Estimados, tengo una extensión sip externa, la misma no funciona ya que se loguea ok pero al llamar la misma suena pero no tiene audio, dejo mi configuración:
Existe un router en la 192.168.0.1 y nateado a la 192.168.0.8
Puertos abiertos y nateados:
5060 al 5065
10000 al 20000
50000 al 65000
35060
[general]
bindaddr=0.0.0.0
context=normal
videosupport=no
language=es
nat=yes
externrefresh=120
qualify=yes
relaxdtmf=yes
localnet=192.168.0.0/255.255.255.0
externip=xxx.210.xx.1xx
directmedia=off
sip_nat.conf
externip=xxx.210.xx.1xx
localhost=192.168.0.8
localnet=192.168.0.0/255.255.255.0
externrefresh=120
nat=yes
language=es
qualify=yes
relaxdtmf=yes
directmedia=off
rtp_additional.conf
[general]
rtpstart=10000
rtpend=20000
Veo que la ip remota la toma como IP local en el debug, entonces entiendo que por eso no encuentra la devolución correcta del paquete, dicho esto, como hago el NAT correctamente ?
DEBUG:
2018-01-17 09:14:27] DEBUG[3735][C-00006522]: chan_sip.c:3374 initialize_initreq: Initializing initreq for method INVITE - callid 6d6129esdsdsdsdsdsdsd34cee33ef868c93e1d18e3@192.168.0.8:5060
Reliably Transmitting (NAT) to 192.168.0.1:58119:
INVITE sip:103@192.168.1.200:58119;rinstance=2822sdsdsdsdsdsdsd67f5c255;transport=UDP SIP/2.0 --------------- (ES MI EXTENSIÓN SIP LA 103, LO QUE ME LLAMA LA ATENCIÓN ES QUE LA IP QUE FIGURA ES MI IP LOCAL QUE ESTA AFUERA DE LA RED Y LA PONE COMO IP PRIVADA, SUPONGO QUE AL DEVOLVER EL PAQUETE LO DEVUELVE HACIA LA IP PRIVADA) ----------------------
Via: SIP/2.0/UDP 192.168.0.8:5060;branch=z9hG4bKjrtr875c2aa;rport
Max-Forwards: 70
From: "0054232544545xx6" <sip:00542325414946@192.168.0.8>;tag=as6c5a2f57 -----------(ENTRA LA LLAMADA DE MI CELULAR) ---------
To: <sip:103@192.168.1.200:58119;rinstance=282225ee67f5c255;transport=UDP> ---------(QUE ATIENDE LA IP 1.200 POR EL PUERTO 58119) ----------
Contact: <sip:00542325414946@192.168.0.8:5060>
Call-ID: 6d6129e46134cee33ef868c93e1d18e3@192.168.0.8:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.25.0)
Date: Wed, 17 Jan 2018 12:14:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Diversion: <sip:+5411521754544545@192.168.0.8>;reason=unconditional
Content-Type: application/sdp
Content-Length: 281
v=0
o=root 1798058053 1798058053 IN IP4 192.168.0.8
s=Asterisk PBX 11.25.0
c=IN IP4 192.168.0.8
t=0 0
m=audio 16652 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[2018-01-17 09:14:27] DEBUG[3735][C-00006522]: chan_sip.c:3731 __sip_xmit: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.0.1:58119
-- Called SIP/103
-- Local/103@from-queue-0000001a;1 is ringing
[2018-01-17 09:14:27] DEBUG[2990]: app_queue.c:2023 extension_state_cb: Extension '103@ext-local' changed to state '6' (Ringing)
<--- SIP read from UDP:192.168.0.1:58119 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.8:5060;branch=z9h546452aa;rport=5060;received=190.210.40.186
To: <sip:103@192.168.1.200:58119;rinstance=2822245645645646565c255;transport=UDP>
From: "00542325414946" <sip:00542325414946@192.168.0.8>;tag=as6c5a2f57
Call-ID: 6d6129e44554545458c93e1d18e3@192.168.0.8:5060
CSeq: 102 INVITE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
[2018-01-17 09:14:27] DEBUG[3051][C-00006522]: chan_sip.c:4470 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '6d6129e445454545454568c93e1d18e3@192.168.0.8:5060' Request 102: Found
<--- SIP read from UDP:192.168.0.1:58119 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.8:5060;branch=z9hG445445454542aa;rport=5060;received=190.210.40.XXX
Contact: <sip:103@192.168.1.200:58119>
To: <sip:103@192.168.1.200:58119;rinstance=282225asdsadasd55;transport=UDP>;tag=0a150b7d
From: "00542325414946" <sip:0054232541xxxx6@192.168.0.8>;tag=as6c5a2f57
Call-ID: 6d612sdfsdfdsfsdfsdfsdfsdfsdfdfsdf3e1d18e3@192.168.0.8:5060
CSeq: 102 INVITE
User-Agent: Z 3.15.40006 rv2.8.20
Allow-Events: presence, kpml, talk
Content-Length: 0