Re: Problema en escuchar algunas llamadas
Hola que tal!
El core show translation me muestra:
g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 siren7 siren14 slin16 g719 speex16 testlaw
g723 - 4002 2 2 4002 2 1 2 4001 8001 - 2 2 - - 4003 - 8004 4001
gsm 8001 - 4001 4001 8001 4001 4000 4001 8000 12000 - 4001 4001 - - 8002 - 12003 8000
ulaw 4002 4002 - 1 4002 2 1 2 4001 8001 - 2 2 - - 4003 - 8004 4001
alaw 4002 4002 1 - 4002 2 1 2 4001 8001 - 2 2 - - 4003 - 8004 4001
g726aal2 4002 4002 2 2 - 2 1 2 4001 8001 - 2 2 - - 4003 - 8004 4001
adpcm 4002 4002 2 2 4002 - 1 2 4001 8001 - 2 2 - - 4003 - 8004 4001
slin 4001 4001 1 1 4001 1 - 1 4000 8000 - 1 1 - - 4002 - 8003 4000
lpc10 8001 8001 4001 4001 8001 4001 4000 - 8000 12000 - 4001 4001 - - 8002 - 12003 8000
g729 4002 4002 2 2 4002 2 1 2 - 8001 - 2 2 - - 4003 - 8004 4001
speex 4002 4002 2 2 4002 2 1 2 4001 - - 2 2 - - 4003 - 8004 4001
ilbc - - - - - - - - - - - - - - - - - - -
g726 8001 8001 4001 4001 8001 4001 4000 4001 8000 12000 - - 4001 - - 8002 - 12003 8000
g722 8001 8001 4001 4001 8001 4001 4000 4001 8000 12000 - 4001 - - - 4001 - 8002 8000
siren7 - - - - - - - - - - - - - - - - - - -
siren14 - - - - - - - - - - - - - - - - - - -
slin16 8002 8002 4002 4002 8002 4002 4001 4002 8001 12001 - 4002 1 - - - - 4001 8001
g719 - - - - - - - - - - - - - - - - - - -
speex16 12002 12002 8002 8002 12002 8002 8001 8002 12001 16001 - 8002 4001 - - 4000 - - 12001
testlaw 4002 4002 2 2 4002 2 1 2 4001 8001 - 2 2 - - 4003 - 8004 -
Parece ser que si esta instalado.
En cuanto al problema solucioné una parte forzando el puerto en el sip.conf y conectando los telefonos a la dirección IP local del servidor, en lugar del dyndns. El problema que presento ahora es con un solo grupo de teléfonos, ese grupo requiero que se conecte desde cualquier otro sitio que no este dentro de la VPN (aeropuertos, hoteles, etc.) El problema es que al tratar de jalar con la IP fija que tenemos o con el dyndns, de un lado se escucha y del otro lado nada. Anexo log de uno de los telefonos con el problema.
srvTelIP*CLI> sip set debug peer 1105
SIP Debugging Enabled for IP: 187.163.25.92
<--- SIP read from UDP:187.163.25.92:1125 --->
INVITE sip:1002@187.177.181.224 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.123:5060;branch=z9hG4bK-a7d08bc3
From: "Cesar" <sip:1105@187.177.181.224>;tag=f3c23f51c65b4e93o0
To: "Edith" <sip:1002@187.177.181.224>
Call-ID: da10c4a9-cd52c91b@192.168.1.123
CSeq: 101 INVITE
Max-Forwards: 70
Contact: "Cesar" <sip:1105@192.168.1.123:5060>
Expires: 240
User-Agent: Cisco/SPA303-7.4.9a
Content-Length: 395
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces
Content-Type: application/sdp
v=0
o=- 15153 15153 IN IP4 192.168.1.123
s=-
c=IN IP4 192.168.1.123
t=0 0
m=audio 16450 RTP/AVP 18 0 2 8 9 96 97 98 101
a=rtpmap:18 G729a/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (14 headers 18 lines) ---
Sending to 187.163.25.92:1125 (NAT)
Using INVITE request as basis request - da10c4a9-cd52c91b@192.168.1.123
Found peer '1105' for '1105' from 187.163.25.92:1125
<--- Reliably Transmitting (NAT) to 187.163.25.92:1125 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.123:5060;branch=z9hG4bK-a7d08bc3;received=187.163.25.92;rport=1125
From: "Cesar" <sip:1105@187.177.181.224>;tag=f3c23f51c65b4e93o0
To: "Edith" <sip:1002@187.177.181.224>;tag=as74af5f97
Call-ID: da10c4a9-cd52c91b@192.168.1.123
CSeq: 101 INVITE
Server: Digitalife PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2596f17f"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'da10c4a9-cd52c91b@192.168.1.123' in 32000 ms (Method: INVITE)
<--- SIP read from UDP:187.163.25.92:1125 --->
ACK sip:1002@187.177.181.224 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.123:5060;branch=z9hG4bK-a7d08bc3
From: "Cesar" <sip:1105@187.177.181.224>;tag=f3c23f51c65b4e93o0
To: "Edith" <sip:1002@187.177.181.224>;tag=as74af5f97
Call-ID: da10c4a9-cd52c91b@192.168.1.123
CSeq: 101 ACK
Max-Forwards: 70
Contact: "Cesar" <sip:1105@192.168.1.123:5060>
User-Agent: Cisco/SPA303-7.4.9a
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from UDP:187.163.25.92:1125 --->
INVITE sip:1002@187.177.181.224 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.123:5060;branch=z9hG4bK-6067f63
From: "Cesar" <sip:1105@187.177.181.224>;tag=f3c23f51c65b4e93o0
To: "Edith" <sip:1002@187.177.181.224>
Call-ID: da10c4a9-cd52c91b@192.168.1.123
CSeq: 102 INVITE
Max-Forwards: 70
Authorization: Digest username="1105",realm="asterisk",nonce="2596f17f",uri="sip:1002@187.177.181.224",algorithm=MD5,response="e3ad20d87ec50a2d9087d1ee9849fa8c"
Contact: "Cesar" <sip:1105@192.168.1.123:5060>
Expires: 240
User-Agent: Cisco/SPA303-7.4.9a
Content-Length: 395
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces
Content-Type: application/sdp
v=0
o=- 15153 15153 IN IP4 192.168.1.123
s=-
c=IN IP4 192.168.1.123
t=0 0
m=audio 16450 RTP/AVP 18 0 2 8 9 96 97 98 101
a=rtpmap:18 G729a/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (15 headers 18 lines) ---
Sending to 187.163.25.92:1125 (NAT)
Using INVITE request as basis request - da10c4a9-cd52c91b@192.168.1.123
Found peer '1105' for '1105' from 187.163.25.92:1125
== Using SIP RTP CoS mark 5
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 101
Found audio description format G729a for ID 18
Found audio description format PCMU for ID 0
Found audio description format G726-32 for ID 2
Found audio description format PCMA for ID 8
Found audio description format G722 for ID 9
Found unknown media description format G726-40 for ID 96
Found unknown media description format G726-24 for ID 97
Found unknown media description format G726-16 for ID 98
Found audio description format telephone-event for ID 101
Capabilities: us - 0xd1e (gsm|ulaw|alaw|g726|g729|ilbc|g726aal2), peer - audio=0x190c (ulaw|alaw|g726|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x90c (ulaw|alaw|g726|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.123:16450
Looking for 1002 in digitalife (domain 187.177.181.224)
list_route: hop: <sip:1105@192.168.1.123:5060>
<--- Transmitting (NAT) to 187.163.25.92:1125 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.123:5060;branch=z9hG4bK-6067f63;received=187.163.25.92;rport=1125
From: "Cesar" <sip:1105@187.177.181.224>;tag=f3c23f51c65b4e93o0
To: "Edith" <sip:1002@187.177.181.224>
Call-ID: da10c4a9-cd52c91b@192.168.1.123
CSeq: 102 INVITE
Server: Digitalife PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:1002@187.177.181.224:5060>
Content-Length: 0
<------------>
-- Executing [1002@digitalife:1] Macro("SIP/1105-00000677", "llamada,SIP/1002")
-- Executing [s@macro-llamada:1] NoOp("SIP/1105-00000677", "Llamada a SIP/1002 en proceso") in new stack
-- Executing [s@macro-llamada:2] Dial("SIP/1105-00000677", "SIP/1002") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/1002
-- SIP/1002-00000678 is ringing
<--- Transmitting (NAT) to 187.163.25.92:1125 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.123:5060;branch=z9hG4bK-6067f63;received=187.163.25.92;rport=1125
From: "Cesar" <sip:1105@187.177.181.224>;tag=f3c23f51c65b4e93o0
To: "Edith" <sip:1002@187.177.181.224>;tag=as614836f4
Call-ID: da10c4a9-cd52c91b@192.168.1.123
CSeq: 102 INVITE
Server: Digitalife PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:1002@187.177.181.224:5060>
Content-Length: 0
<------------>
-- SIP/1002-00000678 answered SIP/1105-00000677
Audio is at 5060
Adding codec 0x100 (g729) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x800 (g726) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (NAT) to 187.163.25.92:1125 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.123:5060;branch=z9hG4bK-6067f63;received=187.163.25.92;rport=1125
From: "Cesar" <sip:1105@187.177.181.224>;tag=f3c23f51c65b4e93o0
To: "Edith" <sip:1002@187.177.181.224>;tag=as614836f4
Call-ID: da10c4a9-cd52c91b@192.168.1.123
CSeq: 102 INVITE
Server: Digitalife PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:1002@187.177.181.224:5060>
Content-Type: application/sdp
Content-Length: 340
v=0
o=root 1819468860 1819468860 IN IP4 187.177.181.224
s=Asterisk PBX 1.8.7.2
c=IN IP4 187.177.181.224
t=0 0
m=audio 13556 RTP/AVP 18 0 8 2 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------>
-- Remotely bridging SIP/1105-00000677 and SIP/1002-00000678
<--- SIP read from UDP:187.163.25.92:1125 --->
ACK sip:1002@187.177.181.224:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.123:5060;branch=z9hG4bK-60f4c054
From: "Cesar" <sip:1105@187.177.181.224>;tag=f3c23f51c65b4e93o0
To: "Edith" <sip:1002@187.177.181.224>;tag=as614836f4
Call-ID: da10c4a9-cd52c91b@192.168.1.123
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username="1105",realm="asterisk",nonce="2596f17f",uri="sip:1002@187.177.181.224",algorithm=MD5,response="e3ad20d87ec50a2d9087d1ee9849fa8c"
Contact: "Cesar" <sip:1105@192.168.1.123:5060>
User-Agent: Cisco/SPA303-7.4.9a
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
set_destination: Parsing <sip:1105@192.168.1.123:5060> for address/port to send to
set_destination: set destination to 192.168.1.123:5060
Audio is at 5060
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 187.163.25.92:1125:
INVITE sip:1105@192.168.1.123:5060 SIP/2.0
Via: SIP/2.0/UDP 187.177.181.224:5060;branch=z9hG4bK4b0da378;rport
Max-Forwards: 70
From: "Edith" <sip:1002@187.177.181.224>;tag=as614836f4
To: "Cesar" <sip:1105@187.177.181.224>;tag=f3c23f51c65b4e93o0
Contact: <sip:1002@187.177.181.224:5060>
Call-ID: da10c4a9-cd52c91b@192.168.1.123
CSeq: 102 INVITE
User-Agent: Digitalife PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 261
v=0
o=root 1819468860 1819468861 IN IP4 192.168.0.164
s=Asterisk PBX 1.8.7.2
c=IN IP4 192.168.0.164
t=0 0
m=audio 16526 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
<--- SIP read from UDP:187.163.25.92:1125 --->
SIP/2.0 200 OK
To: "Cesar" <sip:1105@187.177.181.224>;tag=f3c23f51c65b4e93o0
From: "Edith" <sip:1002@187.177.181.224>;tag=as614836f4
Call-ID: da10c4a9-cd52c91b@192.168.1.123
CSeq: 102 INVITE
Via: SIP/2.0/UDP 187.177.181.224:5060;branch=z9hG4bK4b0da378
Contact: "Cesar" <sip:1105@192.168.1.123:5060>
Server: Cisco/SPA303-7.4.9a
Content-Length: 230
Content-Type: application/sdp
v=0
o=- 15153 15154 IN IP4 192.168.1.123
s=-
c=IN IP4 192.168.1.123
t=0 0
m=audio 16450 RTP/AVP 18 101
a=rtpmap:18 G729a/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (10 headers 12 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729a for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0xd1e (gsm|ulaw|alaw|g726|g729|ilbc|g726aal2), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.123:16450
set_destination: Parsing <sip:1105@192.168.1.123:5060> for address/port to send to
set_destination: set destination to 192.168.1.123:5060
Transmitting (NAT) to 187.163.25.92:1125:
ACK sip:1105@192.168.1.123:5060 SIP/2.0
Via: SIP/2.0/UDP 187.177.181.224:5060;branch=z9hG4bK3d11be85;rport
Max-Forwards: 70
From: "Edith" <sip:1002@187.177.181.224>;tag=as614836f4
To: "Cesar" <sip:1105@187.177.181.224>;tag=f3c23f51c65b4e93o0
Contact: <sip:1002@187.177.181.224:5060>
Call-ID: da10c4a9-cd52c91b@192.168.1.123
CSeq: 102 ACK
User-Agent: Digitalife PBX
Content-Length: 0
---
set_destination: Parsing <sip:1105@192.168.1.123:5060> for address/port to send to
set_destination: set destination to 192.168.1.123:5060
Audio is at 5060
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 187.163.25.92:1125:
INVITE sip:1105@192.168.1.123:5060 SIP/2.0
Via: SIP/2.0/UDP 187.177.181.224:5060;branch=z9hG4bK5a557150;rport
Max-Forwards: 70
From: "Edith" <sip:1002@187.177.181.224>;tag=as614836f4
To: "Cesar" <sip:1105@187.177.181.224>;tag=f3c23f51c65b4e93o0
Contact: <sip:1002@187.177.181.224:5060>
Call-ID: da10c4a9-cd52c91b@192.168.1.123
CSeq: 103 INVITE
User-Agent: Digitalife PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 265
v=0
o=root 1819468860 1819468862 IN IP4 187.177.181.224
s=Asterisk PBX 1.8.7.2
c=IN IP4 187.177.181.224
t=0 0
m=audio 13556 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
<--- SIP read from UDP:187.163.25.92:1125 --->
SIP/2.0 200 OK
To: "Cesar" <sip:1105@187.177.181.224>;tag=f3c23f51c65b4e93o0
From: "Edith" <sip:1002@187.177.181.224>;tag=as614836f4
Call-ID: da10c4a9-cd52c91b@192.168.1.123
CSeq: 103 INVITE
Via: SIP/2.0/UDP 187.177.181.224:5060;branch=z9hG4bK5a557150
Contact: "Cesar" <sip:1105@192.168.1.123:5060>
Server: Cisco/SPA303-7.4.9a
Content-Length: 230
Content-Type: application/sdp
v=0
o=- 15153 15155 IN IP4 192.168.1.123
s=-
c=IN IP4 192.168.1.123
t=0 0
m=audio 16450 RTP/AVP 18 101
a=rtpmap:18 G729a/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (10 headers 12 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729a for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0xd1e (gsm|ulaw|alaw|g726|g729|ilbc|g726aal2), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.123:16450
set_destination: Parsing <sip:1105@192.168.1.123:5060> for address/port to send to
set_destination: set destination to 192.168.1.123:5060
Transmitting (NAT) to 187.163.25.92:1125:
ACK sip:1105@192.168.1.123:5060 SIP/2.0
Via: SIP/2.0/UDP 187.177.181.224:5060;branch=z9hG4bK7672fb2b;rport
Max-Forwards: 70
From: "Edith" <sip:1002@187.177.181.224>;tag=as614836f4
To: "Cesar" <sip:1105@187.177.181.224>;tag=f3c23f51c65b4e93o0
Contact: <sip:1002@187.177.181.224:5060>
Call-ID: da10c4a9-cd52c91b@192.168.1.123
CSeq: 103 ACK
User-Agent: Digitalife PBX
Content-Length: 0
---
== Spawn extension (macro-llamada, s, 2) exited non-zero on 'SIP/1105-00000677' in macro 'llamada'
== Spawn extension (digitalife, 1002, 1) exited non-zero on 'SIP/1105-00000677'
Scheduling destruction of SIP dialog 'da10c4a9-cd52c91b@192.168.1.123' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:1105@192.168.1.123:5060> for address/port to send to
set_destination: set destination to 192.168.1.123:5060
Reliably Transmitting (NAT) to 187.163.25.92:1125:
BYE sip:1105@192.168.1.123:5060 SIP/2.0
Via: SIP/2.0/UDP 187.177.181.224:5060;branch=z9hG4bK79b11927;rport
Max-Forwards: 70
From: "Edith" <sip:1002@187.177.181.224>;tag=as614836f4
To: "Cesar" <sip:1105@187.177.181.224>;tag=f3c23f51c65b4e93o0
Call-ID: da10c4a9-cd52c91b@192.168.1.123
CSeq: 104 BYE
User-Agent: Digitalife PBX
Proxy-Authorization: Digest username="1105", realm="asterisk", algorithm=MD5, uri="sip:187.177.181.224", nonce="", response="b002c53fdd7a2ba4780d2268a542432c"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
<--- SIP read from UDP:187.163.25.92:1125 --->
SIP/2.0 200 OK
To: "Cesar" <sip:1105@187.177.181.224>;tag=f3c23f51c65b4e93o0
From: "Edith" <sip:1002@187.177.181.224>;tag=as614836f4
Call-ID: da10c4a9-cd52c91b@192.168.1.123
CSeq: 104 BYE
Via: SIP/2.0/UDP 187.177.181.224:5060;branch=z9hG4bK79b11927
Server: Cisco/SPA303-7.4.9a
Content-Length: 0
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Really destroying SIP dialog 'da10c4a9-cd52c91b@192.168.1.123' Method: ACK
No se como se podría agregar la red VPN? No sirve unicamente agregar las subredes?
Saludos y gracias .
