No puedo configurar las llamadas

inforemp

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#1
Me canse de leer posts y manuales, y ya estoy muy confundido

No puedo hacer llamadas desde mi extension configurada como "3000" hacia algun numero interno de mi voip, por ejemplo "100000", cada vez q marco me dice "todas las lineas se encuentran ocupadas"

y si marco mi extension "3000" cuyo numero de mi voip seria el "100123", nunca recibo la llamada.


Aca pego todo lo que tengo hasta ahora, espero que alguien pueda iluminarme en lo que estoy haciendo mal. PD: 0 experiencia en esto, hace 1 semana instale el elastix y recien entro en el "mundo voip"




Los puertos abiertos en mi Linksys PAP2 son Linea 1: 5060 y Linea 2: 5061

sip.conf

externip=cuenxxxx.ath.xx
localnet=192.168.1.0/255.255.255.0

sip_nat.conf

nat=yes
externip=cuentxxx.ath.cx
localnet=192.168.1.0/255.255.255.0
externrefresh=5




trunk:

username= ? (extension user, o voip?)
type=peer
secret=xxxxx
insecure=very
host= ? (router ip, elastix, o my voip?)
dtmfmode=rfc2833
disallow=all
allow=alaw&ulaw&gsm
canredirect=no
canreinvite=no

user details

canreinvite=no
context=from-trunk
fromuser= ? (usuario extension o voip?)
qualify=no
secret=xxxx
type=user
username=xxxx


Register String


usuariovoip:passwordvoip@200.69.159.33 (ip voip) no?


outbound routes:

dial pattern: 8|x.

trunk sequence: trunk


my sip debug:
-- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/3000-09f58aa8", "0?bypass|1") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/3000-09f58aa8", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:19] Dial("SIP/3000-09f58aa8", "SIP/citarella/100000|300|") in new stack
Audio is at 192.168.1.100 port 13526
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.1.50:5060:
INVITE sip:100000@192.168.1.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK1f6352ed;rport
From: "1003000" <sip:1003000@192.168.1.100>;tag=as50c053f6
To: <sip:100000@192.168.1.50>
Contact: <sip:1003000@192.168.1.100>
Call-ID: 21483b1906c22465184755a9611a97c7@192.168.1.100
CSeq: 102 INVITE
User-Agent: Elastix
Max-Forwards: 70
Date: Wed, 16 Sep 2009 12:24:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 3125 3125 IN IP4 192.168.1.100
s=session
c=IN IP4 192.168.1.100
t=0 0
m=audio 13526 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:eek:ff - - - -
a=ptime:20
a=sendrecv

---
-- Called citarella/100000
cuentaip*CLI>
<--- SIP read from 192.168.1.50:5060 --->
SIP/2.0 404 Not Found
To: <sip:100000@192.168.1.50>;tag=b81f97b570e4f770i0
From: "1003000" <sip:1003000@192.168.1.100>;tag=as50c053f6
Call-ID: 21483b1906c22465184755a9611a97c7@192.168.1.100
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK1f6352ed
Server: Linksys/PAP2T-3.1.15(LS)
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 192.168.1.50:5060:
ACK sip:100000@192.168.1.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK1f6352ed;rport
From: "1003000" <sip:1003000@192.168.1.100>;tag=as50c053f6
To: <sip:100000@192.168.1.50>;tag=b81f97b570e4f770i0
Contact: <sip:1003000@192.168.1.100>
Call-ID: 21483b1906c22465184755a9611a97c7@192.168.1.100
CSeq: 102 ACK
User-Agent: Elastix
Max-Forwards: 70
Content-Length: 0


---
-- SIP/citarella-09f5fa88 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [s@macro-dialout-trunk:20] Goto("SIP/3000-09f58aa8", "s-CONGESTION|1") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing [s-CONGESTION@macro-dialout-trunk:1] GotoIf("SIP/3000-09f58aa8", "1?noreport") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,3)
-- Executing [s-CONGESTION@macro-dialout-trunk:3] NoOp("SIP/3000-09f58aa8", "TRUNK Dial failed due to CONGESTION - failing through to other trunks") in new stack
-- Executing [8100000@from-internal:5] Macro("SIP/3000-09f58aa8", "outisbusy|") in new stack
-- Executing [s@macro-outisbusy:1] Playback("SIP/3000-09f58aa8", "all-circuits-busy-now|noanswer") in new stack
Audio is at 192.168.1.100 port 11100
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x1 (g723) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
cuentaip*CLI>
<--- Transmitting (NAT) to 192.168.1.50:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-dc72af61;received=192.168.1.50
From: Linea 1 <sip:3000@192.168.1.100>;tag=634d08aa8d51d295o0
To: <sip:8100000@192.168.1.100>;tag=as6734574c
Call-ID: c4612e0e-bb4b4c81@192.168.1.50
CSeq: 102 INVITE
User-Agent: Elastix
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:8100000@192.168.1.100>
Content-Type: application/sdp
Content-Length: 355

v=0
o=root 3125 3125 IN IP4 192.168.1.100
s=session
c=IN IP4 192.168.1.100
t=0 0
m=audio 11100 RTP/AVP 0 8 18 4 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:eek:ff - - - -
a=ptime:20
a=sendrecv

<------------>
-- <SIP/3000-09f58aa8> Playing 'all-circuits-busy-now' (language 'es')
Really destroying SIP dialog '21483b1906c22465184755a9611a97c7@192.168.1.100' Method: INVITE
-- Executing [s@macro-outisbusy:2] Playback("SIP/3000-09f58aa8", "pls-try-call-later|noanswer") in new stack
-- <SIP/3000-09f58aa8> Playing 'pls-try-call-later' (language 'es')
cuentaip*CLI>
<--- SIP read from 192.168.1.50:5060 --->
CANCEL sip:8100000@192.168.1.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-dc72af61
From: Linea 1 <sip:3000@192.168.1.100>;tag=634d08aa8d51d295o0
To: <sip:8100000@192.168.1.100>
Call-ID: c4612e0e-bb4b4c81@192.168.1.50
CSeq: 102 CANCEL
Max-Forwards: 70
Proxy-Authorization: Digest username="3000",realm="asterisk",nonce="65ea3d5a",uri="sip:8100000@192.168.1.100",algorithm=MD5,response="4afdab4c474409369b72ce7ae707f132"
User-Agent: Linksys/PAP2T-3.1.15(LS)
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Sending to 192.168.1.50 : 5060 (NAT)

<--- Reliably Transmitting (NAT) to 192.168.1.50:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-dc72af61;received=192.168.1.50
From: Linea 1 <sip:3000@192.168.1.100>;tag=634d08aa8d51d295o0
To: <sip:8100000@192.168.1.100>;tag=as6734574c
Call-ID: c4612e0e-bb4b4c81@192.168.1.50
CSeq: 102 INVITE
User-Agent: Elastix
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


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

<--- Transmitting (NAT) to 192.168.1.50:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-dc72af61;received=192.168.1.50
From: Linea 1 <sip:3000@192.168.1.100>;tag=634d08aa8d51d295o0
To: <sip:8100000@192.168.1.100>;tag=as6734574c
Call-ID: c4612e0e-bb4b4c81@192.168.1.50
CSeq: 102 CANCEL
User-Agent: Elastix
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------>
== Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/3000-09f58aa8' in macro 'outisbusy'
== Spawn extension (from-internal, 8100000, 5) exited non-zero on 'SIP/3000-09f58aa8'
-- Executing [h@from-internal:1] Macro("SIP/3000-09f58aa8", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/3000-09f58aa8", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4] GotoIf("SIP/3000-09f58aa8", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf("SIP/3000-09f58aa8", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] Hangup("SIP/3000-09f58aa8", "") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/3000-09f58aa8' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/3000-09f58aa8'
cuentaip*CLI>
<--- SIP read from 192.168.1.50:5060 --->
ACK sip:8100000@192.168.1.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-dc72af61
From: Linea 1 <sip:3000@192.168.1.100>;tag=634d08aa8d51d295o0
To: <sip:8100000@192.168.1.100>;tag=as6734574c
Call-ID: c4612e0e-bb4b4c81@192.168.1.50
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest username="3000",realm="asterisk",nonce="65ea3d5a",uri="sip:8100000@192.168.1.100",algorithm=MD5,response="fc9351980db59df69276c8325bb1ebff"
Contact: Linea 1 <sip:3000@192.168.1.50:5060>
User-Agent: Linksys/PAP2T-3.1.15(LS)
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog 'c4612e0e-bb4b4c81@192.168.1.50' Method: ACK
 

telecomtechnician

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#2
Hola, empecemos por el principio

1) Las llamadas que quieres hacer son internas? (de una extensión IP a otra dentro de la misma red lan?)

2) Si una o ambas extensiones IP son remotas (fuera de la red lan) abriste los puertos correctos en el router?, configuraste el archivo sip_nat.conf?

2) Verificaste que cuando creaste las extensiones, tienen los datos importantes como creación de buzón de voz y que se encuentran en el mismo contexto?

Espero tus comentarios

David Medina
 

inforemp

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#3
ya lo solucione, ahora tengo otra duda, creo otro post, por favor, respondeme ahi :) gracias.
 

gamba47

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#4
inforemp said:
ya lo solucione, ahora tengo otra duda, creo otro post, por favor, respondeme ahi :) gracias.
Sería muy bueno que comentes en que te equivocastes y como lo solucionastes, más que nada para que quede el problema y la solución, para futuros usuarios que pasen por lo mismo que vos, aunque te parezca algo fácil ahora, a otros le puede pasar lo mismo!

gracias.

gamba47
 

ElasMex

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#5
inforemp

Coloca tu solución o el error que tenías.

Hay que dar seguimiento a los Topic que uno genera.

Saludos
 

inforemp

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#6
Recien vuelvo a entrar a los foros, sepan disculpar, pongo mi solucion, estaban mal hecho los trunks... les pego como los deje.

Peer Details
username=XXXXX (Aqui va el nombre de usuario del servicio VOIP)
type=peer
secret=1234 (password del servicio voip)
insecure=very
host=200.69.159.33 (ip del servicio voip)
dtmfmode=rfc2833
disallow=all
allow=alaw&ulaw&gsm
canredirect=no
canreinvite=no

User Details
canreinvite=no
context=from-trunk
fromuser=XXXX (nombre de usuario de una extension creada)
qualify=no
secret=1234 (password extension creada)
type=user
username=3000 (nuevamente nombre de usuario)

Register String:

usuario:1234@200.xxx.xxx.xxx (esto es lo mas importante, la forma de conectarse al servicio voip es "NOMBREDUSUARIOVOIP:pASSWORDVOIP@IPDELSERVIDORVOIP
 

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