[RESUELTO] Ayuda para conectar dos centrales asterisk con troncal sip

**Marco Zink** · 04-28-2013, 01:06 PM

Creo que es más fácil si haces troncales y los pones como friend:

en sip.conf o sip_additional.conf

En servidor A

Código:

[servidorB]
host=(la ip del servidor B)
name=servidorA
secret=password compartido
type=friend
allow=all
context=el contexto al que permitirás acceso // si vas a homologarlos, probablemente quieras utilizar from-internal

En el servidor B

Código:

[servidorA]
host=(ip del servidor A)
name=servidorB
secret=password compartido
type=friend
allow=all
context=el contexto al que permitirás acceso // mismo caso que el anterior

Chécalo y dinos si pudiste.

Saludos

**luquetty** · 04-28-2013, 04:04 PM

En sip.conf cree

central1

[192.168.1.202]
host=192.168.1.202
name=192.168.1.201
secret=password
type=friend
allow=all
context=default

central2

[192.168.1.201]
host=192.168.1.201
name=192.168.1.202
secret=password
type=friend
allow=all
context=default

en extension.conf
[default]
exten = _XX.,1,Dial(sip/${EXTEN}@192.168.1.202,30,Ttm)

en que le estoy errando ???

**Marco Zink** · 04-28-2013, 04:56 PM

El dial correcto sería así:

exten= _XX.,1,Dial(SIP/192.168.1.202/${EXTEN},30,Ttm)

aunque te recomendaría que utilizaras nombres en vez de direcciones en los nombres de las troncales.

tu dial podría quedar así:

exten= _XX.,1,Dial(SIP/central2/${EXTEN},30,Ttm)

**luquetty** · 04-28-2013, 05:10 PM

Desde ya gracias, creo que vamos avansando pero me responde congestion

-- Executing [2000@default:1] Dial("SIP/3000-00000002", "SIP/192.168.1.201/2000,30,Ttm") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/192.168.1.201/2000
-- Started music on hold, class 'default', on SIP/3000-00000002
[Apr 28 19:07:24] NOTICE[1179]: chan_sip.c:20959 handle_response_invite: Failed to authenticate on INVITE to '"3000" <sip:3000@192.168.1.202>;tag=as166f1e29'
-- SIP/192.168.1.201-00000003 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Stopped music on hold on SIP/3000-00000002
-- Auto fallthrough, channel 'SIP/3000-00000002' status is 'CONGESTION'

**Marco Zink** · 04-28-2013, 05:28 PM

Tienes declarado el parámetro localnet en [general] de sip.conf?

**luquetty** · 04-28-2013, 05:30 PM

no lo creo, no lo hice, como se hace??

[general]
localnet=192.168.1.0/255.255.255.0

**Marco Zink** · 04-28-2013, 06:09 PM

Si, así es como se hace, pruébalo y pega aquí un debug sip de ambos servidores cuando intentar realizar la llamada.

**luquetty** · 04-28-2013, 06:19 PM

si ya lo hice,

intento llamar de la central1 a la central2

en la central1 me dice

[Apr 28 14:06:48] WARNING[1859]: chan_sip.c:10067 process_sdp_a_audio: Got Siren7 offer at 240 00 bps, but only 32000 bps supported; ignoring.
[Apr 28 14:06:48] WARNING[2194]: pbx.c:4250 pbx_substitute_variables_helper_full: Error in ext ension logic (missing '}')
-- Executing [3000@default:1] Dial("SIP/2001-0000002c", "sip/central2/") in new stack
== Using SIP RTP CoS mark 5
-- Called sip/central2/
[Apr 28 14:06:48] NOTICE[1859]: chan_sip.c:20959 handle_response_invite: Failed to authenticat e on INVITE to '"2001" <sip:2001@192.168.1.201>;tag=as1c817489'
-- SIP/central2-0000002d is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Auto fallthrough, channel 'SIP/2001-0000002c' status is 'CONGESTION'

en la central2 nada,

**luquetty** · 04-28-2013, 06:50 PM

central 1

sip.conf

[plantilla](!)
type=friend
secret=lucas
qualify=yes
nat=yes
host=dynamic
canreinvite=no
mailbox= prueba@hotmail.com
context= default

[2000](plantilla)
[2001](plantilla)

[central2]
host=192.168.1.202
name=central1
secret=password
type=friend
allow=all
context=default
qualify=yes

[general]
localnet=192.168.1.0/255.255.255.0

extensions.conf

[default]
; A internos
exten = _2XX.,1,Dial(sip/${EXTEN},30,Ttm)
exten = _2XX.,2,Playback(ss-noservice)
exten = _2XX.,3,Hangup
exten = _3XX.,1,Dial(sip/central2/${EXTEN],30,tTM)

central2

sip.conf

[3000]
type=friend
secret=lucas
qualify=yes
nat=no
host=dynamic
canreinvite=no
mailbox= prueba@hotmail.com

[central1]
host=192.168.1.201
name=central2
secret=password
type=friend
allow=all
context=default
qualify=yes

[general]
localnet=192.168.1.0/255.255.255.0

extensions.conf

[default]
exten= _2XX.,1,Dial(SIP/central1/${EXTEN},30,Ttm)
exten= _3XX.,1,Dial(SIP/${EXTEN},30,Ttm)

**Marco Zink** · 04-28-2013, 06:54 PM

Ejecuta

Código:

sip set debug on

En ambos servidores y pega lo que te salga en el cli de Asterisk.

**luquetty** · 04-28-2013, 07:01 PM

Central 1

<--- Reliably Transmitting (NAT) to 192.168.1.4:5070 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.1.4:5070;branch=z9hG4bKPjf12010730d344c828 bb1af043bc90bcb;received=19 2.168.1.4;rport=5070
From: sip:2001@192.168.1.201;tag=94fa21886bff4db39110993 2020f7652
To: sip:3000@192.168.1.201;tag=as2f5f21fe
Call-ID: 13a4f2a9a52447db9d63f8faa04ec2bb
CSeq: 13950 INVITE
Server: Asterisk PBX 1.8.21.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21
Content-Length: 0

<------------>
Really destroying SIP dialog '69de00a801ce6cc443dce0e674aee3a0@192.168.1.201:50 60' Method: INV ITE
Retransmitting #1 (NAT) to 192.168.1.4:5070:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.1.4:5070;branch=z9hG4bKPjf12010730d344c828 bb1af043bc90bcb;received=19 2.168.1.4;rport=5070
From: sip:2001@192.168.1.201;tag=94fa21886bff4db39110993 2020f7652
To: sip:3000@192.168.1.201;tag=as2f5f21fe
Call-ID: 13a4f2a9a52447db9d63f8faa04ec2bb
CSeq: 13950 INVITE
Server: Asterisk PBX 1.8.21.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21
Content-Length: 0

---

<--- SIP read from UDP:192.168.1.4:5070 --->
ACK sip:3000@192.168.1.201 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:5070;rport;branch=z9hG4bKPjf12010730d3 44c828bb1af043bc90bcb
Max-Forwards: 70
From: sip:2001@192.168.1.201;tag=94fa21886bff4db39110993 2020f7652
To: sip:3000@192.168.1.201;tag=as2f5f21fe
Call-ID: 13a4f2a9a52447db9d63f8faa04ec2bb
CSeq: 13950 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '13a4f2a9a52447db9d63f8faa04ec2bb' Method: ACK

<--- SIP read from UDP:192.168.1.4:5070 --->

CENTRAL2

<--- SIP read from UDP:192.168.1.201:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK1916b532;received =192.168.1.202;rport=5060
From: "asterisk" <sip:asterisk@192.168.1.202>;tag=as68311fa5
To: <sip:192.168.1.201>;tag=as441827c4
Call-ID: 6b0f7b3e11623852480d12653a74a8c9@192.168.1.202:506 0
CSeq: 102 OPTIONS
Server: Asterisk PBX 1.8.21.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:192.168.1.201:5060>
Accept: application/sdp
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '6b0f7b3e11623852480d12653a74a8c9@192.168.1.202:50 60' Method: OPTIONS
Retransmitting #3 (no NAT) to 190.122.72.253:53321:
OPTIONS sip:3000@190.122.72.253:53321;rinstance=5add4181bb f19751;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK26a4ec60
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.1.202>;tag=as0d2ae947
To: <sip:3000@190.122.72.253:53321;rinstance=5add4181b bf19751;transport=UDP>
Contact: <sip:asterisk@192.168.1.202:5060>
Call-ID: 7bcecbf206c7bc9320e04376542d5b78@192.168.1.202:506 0
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.21.0
Date: Mon, 29 Apr 2013 00:00:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

---
Retransmitting #4 (no NAT) to 190.122.72.253:53321:
OPTIONS sip:3000@190.122.72.253:53321;rinstance=5add4181bb f19751;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK26a4ec60
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.1.202>;tag=as0d2ae947
To: <sip:3000@190.122.72.253:53321;rinstance=5add4181b bf19751;transport=UDP>
Contact: <sip:asterisk@192.168.1.202:5060>
Call-ID: 7bcecbf206c7bc9320e04376542d5b78@192.168.1.202:506 0
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.21.0
Date: Mon, 29 Apr 2013 00:00:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

---
Really destroying SIP dialog '7bcecbf206c7bc9320e04376542d5b78@192.168.1.202:50 60' Method: OPTIONS

**navaismo** · 04-28-2013, 07:33 PM

Tienes varias cosas ahí:

En tus trunk usa disallow=all y luego usa allow=ulaw y añade alas y GSM.
En tu dial plan de la central 1 tienes un corchete en lugar de una llave en el dial.
Y finalmente al parecer esta fallando al utenttificar el Invite. Checa las contraseñas y que ambos realms sean los mismos si usas tipo friend.

El debug parece incompleto trata de capturarlo todo.

**luquetty** · 04-28-2013, 07:50 PM

Ya hice los cambios
[central2]
host=192.168.1.202
name=central1
secret=password
type=friend
context=default
qualify=yes
disallow=all
allow=ulaw
allow=gsm

[central1]
host=192.168.1.201
name=central2
secret=password
type=friend
context=default
qualify=yes
disallow=all
allow=ulaw
allow=gsm

Central1

<--- SIP read from UDP:192.168.1.202:5060 --->
ACK sip:2001@192.168.1.201 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK3b42e3cb;rport
Max-Forwards: 70
From: "3000" <sip:3000@192.168.1.202>;tag=as198d19bd
To: <sip:2001@192.168.1.201>;tag=as0df947c0
Contact: <sip:3000@192.168.1.202:5060>
Call-ID: 3c036b6a5b32dddb58548e0e6613a673@192.168.1.202:506 0
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.21.0
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '3c036b6a5b32dddb58548e0e6613a673@192.168.1.202:50 60' Method: ACK

<--- SIP read from UDP:192.168.1.4:5070 --->
REGISTER sip:2001@192.168.1.201 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:5070;rport;branch=z9hG4bKPj4f61ce1ec83 b4ec59106cd34987a5cd7
Max-Forwards: 70
From: <sip:2001@192.168.1.201>;tag=d29741af4fcf41978fb7e 7670384ee08
To: <sip:2001@192.168.1.201>
Call-ID: 802c20bc482943299346cb4130d258d6
CSeq: 18203 REGISTER
User-Agent: AdoreSoftphone
Contact: <sip:2001@192.168.1.4:5070>
Expires: 300
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 192.168.1.4:5070 (NAT)

<--- Transmitting (NAT) to 192.168.1.4:5070 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.4:5070;branch=z9hG4bKPj4f61ce1ec83b4ec59 106cd34987a5cd7;received=19 2.168.1.4;rport=5070
From: <sip:2001@192.168.1.201>;tag=d29741af4fcf41978fb7e 7670384ee08
To: <sip:2001@192.168.1.201>;tag=as5c4a7869
Call-ID: 802c20bc482943299346cb4130d258d6
CSeq: 18203 REGISTER
Server: Asterisk PBX 1.8.21.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2399f341"
Content-Length: 0

Scheduling destruction of SIP dialog '802c20bc482943299346cb4130d258d6' in 32000 ms (Method: R EGISTER)
<--- SIP read from UDP:192.168.1.4:5070 --->
REGISTER sip:2001@192.168.1.201 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:5070;rport;branch=z9hG4bKPjb493013deff 340c6a4b9da8e0fb5bac0
Max-Forwards: 70
From: <sip:2001@192.168.1.201>;tag=d29741af4fcf41978fb7e 7670384ee08
To: <sip:2001@192.168.1.201>
Call-ID: 802c20bc482943299346cb4130d258d6
CSeq: 18204 REGISTER
User-Agent: AdoreSoftphone
Contact: <sip:2001@192.168.1.4:5070>
Expires: 300
Authorization: Digest username="2001", realm="asterisk", nonce="2399f341", uri="sip:2001@192.1 68.1.201", response="8717306df7315613303d000e4f037aee", algorithm=MD5
Content-Length: 0

--- (12 headers 0 lines) ---
Sending to 192.168.1.4:5070 (NAT)
Reliably Transmitting (NAT) to 192.168.1.4:5070:
OPTIONS sip:2001@192.168.1.4:5070 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK51f7ae44;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.1.201>;tag=as7106eed6
To: <sip:2001@192.168.1.4:5070>
Contact: <sip:asterisk@192.168.1.201:5060>
Call-ID: 0780b7270634f0755f44a67a3844ebee@192.168.1.201:506 0
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.21.0
Date: Sun, 28 Apr 2013 17:56:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

---

<--- Transmitting (NAT) to 192.168.1.4:5070 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.4:5070;branch=z9hG4bKPjb493013deff340c6a 4b9da8e0fb5bac0;received=19 2.168.1.4;rport=5070
From: <sip:2001@192.168.1.201>;tag=d29741af4fcf41978fb7e 7670384ee08
To: <sip:2001@192.168.1.201>;tag=as5c4a7869
Call-ID: 802c20bc482943299346cb4130d258d6
CSeq: 18204 REGISTER
Server: Asterisk PBX 1.8.21.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 300
Contact: <sip:2001@192.168.1.4:5070>;expires=300
Date: Sun, 28 Apr 2013 17:56:27 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '1ec8f97c6c7357e05e64368135353add@192.168.1.201:50 60' in 6400 ms (Method: NOTIFY)
Reliably Transmitting (NAT) to 192.168.1.4:5070:
NOTIFY sip:2001@192.168.1.4:5070 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK2bff97fd;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.1.201>;tag=as512a745c
To: <sip:2001@192.168.1.4:5070>
Contact: <sip:asterisk@192.168.1.201:5060>
Call-ID: 1ec8f97c6c7357e05e64368135353add@192.168.1.201:506 0
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX 1.8.21.0
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 93

Messages-Waiting: no
Message-Account: sip:asterisk@192.168.1.201
Voice-Message: 0/0 (0/0)

---
Scheduling destruction of SIP dialog '802c20bc482943299346cb4130d258d6' in 32000 ms (Method: R EGISTER)

<--- SIP read from UDP:192.168.1.4:5070 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.201:5060;rport=5060;received=192.168.1.2 01;branch=z9hG4bK51f7ae44
Call-ID: 0780b7270634f0755f44a67a3844ebee@192.168.1.201:506 0
From: "asterisk" <sip:asterisk@192.168.1.201>;tag=as7106eed6
To: <sip:2001@192.168.1.4>
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, message/sipfrag;version= 2.0, application/im-iscomposing+xml, text/plain
Supported: 100rel, norefersub
Allow-Events: presence, refer
User-Agent: AdoreSoftphone
Content-Type: application/sdp
Content-Length: 718

v=0
o=- 3576174092 3576174092 IN IP4 192.168.1.4
s=pjmedia
c=IN IP4 192.168.1.4
t=0 0
m=audio 4000 RTP/AVP 103 102 104 117 3 0 8 9 118 119 18 4 2 15 125 126 101
a=rtcp:4001 IN IP4 192.168.1.4
a=rtpmap:103 speex/16000
a=rtpmap:102 speex/8000
a=rtpmap:104 speex/32000
a=rtpmap:117 iLBC/8000
a=fmtp:117 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:118 AMR/8000
a=rtpmap:119 AMR-WB/16000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:15 G728/8000
a=rtpmap:125 G7221/16000
a=fmtp:125 bitrate=24000
a=rtpmap:126 G7221/16000
a=fmtp:126 bitrate=32000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (13 headers 29 lines) ---
Really destroying SIP dialog '0780b7270634f0755f44a67a3844ebee@192.168.1.201:50 60' Method: OPT IONS

<--- SIP read from UDP:192.168.1.4:5070 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.201:5060;rport=5060;received=192.168.1.2 01;branch=z9hG4bK2bff97fd
Call-ID: 1ec8f97c6c7357e05e64368135353add@192.168.1.201:506 0
From: "asterisk" <sip:asterisk@192.168.1.201>;tag=as512a745c
To: <sip:2001@192.168.1.4>
CSeq: 102 NOTIFY
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '1ec8f97c6c7357e05e64368135353add@192.168.1.201:50 60' Method: NOT IFY
home*CLI> sip set debug off

Central2

<--- Transmitting (NAT) to 192.168.1.9:5066 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.9:5066;branch=z9hG4bK-e126f7e4;received=192.168.1.9;rport=5066
From: "3001" <sip:3001@192.168.1.202>;tag=5138e8786dd0a5d3o0
To: "3001" <sip:3001@192.168.1.202>;tag=as2fe25d1e
Call-ID: ccf9627c-fa28d39f@192.168.1.9
CSeq: 84 REGISTER
Server: Asterisk PBX 1.8.21.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 60
Contact: <sip:3001@192.168.1.9:5066>;expires=60
Date: Mon, 29 Apr 2013 00:41:29 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '52fc475669f24fce125544f32519c96e@192.168.1.202:50 60' in 6400 ms (Method: NOTIFY)
Reliably Transmitting (NAT) to 192.168.1.9:5066:
NOTIFY sip:3001@192.168.1.9:5066 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK49b80ec6;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.1.202>;tag=as53b7d78c
To: <sip:3001@192.168.1.9:5066>
Contact: <sip:asterisk@192.168.1.202:5060>
Call-ID: 52fc475669f24fce125544f32519c96e@192.168.1.202:506 0
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX 1.8.21.0
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 93

<--- SIP read from UDP:192.168.1.9:5066 --->
SIP/2.0 200 OK
To: <sip:3001@192.168.1.9:5066>;tag=3730351cb34f217f i0
From: "asterisk" <sip:asterisk@192.168.1.202>;tag=as53b7d78c
Call-ID: 52fc475669f24fce125544f32519c96e@192.168.1.202:506 0
CSeq: 102 NOTIFY
Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK49b80ec6
Server: Sipura/SPA941-4.1.8
Content-Length: 0

<--- Transmitting (NAT) to 192.168.1.9:5066 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.9:5066;branch=z9hG4bK-9081e9f;received=192.168.1.9;rport=5066
From: "3001" <sip:3001@192.168.1.202>;tag=5138e8786dd0a5d3o0
To: <sip:192.168.1.202>;tag=as6d44df28
Call-ID: 554ddcd8-5f93cf3@192.168.1.9
CSeq: 93 NOTIFY
Server: Asterisk PBX 1.8.21.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

**navaismo** · 04-28-2013, 08:09 PM

Siguen incompletos esos debugs no se ven los invites. Para que los tengas completos en un archivo has:

asterisk -rnvvvvdd | tee debugc1.txt

En ambos pbx y sube los archivos resultantes a pastebin.com

[RESUELTO] Ayuda para conectar dos centrales asterisk con troncal sip

Anuncio