Llamada se desconecta

**rickygm** · 11-25-2011, 10:45 PM

Re: Llamada se desconecta

comienza poniendo la ip genuina y no falsa , y veo que el usuario creado no pones el nat=yes prueba con eso , si no te funciona pon aqui el debug para ver que pasa cuando se corta , pero esto huele a NAT!!

**irvingddp** · 11-30-2011, 06:28 PM

Re: Llamada se desconecta

Aqui te va el debug.
usando esta configuracion en sip.conf
-----------------------------------------
[general]
context=CCSIMURPHY
allowoverlap=no
nat=yes
externip=71.169.54.213
localnet=0.0.0.0/255.255.255.0

[test.user.mob]
Type = peer
Username = test.user.mob
Secret = secret
nat=yes
port:6512
canreinvite=very
Qualify=yes
Host = dynamic
Context = phones

[test.user2.mob]
Type = peer
Username = test.user.mob
Secret = secret
nat=yes
port:6512
canreinvite=very
Qualify=yes
Host = dynamic
Context = phones
--------------------------------------------------------------------------------------------------------

WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="66c6bcb4"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog 'ZmRmYzVlZmI0NDZiZTU4N2E5MWUzYjE2ZTY5Mzk0MmE.' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:187.184.21.19:41846 --->
REGISTER sip:71.169.54.213:6512 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.109:41846;branch=z9hG4bK-d8754z-6fbbdd0d41f89eed-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:test.user.mob@187.184.21.19:41846;transport=u dp;rinstance=c430d07101cb7f28>
To: <sip:test.user.mob@71.169.54.213:6512>
From: <sip:test.user.mob@71.169.54.213:6512>;tag=687347f e
Call-ID: ZmRmYzVlZmI0NDZiZTU4N2E5MWUzYjE2ZTY5Mzk0MmE.
CSeq: 12 REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite 4 release 4.1 stamp 63214
Authorization: Digest username="test.user.mob",realm="asterisk",nonce="6 6c6bcb4",uri="sip:71.169.54.213:6512",response="56 a4a01926bb5bff732bd2fc7e543051",algorithm=MD5
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 187.184.21.19 : 41846 (NAT)
Reliably Transmitting (NAT) to 187.184.21.19:41846:
OPTIONS sip:test.user.mob@187.184.21.19:41846;transport=ud p;rinstance=c430d07101cb7f28 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.16:6512;branch=z9hG4bK5e0c85d4;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.0.16:6512>;tag=as43e9ccbd
To: <sip:test.user.mob@187.184.21.19:41846;transport=u dp;rinstance=c430d07101cb7f28>
Contact: <sip:asterisk@192.168.0.16:6512>
Call-ID: 516e4a591fa261ab2a90138448ddcee3@192.168.0.16
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.20
Date: Thu, 01 Dec 2011 01:20:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

---

<--- Transmitting (NAT) to 187.184.21.19:41846 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.109:41846;branch=z9hG4bK-d8754z-6fbbdd0d41f89eed-1---d8754z-;received=187.184.21.19;rport=41846
From: <sip:test.user.mob@71.169.54.213:6512>;tag=687347f e
To: <sip:test.user.mob@71.169.54.213:6512>;tag=as65b53 03a
Call-ID: ZmRmYzVlZmI0NDZiZTU4N2E5MWUzYjE2ZTY5Mzk0MmE.
CSeq: 12 REGISTER
Server: Asterisk PBX 1.6.2.20
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 60
Contact: <sip:test.user.mob@187.184.21.19:41846;transport=u dp;rinstance=c430d07101cb7f28>;expires=60
Date: Thu, 01 Dec 2011 01:20:09 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog 'ZmRmYzVlZmI0NDZiZTU4N2E5MWUzYjE2ZTY5Mzk0MmE.' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:187.184.21.19:41846 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.16:6512;branch=z9hG4bK5e0c85d4;rport=651 2;received=71.169.54.213
Contact: <sip:192.168.1.109:41846>
To: <sip:test.user.mob@187.184.21.19:41846;transport=u dp;rinstance=c430d07101cb7f28>;tag=a8ea38b2
From: "asterisk"<sip:asterisk@192.168.0.16:6512>;tag=as4 3e9ccbd
Call-ID: 516e4a591fa261ab2a90138448ddcee3@192.168.0.16
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Supported: replaces
User-Agent: X-Lite 4 release 4.1 stamp 63214
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Really destroying SIP dialog '516e4a591fa261ab2a90138448ddcee3@192.168.0.16' Method: OPTIONS
Retransmitting #4 (NAT) to 187.184.21.19:41846:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.109:41846;branch=z9hG4bK-d8754z-02afe691a1064cfa-1---d8754z-;received=187.184.21.19;rport=41846
From: <sip:test.user.mob@71.169.54.213:6512>;tag=33fb229 c
To: <sip:406@71.169.54.213:6512>;tag=as6e88e372
Call-ID: ZGJjNWEyZmE5YzFkMzE0OWQxYmZlMzMxNjlkMDYyOGI.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.20
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:406@192.168.0.16:6512>
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 1038569553 1038569553 IN IP4 192.168.0.16
s=Asterisk PBX 1.6.2.20
c=IN IP4 192.168.0.16
t=0 0
m=audio 10794 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
Retransmitting #5 (NAT) to 187.184.21.19:41846:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.109:41846;branch=z9hG4bK-d8754z-02afe691a1064cfa-1---d8754z-;received=187.184.21.19;rport=41846
From: <sip:test.user.mob@71.169.54.213:6512>;tag=33fb229 c
To: <sip:406@71.169.54.213:6512>;tag=as6e88e372
Call-ID: ZGJjNWEyZmE5YzFkMzE0OWQxYmZlMzMxNjlkMDYyOGI.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.20
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:406@192.168.0.16:6512>
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 1038569553 1038569553 IN IP4 192.168.0.16
s=Asterisk PBX 1.6.2.20
c=IN IP4 192.168.0.16
t=0 0
m=audio 10794 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
Really destroying SIP dialog 'IEigEG.ek-TmsS3YyGTTsaVEz0vxrsec' Method: REGISTER
Retransmitting #6 (NAT) to 187.184.21.19:41846:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.109:41846;branch=z9hG4bK-d8754z-02afe691a1064cfa-1---d8754z-;received=187.184.21.19;rport=41846
From: <sip:test.user.mob@71.169.54.213:6512>;tag=33fb229 c
To: <sip:406@71.169.54.213:6512>;tag=as6e88e372
Call-ID: ZGJjNWEyZmE5YzFkMzE0OWQxYmZlMzMxNjlkMDYyOGI.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.20
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:406@192.168.0.16:6512>
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 1038569553 1038569553 IN IP4 192.168.0.16
s=Asterisk PBX 1.6.2.20
c=IN IP4 192.168.0.16
t=0 0
m=audio 10794 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
[Nov 30 17:20:18] WARNING[12335]: chan_sip.c:4051 retrans_pkt: Maximum retries exceeded on transmission ZGJjNWEyZmE5YzFkMzE0OWQxYmZlMzMxNjlkMDYyOGI. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/ ... nsmissions
[Nov 30 17:20:18] WARNING[12335]: chan_sip.c:4078 retrans_pkt: Hanging up call ZGJjNWEyZmE5YzFkMzE0OWQxYmZlMzMxNjlkMDYyOGI. - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/ ... nsmissions ).
Scheduling destruction of SIP dialog '1f1866e973bc574b36d9c61c41e7e676@192.168.0.16' in 79296 ms (Method: INVITE)
set_destination: Parsing <sip:test.user2.mob@200.79.138.41:19525;transport= UDP;ob> for address/port to send to
set_destination: set destination to 200.79.138.41, port 19525
Reliably Transmitting (NAT) to 200.79.138.41:19525:
BYE sip:test.user2.mob@200.79.138.41:19525;transport=U DP;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.0.16:6512;branch=z9hG4bK10e78496;rport
Max-Forwards: 70
From: "test.user.mob" <sip:test.user.mob@192.168.0.16:6512>;tag=as5dcbbc bb
To: <sip:test.user2.mob@200.79.138.41:19525;transport= UDP;ob>;tag=.Ioymo16hf.9D0fUAdIVltERfIzepInW
Call-ID: 1f1866e973bc574b36d9c61c41e7e676@192.168.0.16
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.6.2.20
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

---
== Spawn extension (phones, 406, 1) exited non-zero on 'SIP/test.user.mob-00000059'

<--- SIP read from UDP:200.79.138.41:19525 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.16:6512;rport=6512;received=71.169.54.21 3;branch=z9hG4bK10e78496
Call-ID: 1f1866e973bc574b36d9c61c41e7e676@192.168.0.16
From: "test.user.mob" <sip:test.user.mob@192.168.0.16>;tag=as5dcbbcbb
To: <sip:test.user2.mob@200.79.138.41;ob>;tag=.Ioymo16 hf.9D0fUAdIVltERfIzepInW
CSeq: 103 BYE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog 'ZGJjNWEyZmE5YzFkMzE0OWQxYmZlMzMxNjlkMDYyOGI.' Method: INVITE
Really destroying SIP dialog '1f1866e973bc574b36d9c61c41e7e676@192.168.0.16' Method: INVITE

<--- SIP read from UDP:200.79.138.41:19525 --->

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

<--- SIP read from UDP:187.184.21.19:41846 --->

<------------->
localhost*CLI>

**Christian Cabrera** · 11-30-2011, 06:41 PM

Re: Llamada se desconecta

Cambia el localnet para que coincida con tu red local real y prueba de nuevo. Segun veo, tienes el localnet abierto a todo.

**irvingddp** · 12-02-2011, 06:55 PM

Re: Llamada se desconecta

Ya lo he intentado inclusive dando de altas varias redes en localnet se sigues desconectando alos 10 segundos, si le quito la opcion de localnet solo tengo audio en un lado pero la llamada nos e corta.

**irvingddp** · 12-02-2011, 07:36 PM

Re: Llamada se desconecta

LO ENCONTREEEE!!!
EL DETALLE ERA QUE EN EL EXTERN IP NO TENIA DEFINIDO EL PUERTO!!
Ya funciona perfectamente.

Gracias por su tiempo y aportaciones!!

**Christian Cabrera** · 12-02-2011, 09:04 PM

Re: Llamada se desconecta

PLOP! jajaja, muy bien!

**yannier** · 12-10-2013, 02:24 PM

me pasa algo parecido, configuré lo que antes era una extensión de un servidor remoto (en brazil) de la misma compañía en México, como troncal SIP, me había funcionado inidireccionalmente (solo podían hacer llamadas, no recibirlas), pero ahora aunque escucho que timbra allá no suena. Al ver el ip show peers decía unmanaged, añadí qualify y ahora dice UNREACHABLE.
Por el mensaje supongo que no estoy alcanzando el puerto del servidor remoto, pero es raro, puedo hacer telnet a ese puerto desde mi servidor local.

Ayudaaa.

**hector** · 12-10-2013, 03:26 PM

Recuerda que SIP (señalización) viaja por el puerto 5060 y el RTP (audio) puede elegir cualquier puerto de los definidos en rtp.conf.. probablemente puedas describir un poco mas la infraestructura y como configuraste el nateo para ver por donde se ataca

Llamada se desconecta

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