Extensiones remotas no timbran en Asterisk 11.4.0

peponciobot

Buen día, el motivo de mi participación en el foro es el siguiente:

Tengo instalado Asterisk 11.4.0 en Ubuntu Server 12.04 dentro de la red local 10.1.2.0, Asterisk tiene asignada la ip 10.1.2.13, tengo 6 extensiones locales y 4 remotas, tengo ademas funcionando un sistema de mensajería instantánea mediante lo siguiente en el dialplan:

exten => _.,1,NoOp(SMS receiving dialplan invoked)
exten => _.,n,NoOp(To ${MESSAGE(to)})
exten => _.,n,NoOp(From ${MESSAGE(from)})
exten => _.,n,NoOp(Body ${MESSAGE(body)})
exten => _.,n,Set(ACTUALTO=${CUT(MESSAGE(to),@,1)})
exten => _.,n,MessageSend(${ACTUALTO},${MESSAGE(from)})
exten => _.,n,NoOp(Send status is ${MESSAGE_SEND_STATUS})
exten => _.,n,GotoIf($["${MESSAGE_SEND_STATUS}" != "SUCCESS"]?sendfailedmsg)
exten => _.,n,Hangup()

Cuando se envían mensajes desde extensiones remotas a extensiones locales,entre locales y entre remotas, los mensajes llegan sin ningún problema, no así el caso de las llamadas, en el caso de que una extensión remota llame a una local todo funciona bien, con un poco de retraso en el audio pero es mínimo. En caso de una llamada de una extensión local a una remota todo cambia, en la extensión local se tiene el tono de que la otra extensión esta timbrando y se espera que sea contestada, pero no es así porque la otra extensión ni siquiera timbra. Es el mismo caso entre extensiones remotas, las extensiones no timbran.

Tengo habilitado también el canal motif para conexión con g-talk, y aqui si funciona perfecto el audio y el timbrado.
Aquí lo que me aparece en la cli.

CLI> == Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
-- Executing [556@internas:1] Dial("SIP/102-00000002", "Motif/google/cuenta@gmail.com,,r") in new stack
-- Called Motif/google/cuentacosmica1000@gmail.com
-- Motif/cuenta@gmail.com/Talk.v105A85A48CB-1534 is proceeding passing it to SIP/102-00000002
-- Motif/cuenta@gmail.com/Talk.v105A85A48CB-1534 answered SIP/102-00000002
-- Locally bridging SIP/102-00000002 and Motif/cuenta@gmail.com/Talk.v105A85A48CB-1534
== Spawn extension (internas, 556, 1) exited non-zero on 'SIP/102-00000002'
[Feb 20 11:59:45] WARNING: chan_sip.c:4169 retrans_pkt: Retransmission timeout reached on transmission NWNjYTgyMmYwMDVlYjMwOTU4OTJhZjczMDMxY2NkZGM. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 16960ms with no response

Entre llamadas entre extensiones remotas esto es lo que aparece es una llamada de la 101 a 102

*CLI> == Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
-- Executing [102@internas:1] Dial("SIP/101-00000000", "SIP/102,20,Ttr") in new stack
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
-- Called SIP/102
-- Nobody picked up in 20000 ms

El servidor esta conectado a un router tp-link TL-R4000+ el cual a su vez recibe la conexion de un modem de telmex que solo actua como traductor de adsl hacia el router el cual tiene la red privada 10.1.2.0, tengo midominio.dyndns.org.

En el router he abierto los puertos 5060 TCP/UDP y del 10000 al 20000 TCP/UDP direccionados a la ip de Asterisk.

Asi es como esta configurado mi archivo sip_nat.conf

nat = auto_force_rport,force_rport,comedia
localnet = 10.1.2.0/255.255.255.0
externhost = midominio.dyndns.org
domain = midominio.dyndns.org
localhost = 10.1.2.13

Y el archivo rtp.conf

icesupport = yes
rtpstart = 10000
rtpend = 20000

De antemano muchas gracias, si es necesario mas datos estaré pendiente.

navaismo

Hola, tu log muestra que si esta llamando a la extensión, quizas el paquete se pierde en algún salto de tu ruta ip. También si puedes agregar el SIP DEBUG podriamos ver hacia que IP esta enviando los INVITES y si estos son contestados.

peponciobot

Hola, gracias por tu respuesta aqui te dejo el log del SIP DEBUG lo puse en dos partes porque me aparece demasiado log.

Via: SIP/2.0/UDP x.x.x.x:44786;branch=z9hG4bK-d8754z-75d8daeda92539c1-1---d8754z-;received=x.x.x.x
From: <sip:101@midominio.dyndns.org:5060;transport=UDP>;tag=ef4fe13a
To: <sip:102@midominio.dyndns.org:5060;transport=UDP>;tag=as1be4137f
Call-ID: YzM3YTAxMDVmZDk2ZDZjZWY4MzVmOTg3NWU5ZmZhMWE.
CSeq: 2 INVITE
Server: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

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

<--- Transmitting (no NAT) to x.x.x.x:44786 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP x.x.x.x:44786;branch=z9hG4bK-d8754z-75d8daeda92539c1-1---d8754z-;received=x.x.x.x
From: <sip:101@midominio.dyndns.org:5060;transport=UDP>;tag=ef4fe13a
To: <sip:102@midominio.dyndns.org:5060;transport=UDP>;tag=as1be4137f
Call-ID: YzM3YTAxMDVmZDk2ZDZjZWY4MzVmOTg3NWU5ZmZhMWE.
CSeq: 2 CANCEL
Server: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '6cb6e2831ed1a194710a404b54822727@10.1.2.10:5060' in 21952 ms (Method: INVITE)
== Spawn extension (internas, 102, 1) exited non-zero on 'SIP/101-00000000'
Retransmitting #4 (no NAT) to x.x.x.x:5060:
INVITE sip:102@x.x.x.x:5060;rinstance=b0290732f7a86536;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.1.2.10:5060;branch=z9hG4bK7005bb37
Max-Forwards: 70
From: <sip:Jose@10.1.2.10>;tag=as0faa60c0
To: <sip:102@x.x.x.x:5060;rinstance=b0290732f7a86536;transport=UDP>
Contact: <sip:Jose@10.1.2.10:5060>
Call-ID: 6cb6e2831ed1a194710a404b54822727@10.1.2.10:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.4.0
Date: Thu, 20 Feb 2014 22:42:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 1915

v=0
o=root 918471766 918471766 IN IP4 10.1.2.10
s=Asterisk PBX 11.4.0
c=IN IP4 10.1.2.10
b=CT:384
t=0 0
m=audio 13160 RTP/AVP 0 4 3 3 8 8 112 5 7 18 110 97 111 9 102 115 116 117 96 100 107 108 10 118 119 101
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:3 GSM/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:117 speex/16000
a=rtpmap:96 SILK/8000
a=fmtp:96 maxaveragebitrate=10000
a=fmtp:96 usedtx=0
a=fmtp:96 useinbandfec=1
a=rtpmap:100 SILK/12000
a=fmtp:100 maxaveragebitrate=12000
a=fmtp:100 usedtx=0
a=fmtp:100 useinbandfec=1
a=rtpmap:107 SILK/16000
a=fmtp:107 maxaveragebitrate=20000
a=fmtp:107 usedtx=0
a=fmtp:107 useinbandfec=1
a=rtpmap:108 SILK/24000
a=fmtp:108 maxaveragebitrate=30000
a=fmtp:108 usedtx=0
a=fmtp:108 useinbandfec=1
a=rtpmap:10 L16/8000
a=rtpmap:118 L16/16000
a=rtpmap:119 speex/32000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 18498 RTP/AVP 31 34 34 98 99 104
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=fmtp:34 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:34 H263/90000
a=fmtp:34 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:98 h263-1998/90000
a=fmtp:98 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:99 H264/90000
a=fmtp:99 redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0
a=rtpmap:104 MP4V-ES/90000
a=sendrecv
m=text 18452 RTP/AVP 105 106
a=rtpmap:105 RED/1000
a=fmtp:105 106/106/106
a=rtpmap:106 T140/1000
a=sendrecv

<--- SIP read from UDP:x.x.x.x:44786 --->
PUBLISH sip:101@midominio.dyndns.org:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:44786;branch=z9hG4bK-d8754z-a781af2eda53ac08-1---d8754z-
Max-Forwards: 70
Contact: <sip:101@x.x.x.x:44786;transport=UDP>
To: <sip:101@midominio.dyndns.org:5060;transport=UDP>
From: <sip:101@midominio.dyndns.org:5060;transport=UDP>;tag=ec370725
Call-ID: ZWYxMDI1MTQ0YzljMjFjYjgxMTcxMWQ3ODU1ZGQ2MzI.
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.2.21357 r21367
Event: presence
Allow-Events: presence, kpml
Content-Length: 272

<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf" entity="sip:101@midominio.dyndns.org:5060;transport=UDP"> <tuple id="101" > <status><basic>open</basic></status> <note>Online</note> </tuple>
</presence>
<------------->
--- (16 headers 3 lines) ---
Sending to x.x.x.x:44786 (no NAT)

<--- Transmitting (no NAT) to x.x.x.x:44786 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP x.x.x.x:44786;branch=z9hG4bK-d8754z-a781af2eda53ac08-1---d8754z-;received=x.x.x.x
From: <sip:101@midominio.dyndns.org:5060;transport=UDP>;tag=ec370725
To: <sip:101@midominio.dyndns.org:5060;transport=UDP>;tag=as67ebbcaf
Call-ID: ZWYxMDI1MTQ0YzljMjFjYjgxMTcxMWQ3ODU1ZGQ2MzI.
CSeq: 1 PUBLISH
Server: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
Really destroying SIP dialog 'ZWYxMDI1MTQ0YzljMjFjYjgxMTcxMWQ3ODU1ZGQ2MzI.' Method: PUBLISH

<--- SIP read from UDP:x.x.x.x:44786 --->
SUBSCRIBE sip:101@midominio.dyndns.org:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:44786;branch=z9hG4bK-d8754z-50de3ecfd6b237b6-1---d8754z-
Max-Forwards: 70
Contact: <sip:101@x.x.x.x:44786;transport=UDP>
To: <sip:101@midominio.dyndns.org:5060;transport=UDP>
From: <sip:101@midominio.dyndns.org:5060;transport=UDP>;tag=ae6e1237
Call-ID: YzQwYmI5YjAyN2M1OTY1NzdjZGYyMjEyY2QzZDIwMjY.
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.2.21357 r21367
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (16 headers 0 lines) ---
Sending to x.x.x.x:44786 (no NAT)
Creating new subscription
Sending to x.x.x.x:44786 (no NAT)
list_route: hop: <sip:101@x.x.x.x:44786;transport=UDP>
Found peer '101' for '101' from x.x.x.x:44786

<--- Transmitting (no NAT) to x.x.x.x:44786 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP x.x.x.x:44786;branch=z9hG4bK-d8754z-50de3ecfd6b237b6-1---d8754z-;received=x.x.x.x
From: <sip:101@midominio.dyndns.org:5060;transport=UDP>;tag=ae6e1237
To: <sip:101@midominio.dyndns.org:5060;transport=UDP>;tag=as30f496da
Call-ID: YzQwYmI5YjAyN2M1OTY1NzdjZGYyMjEyY2QzZDIwMjY.
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="28329a19"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog 'YzQwYmI5YjAyN2M1OTY1NzdjZGYyMjEyY2QzZDIwMjY.' in 11968 ms (Method: SUBSCRIBE)
Retransmitting #1 (no NAT) to x.x.x.x:44786:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP x.x.x.x:44786;branch=z9hG4bK-d8754z-75d8daeda92539c1-1---d8754z-;received=x.x.x.x
From: <sip:101@midominio.dyndns.org:5060;transport=UDP>;tag=ef4fe13a
To: <sip:102@midominio.dyndns.org:5060;transport=UDP>;tag=as1be4137f
Call-ID: YzM3YTAxMDVmZDk2ZDZjZWY4MzVmOTg3NWU5ZmZhMWE.
CSeq: 2 INVITE
Server: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<--- SIP read from UDP:x.x.x.x:44786 --->
ACK sip:102@midominio.dyndns.org:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:44786;branch=z9hG4bK-d8754z-75d8daeda92539c1-1---d8754z-
Max-Forwards: 70
To: <sip:102@midominio.dyndns.org:5060;transport=UDP>;tag=as1be4137f
From: <sip:101@midominio.dyndns.org:5060;transport=UDP>;tag=ef4fe13a
Call-ID: YzM3YTAxMDVmZDk2ZDZjZWY4MzVmOTg3NWU5ZmZhMWE.
CSeq: 2 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog 'YzM3YTAxMDVmZDk2ZDZjZWY4MzVmOTg3NWU5ZmZhMWE.' Method: ACK

<--- SIP read from UDP:x.x.x.x:44786 --->
SUBSCRIBE sip:101@midominio.dyndns.org:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:44786;branch=z9hG4bK-d8754z-535185b278bcb8b6-1---d8754z-
Max-Forwards: 70
Contact: <sip:101@x.x.x.x:44786;transport=UDP>
To: <sip:101@midominio.dyndns.org:5060;transport=UDP>
From: <sip:101@midominio.dyndns.org:5060;transport=UDP>;tag=ae6e1237
Call-ID: YzQwYmI5YjAyN2M1OTY1NzdjZGYyMjEyY2QzZDIwMjY.
CSeq: 2 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.2.21357 r21367
Authorization: Digest username="101",realm="asterisk",nonce="28329a19",uri="sip:101@midominio.dyndns.org:5060;transport=UDP",response="121418713178869e270387bbe2690378",algorithm=MD5
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

peponciobot

<------------->
--- (17 headers 0 lines) ---
Creating new subscription
Sending to x.x.x.x:44786 (no NAT)
Found peer '101' for '101' from x.x.x.x:44786

<--- Transmitting (no NAT) to x.x.x.x:44786 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP x.x.x.x:44786;branch=z9hG4bK-d8754z-535185b278bcb8b6-1---d8754z-;received=x.x.x.x
From: <sip:101@midominio.dyndns.org:5060;transport=UDP>;tag=ae6e1237
To: <sip:101@midominio.dyndns.org:5060;transport=UDP>;tag=as30f496da
Call-ID: YzQwYmI5YjAyN2M1OTY1NzdjZGYyMjEyY2QzZDIwMjY.
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
Really destroying SIP dialog 'YzQwYmI5YjAyN2M1OTY1NzdjZGYyMjEyY2QzZDIwMjY.' Method: SUBSCRIBE

<------------->
--- (8 headers 0 lines) ---
Retransmitting #5 (no NAT) to x.x.x.x:5060:
INVITE sip:102@x.x.x.x:5060;rinstance=b0290732f7a86536;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.1.2.10:5060;branch=z9hG4bK7005bb37
Max-Forwards: 70
From: <sip:Jose@10.1.2.10>;tag=as0faa60c0
To: <sip:102@x.x.x.x:5060;rinstance=b0290732f7a86536;transport=UDP>
Contact: <sip:Jose@10.1.2.10:5060>
Call-ID: 6cb6e2831ed1a194710a404b54822727@10.1.2.10:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.4.0
Date: Thu, 20 Feb 2014 22:42:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 1915

Reliably Transmitting (no NAT) to x.x.x.x:44786:
OPTIONS sip:101@x.x.x.x:44786;rinstance=2fb6b7fd6b6e981f;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.1.2.10:5060;branch=z9hG4bK040f2ab2
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.1.2.10>;tag=as7998783e
To: <sip:101@x.x.x.x:44786;rinstance=2fb6b7fd6b6e981f;transport=UDP>
Contact: <sip:asterisk@10.1.2.10:5060>
Call-ID: 5d0d91eb4fc3a28159115d7a437df4b4@10.1.2.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.4.0
Date: Thu, 20 Feb 2014 22:43:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<--- SIP read from UDP:x.x.x.x:44786 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.2.10:5060;branch=z9hG4bK040f2ab2;received=189.243.65.153
Contact: <sip:192.168.0.5:44786>
To: <sip:101@x.x.x.x:44786;rinstance=2fb6b7fd6b6e981f;transport=UDP>;tag=650f284a
From: "asterisk"<sip:asterisk@10.1.2.10>;tag=as7998783e
Call-ID: 5d0d91eb4fc3a28159115d7a437df4b4@10.1.2.10:5060
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, timer, X-cisco-serviceuri
User-Agent: Z 3.2.21357 r21367
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog '5d0d91eb4fc3a28159115d7a437df4b4@10.1.2.10:5060' Method: OPTIONS
Retransmitting #6 (no NAT) to x.x.x.x:5060:
INVITE sip:102@x.x.x.x:5060;rinstance=b0290732f7a86536;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.1.2.10:5060;branch=z9hG4bK7005bb37
Max-Forwards: 70
From: <sip:Jose@10.1.2.10>;tag=as0faa60c0
To: <sip:102@x.x.x.x:5060;rinstance=b0290732f7a86536;transport=UDP>
Contact: <sip:Jose@10.1.2.10:5060>
Call-ID: 6cb6e2831ed1a194710a404b54822727@10.1.2.10:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.4.0
Date: Thu, 20 Feb 2014 22:42:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 1915

[Feb 20 16:43:21] WARNING: chan_sip.c:4169 retrans_pkt: Retransmission timeout reached on transmission6cb6e2831ed1a194710a404b54822727@10.1.2.10:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 21953ms with no response
Really destroying SIP dialog '6cb6e2831ed1a194710a404b54822727@10.1.2.10:5060' Method: INVITE

<--- SIP read from UDP:x.x.x.x:5060 --->

<------------->
Really destroying SIP dialog 'YjY1MjMyMjJjMzJkMjIwMzBlMDExZTc1YjBkNTQyMmQ.' Method: REGISTER

navaismo

Habilita el NAT=yes en las extensiones remotas. Intenta de nuevo y peganos el debug otra vez por favor.

peponciobot

Muchas gracias por tu atencion por el momento ya no tengo acceso al servidor Asterisk, lo curios es que aunque no realice llamadas el SIP DEBUG sigue apareciendo. Mañana subiré el log que me has solicitado. Muchas Gracias.

peponciobot

Aqui esta el SIP DEBUG con nat = yes, mi servidor esta en la 10.1.2.10, de hecho siempre ha tenido esa dirección, una disculpa.

<------------>
-- Executing [102@internas:1] Dial("SIP/101-00000006", "SIP/102,20,Ttr") in new stack
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
We think we can do text
And we have a text rtp object
Audio is at 13644
Video is at 10.1.2.10:11998
Lets set up the text sdp
Text is at 0.0.0.0:11628
Adding codec 100003 (ulaw) to SDP
Adding codec 100001 (g723) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100005 (g726aal2) to SDP
Adding codec 100006 (adpcm) to SDP
Adding codec 100007 (lpc10) to SDP
Adding codec 100008 (g729) to SDP
Adding codec 100009 (speex) to SDP
Adding codec 100010 (ilbc) to SDP
Adding codec 100011 (g726) to SDP
Adding codec 100012 (g722) to SDP
Adding codec 100013 (siren7) to SDP
Adding codec 100014 (siren14) to SDP
Adding codec 100015 (g719) to SDP
Adding codec 100016 (speex16) to SDP
Adding codec 100017 (testlaw) to SDP
Adding codec 100017 (testlaw) to SDP
Adding codec 100018 (silk8) to SDP
Adding codec 100018 (silk12) to SDP
Adding codec 100018 (silk16) to SDP
Adding codec 100018 (silk24) to SDP
Adding codec 100019 (slin) to SDP
Adding codec 100020 (slin12) to SDP
Adding codec 100021 (slin16) to SDP
Adding text codec 400001 (red) to SDP
Adding text codec 400002 (t140) to SDP
Adding codec 100022 (slin24) to SDP
Adding codec 100023 (slin32) to SDP
Adding codec 100024 (slin44) to SDP
Adding codec 100025 (slin48) to SDP
Adding codec 100026 (slin96) to SDP
Adding codec 100027 (slin192) to SDP
Adding codec 100028 (speex32) to SDP
Adding video codec 200001 (h261) to SDP
Adding video codec 200002 (h263) to SDP
Adding video codec 200002 (h263) to SDP
Adding video codec 200003 (h263p) to SDP
Adding video codec 200004 (h264) to SDP
Adding video codec 200005 (mpeg4) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 189.215.218.127:5060:
INVITE sip:102@189.215.218.127:5060;rinstance=6b4146651f3716b8;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.1.2.10:5060;branch=z9hG4bK02a34589
Max-Forwards: 70
From: <sip:Jose@10.1.2.10>;tag=as4de0d6d7
To: <sip:102@189.215.218.127:5060;rinstance=6b4146651f3716b8;transport=UDP>
Contact: <sip:Jose@10.1.2.10:5060>
Call-ID: 51484ae332268f7a17c3a5190feeb6ef@10.1.2.10:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.4.0
Date: Fri, 21 Feb 2014 18:52:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 1915

v=0
o=root 189381203 189381203 IN IP4 10.1.2.10
s=Asterisk PBX 11.4.0
c=IN IP4 10.1.2.10
b=CT:384
t=0 0
m=audio 13644 RTP/AVP 0 4 3 3 8 8 112 5 7 18 110 97 111 9 102 115 116 117 96 100 107 108 10 118 119 101
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:3 GSM/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:117 speex/16000
a=rtpmap:96 SILK/8000
a=fmtp:96 maxaveragebitrate=10000
a=fmtp:96 usedtx=0
a=fmtp:96 useinbandfec=1
a=rtpmap:100 SILK/12000
a=fmtp:100 maxaveragebitrate=12000
a=fmtp:100 usedtx=0
a=fmtp:100 useinbandfec=1
a=rtpmap:107 SILK/16000
a=fmtp:107 maxaveragebitrate=20000
a=fmtp:107 usedtx=0
a=fmtp:107 useinbandfec=1
a=rtpmap:108 SILK/24000
a=fmtp:108 maxaveragebitrate=30000
a=fmtp:108 usedtx=0
a=fmtp:108 useinbandfec=1
a=rtpmap:10 L16/8000
a=rtpmap:118 L16/16000
a=rtpmap:119 speex/32000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 11998 RTP/AVP 31 34 34 98 99 104
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=fmtp:34 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:34 H263/90000
a=fmtp:34 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:98 h263-1998/90000
a=fmtp:98 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:99 H264/90000
a=fmtp:99 redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0
a=rtpmap:104 MP4V-ES/90000
a=sendrecv
m=text 11628 RTP/AVP 105 106
a=rtpmap:105 RED/1000
a=fmtp:105 106/106/106
a=rtpmap:106 T140/1000
a=sendrecv

-- Called SIP/102

<--- Transmitting (no NAT) to 189.243.80.116:6275 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 189.243.80.116:6275;branch=z9hG4bK-d8754z-fed0cdc1e985fd41-1---d8754z-;received=189.243.80.116
From: <sip:101@midominio.dyndns.org:5060;transport=UDP>;tag=2f56ff7f
To: <sip:102@midominio.dyndns.org:5060;transport=UDP>;tag=as61179916
Call-ID: ZGUxN2M5NWNiZTUyY2VlZGIzZTc2NjA5YWIyZDIzZDE.
CSeq: 2 INVITE
Server: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:102@10.1.2.10:5060>
Content-Length: 0

<------------>
Retransmitting #1 (no NAT) to 189.215.218.127:5060:
INVITE sip:102@189.215.218.127:5060;rinstance=6b4146651f3716b8;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.1.2.10:5060;branch=z9hG4bK02a34589
Max-Forwards: 70
From: <sip:Jose@10.1.2.10>;tag=as4de0d6d7
To: <sip:102@189.215.218.127:5060;rinstance=6b4146651f3716b8;transport=UDP>
Contact: <sip:Jose@10.1.2.10:5060>
Call-ID: 51484ae332268f7a17c3a5190feeb6ef@10.1.2.10:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.4.0
Date: Fri, 21 Feb 2014 18:52:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 1915

Retransmitting #2 (no NAT) to 189.215.218.127:5060:
INVITE sip:102@189.215.218.127:5060;rinstance=6b4146651f3716b8;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.1.2.10:5060;branch=z9hG4bK02a34589
Max-Forwards: 70
From: <sip:Jose@10.1.2.10>;tag=as4de0d6d7
To: <sip:102@189.215.218.127:5060;rinstance=6b4146651f3716b8;transport=UDP>
Contact: <sip:Jose@10.1.2.10:5060>
Call-ID: 51484ae332268f7a17c3a5190feeb6ef@10.1.2.10:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.4.0
Date: Fri, 21 Feb 2014 18:52:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 1915

peponciobot

Retransmitting #3 (no NAT) to x.x.x.x:5060:
INVITE sip:102@x.x.x.x:5060;rinstance=6b4146651f3716b8;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.1.2.10:5060;branch=z9hG4bK02a34589
Max-Forwards: 70
From: <sip:Jose@10.1.2.10>;tag=as4de0d6d7
To: <sip:102@x.x.x.x:5060;rinstance=6b4146651f3716b8;transport=UDP>
Contact: <sip:Jose@10.1.2.10:5060>
Call-ID: 51484ae332268f7a17c3a5190feeb6ef@10.1.2.10:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.4.0
Date: Fri, 21 Feb 2014 18:52:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 1915

Reliably Transmitting (no NAT) to 187.237.105.27:50358:
OPTIONS sip:130@10.25.87.170:5060;rinstance=a9a45af2a6ce9dd9;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.1.2.10:5060;branch=z9hG4bK695b4e9f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.1.2.10>;tag=as521bb251
To: <sip:130@10.25.87.170:5060;rinstance=a9a45af2a6ce9dd9;transport=UDP>
Contact: <sip:asterisk@10.1.2.10:5060>
Call-ID: 59e91e12035dde223a03f9c660033b4a@10.1.2.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.4.0
Date: Fri, 21 Feb 2014 18:52:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

ubuntuCLI> sip set debug off
SIP Debugging Disabled
[Feb 21 12:52:12] WARNING: chan_sip.c:4169 retrans_pkt: Retransmission timeout reached on transmission51484ae332268f7a17c3a5190feeb6ef@10.1.2.10:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 10751ms with no response
[Feb 21 12:52:12] WARNING: chan_sip.c:4198 retrans_pkt: Hanging up call 51484ae332268f7a17c3a5190feeb6ef@10.1.2.10:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [102@internas:2] VoiceMail("SIP/101-00000006", "102@default") in new stack
-- <SIP/101-00000006> Playing 'vm-intro.gsm' (language 'en')
[Feb 21 12:52:12] NOTICE[C-00000003]: channel.c:4257 __ast_read: Dropping incompatible voice frame on SIP/101-00000006 of format ulaw since our native format has changed to (gsm)
-- <SIP/101-00000006> Playing 'beep.gsm' (language 'en')
-- Recording the message
-- x=0, open writing: /var/spool/asterisk/voicemail/default/102/tmp/lcYof6 format: wav, 0x7fe6d4097e98
ubuntuCLI>

navaismo

peponciobot escribió:

Mlo curios es que aunque no realice llamadas el SIP DEBUG sigue apareciendo.

El SIP debug muestra todos los mensajes SIP que recibe el PBX, es por eso que ves muchos mensajes y se incrementa proporcionalmente a la cantidad de teléfonos que poseas, todos estos envían mensajes de ACK, OK, OPTIONS REGISTER etc.

Sobre el problema parece que el PBX está tratando la IP pública como privada:

Retransmitting #3 (no NAT) to 189.215.218.127:5060:

Ya que envía sin NAT a la IP pública del teléfono los INVITES y estos jamás reciben un ACK de la parte remota. Intenta quitando el domain y el externhost y usando externip solo para ver si rutea los paquetes con NAT, y también checa tu ruta IP del server.

Saludos.

peponciobot

Ya lo hice con solo externip y sigue sin timbrar. Con respecto lo de la ruta IP del server ya esta definida, ya le he movido bastante y siempre con los mismos resultados. seguiré intentando

peponciobot

Estaba experimentando con varias modificaciones al archivo sip_nat.conf, intente quitando todo, elimine sip_nat.conf , y el comportamiento es el mismo, como si Asterisk no detectara para nada la opción NAT. mis extensiones remotas se siguen registrando y eso que no tengo especificado el dominio en ninguna otra parte mas que en sip_nat.conf.

¿Qué podrá ser?

navaismo

Ese archivo es algo que introdujeron las versiones enlatadas de FreePBX si no estas usando GUI setealo en el sip.conf, puede ser que tu sip.conf no este incluyendo ese archivo. Y si usas una GUI intenta cambiar esos settings en el freepbx.

Saludos,

peponciobot

Ya incrusté los parametros de sip_nat.conf dentro de sip.conf, pero sigo con los mismos resultados.

navaismo

Pega la salida del comando: sip show settings

peponciobot

Aquí esta

sip show settings

Global Settings:

UDP Bindaddress: 0.0.0.0:5060
TCP SIP Bindaddress: Disabled
TLS SIP Bindaddress: Disabled
Videosupport: Yes
Textsupport: Yes
Ignore SDP sess. ver.: No
AutoCreate Peer: Off
Match Auth Username: No
Allow unknown access: No
Allow subscriptions: Yes
Allow overlap dialing: Yes
Allow promisc. redir: No
Enable call counters: No
SIP domain support: Yes
Realm. auth: No
Our auth realm asterisk
Use domains as realms: No
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: No
User Agent: Asterisk PBX 11.4.0
SDP Session Name: Asterisk PBX 11.4.0
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Trust RPID: No
Send RPID: No
Legacy userfield parse: No
Send Diversion: Yes
Caller ID: asterisk
From: Domain:
Record SIP history: Off
Call Events: On
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: -1
SIP realtime: Disabled
Qualify Freq : 60000 ms
Q.850 Reason header: No
Store SIP_CAUSE: No

Network Settings:

SIP address remapping: Enabled using externhost
Externhost: midominio.dvrdns.org
Externaddr: 189.215.218.127:0
Externrefresh: 10
Localnet: 10.1.2.0/255.255.255.0

Global Signalling Settings:

Default Settings:

Allowed transports: UDP
Outbound transport: UDP
Context: internas
Record on feature: automon
Record off feature: automon
Force rport: Yes
DTMF: rfc2833
Qualify: 2000
Keepalive: 0
Use ClientCode: No
Progress inband: Never
Language:
Tone zone: <Not set>
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: asterisk

navaismo

Estoy viendo que la IP a donde esta enviando la retransmisión es la misma de asterisk, si le das el comando sip show peers las remotas tienen la ip de tu dominio? Si es así lo mas probable es que tu router/firewall este enmascarando la IP de los clientes, si usas iptables es fácil pero si tu router no lo soporta creo que lo conveniente será poner una vpn.

peponciobot

Muchas gracias, voy a tratar reinstalando todo, luego comento.

peponciobot

Hola, lo he solucionado cambiando de router, ahora las las extensiones remotas timbran y el audio esta bien.

Muchas gracias por la ayuda.