LLamadas de entrada

[Gendo Ikari]

Buenas tardes amigos, tengo una duda, eh configurado conforme a lo que encontre en internet y aqui, las llamadas entrantes, mas sin embargo, no logro echarlo a andar, alguien tiene una idea de como puedo hacerlo? les dejo mi extensions.conf y mi sip.conf

sip.conf

[63877770]
disallow=all
username=XXXXXXXXXip
type=friend
secret=XXXXX
qualify=yes
insecure=port,invite
host=mex14.axtel.net
fromuser=XXXXXXXXXip
fromdomain=0.0.0.0
dtmfmode=auto
context=from-ax70
canreinvite=no
allow=gsm
port=5060
rfc2833compensate=yes
register=XXXXXXXXip:XXXX@mex14.axtel.net/XXXXXXXXXX

extensions.conf

[from-ax70]
exten => _63877770,1,Answer
exten => _63877770,2,monitor(gsm,${EXTEN}-${CALLFILENAME}-${CALLERIDNUM}-${STRFTIME(${EPOCH},gmt+6,%C%y%m%d-%H%M%S)},m)
exten => _63877770,3,Dial(SIP/108,30,Ttr)
same => n,Hangup

Ayudenme por favor.

[hector]

Revisa que en el [globals] de tu sip.conf estés registrando tu linea (register=>) y posteriormente en el CLI verifica que esté registrado sip show registry... probablemente el carrier no te esta asignando las llamadas de entrada... dudo que te este llegando el INVITE desde la central (puedes verificarlo con el debug de SIP o con algo externo como wireshark)

[Gendo Ikari]

El CLI me muestra esto:

BALTAZAR*CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
0 SIP registrations.
BALTAZAR*CLI>


El globals como se configura? yo solo configure el de [general]

[hector]

[globals]
register => fromuser@fromdomain:secret@host

[Gendo Ikari]

Lo he configurado como me dices y me marca esto el CLI

Section 'globals' lacks type

y las llamadas siguen sin entrar, por favor ayúdame. =(

[navaismo]

La linea Register va en General.

[Gendo Ikari]

Asi esta el general

[general]
port=5060
bindaddr=0.0.0.0
type=friend
allowguest=no
disallow=all
allow=gsm
callevents=yes
useragent=zoiper
nat=yes
register => XXXXXXXXip@192.168.1.200:XXXX@mex14.axtel.net

pero aun asi no logro recibir llamadas, alguna idea amigos?

[navaismo]

Has intentado cambiar tu Register string por:

register => nombredeusuarioass@hostprovider:5060/DID

Y bueno para poder ayudarte necesitamos ver el sip debug completo cuando haces un sip reload. Y la salida nuevamente de sip show registry.

[Gendo Ikari]

Esto es lo que me muestra:

BALTAZAR*CLI> sip reload
Reloading SIP
BALTAZAR*CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
0 SIP registrations.
BALTAZAR*CLI>

Aparece sin actividad como puedes ver, tienes alguna idea si me falta algo en el sip.conf y extensions.conf para levantar la troncal sip?

[navaismo]

Cambia tu linea de Register por lo que te comente y no uses direcciones IP privadas, si vas a especificar la Ip de tu server usa la IP Pública no la privada.

Por otro lado esos mensajes no contienen el SIP DEBUG y es lo que necesitamos ver para poder ayudarte. se habilita con "sip set debug on" y son muchos mensajes copia todo despues de hacer un sip reload.

[hector]

Si van en [general] perdon por el error... y lo que pasa es que tu linea se ve un poco rara... prueba usando solo
register => XXXXip:XXXX@mex14.axtel.net

habilita el debug en sip (sip set debug on) y haz el sip reload despues de eso... para ver que esta pasando

[Gendo Ikari]

Esto es lo que me arrojo el sip

<--- SIP read from UDP:192.168.1.55:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK776012cf;rport=50 60
Contact: <sip:192.168.1.55:5060>
To: <sip:110@192.168.1.55:5060;rinstance=19689eb292628 5d1;transport=UDP>;tag=7b3a0874
From: "asterisk"<sip:asterisk@192.168.1.200>;tag=as07f05 ef7
Call-ID: 1435651c208386f86d353281068a8453@192.168.1.200:506 0
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper rev.14736
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog '1435651c208386f86d353281068a8453@192.168.1.200:50 60' Method: OPTIONS
Reliably Transmitting (NAT) to 207.249.182.154:5060:
OPTIONS sip:mex14.axtel.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK0faea329;rport
Max-Forwards: 70
From: "asterisk" <sip:10003361548ip@192.168.1.200>;tag=as79aa0001
To: <sip:mex14.axtel.net>
Contact: <sip:10003361548ip@192.168.1.200:5060>
Call-ID: 18ba707945efbd7d706a2f81224fb658@192.168.1.200:506 0
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.15.0
Date: Thu, 25 Apr 2013 19:03:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Reliably Transmitting (NAT) to 207.249.182.154:5060:
OPTIONS sip:mex15.axtel.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK543cd903;rport
Max-Forwards: 70
From: "asterisk" <sip:10003361751ip@192.168.1.200>;tag=as233bc4c3
To: <sip:mex15.axtel.net>
Contact: <sip:10003361751ip@192.168.1.200:5060>
Call-ID: 7bbfef00023213fa23c0ba6126a6d01a@192.168.1.200:506 0
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.15.0
Date: Thu, 25 Apr 2013 19:03:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

[navaismo]

No esta completo no se ve ninguna mensaje para registrarse a un proveedor sip, muestranos como esta tu linea de REGISTER ahora, y trata de recolctar el sip debug completo.

[Gendo Ikari]

Aqui esta:

Reliably Transmitting (NAT) to 192.168.1.55:5060:
OPTIONS sip:110@192.168.1.55:5060;rinstance=19689eb2926285 d1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK7b58fa9b;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.1.200>;tag=as6df67cf1
To: <sip:110@192.168.1.55:5060;rinstance=19689eb292628 5d1;transport=UDP>
Contact: <sip:asterisk@192.168.1.200:5060>
Call-ID: 3c557fb07f9816e160f8a71e4e36feb7@192.168.1.200:506 0
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.15.0
Date: Thu, 25 Apr 2013 20:19:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.1.55:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK7b58fa9b;rport=50 60
Contact: <sip:192.168.1.55:5060>
To: <sip:110@192.168.1.55:5060;rinstance=19689eb292628 5d1;transport=UDP>;tag=27f41f0c
From: "asterisk"<sip:asterisk@192.168.1.200>;tag=as6df67 cf1
Call-ID: 3c557fb07f9816e160f8a71e4e36feb7@192.168.1.200:506 0
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper rev.14736
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog '3c557fb07f9816e160f8a71e4e36feb7@192.168.1.200:50 60' Method: OPTIONS
Reliably Transmitting (NAT) to 207.249.182.154:5060:
OPTIONS sip:mex14.axtel.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK2c715a97;rport
Max-Forwards: 70
From: "asterisk" <sip:10003361548ip@192.168.1.200>;tag=as1ecf15ab
To: <sip:mex14.axtel.net>
Contact: <sip:10003361548ip@192.168.1.200:5060>
Call-ID: 741235064e0a83016f3dc41f53150ab1@192.168.1.200:506 0
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.15.0
Date: Thu, 25 Apr 2013 20:19:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Reliably Transmitting (NAT) to 207.249.182.154:5060:
OPTIONS sip:mex15.axtel.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK69e9ec74;rport
Max-Forwards: 70
From: "asterisk" <sip:10003361751ip@192.168.1.200>;tag=as2d55bd3b
To: <sip:mex15.axtel.net>
Contact: <sip:10003361751ip@192.168.1.200:5060>
Call-ID: 65faf4f52c828dec5e12f0ce74f796c5@192.168.1.200:506 0
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.15.0
Date: Thu, 25 Apr 2013 20:19:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:207.249.182.154:5060 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK2c715a97;rport=50 60;received=201.156.15.87
From: "asterisk" <sip:10003361548ip@192.168.1.200>;tag=as1ecf15ab
To: <sip:mex14.axtel.net>;tag=e02BFvNjNcpmc
Call-ID: 741235064e0a83016f3dc41f53150ab1@192.168.1.200:506 0
CSeq: 102 OPTIONS
Contact: <sip:10003361548ip@201.156.15.87:5060>
Server: Flexisip/0.5.0 (sofia-sip-nta/2.0)
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '741235064e0a83016f3dc41f53150ab1@192.168.1.200:50 60' Method: OPTIONS

<--- SIP read from UDP:207.249.182.154:5060 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK69e9ec74;rport=50 60;received=201.156.15.87
From: "asterisk" <sip:10003361751ip@192.168.1.200>;tag=as2d55bd3b
To: <sip:mex15.axtel.net>;tag=F9U4gQ6NjNc7Q
Call-ID: 65faf4f52c828dec5e12f0ce74f796c5@192.168.1.200:506 0
CSeq: 102 OPTIONS
Contact: <sip:10003361751ip@201.156.15.87:5060>
Server: Flexisip/0.5.0 (sofia-sip-nta/2.0)
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '65faf4f52c828dec5e12f0ce74f796c5@192.168.1.200:50 60' Method: OPTIONS

<--- SIP read from UDP:192.168.1.55:5060 --->

[Christian Cabrera]

Hay un error 488 Not Acceptable here.

Te recomiendo cambies tu troncal y quites el disallow=all y agregues el allow=all para que recibas todos los codecs.

Saludos,