error en llamada saliente Everyone is busy/congested at this time (1:0/0/1)

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rochasystem

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Jan 4, 2016, 3:53:29 PM1/4/16
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Buenas tardes tengo una planta en asterisk 11 y una troncal sip con el proveedor UNE (Colomabia), la cuestion es que estoy intentando sacar llamadas por esta troncal pero no he podido y me sale el siguiente mensage:

 == Using SIP RTP CoS mark 5
    -- Executing [3632542@ADMIN:1] Set("SIP/90-00000012", "CALLERID(num)=53100970") in new stack
    -- Executing [3632542@ADMIN:2] Dial("SIP/90-00000012", "SIP/UNE1/3222222,90,Ttr") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/UNE1/3632542
    -- Got SIP response 484 "Address Incomplete" back from 172.17.179.166:5060
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [3632542@ADMIN:3] Congestion("SIP/90-00000012", "") in new stack
  == Spawn extension (ADMIN, 3632542, 3) exited non-zero on 'SIP/90-00000012'


mi troncal esta asi:


register=>5XXX...@XXX.XX.XXX.XXX/5XXXXXXXX

; UNE SIP


[UNE1]
username=*******
fromuser=*******
type=friend
host=XXX.XX.XX.XXX
secret=**********
qualify=yes
directmedia=no
disallow=all
;allow=gsm
allow=ulaw
allow=alaw
context=from-pstn

mi contexto:

[SALIENTES_UNE1]
exten=>_XXXXXXX,1,set(CALLERID(num)=5XXXXXXX)
same=>n,Dial(SIP/UNE1/${EXTEN},90,Ttr)
same=>n,Congestion()
same=>n,Hangup()


si ejecuto el sip debug ip sale esto:

 == Using SIP RTP CoS mark 5
    -- Executing [XXXXXXX@ADMIN:1] Set("SIP/90-00000010", "CALLERID(num)=5XXXXXXX") in new stack
    -- Executing [XXXXXXX@ADMIN:2] Dial("SIP/90-00000010", "SIP/UNE1/xxxxxxx,90,Ttr") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 13236
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to xxx.xx.xxx.xxx:5060:
INVITE sip:xxx...@xxx.xx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP xxx.xx.xx.xxx:5060;branch=z9hG4bK5806dbc8;rport
Max-Forwards: 70
From: "Servidores" <sip:ri...@xxx.xx.xx.xxx>;tag=as240328e2
To: <sip:xxx...@xxx.xx.xxx.xx>
Contact: <sip:xx...@xxx.xx.xx.xx:5060>
Call-ID: 438956893746...@xxx.xx.xx.xxx:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.14.1
Date: Mon, 04 Jan 2016 20:29:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "Servidores" <sip:xxx...@xxx.xx.xx.xxx>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 288361789 288361789 IN IP4 xxx.xx.xx.xxx
s=Asterisk PBX 11.14.1
c=IN IP4 xxx.xx.xx.xxx
t=0 0
m=audio 13236 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/UNE1/xxxxxxx

<--- SIP read from UDP:xxx.xx.xxx.xxx:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP xxx.xx.xx.xxx:5060;received=xxx.xx.xx.xxx;branch=z9hG4bK5806dbc8;rport=5060
From: "Servidores" <sip:xx...@xxx.xx.xx.xxx>;tag=as240328e2
To: <sip:xxx...@xxx.xx.xxx.xxx>
Call-ID: 438956893746...@xxx.xx.xx.xxx:5060
CSeq: 102 INVITE

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

<--- SIP read from UDP:xxx.xx.xxx.xxx:5060 --->
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP xxx.xx.xx.xxx:5060;received=xxx.xx.xx.xxx;branch=z9hG4bK5806dbc8;rport=5060
From: "Servidores" <sip:xx...@xxx.xx.xx.xxx>;tag=as240328e2
To: <sip:xx...@xxx.xx.xxx.xxx>;tag=5ieye5hg
Call-ID: 438956893746...@xxx.xx.xx.xxx:5060
CSeq: 102 INVITE
Reason: Q.850;cause=28;text="address incomplete"
Warning: 399 - "SoftX3000 R601-CCU Rel POS:[10680] Release from CCB PP[0xe0] CS[0x4] AS[0x4] FR[0x8] TR[0x9] TYPE[0x2]"
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
    -- Got SIP response 484 "Address Incomplete" back from xxx.xx.xxx.xxx:5060
Transmitting (NAT) to xxx.xx.xxx.xxx:5060:
ACK sip:xxx...@xxx.xx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP xxx.xx.xx.xxx:5060;branch=z9hG4bK5806dbc8;rport
Max-Forwards: 70
From: "Servidores" <sip:xx...@xxx.xx.xx.xxx>;tag=as240328e2
To: <sip:xxx...@xxx.xx.xxx.xxx>;tag=5ieye5hg
Contact: <sip:xx...@xxx.xx.xx.xxx:5060>
Call-ID: 438956893746...@xxx.xx.xx.xxx:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.14.1
Content-Length: 0


---
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [xxxxxxx@ADMIN:3] Congestion("SIP/90-00000010", "") in new stack
Really destroying SIP dialog '438956893746...@xxx.xx.xx.xxx:5060' Method: INVITE
  == Spawn extension (ADMIN, xxxxxxx, 3) exited non-zero on 'SIP/90-00000010'
Reliably Transmitting (NAT) to xxx.xx.xxx.xxx:5060:
OPTIONS sip:xxx.xx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP xxx.xx.xx.xxx:5060;branch=z9hG4bK45004361;rport
Max-Forwards: 70
From: "asterisk" <sip:xx...@xxx.xx.xx.xxx>;tag=as0d7761c0
To: <sip:xxx.xx.xxx.xxx>
Contact: <sip:xx...@xxx.xx.xx.xxx:5060>
Call-ID: 438956893746...@xxx.xx.xx.xxx:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.14.1
Date: Mon, 04 Jan 2016 20:29:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:xxx.xx.xxx.xxx:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.xx.xx.xxx:5060;received=xxx.xx.xx.xxx;branch=z9hG4bK45004361;rport=5060
From: "asterisk" <sip:xx...@xxx.xx.xx.xxx>;tag=as0d7761c0
To: <sip:xxx.xx.xxx.xxx>;tag=aprqjd28kp0-5d0mt330000c6
Call-ID: 438956893746...@xxx.xx.xx.xxx:5060
CSeq: 102 OPTIONS

<------------->
--- (6 headers 0 lines) ---
Really destroying SIP dialog '4389568937465...@xxx.xx.xx.xxx:5060' Method: OPTIONS
ivr*CLI>


=====================================================================================

de antemano les agradezco su colaboracion

Rodrigo Ramírez Norambuena

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Jan 5, 2016, 9:23:48 AM1/5/16
to aster...@googlegroups.com
January 4 2016 5:53 PM, "rochasystem" <zenk...@hotmail.com> wrote:
> Buenas tardes tengo una planta en asterisk 11 y una troncal sip con el proveedor UNE (Colomabia),
> la cuestion es que estoy intentando sacar llamadas por esta troncal pero no he podido y me sale el
> siguiente mensage:
>
> == Using SIP RTP CoS mark 5
> -- Executing [3632542@ADMIN:1] Set("SIP/90-00000012", "CALLERID(num)=53100970") in new stack
> -- Executing [3632542@ADMIN:2] Dial("SIP/90-00000012", "SIP/UNE1/3222222,90,Ttr") in new stack
> == Using SIP RTP CoS mark 5
> -- Called SIP/UNE1/3632542
> -- Got SIP response 484 "Address Incomplete" back from 172.17.179.166:5060
> == Everyone is busy/congested at this time (1:0/0/1)
> -- Executing [3632542@ADMIN:3] Congestion("SIP/90-00000012", "") in new stack
> == Spawn extension (ADMIN, 3632542, 3) exited non-zero on 'SIP/90-00000012'

Pero es el proveedor que te da eso, lo has visto con él?

> Call-ID: 438956893746...@xxx.xx.xx.xxx:5060

Este call-id está raro, no te estarán no completando la llamda por eso.

--
Rodrigo Ramírez Norambuena
http://www.rodrigoramirez.com

Juan Felipe

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Jan 5, 2016, 11:45:08 AM1/5/16
to aster...@googlegroups.com
Normalmente con las troncales de une en colombia ahí que enviar la llamada con el caller id asignado a la troncal sip.

Enviado desde mi iPhone
> --
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> ---
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Carlos Andrés Tapasco Viera

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Jan 5, 2016, 12:07:49 PM1/5/16
to aster...@googlegroups.com
Adicionalmente, ellos deben registrar la MAC de tu NIC desde el equipo que entregan en sitio.

rochasystem

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Jan 5, 2016, 5:35:04 PM1/5/16
to asterisk-es

Hola a todos y gracias por su ayuda, el caller id lo mando en el contesto de llamado asi: exten=>_XXXXXXX,1,set(CALLERID(num)=53100970)
no se si es la forma correcta.
Call-ID: 438956893746534dfshgdsgdg@xxx.xx.xx.xxx:5060
Call-ID: 438956893746534dfshgdsgdg@xxx.xx.xx.xxx:5060

CSeq: 102 INVITE

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

<--- SIP read from UDP:xxx.xx.xxx.xxx:5060 --->
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP xxx.xx.xx.xxx:5060;received=xxx.xx.xx.xxx;branch=z9hG4bK5806dbc8;rport=5060
From: "Servidores" <sip:xx...@xxx.xx.xx.xxx>;tag=as240328e2
To: <sip:xx...@xxx.xx.xxx.xxx>;tag=5ieye5hg
Call-ID: 438956893746534dfshgdsgdg@xxx.xx.xx.xxx:5060

CSeq: 102 INVITE
Reason: Q.850;cause=28;text="address incomplete"
Warning: 399 - "SoftX3000 R601-CCU Rel POS:[10680] Release from CCB PP[0xe0] CS[0x4] AS[0x4] FR[0x8] TR[0x9] TYPE[0x2]"
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
    -- Got SIP response 484 "Address Incomplete" back from xxx.xx.xxx.xxx:5060
Transmitting (NAT) to xxx.xx.xxx.xxx:5060:
ACK sip:xxx...@xxx.xx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP xxx.xx.xx.xxx:5060;branch=z9hG4bK5806dbc8;rport
Max-Forwards: 70
From: "Servidores" <sip:xx...@xxx.xx.xx.xxx>;tag=as240328e2
To: <sip:xxx...@xxx.xx.xxx.xxx>;tag=5ieye5hg
Contact: <sip:xx...@xxx.xx.xx.xxx:5060>
Call-ID: 438956893746534dfshgdsgdg@xxx.xx.xx.xxx:5060

CSeq: 102 ACK
User-Agent: Asterisk PBX 11.14.1
Content-Length: 0


---
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [xxxxxxx@ADMIN:3] Congestion("SIP/90-00000010", "") in new stack
Really destroying SIP dialog '438956893746534dfshgdsgdg@xxx.xx.xx.xxx:5060' Method: INVITE

  == Spawn extension (ADMIN, xxxxxxx, 3) exited non-zero on 'SIP/90-00000010'
Reliably Transmitting (NAT) to xxx.xx.xxx.xxx:5060:
OPTIONS sip:xxx.xx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP xxx.xx.xx.xxx:5060;branch=z9hG4bK45004361;rport
Max-Forwards: 70
From: "asterisk" <sip:xx...@xxx.xx.xx.xxx>;tag=as0d7761c0
To: <sip:xxx.xx.xxx.xxx>
Contact: <sip:xx...@xxx.xx.xx.xxx:5060>
Call-ID: 438956893746534dfshgdsgdg@xxx.xx.xx.xxx:5060

CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.14.1
Date: Mon, 04 Jan 2016 20:29:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:xxx.xx.xxx.xxx:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.xx.xx.xxx:5060;received=xxx.xx.xx.xxx;branch=z9hG4bK45004361;rport=5060
From: "asterisk" <sip:xx...@xxx.xx.xx.xxx>;tag=as0d7761c0
To: <sip:xxx.xx.xxx.xxx>;tag=aprqjd28kp0-5d0mt330000c6
Call-ID: 438956893746534dfshgdsgdg@xxx.xx.xx.xxx:5060

CSeq: 102 OPTIONS

<------------->
--- (6 headers 0 lines) ---
Really destroying SIP dialog '438956893746534dfshgdsgdg8@xxx.xx.xx.xxx:5060' Method: OPTIONS

Carlos Andrés Tapasco Viera

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Jan 5, 2016, 6:11:29 PM1/5/16
to aster...@googlegroups.com
¿Ya hablaste con tu proveedor y le consultaste cómo lo debes enviar?. A veces lo piden con inidicativo de ciudad, otras veces con prefijo de país, depende de qué esperen del otro lado.

Call-ID: 438956893746...@xxx.xx.xx.xxx:5060
Call-ID: 438956893746...@xxx.xx.xx.xxx:5060

CSeq: 102 INVITE

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

<--- SIP read from UDP:xxx.xx.xxx.xxx:5060 --->
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP xxx.xx.xx.xxx:5060;received=xxx.xx.xx.xxx;branch=z9hG4bK5806dbc8;rport=5060
From: "Servidores" <sip:xx...@xxx.xx.xx.xxx>;tag=as240328e2
To: <sip:xx...@xxx.xx.xxx.xxx>;tag=5ieye5hg
Call-ID: 438956893746...@xxx.xx.xx.xxx:5060

CSeq: 102 INVITE
Reason: Q.850;cause=28;text="address incomplete"
Warning: 399 - "SoftX3000 R601-CCU Rel POS:[10680] Release from CCB PP[0xe0] CS[0x4] AS[0x4] FR[0x8] TR[0x9] TYPE[0x2]"
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
    -- Got SIP response 484 "Address Incomplete" back from xxx.xx.xxx.xxx:5060
Transmitting (NAT) to xxx.xx.xxx.xxx:5060:
ACK sip:xxx...@xxx.xx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP xxx.xx.xx.xxx:5060;branch=z9hG4bK5806dbc8;rport
Max-Forwards: 70
From: "Servidores" <sip:xx...@xxx.xx.xx.xxx>;tag=as240328e2
To: <sip:xxx...@xxx.xx.xxx.xxx>;tag=5ieye5hg
Contact: <sip:xx...@xxx.xx.xx.xxx:5060>
Call-ID: 438956893746...@xxx.xx.xx.xxx:5060

CSeq: 102 ACK
User-Agent: Asterisk PBX 11.14.1
Content-Length: 0


---
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [xxxxxxx@ADMIN:3] Congestion("SIP/90-00000010", "") in new stack
Really destroying SIP dialog '438956893746...@xxx.xx.xx.xxx:5060' Method: INVITE

  == Spawn extension (ADMIN, xxxxxxx, 3) exited non-zero on 'SIP/90-00000010'
Reliably Transmitting (NAT) to xxx.xx.xxx.xxx:5060:
OPTIONS sip:xxx.xx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP xxx.xx.xx.xxx:5060;branch=z9hG4bK45004361;rport
Max-Forwards: 70
From: "asterisk" <sip:xx...@xxx.xx.xx.xxx>;tag=as0d7761c0
To: <sip:xxx.xx.xxx.xxx>
Contact: <sip:xx...@xxx.xx.xx.xxx:5060>
Call-ID: 438956893746...@xxx.xx.xx.xxx:5060

CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.14.1
Date: Mon, 04 Jan 2016 20:29:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:xxx.xx.xxx.xxx:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.xx.xx.xxx:5060;received=xxx.xx.xx.xxx;branch=z9hG4bK45004361;rport=5060
From: "asterisk" <sip:xx...@xxx.xx.xx.xxx>;tag=as0d7761c0
To: <sip:xxx.xx.xxx.xxx>;tag=aprqjd28kp0-5d0mt330000c6
Call-ID: 438956893746...@xxx.xx.xx.xxx:5060

CSeq: 102 OPTIONS

<------------->
--- (6 headers 0 lines) ---
Really destroying SIP dialog '4389568937465...@xxx.xx.xx.xxx:5060' Method: OPTIONS

ivr*CLI>


=====================================================================================

de antemano les agradezco su colaboracion

--
Este email pertenece a la lista de Asterisk-ES (http://www.asterisk-es.org)
Normas de la lista Asterisk-ES: http://comunidad.asterisk-es.org/index.php?title=Lista:normas-asterisk-es
---
Has recibido este mensaje porque estás suscrito al grupo "asterisk-es" de Grupos de Google.
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Para publicar en este grupo, envía un correo electrónico a aster...@googlegroups.com.

Visita este grupo en https://groups.google.com/group/asterisk-es.
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