(“SIP/2345-00000090”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,19)
– Executing [s@macro-user-callerid:19] NoOp(“SIP/2345-00000090”, “Using CallerID “Jonathan Martinez” <2345>”) in new stack
– Executing [88092270399@from-internal:2] Set(“SIP/2345-00000090”, “_NODEST=”) in new stack
– Executing [88092270399@from-internal:3] Macro(“SIP/2345-00000090”, “record-enable,2345,OUT,”) in new stack
– Executing [s@macro-record-enable:1] GotoIf(“SIP/2345-00000090”, “1?check”) in new stack
– Goto (macro-record-enable,s,4)
– Executing [s@macro-record-enable:4] ExecIf(“SIP/2345-00000090”, “0?MacroExit()”) in new stack
– Executing [s@macro-record-enable:5] GotoIf(“SIP/2345-00000090”, “0?Group:OUT”) in new stack
– Goto (macro-record-enable,s,15)
– Executing [s@macro-record-enable:15] GotoIf(“SIP/2345-00000090”, “0?IN”) in new stack
– Executing [s@macro-record-enable:16] ExecIf(“SIP/2345-00000090”, “1?MacroExit()”) in new stack
– Executing [88092270399@from-internal:4] Macro(“SIP/2345-00000090”, “dialout-trunk,2,8092270399,”) in new stack
– Executing [s@macro-dialout-trunk:1] Set(“SIP/2345-00000090”, “DIAL_TRUNK=2”) in new stack
– Executing [s@macro-dialout-trunk:2] GosubIf(“SIP/2345-00000090”, “0?sub-pincheck,s,1”) in new stack
– Executing [s@macro-dialout-trunk:3] GotoIf(“SIP/2345-00000090”, “0?disabletrunk,1”) in new stack
– Executing [s@macro-dialout-trunk:4] Set(“SIP/2345-00000090”, “DIAL_NUMBER=8092270399”) in new stack
– Executing [s@macro-dialout-trunk:5] Set(“SIP/2345-00000090”, “DIAL_TRUNK_OPTIONS=tr”) in new stack
– Executing [s@macro-dialout-trunk:6] Set(“SIP/2345-00000090”, “OUTBOUND_GROUP=OUT_2”) in new stack
– Executing [s@macro-dialout-trunk:7] GotoIf(“SIP/2345-00000090”, “0?nomax”) in new stack
– Executing [s@macro-dialout-trunk:8] GotoIf(“SIP/2345-00000090”, “0?chanfull”) in new stack
– Executing [s@macro-dialout-trunk:9] GotoIf(“SIP/2345-00000090”, “0?skipoutcid”) in new stack
– Executing [s@macro-dialout-trunk:10] Set(“SIP/2345-00000090”, “DIAL_TRUNK_OPTIONS=r”) in new stack
– Executing [s@macro-dialout-trunk:11] Macro(“SIP/2345-00000090”, “outbound-callerid,2”) in new stack
– Executing [s@macro-outbound-callerid:1] ExecIf(“SIP/2345-00000090”, “0?Set(CALLERPRES()=)”) in new stack
– Executing [s@macro-outbound-callerid:2] ExecIf(“SIP/2345-00000090”, “0?Set(REALCALLERIDNUM=2345)”) in new stack
– Executing [s@macro-outbound-callerid:3] GotoIf(“SIP/2345-00000090”, “1?normcid”) in new stack
– Goto (macro-outbound-callerid,s,6)
– Executing [s@macro-outbound-callerid:6] Set(“SIP/2345-00000090”, “USEROUTCID=”) in new stack
– Executing [s@macro-outbound-callerid:7] Set(“SIP/2345-00000090”, “EMERGENCYCID=”) in new stack
– Executing [s@macro-outbound-callerid:8] Set(“SIP/2345-00000090”, “TRUNKOUTCID=8092892952”) in new stack
– Executing [s@macro-outbound-callerid:9] GotoIf(“SIP/2345-00000090”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,12)
– Executing [s@macro-outbound-callerid:12] ExecIf(“SIP/2345-00000090”, “1?Set(CALLERID(all)=8092892952)”) in new stack
– Executing [s@macro-outbound-callerid:13] ExecIf(“SIP/2345-00000090”, “0?Set(CALLERID(all)=)”) in new stack
– Executing [s@macro-outbound-callerid:14] ExecIf(“SIP/2345-00000090”, “0?Set(CALLERID(all)=)”) in new stack
– Executing [s@macro-outbound-callerid:15] ExecIf(“SIP/2345-00000090”, “0?Set(CALLERPRES()=prohib_passed_screen)”) in new stack
– Executing [s@macro-dialout-trunk:12] ExecIf(“SIP/2345-00000090”, “0?AGI(fixlocalprefix)”) in new stack
– Executing [s@macro-dialout-trunk:13] Set(“SIP/2345-00000090”, “OUTNUM=8092270399”) in new stack
– Executing [s@macro-dialout-trunk:14] Set(“SIP/2345-00000090”, “custom=SIP/Claroout”) in new stack
– Executing [s@macro-dialout-trunk:15] ExecIf(“SIP/2345-00000090”, “0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^)r)”) in new stack
– Executing [s@macro-dialout-trunk:16] Macro(“SIP/2345-00000090”, “dialout-trunk-predial-hook,”) in new stack
– Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit(“SIP/2345-00000090”, “”) in new stack
– Executing [s@macro-dialout-trunk:17] GotoIf(“SIP/2345-00000090”, “0?bypass,1”) in new stack
– Executing [s@macro-dialout-trunk:18] GotoIf(“SIP/2345-00000090”, “0?customtrunk”) in new stack
– Executing [s@macro-dialout-trunk:19] Dial(“SIP/2345-00000090”, “SIP/Claroout/8092270399,300,r”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [s@macro-dialout-trunk:20] NoOp(“SIP/2345-00000090”, “Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 20”) in new stack
– Executing [s@macro-dialout-trunk:21] Goto(“SIP/2345-00000090”, “s-CHANUNAVAIL,1”) in new stack
– Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
– Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set(“SIP/2345-00000090”, “RC=20”) in new stack
– Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto(“SIP/2345-00000090”, “20,1”) in new stack
– Goto (macro-dialout-trunk,20,1)
– Executing [20@macro-dialout-trunk:1] Goto(“SIP/2345-00000090”, “continue,1”) in new stack
– Goto (macro-dialout-trunk,continue,1)
– Executing [continue@macro-dialout-trunk:1] GotoIf(“SIP/2345-00000090”, “1?noreport”) in new stack
– Goto (macro-dialout-trunk,continue,3)
– Executing [continue@macro-dialout-trunk:3] NoOp(“SIP/2345-00000090”, “TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 20 - failing through to other trunks”) in new stack
– Executing [continue@macro-dialout-trunk:4] Set(“SIP/2345-00000090”, “CALLERID(number)=2345”) in new stack
– Executing [88092270399@from-internal:5] Macro(“SIP/2345-00000090”, “outisbusy,”) in new stack
– Executing [s@macro-outisbusy:1] Progress(“SIP/2345-00000090”, “”) in new stack
– Executing [s@macro-outisbusy:2] GotoIf(“SIP/2345-00000090”, “0?emergency,1”) in new stack
– Executing [s@macro-outisbusy:3] GotoIf(“SIP/2345-00000090”, “0?intracompany,1”) in new stack
– Executing [s@macro-outisbusy:4] Playback(“SIP/2345-00000090”, “all-circuits-busy-now&pls-try-call-later, noanswer”) in new stack
– <SIP/2345-00000090> Playing ‘all-circuits-busy-now.gsm’ (language ‘en’)
== Spawn extension (macro-outisbusy, s, 4) exited non-zero on ‘SIP/2345-00000090’ in macro ‘outisbusy’
== Spawn extension (from-internal, 88092270399, 5) exited non-zero on ‘SIP/2345-00000090’
– Executing [h@from-internal:1] Macro(“SIP/2345-00000090”, “hangupcall”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“SIP/2345-00000090”, “1?noautomon”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [s@macro-hangupcall:3] NoOp(“SIP/2345-00000090”, “TOUCH_MONITOR_OUTPUT=”) in new stack
– Executing [s@macro-hangupcall:4] GotoIf(“SIP/2345-00000090”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,7)
– Executing [s@macro-hangupcall:7] GotoIf(“SIP/2345-00000090”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,10)
– Executing [s@macro-hangupcall:10] GotoIf(“SIP/2345-00000090”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,12)
– Executing [s@macro-hangupcall:12] Hangup(“SIP/2345-00000090”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 12) exited non-zero on ‘SIP/2345-00000090’ in macro ‘hangupcall’
== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/2345-00000090’

#3

Hola, para obtener más ayuda en este foro debes poner tus mensajes en Inglés.

Dicho eso revisa que tu server pueda al menos hacer ping al server remoto(esperando que no los tenga bloqueados). Las llamadas fallan porque todo indica que no tienes contacto con ese server. checha como construiste el peer.

#4

You are using a version of Asterisk that is over a year past end of life.

You have the over-used nat=yes, but you don’t have any of the settings actually needed if you are inside NAT. In that case, you need a t least one of externip, externhost or stunaddr.

Also you have insecure=port,invite, but you have no secret. Few people need port, and the purpose of invite is avoid your trying t authenticate the ITSP using the secret they use to authenticate you. remotesecret is a better way of doing the latter on the latest versions of Asterisk, but not in the obsolete version you are using.

canreinvite is deprecated. Use directmedia. I think this was available in 1.6.2.

The most likely reason for your problem is failure to announce your public address when you are inside NAT. A SIP trace would be needed to confirm this. Howeveer, the exact diagnostic you have is that you have qualify=yes, but you are getting no replies to OPTION requests, so there could also be a bug in the ITSPs handling of options (they should be accepted or rejected, but not ignored).

#5

Saludos,

Lograste configurar tu sip trunking de claro. Estoy en la misma situación que tu…

Gracias/

#6

[quote=“Luis538”]Saludos,

Lograste configurar tu sip trunking de claro. Estoy en la misma situación que tu…

Gracias/[/quote]

Configurar un SIP Trunk de claro es bastante facil, en mi blog tengo la informacion tanto para el de claro como el de Tricom y los problemas mas comunes son derivados de que no especifican la ruta adecuada para contactar al Host de Claro o Tricom