Falla llamadas salientes

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Falla llamadas salientes

RC_ANDRESS

Buen dia,

De la manera mas atenta solicito su ayuda ya que tengo un problema con las llamadas salientes tanto a numeros de 7 digitos como de 10 digitos.

Mi configuracion es:

Issabel 4 con asterisk 16.
Troncal SIP suministrada por CLARO
Sofpthone 3CX
Configuracion de la troncal

host=10.8.18.XXX
type=friend
disallow=all
allow=ulaw&alaw
qualify=yes
insecure=very
context=from-trunk

*Configuracion ruta saliente

He probado dejar vacio el campo CID, agregar 7 digitos, inclusive 10 digitos pero la falla persiste

Las llamadas entrantes no tienen ningun problema pero al momento de generar llamadas externas obtengo como resultado el auido you-dialed-wrong-number.gsm, las llamadas entre extensiones funcionan correctamente

-Si ejecuto un sip show peers puedo ver mis extensiones y la troncal OK

6005/6005 170.116.11.45 D No No A 5060 OK (17 ms)

6006/6006 170.116.11.10 D No No A 5360 OK (22 ms)

6007/6007 170.116.11.10 D No No A 5060 OK (20 ms)

6008/6008 170.116.11.10 D No No A 5260 OK (21 ms)

6010/6010 170.116.10.3 D Yes Yes A 56588 OK (106 ms)

CLARO 10.8.18.XXX Auto (No) No 5060 OK (26 ms)

TELEAMIGO 23.29.116.121 Yes Yes 5060 OK (93 ms)

-Si ejecuto un sip show registry obtengo este resultado

Host dnsmgr Username Refresh State Reg.Time
0 SIP registrations.

Cabe aclarar que tengo conexion hacia mi troncal y tengo respuesta de ping ok

-Si ejecuto un sip set debug on obtengo este resultado

<--- SIP read from UDP:170.116.10.3:56588 --->
ACK sip:97423470@170.116.10.5:5060 SIP/2.0
Via: SIP/2.0/UDP 170.116.10.3:56588;branch=z9hG4bK-d8754z-713c3734df13bd5a-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:6010@170.116.10.3:56588;rinstance=3fc565130d14f066>
To: <sip:97423470@170.116.10.5:5060>;tag=as7fc4f8a9
From: "6010"<sip:6010@170.116.10.5:5060>;tag=9b5a1f7a
Call-ID: ODE5NmJjM2QzMjNiZTdjM2Q1ZjFhY2E0MjEzYzFlNGY.
CSeq: 2 ACK
User-Agent: 3CXPhone 6.0.26523.0
Authorization: Digest username="6010",realm="asterisk",nonce="3944f6d5",uri="sip:97423470@170.116.10.5:5060",response="8c5fc847b6510595f55e7a3543cb3b46",algorithm=MD5
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---

<--- SIP read from UDP:170.116.10.3:56588 --->
BYE sip:97423470@170.116.10.5:5060 SIP/2.0
Via: SIP/2.0/UDP 170.116.10.3:56588;branch=z9hG4bK-d8754z-9c3e2830b040c530-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:6010@170.116.10.3:56588;rinstance=3fc565130d14f066>
To: <sip:97423470@170.116.10.5:5060>;tag=as7fc4f8a9
From: "6010"<sip:6010@170.116.10.5:5060>;tag=9b5a1f7a
Call-ID: ODE5NmJjM2QzMjNiZTdjM2Q1ZjFhY2E0MjEzYzFlNGY.
CSeq: 3 BYE
User-Agent: 3CXPhone 6.0.26523.0
Authorization: Digest username="6010",realm="asterisk",nonce="3944f6d5",uri="sip:97423470@170.116.10.5:5060",response="2ad0ade453cde8a45859b308fcf75b14",algorithm=MD5
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 170.116.10.3:56588 (NAT)
Scheduling destruction of SIP dialog 'ODE5NmJjM2QzMjNiZTdjM2Q1ZjFhY2E0MjEzYzFlNGY.' in 7040 ms (Method: BYE)

<--- Transmitting (NAT) to 170.116.10.3:56588 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 170.116.10.3:56588;branch=z9hG4bK-d8754z-9c3e2830b040c530-1---d8754z-;received=170.116.10.3;rport=56588
From: "6010"<sip:6010@170.116.10.5:5060>;tag=9b5a1f7a
To: <sip:97423470@170.116.10.5:5060>;tag=as7fc4f8a9
Call-ID: ODE5NmJjM2QzMjNiZTdjM2Q1ZjFhY2E0MjEzYzFlNGY.
CSeq: 3 BYE
Server: IPBX-2.11.0(16.16.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>

Si valido el log de asterisk se evidencia lo siguiente

[2021-07-29 13:32:37] VERBOSE[22623][C-000001fa] pbx.c: Executing [s@macro-dialout-trunk:23] NoOp("SIP/6010-0000028f", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 1") in new stack
[2021-07-29 13:32:37] VERBOSE[22623][C-000001fa] pbx.c: Executing [s@macro-dialout-trunk:24] GotoIf("SIP/6010-0000028f", "0?continue,1:s-CHANUNAVAIL,1") in new stack
[2021-07-29 13:32:37] VERBOSE[22623][C-000001fa] pbx.c: Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/6010-0000028f", "RC=1") in new stack
[2021-07-29 13:32:37] VERBOSE[22623][C-000001fa] pbx.c: Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/6010-0000028f", "1,1") in new stack
[2021-07-29 13:32:37] VERBOSE[22623][C-000001fa] pbx.c: Executing [1@macro-dialout-trunk:1] Goto("SIP/6010-0000028f", "continue,1") in new stack
[2021-07-29 13:32:37] VERBOSE[22623][C-000001fa] pbx.c: Executing [continue@macro-dialout-trunk:1] NoOp("SIP/6010-0000028f", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 1 - failing through to other trunks") in new stack
[2021-07-29 13:32:37] VERBOSE[22623][C-000001fa] pbx.c: Executing [continue@macro-dialout-trunk:2] Set("SIP/6010-0000028f", "CALLERID(number)=6010") in new stack
[2021-07-29 13:32:37] VERBOSE[22623][C-000001fa] pbx.c: Executing [97423470@from-internal:6] Macro("SIP/6010-0000028f", "outisbusy,") in new stack
[2021-07-29 13:32:37] VERBOSE[22623][C-000001fa] pbx.c: Executing [s@macro-outisbusy:1] Progress("SIP/6010-0000028f", "") in new stack
[2021-07-29 13:32:37] VERBOSE[22623][C-000001fa] pbx.c: Executing [s@macro-outisbusy:2] GotoIf("SIP/6010-0000028f", "0?emergency,1") in new stack
[2021-07-29 13:32:37] VERBOSE[22623][C-000001fa] pbx.c: Executing [s@macro-outisbusy:3] GotoIf("SIP/6010-0000028f", "0?intracompany,1") in new stack
[2021-07-29 13:32:37] VERBOSE[22623][C-000001fa] pbx.c: Executing [s@macro-outisbusy:4] GotoIf("SIP/6010-0000028f", "1?unallocated,1") in new stack
[2021-07-29 13:32:37] VERBOSE[22623][C-000001fa] pbx.c: Executing [unallocated@macro-outisbusy:1] Playback("SIP/6010-0000028f", "you-dialed-wrong-number") in new stack
[2021-07-29 13:32:38] VERBOSE[22623][C-000001fa] file.c: <SIP/6010-0000028f> Playing 'you-dialed-wrong-number.gsm' (language 'es')
[2021-07-29 13:32:39] VERBOSE[22623][C-000001fa] pbx.c: Executing [unallocated@macro-outisbusy:2] Congestion("SIP/6010-0000028f", "20") in new stack
[2021-07-29 13:32:40] VERBOSE[22623][C-000001fa] app_macro.c: Spawn extension (macro-outisbusy, unallocated, 2) exited non-zero on 'SIP/6010-0000028f' in macro 'outisbusy'
[2021-07-29 13:32:40] VERBOSE[22623][C-000001fa] pbx.c: Spawn extension (from-internal, 97423470, 6) exited non-zero on 'SIP/6010-0000028f'
[2021-07-29 13:32:40] VERBOSE[22623][C-000001fa] pbx.c: Executing [h@from-internal:1] Hangup("SIP/6010-0000028f", "") in new stack
[2021-07-29 13:32:40] VERBOSE[22623][C-000001fa] pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/6010-0000028f

Ya consulte con el proveedor de la troncal y no me da alguna solucion, espero logren ayudarme. Gracias.

Sippy

what are the settings in outbound routes?
I have learned to keep the dial plan in outbound routes only and nothing in the trunks. If you have sip trunks

Sippy

Also most sip providers make you dial 11 digits like this 1-555-555-5555 so to make 7 digit dialing you have to prepend a seven digit call with 1555 then NXXXXXX in the dial pattern of the outbound route if your local area code is 555. .

RC_ANDRESS

Hello,

I tried both ways, leaving the trunk without any dial plan and also indicating the DID provided by the provider, but I was unsuccessful.

In the outgoing route in the CID section I indicate the DID that the provider indicates to me, in the dialing pattern indicate that to generate calls you must enter 9 + XXXXXXX and the sequence of the trunks is indicated only my SIP trunk.

Also try to leave the dial plan without prefix 9 but I can't generate outgoing calls in any way

Sippy

RC_ANDRESS post your settings for outbound routes and clear any dial plans for trunks. What is your extensions digits. Also you can place a |. in an outbound route and it will let you dial anything. make sure dtmf setting is rfc2833

Sippy

that dial plan 9-XXXXXXX will never work with sip trunks you need to prepend it like this 1555- 9- NXXXXXX use your area code for the 555

RC_ANDRESS

Okay this is my SIP trunk data

Nombre de la línea troncal: CLARO
CallerID Saliente:
Opciones CID: Permitir Cualquier CID
Canales máximos:
Opciones Dial de Asterisk:
Continuar si da Ocupado:
Deshabilitar línea troncal:

Asistente de reglas de marcación: (seleccione uno)
Prefijo de marcación externa:

Opciones salientes

Nombre de la línea troncal: CLARO
host=10.8.18.218
type=friend
disallow=all
allow=ulaw&alaw
dtmfmode=rfc2833
qualify=yes
insecure=very
context=from-trunk

And this is the data of my outbound route

Nombre de la ruta: LOCAL
Route CID: 7431212
Override Extension:
Contraseña de la ruta:
Route Type:
Música en espera: default
Time Group: ---Permanent Route---
Route Position: ---No Change---

Additional Settings

Call Recording: Permitir
PIN Set: None

Dial Patterns that will use this Route

() + 9 | [ XXXXXXX /

Trunk Sequence for Matched Routes
0 CLARO

I do not agree that the 9 + XXXXXXX dial plan will never work since I have 4 other Issabel PBXs with SIP trunk and they work correctly with the 9 + XXXXXXX dial plan. Additional forget to mention that this failure has been presented 2 weeks ago, the server before that time was working correctly

Sippy

put |. in the dial pattern by itself and make test call try seven digits, 10 digits and 11 digits. If none go through they have something wrong on their end. ( ) ( ) (|.) is what I mean for the outgoing route Leave the trunk dial plan empty.