Error softphone en android al realizar una llamada

rvazkez

Hola!

Me encuentro con este problema a ver si me podrian apoyar.

Lo que pasa es que quiero instalar un softphone en mi telefono movil y descargue el Zoiper para android lo configure y se logro registrar en el asterisk con protocolo SIP. Lo que pasa es que cuando realizo una llamada a un telefono cisco, si suena y todo pero al contestar cuando hablo del telefono cisco se escucha en la bocina de mi telefono movil pero del movil al telefono no se escucha, se escucha puro ruido en la bocina y hace que cuelgue y termine la llamada. A y tambien cuando quiero llamar del cisco no entra la llamada en mi celular.

Anexo lo que muestra en la consola de asterisk:

== Using SIP RTP CoS mark 5
-- Executing [201@interno:1] Answer("SIP/204-00000042", "") in new stack
-- Executing [201@interno:2] Dial("SIP/204-00000042", "Sip/201,25,Tt") in new stack
== Using SIP RTP CoS mark 5
-- Called 201
-- SIP/201-00000043 is ringing
-- SIP/201-00000043 answered SIP/204-00000042
[Jan 16 17:44:51] WARNING: chan_sip.c:4051 retrans_pkt: Maximum retries exceeded on transmission YTIyNjY4MmUxYjU1MTNjOTAzZjFkMjY5OGM2ZmYwYTg. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
[Jan 16 17:44:51] WARNING: chan_sip.c:4078 retrans_pkt: Hanging up call YTIyNjY4MmUxYjU1MTNjOTAzZjFkMjY5OGM2ZmYwYTg. - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions ).
== Spawn extension (interno, 201, 2) exited non-zero on 'SIP/204-00000042'

Asi tengo mi cuentas SIP:

; teléfono cisco 7911
context=interno
callerid="201" <201>
secret=201
mailbox=201@default
type=friend
host=dynamic
canreinvite=no
call-limit=4
callgroup=1
pickupgroup=1
disallow=all
allow=gsm
allow=g729
allow=ulaw
allow=alaw
nat=no
dtfmode=rfc2833

;telefono celular con android
context=interno
callerid="204" <204>
secret=204
mailbox=204@default
type=friend
host=dynamic
canreinvite=no
call-limit=4
callgroup=1
pickupgroup=1
disallow=all
allow=gsm
allow=ulaw
allow=alaw
nat=yes

Y este es mi dialplan:

exten => 201,1,Answer()
exten => 201,n,Dial(Sip/201,25,Tt)
exten => 201,n,Voicemail(201@default)
exten => 201,n,Hangup()

exten => 204,1,Answer()
exten => 204,n,Dial(Sip/204,25,Tt)
exten => 204,n,Voicemail(204@default)
exten => 204,n,Hangup()

Que podrá ser lo que pueda estar haciendo conflicto?

Gracias, espero respuestas!

Christian_Cabrera

¿Podrias generar una llamada, descolgarla, y mientras escuchas el ruido ejecutar sip show channels para ver como se establece la llamada entre los teléfonos?

navaismo

Este error es muy común cuando no tienes configurado correctamente los settings de nat:

Jan 16 17:44:51] WARNING: chan_sip.c:4051 retrans_pkt: Maximum retries exceeded on transmission YTIyNjY4MmUxYjU1MTNjOTAzZjFkMjY5OGM2ZmYwYTg. for seqno 2 (Critical Response) -- See

Checa que tengas configurado externhost o externip, localnet etc.

rvazkez

Christian, ya ejecute el sip show channels y este fue el resultado:

Peer User/ANR Call ID Format Hold Last Message Expiry
10.1.2.7 (None) NDZjYzg0YTIxODc 0x0 (nothing) No Rx: REGISTER
10.1.2.101 (None) 0024c445-815300 0x0 (nothing) No Rx: REGISTER
10.1.2.7 204 NzJiZmY3NWI2OWN 0x8 (alaw) No Rx: INVITE
10.1.2.102 (None) 0024c444-ad7d00 0x0 (nothing) No Rx: REGISTER
10.1.2.117 201 129624ca26f6215 0x4 (ulaw) No Tx: ACK

Navaismo que valores debe de llevar los parametros externhost o externip, localnet?

navaismo

Los datos de tu red, externip es tu IP pública, si es dinámica entonces debes usar externhost tu dominio publico que apunta al pbx y localnet debe ser tu red local por ejemplo: 192.168.1.0/255.255.255.0

rvazkez

ok navaismo... ya se los puse pero no cambio en nada. :/

Aprovechando tengo una pregunta.. como le puedo hacer para que cuando yo marque una extensión salga el nombre de dicha extensión, ya que cuando marco por el ejemplo 201 en la pantalla nada mas dice 201 y quiero que aparezca el nombre del usuario. Se te cambiar la configuracion en el asterisk o en el teléfono? Tengo teléfonos cisco 7911.

navaismo

Si estas en un ambiente externo a tu LAN tambien tienes que checar los puertos de tu firewall en tu modem/router e incluso en tu PBX. Támbien sería bueno que pegaras la salida del SIP debug completo de la llamada faliida.

rvazkez

Ejecute el rtp y este fue el resultado

Sent RTP packet to 10.1.2.117:25098 (type 00, seq 035183, ts 958402496, len 000160)
Got RTP packet from 10.1.2.117:25098 (type 00, seq 035540, ts 3763232469, len 000160)
Sent RTP packet to 10.1.2.7:33000 (type 00, seq 014244, ts 3763232464, len 000160)
Got RTP packet from 10.1.2.117:25098 (type 00, seq 035541, ts 3763232629, len 000160)
Sent RTP packet to 10.1.2.7:33000 (type 00, seq 014245, ts 3763232624, len 000160)
Got RTP packet from 10.1.2.117:25098 (type 00, seq 035542, ts 3763232789, len 000160)
Sent RTP packet to 10.1.2.7:33000 (type 00, seq 014246, ts 3763232784, len 000160)
Got RTP packet from 10.1.2.117:25098 (type 00, seq 035543, ts 3763232949, len 000160)
Sent RTP packet to 10.1.2.7:33000 (type 00, seq 014247, ts 3763232944, len 000160)
Got RTP packet from 10.1.2.117:25098 (type 00, seq 035544, ts 3763233109, len 000160)
Sent RTP packet to 10.1.2.7:33000 (type 00, seq 014248, ts 3763233104, len 000160)
Got RTP packet from 10.1.2.117:25098 (type 00, seq 035545, ts 3763233269, len 000160)
Sent RTP packet to 10.1.2.7:33000 (type 00, seq 014249, ts 3763233264, len 000160)
Got RTP packet from 10.1.2.117:25098 (type 00, seq 035546, ts 3763233429, len 000160)
Sent RTP packet to 10.1.2.7:33000 (type 00, seq 014250, ts 3763233424, len 000160)
Got RTP packet from 10.1.2.117:25098 (type 00, seq 035547, ts 3763233589, len 000160)
Sent RTP packet to 10.1.2.7:33000 (type 00, seq 014251, ts 3763233584, len 000160)
== Spawn extension (interno, 201, 2) exited non-zero on 'SIP/204-0000008f'

y esto es cuando ejecute el sip debug:

-- SIP/201-00000094 answered SIP/204-00000093
Retransmitting #4 (no NAT) to 10.1.2.7:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.2.7:5060;branch=z9hG4bK-d8754z-c4740348d1a40479-1---d8754z-;received=10.1.2.7
From: <sip:204@10.1.2.37;transport=UDP>;tag=d084c77b
To: <sip:201@10.1.2.37;transport=UDP>;tag=as44fff995
Call-ID: OGIxZDcwY2ExMmY3NmZmM2Y4MmM5NjUyYWJlMzNmYmU.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.19
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:201@10.1.2.37>
Content-Type: application/sdp
Content-Length: 199

v=0
o=root 1930319287 1930319287 IN IP4 10.1.2.37
s=Asterisk PBX 1.6.2.19
c=IN IP4 10.1.2.37
t=0 0
m=audio 19448 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

Retransmitting #5 (no NAT) to 10.1.2.7:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.2.7:5060;branch=z9hG4bK-d8754z-c4740348d1a40479-1---d8754z-;received=10.1.2.7
From: <sip:204@10.1.2.37;transport=UDP>;tag=d084c77b
To: <sip:201@10.1.2.37;transport=UDP>;tag=as44fff995
Call-ID: OGIxZDcwY2ExMmY3NmZmM2Y4MmM5NjUyYWJlMzNmYmU.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.19
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:201@10.1.2.37>
Content-Type: application/sdp
Content-Length: 199

v=0
o=root 1930319287 1930319287 IN IP4 10.1.2.37
s=Asterisk PBX 1.6.2.19
c=IN IP4 10.1.2.37
t=0 0
m=audio 19448 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

Retransmitting #6 (no NAT) to 10.1.2.7:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.2.7:5060;branch=z9hG4bK-d8754z-c4740348d1a40479-1---d8754z-;received=10.1.2.7
From: <sip:204@10.1.2.37;transport=UDP>;tag=d084c77b
To: <sip:201@10.1.2.37;transport=UDP>;tag=as44fff995
Call-ID: OGIxZDcwY2ExMmY3NmZmM2Y4MmM5NjUyYWJlMzNmYmU.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.19
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:201@10.1.2.37>
Content-Type: application/sdp
Content-Length: 199

v=0
o=root 1930319287 1930319287 IN IP4 10.1.2.37
s=Asterisk PBX 1.6.2.19
c=IN IP4 10.1.2.37
t=0 0
m=audio 19448 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

[Jan 17 13:27:47] WARNING: chan_sip.c:4051 retrans_pkt: Maximum retries exceeded on transmission OGIxZDcwY2ExMmY3NmZmM2Y4MmM5NjUyYWJlMzNmYmU. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
[Jan 17 13:27:47] WARNING: chan_sip.c:4078 retrans_pkt: Hanging up call OGIxZDcwY2ExMmY3NmZmM2Y4MmM5NjUyYWJlMzNmYmU. - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions ).
Scheduling destruction of SIP dialog '5b58928a5e4b16034bb3ee9b38330657@10.1.2.37' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:201@10.1.2.117:5060;transport=udp> for address/port to send to
set_destination: set destination to 10.1.2.117, port 5060
Reliably Transmitting (no NAT) to 10.1.2.117:5060:
BYE sip:201@10.1.2.117:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.1.2.37:5060;branch=z9hG4bK229cae3f;rport
Max-Forwards: 70
From: "Cel Rafa" <sip:204@10.1.2.37>;tag=as6d31e3b1
To: <sip:201@10.1.2.117:5060;transport=udp>;tag=0024c444a2fa0456879b915a-46ce895b
Call-ID: 5b58928a5e4b16034bb3ee9b38330657@10.1.2.37
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.6.2.19
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

== Spawn extension (interno, 201, 2) exited non-zero on 'SIP/204-00000093'

<--- SIP read from UDP:10.1.2.117:51723 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.2.37:5060;branch=z9hG4bK229cae3f;rport
From: "Cel Rafa" <sip:204@10.1.2.37>;tag=as6d31e3b1
To: <sip:201@10.1.2.117:5060;transport=udp>;tag=0024c444a2fa0456879b915a-46ce895b
Call-ID: 5b58928a5e4b16034bb3ee9b38330657@10.1.2.37
Date: Thu, 17 Jan 2013 19:27:48 GMT
CSeq: 103 BYE
Server: Cisco-CP7911G/8.3.0
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '5b58928a5e4b16034bb3ee9b38330657@10.1.2.37' Method: INVITE
Really destroying SIP dialog 'OGIxZDcwY2ExMmY3NmZmM2Y4MmM5NjUyYWJlMzNmYmU.' Method: INVITE
Really destroying SIP dialog 'ODk1MTQzYzY3ZGY5ODhkNTI3YzFmOGQ4MGVkMTEzOTk.' Method: REGISTER

navaismo

Los mensajes que ves que empiezan con

Retransmitting #

Son las veces que el asterisk ha intentado hacer contacto con tu peer, y como no recibe respuesta asterisk cuelga la llamada. Para poder un poco mas a fondo se necesita el sip debug completo de la llamada.

Intenta también estableciendo: pedantic=no en el sip.conf ya que android tiene algunos detalles.

rvazkez

Ya le agregue pedantic=no y sigue igual, aqui esta el sip debug completo:

<--- SIP read from UDP:10.1.2.7:5060 --->
INVITE sip:201@10.1.2.37;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.1.2.7:5060;branch=z9hG4bK-d8754z-f144f5b432752fa3-1---d8754z-
Max-Forwards: 70
Contact: <sip:204@10.1.2.7:5060;transport=UDP>
To: <sip:201@10.1.2.37;transport=UDP>
From: <sip:204@10.1.2.37;transport=UDP>;tag=d1fae21b
Call-ID: NWQzMTZmMzA0ZmE5Yzc4ZGZjZjlkMzM5NjI1NTRiZDM.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper rev.16688
Allow-Events: presence, kpml
Content-Length: 251

v=0
o=Zoiper 0 0 IN IP4 10.1.2.7
s=Zoiper
c=IN IP4 10.1.2.7
t=0 0
m=audio 33000 RTP/AVP 0 110 3 8 98
a=rtpmap:0 PCMU/8000
a=rtpmap:110 speex/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=sendrecv

<------------->
--- (14 headers 13 lines) ---
== Using SIP RTP CoS mark 5
Sending to 10.1.2.7 : 5060 (no NAT)
Using INVITE request as basis request - NWQzMTZmMzA0ZmE5Yzc4ZGZjZjlkMzM5NjI1NTRiZDM.
Found peer '204' for '204' from 10.1.2.7:5060

<--- Reliably Transmitting (no NAT) to 10.1.2.7:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.1.2.7:5060;branch=z9hG4bK-d8754z-f144f5b432752fa3-1---d8754z-;received=10.1.2.7
From: <sip:204@10.1.2.37;transport=UDP>;tag=d1fae21b
To: <sip:201@10.1.2.37;transport=UDP>;tag=as1e901650
Call-ID: NWQzMTZmMzA0ZmE5Yzc4ZGZjZjlkMzM5NjI1NTRiZDM.
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.19
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3e91df2d"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog 'NWQzMTZmMzA0ZmE5Yzc4ZGZjZjlkMzM5NjI1NTRiZDM.' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:10.1.2.7:5060 --->
ACK sip:201@10.1.2.37;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.1.2.7:5060;branch=z9hG4bK-d8754z-f144f5b432752fa3-1---d8754z-
Max-Forwards: 70
To: <sip:201@10.1.2.37;transport=UDP>;tag=as1e901650
From: <sip:204@10.1.2.37;transport=UDP>;tag=d1fae21b
Call-ID: NWQzMTZmMzA0ZmE5Yzc4ZGZjZjlkMzM5NjI1NTRiZDM.
CSeq: 1 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:10.1.2.7:5060 --->
INVITE sip:201@10.1.2.37;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.1.2.7:5060;branch=z9hG4bK-d8754z-2bf536b143c76348-1---d8754z-
Max-Forwards: 70
Contact: <sip:204@10.1.2.7:5060;transport=UDP>
To: <sip:201@10.1.2.37;transport=UDP>
From: <sip:204@10.1.2.37;transport=UDP>;tag=d1fae21b
Call-ID: NWQzMTZmMzA0ZmE5Yzc4ZGZjZjlkMzM5NjI1NTRiZDM.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper rev.16688
Authorization: Digest username="204",realm="asterisk",nonce="3e91df2d",uri="sip:201@10.1.2.37;transport=UDP",response="5b7c0ea6230464ad2bc6d194e1729f16",algorithm=MD5
Allow-Events: presence, kpml
Content-Length: 251

<------------->
--- (15 headers 13 lines) ---
Sending to 10.1.2.7 : 5060 (no NAT)
Using INVITE request as basis request - NWQzMTZmMzA0ZmE5Yzc4ZGZjZjlkMzM5NjI1NTRiZDM.
Found peer '204' for '204' from 10.1.2.7:5060
Found RTP audio format 0
Found RTP audio format 110
Found RTP audio format 3
Found RTP audio format 8
Found RTP audio format 98
Found audio description format PCMU for ID 0
Found audio description format speex for ID 110
Found audio description format GSM for ID 3
Found audio description format PCMA for ID 8
Found audio description format iLBC for ID 98
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 10.1.2.7:33000
Looking for 201 in interno (domain 10.1.2.37)
list_route: hop: <sip:204@10.1.2.7:5060;transport=UDP>

<--- Transmitting (no NAT) to 10.1.2.7:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.2.7:5060;branch=z9hG4bK-d8754z-2bf536b143c76348-1---d8754z-;received=10.1.2.7
From: <sip:204@10.1.2.37;transport=UDP>;tag=d1fae21b
To: <sip:201@10.1.2.37;transport=UDP>
Call-ID: NWQzMTZmMzA0ZmE5Yzc4ZGZjZjlkMzM5NjI1NTRiZDM.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.19
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:201@10.1.2.37>
Content-Length: 0

<------------>
-- Executing [201@interno:1] Answer("SIP/204-000000a6", "") in new stack
Audio is at 10.1.2.37 port 14344
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP

<--- Reliably Transmitting (no NAT) to 10.1.2.7:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.2.7:5060;branch=z9hG4bK-d8754z-2bf536b143c76348-1---d8754z-;received=10.1.2.7
From: <sip:204@10.1.2.37;transport=UDP>;tag=d1fae21b
To: <sip:201@10.1.2.37;transport=UDP>;tag=as0eb08b33
Call-ID: NWQzMTZmMzA0ZmE5Yzc4ZGZjZjlkMzM5NjI1NTRiZDM.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.19
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:201@10.1.2.37>
Content-Type: application/sdp
Content-Length: 199

v=0
o=root 1256691965 1256691965 IN IP4 10.1.2.37
s=Asterisk PBX 1.6.2.19
c=IN IP4 10.1.2.37
t=0 0
m=audio 14344 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

<------------>
-- Executing [201@interno:2] Dial("SIP/204-000000a6", "Sip/201,20,Tt") in new stack
== Using SIP RTP CoS mark 5
Audio is at 10.1.2.37 port 18656
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.1.2.117:5060:
INVITE sip:201@10.1.2.117:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.1.2.37:5060;branch=z9hG4bK71ab2aee;rport
Max-Forwards: 70
From: "Cel Rafa" <sip:204@10.1.2.37>;tag=as65415c05
To: <sip:201@10.1.2.117:5060;transport=udp>
Contact: <sip:204@10.1.2.37>
Call-ID: 6ceb3a4e4ee24a9910523ebe57277968@10.1.2.37
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.19
Date: Thu, 17 Jan 2013 23:27:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 278

v=0
o=root 1858409230 1858409230 IN IP4 10.1.2.37
s=Asterisk PBX 1.6.2.19
c=IN IP4 10.1.2.37
t=0 0
m=audio 18656 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

-- Called 201

<--- SIP read from UDP:10.1.2.117:51919 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.2.37:5060;branch=z9hG4bK71ab2aee;rport
From: "Cel Rafa" <sip:204@10.1.2.37>;tag=as65415c05
To: <sip:201@10.1.2.117:5060;transport=udp>
Call-ID: 6ceb3a4e4ee24a9910523ebe57277968@10.1.2.37
Date: Thu, 17 Jan 2013 23:27:20 GMT
CSeq: 102 INVITE
Server: Cisco-CP7911G/8.3.0
Contact: <sip:201@10.1.2.117:5060;transport=udp>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Allow-Events: dialog
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---

<--- SIP read from UDP:10.1.2.117:51920 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.1.2.37:5060;branch=z9hG4bK71ab2aee;rport
From: "Cel Rafa" <sip:204@10.1.2.37>;tag=as65415c05
To: <sip:201@10.1.2.117:5060;transport=udp>;tag=0024c444a2fa04b09933761d-88196ad5
Call-ID: 6ceb3a4e4ee24a9910523ebe57277968@10.1.2.37
Date: Thu, 17 Jan 2013 23:27:21 GMT
CSeq: 102 INVITE
Server: Cisco-CP7911G/8.3.0
Contact: <sip:201@10.1.2.117:5060;transport=udp>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Allow-Events: dialog
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
-- SIP/201-000000a7 is ringing
Retransmitting #1 (no NAT) to 10.1.2.7:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.2.7:5060;branch=z9hG4bK-d8754z-2bf536b143c76348-1---d8754z-;received=10.1.2.7
From: <sip:204@10.1.2.37;transport=UDP>;tag=d1fae21b
To: <sip:201@10.1.2.37;transport=UDP>;tag=as0eb08b33
Call-ID: NWQzMTZmMzA0ZmE5Yzc4ZGZjZjlkMzM5NjI1NTRiZDM.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.19
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:201@10.1.2.37>
Content-Type: application/sdp
Content-Length: 199

v=0
o=root 1256691965 1256691965 IN IP4 10.1.2.37
s=Asterisk PBX 1.6.2.19
c=IN IP4 10.1.2.37
t=0 0
m=audio 14344 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

Retransmitting #2 (no NAT) to 10.1.2.7:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.2.7:5060;branch=z9hG4bK-d8754z-2bf536b143c76348-1---d8754z-;received=10.1.2.7
From: <sip:204@10.1.2.37;transport=UDP>;tag=d1fae21b
To: <sip:201@10.1.2.37;transport=UDP>;tag=as0eb08b33
Call-ID: NWQzMTZmMzA0ZmE5Yzc4ZGZjZjlkMzM5NjI1NTRiZDM.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.19
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:201@10.1.2.37>
Content-Type: application/sdp
Content-Length: 199

v=0
o=root 1256691965 1256691965 IN IP4 10.1.2.37
s=Asterisk PBX 1.6.2.19
c=IN IP4 10.1.2.37
t=0 0
m=audio 14344 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

rvazkez

<--- SIP read from UDP:10.1.2.117:51921 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.2.37:5060;branch=z9hG4bK71ab2aee;rport
From: "Cel Rafa" <sip:204@10.1.2.37>;tag=as65415c05
To: <sip:201@10.1.2.117:5060;transport=udp>;tag=0024c444a2fa04b09933761d-88196ad5
Call-ID: 6ceb3a4e4ee24a9910523ebe57277968@10.1.2.37
Date: Thu, 17 Jan 2013 23:27:22 GMT
CSeq: 102 INVITE
Server: Cisco-CP7911G/8.3.0
Contact: <sip:201@10.1.2.117:5060;transport=udp>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Supported: replaces,join,norefersub
Allow-Events: dialog
Content-Length: 201
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 16520 0 IN IP4 10.1.2.117
s=SIP Call
t=0 0
m=audio 27448 RTP/AVP 0 101
c=IN IP4 10.1.2.117
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

<------------->
--- (15 headers 10 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.1.2.117:27448
list_route: hop: <sip:201@10.1.2.117:5060;transport=udp>
set_destination: Parsing <sip:201@10.1.2.117:5060;transport=udp> for address/port to send to
set_destination: set destination to 10.1.2.117, port 5060
Transmitting (no NAT) to 10.1.2.117:5060:
ACK sip:201@10.1.2.117:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.1.2.37:5060;branch=z9hG4bK5cc4cb74;rport
Max-Forwards: 70
From: "Cel Rafa" <sip:204@10.1.2.37>;tag=as65415c05
To: <sip:201@10.1.2.117:5060;transport=udp>;tag=0024c444a2fa04b09933761d-88196ad5
Contact: <sip:204@10.1.2.37>
Call-ID: 6ceb3a4e4ee24a9910523ebe57277968@10.1.2.37
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.19
Content-Length: 0

-- SIP/201-000000a7 answered SIP/204-000000a6
Retransmitting #3 (no NAT) to 10.1.2.7:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.2.7:5060;branch=z9hG4bK-d8754z-2bf536b143c76348-1---d8754z-;received=10.1.2.7
From: <sip:204@10.1.2.37;transport=UDP>;tag=d1fae21b
To: <sip:201@10.1.2.37;transport=UDP>;tag=as0eb08b33
Call-ID: NWQzMTZmMzA0ZmE5Yzc4ZGZjZjlkMzM5NjI1NTRiZDM.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.19
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:201@10.1.2.37>
Content-Type: application/sdp
Content-Length: 199

v=0
o=root 1256691965 1256691965 IN IP4 10.1.2.37
s=Asterisk PBX 1.6.2.19
c=IN IP4 10.1.2.37
t=0 0
m=audio 14344 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

Retransmitting #4 (no NAT) to 10.1.2.7:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.2.7:5060;branch=z9hG4bK-d8754z-2bf536b143c76348-1---d8754z-;received=10.1.2.7
From: <sip:204@10.1.2.37;transport=UDP>;tag=d1fae21b
To: <sip:201@10.1.2.37;transport=UDP>;tag=as0eb08b33
Call-ID: NWQzMTZmMzA0ZmE5Yzc4ZGZjZjlkMzM5NjI1NTRiZDM.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.19
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:201@10.1.2.37>
Content-Type: application/sdp
Content-Length: 199

v=0
o=root 1256691965 1256691965 IN IP4 10.1.2.37
s=Asterisk PBX 1.6.2.19
c=IN IP4 10.1.2.37
t=0 0
m=audio 14344 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

Retransmitting #5 (no NAT) to 10.1.2.7:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.2.7:5060;branch=z9hG4bK-d8754z-2bf536b143c76348-1---d8754z-;received=10.1.2.7
From: <sip:204@10.1.2.37;transport=UDP>;tag=d1fae21b
To: <sip:201@10.1.2.37;transport=UDP>;tag=as0eb08b33
Call-ID: NWQzMTZmMzA0ZmE5Yzc4ZGZjZjlkMzM5NjI1NTRiZDM.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.19
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:201@10.1.2.37>
Content-Type: application/sdp
Content-Length: 199

v=0
o=root 1256691965 1256691965 IN IP4 10.1.2.37
s=Asterisk PBX 1.6.2.19
c=IN IP4 10.1.2.37
t=0 0
m=audio 14344 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

Really destroying SIP dialog 'MDU2YTcxNmNmZWRmNjhjOWEwNzdiYTRkYzg0MmIyOTU.' Method: REGISTER
Really destroying SIP dialog 'MjFhZDlkZmNmZmFmYTg0ZDE0NjdjZGM4MGFhYTE5Y2M.' Method: REGISTER
Retransmitting #6 (no NAT) to 10.1.2.7:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.2.7:5060;branch=z9hG4bK-d8754z-2bf536b143c76348-1---d8754z-;received=10.1.2.7
From: <sip:204@10.1.2.37;transport=UDP>;tag=d1fae21b
To: <sip:201@10.1.2.37;transport=UDP>;tag=as0eb08b33
Call-ID: NWQzMTZmMzA0ZmE5Yzc4ZGZjZjlkMzM5NjI1NTRiZDM.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.19
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:201@10.1.2.37>
Content-Type: application/sdp
Content-Length: 199

v=0
o=root 1256691965 1256691965 IN IP4 10.1.2.37
s=Asterisk PBX 1.6.2.19
c=IN IP4 10.1.2.37
t=0 0
m=audio 14344 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

[Jan 17 17:27:40] WARNING: chan_sip.c:4051 retrans_pkt: Maximum retries exceeded on transmission NWQzMTZmMzA0ZmE5Yzc4ZGZjZjlkMzM5NjI1NTRiZDM. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
[Jan 17 17:27:40] WARNING: chan_sip.c:4078 retrans_pkt: Hanging up call NWQzMTZmMzA0ZmE5Yzc4ZGZjZjlkMzM5NjI1NTRiZDM. - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions ).
Scheduling destruction of SIP dialog '6ceb3a4e4ee24a9910523ebe57277968@10.1.2.37' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:201@10.1.2.117:5060;transport=udp> for address/port to send to
set_destination: set destination to 10.1.2.117, port 5060
Reliably Transmitting (no NAT) to 10.1.2.117:5060:
BYE sip:201@10.1.2.117:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.1.2.37:5060;branch=z9hG4bK7bde824d;rport
Max-Forwards: 70
From: "Cel Rafa" <sip:204@10.1.2.37>;tag=as65415c05
To: <sip:201@10.1.2.117:5060;transport=udp>;tag=0024c444a2fa04b09933761d-88196ad5
Call-ID: 6ceb3a4e4ee24a9910523ebe57277968@10.1.2.37
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.6.2.19
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

== Spawn extension (interno, 201, 2) exited non-zero on 'SIP/204-000000a6'

<--- SIP read from UDP:10.1.2.117:51922 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.2.37:5060;branch=z9hG4bK7bde824d;rport
From: "Cel Rafa" <sip:204@10.1.2.37>;tag=as65415c05
To: <sip:201@10.1.2.117:5060;transport=udp>;tag=0024c444a2fa04b09933761d-88196ad5
Call-ID: 6ceb3a4e4ee24a9910523ebe57277968@10.1.2.37
Date: Thu, 17 Jan 2013 23:27:40 GMT
CSeq: 103 BYE
Server: Cisco-CP7911G/8.3.0
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '6ceb3a4e4ee24a9910523ebe57277968@10.1.2.37' Method: INVITE
Really destroying SIP dialog 'NWQzMTZmMzA0ZmE5Yzc4ZGZjZjlkMzM5NjI1NTRiZDM.' Method: INVITE

navaismo

Hola, no logro ver que haya algo mal en el sip debug. Cuando el pbx te manda el mensaje SIP de "sonando"(ringing) tu teléfono jamás contesta con el ack y de ahi se derivan los problemas y por eso cuelga.

En lo que otros compañeros lo revisan mas a fondo podrías intentar con el softphone Linphone en tu celular solo para verificar.

rvazkez

Bueno el Linphone no lo encontré disponible para mi teléfono pero instale otra aplicación y funciono perfectamente. Esto quiere decir que el problema era la aplicación. Muchas gracias. Ahora por que cuando marco a la extension de mi android no marca.. no sale la llamada?

Christian_Cabrera

¿Estás seguro de que tu android está bien registrado? Pasa frecuentemente de que los teléfonos móviles se "duermen" para ahorrar energía y datos y por ello, las llamadas nunca entran.

Asegúrate de que esté bien registrado antes de marcarle, solo para evitar esta posibilidad.

Saludos,

navaismo

rvazkez escribió:

Bueno el Linphone no lo encontré disponible para mi teléfono pero instale otra aplicación y funciono perfectamente. Esto quiere decir que el problema era la aplicación. Muchas gracias. Ahora por que cuando marco a la extension de mi android no marca.. no sale la llamada?

Hola,

Si marca, pero tu aplicación(softphone) de android no esta respondiendo correctamente, como dice Christian puede que no este bien registrado o de plano si sea un bug de la aplicación.

A grandes rasgos y sin ser el protocolo detallado lo que pasa es:

Peer-------------------------PBX
Quiero marcar -------------->
<----------------------------- Identificate primero
Me identifico ----------------->
<----------------------------- Ok, identificado correctamente
<----------------------------- Intentando Marcar
OK ------------------------------>
** <----------------------------- Ya esta sonando el lado remoto**
Ok ------------------------------->
<----------------------------- Contesto
OK ------------------------------>
RTP MEDIA <-----------------> PEER/PBX
hangup--------------------------->
<-----------------------------OK

Y el error viene cuando el PBX te envia lo del color azul, esta esperando el ACK de tu peer(color rojo), pero nunca llega y empieza a retransmitir el paquete hasta que se da por vencido y cuelga la llamada al no haber respueta desde el peer(tu softphone).

rvazkez

Lo mas seguro es que sea la aplicación por que marco y no suena ni nada.. muy hacer pruebas con otros dispositivos y otras aplicaciones a ver como me va y les comento. Muchas gracias por su ayuda, muy buen foro y buenas respuestas.