Request 102: Match Found

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Sergio Aparicio

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Jan 2, 2007, 9:43:25 AM1/2/07
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Hola,
 

         en el log de debug me encuentro todo el rato las secuencias que adjunto. ¿ Es normal ?, la verdad es que no he tenido ningún problema pero me preocupa.
 

Dec 31 20:37:27 DEBUG[3014] chan_sip.c: Stopping retransmission on ' 194a3cb14fa42be0...@192.168.123.64' of Request 102: Match Found

Dec 31 20:37:29 DEBUG[3014] chan_sip.c: Stopping retransmission on ' 706562c16b9e22b0...@192.168.123.64' of Request 102: Match Found

Dec 31 20:37:29 DEBUG[3014] chan_sip.c: Stopping retransmission on ' 3952babb5a1f1de2...@192.168.123.64' of Request 102: Match Found

Dec 31 20:37:34 DEBUG[3014] chan_sip.c: Stopping retransmission on ' 19b4276339128045...@192.168.123.64' of Request 102: Match Found

Dec 31 20:37:35 DEBUG[3014] chan_sip.c: Stopping retransmission on ' 5acb9d104f2611e3...@192.168.123.64' of Request 102: Match Found

Dec 31 20:37:45 DEBUG[3014] chan_sip.c: Stopping retransmission on ' 3f6fabe17a627cc1...@192.168.123.64' of Request 102: Match Found

Dec 31 20:37:49 DEBUG[3014] chan_sip.c: Stopping retransmission on ' 01c59e5d14e23d39...@192.168.123.64' of Request 102: Match Found

Dec 31 20:37:49 DEBUG[3014] chan_sip.c: Stopping retransmission on ' 38a0933858922916...@192.168.123.64' of Request 102: Match Found

Dec 31 20:37:49 DEBUG[3014] chan_sip.c: Stopping retransmission on ' 5312f7753058ce56...@192.168.123.64' of Request 102: Match Found

Dec 31 20:37:49 DEBUG[3014] chan_sip.c: Stopping retransmission on ' 39665b8505bcb854...@192.168.123.64' of Request 102: Match Found

Dec 31 20:37:49 DEBUG[3014] chan_sip.c: Stopping retransmission on ' 4a4e2cd2630fa29d...@192.168.123.64' of Request 102: Match Found

Dec 31 20:37:50 DEBUG[3014] chan_sip.c: Stopping retransmission on ' 16b1da4756642303...@192.168.123.64' of Request 102: Match Found

Dec 31 20:37:50 DEBUG[3014] chan_sip.c: Stopping retransmission on ' 6668bd5d6faeb5c8...@192.168.123.64' of Request 102: Match Found

Dec 31 20:37:53 DEBUG[3014] chan_sip.c: Stopping retransmission on ' 3201f9ad568c2686...@192.168.123.64' of Request 102: Match Found

Dec 31 20:37:57 DEBUG[3014] chan_sip.c: Stopping retransmission on ' 379ca32e40564b85...@192.168.123.64' of Request 102: Match Found

Dec 31 20:37:57 DEBUG[3014] chan_sip.c: Stopping retransmission on ' 735066b07818147c...@192.168.123.64' of Request 102: Match Found

Dec 31 20:38:01 DEBUG[3014] chan_sip.c: Stopping retransmission on ' 7ca98c49-c0...@192.168.123.111' of Response 2: Match Found

Dec 31 20:38:04 DEBUG[3014] chan_sip.c: Auto destroying call ' 41455b06-c0...@192.168.123.103'

Dec 31 20:38:07 DEBUG[3014] chan_sip.c: Setting NAT on RTP to 0

Dec 31 20:38:07 DEBUG[3014] chan_sip.c: Stopping retransmission on ' 4b170e06-c0...@192.168.123.104' of Response 1: Match Found

Dec 31 20:38:07 DEBUG[3014] chan_sip.c: Setting NAT on RTP to 0

Dec 31 20:38:07 DEBUG[3014] chan_sip.c: Checking SIP call limits for device 104

Dec 31 20:38:07 DEBUG[3014] chan_sip.c: build_route: Contact hop: <sip:1...@192.168.123.104 :5060;user=phone>


--
Saludos / Sam

TelecoSilvia

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Jan 3, 2007, 10:34:08 AM1/3/07
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No lo recuerdo exactamente, pero es normal y no pasa nada. No se pero
me parece que era un problema de algun tipo de telefono como
grandstream que no se registraba correctamente con la opción qualify,
enviaba las respuestas al puerto 0 en vez de al 5060. Se resuelve con
un nat del puerto, o tirando el telefono :P

Sergio Aparicio

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Jan 4, 2007, 6:51:40 AM1/4/07
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Hola,
 
             hombre es que este log me ocurre en todos los telefonos (ST2030), no los voy a tirar :(
 
 
                        Saludos / Sam
 
2007/1/3, TelecoSilvia <teleco...@gmail.com>:

Elio Rojano

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Jan 4, 2007, 8:37:18 AM1/4/07
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Es por que no está registrado bien.

A ver, un teléfono no tiene porque está registrado o no está registrado. También puede darse el caso que esté registrado mal.

Comprueba el contexto del registro de la extensión en el sip.conf

Haz un sip debug y comprueba el mensaje que te devuelve cuando registras un teléfono, ya verás como hay algo que no está bien.

Seguramente te falte el campo 'realm' en el sip con el campo "dominio" en el teléfono, o alguna tontería similar que no le guste al protocolo SIP y por eso te este soltando esos mensajes.

Un saludo,



2007/1/4, Sergio Aparicio <sergio....@gmail.com>:

Sergio Aparicio

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Jan 7, 2007, 11:38:26 AM1/7/07
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Hola Elio,
 
 
         el realm lo cambie como me dijiste pero sigo teniendo el problema del  Request 102: Match Found, cuando el telefono intenta registrarse veo un  "SIP/2.0 401 Unauthorized", pero haciendo un sip show peers aparece registrado y es totalmente funcional salvo el Request 102: Match Found.
 
           Saludos /Sam

 

Transmitting (no NAT) to 192.168.123.102:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.123.102:5060;branch=z9hG4bK7929148681981435932;received= 192.168.123.102
From: <sip:102@F1CONNECTING;user=phone>;tag=c0a80101-84d3
To: <sip:102@F1CONNECTING;user=phone>;tag=as363c0e19
Call-ID: 84d3-c0a...@192.168.123.102
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm="F1CONNECTING", nonce="3a79f58b"
Content-Length: 0


---
Scheduling destruction of call '84d3-c0a...@192.168.123.102' in 15000 ms
Obelisk*CLI>
<-- SIP read from 192.168.123.102:5060:
REGISTER sip:192.168.123.65 SIP/2.0
Via: SIP/2.0/UDP 192.168.123.102:5060;branch=z9hG4bK9142030814803754864
From: <sip:102@F1CONNECTING;user=phone>;tag=c0a80101-84d3
To: <sip:102@F1CONNECTING;user=phone>
Call-ID: 84d3-c0a...@192.168.123.102
CSeq: 2 REGISTER
Max-Forwards: 70
Expires: 0
Contact: *
Authorization: Digest username="102", realm="F1CONNECTING", nonce="3a79f58b", uri="sip:192.168.123.65 ", response="83d965f148d9a0edb418356a970cc33f", algorithm=MD5
User-Agent: THOMSON ST2030 hw3 fw1.50 00-0E-50-4E-62-F5
Content-Length: 0


--- (12 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.123.102 : 5060 (non-NAT)
Transmitting (no NAT) to 192.168.123.102:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.123.102:5060;branch=z9hG4bK9142030814803754864;received=192.168.123.102
From: <sip:102@F1CONNECTING;user=phone>;tag=c0a80101-84d3
To: <sip:102@F1CONNECTING;user=phone>
Call-ID: 84d3-c0a...@192.168.123.102
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1...@192.168.123.65>
Content-Length: 0


---
    -- Unregistered SIP '102'
Transmitting (no NAT) to 192.168.123.102:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.123.102:5060;branch=z9hG4bK9142030814803754864;received=192.168.123.102
From: <sip:102@F1CONNECTING;user=phone>;tag=c0a80101-84d3
To: <sip:102@F1CONNECTING;user=phone>;tag=as363c0e19
Call-ID: 84d3-c0a...@192.168.123.102
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 0
Date: Sun, 07 Jan 2007 16:23:26 GMT
Content-Length: 0


---
Scheduling destruction of call '84d3-c0a...@192.168.123.102' in 15000 ms
Obelisk*CLI>
<-- SIP read from 192.168.123.102:5060:
REGISTER sip:192.168.123.65 SIP/2.0
Via: SIP/2.0/UDP 192.168.123.102:5060;branch=z9hG4bK6519707547536421531
From: <sip:102@F1CONNECTING;user=phone>;tag=c0a80101-84df
To: <sip:102@F1CONNECTING;user=phone>
Call-ID: 84d3-c0a...@192.168.123.102
CSeq: 3 REGISTER
Max-Forwards: 70
Expires: 600
Contact: <sip:1...@192.168.123.102:5060;user=phone>
User-Agent: THOMSON ST2030 hw3 fw1.50 00-0E-50-4E-62-F5
Content-Length: 0


--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.123.102 : 5060 (non-NAT)
Transmitting (no NAT) to 192.168.123.102:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.123.102:5060;branch=z9hG4bK6519707547536421531;received=192.168.123.102
From: <sip:102@F1CONNECTING;user=phone>;tag=c0a80101-84df
To: <sip:102@F1CONNECTING;user=phone>
Call-ID: 84d3-c0a...@192.168.123.102
CSeq: 3 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1...@192.168.123.65>
Content-Length: 0


---
Transmitting (no NAT) to 192.168.123.102:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.123.102:5060;branch=z9hG4bK6519707547536421531;received= 192.168.123.102
From: <sip:102@F1CONNECTING;user=phone>;tag=c0a80101-84df
To: <sip:102@F1CONNECTING;user=phone>;tag=as363c0e19
Call-ID: 84d3-c0a...@192.168.123.102
CSeq: 3 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm="F1CONNECTING", nonce="7e4ba2a5"
Content-Length: 0


---
Scheduling destruction of call '84d3-c0a...@192.168.123.102' in 15000 ms
Obelisk*CLI>
<-- SIP read from 192.168.123.102:5060:
REGISTER sip:192.168.123.65 SIP/2.0
Via: SIP/2.0/UDP 192.168.123.102:5060;branch=z9hG4bK3286474214203158208
From: <sip:102@F1CONNECTING;user=phone>;tag=c0a80101-84df
To: <sip:102@F1CONNECTING;user=phone>
Call-ID: 84d3-c0a...@192.168.123.102
CSeq: 4 REGISTER
Max-Forwards: 70
Expires: 600
Contact: <sip:1...@192.168.123.102:5060;user=phone>
Authorization: Digest username="102", realm="F1CONNECTING", nonce="7e4ba2a5", uri="sip: 192.168.123.65", response="dfce3da0456da50f71f0f742c1d67369", algorithm=MD5
User-Agent: THOMSON ST2030 hw3 fw1.50 00-0E-50-4E-62-F5
Content-Length: 0


--- (12 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.123.102 : 5060 (non-NAT)
Transmitting (no NAT) to 192.168.123.102:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.123.102:5060;branch=z9hG4bK3286474214203158208;received=192.168.123.102
From: <sip:102@F1CONNECTING;user=phone>;tag=c0a80101-84df
To: <sip:102@F1CONNECTING;user=phone>
Call-ID: 84d3-c0a...@192.168.123.102
CSeq: 4 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1...@192.168.123.65>
Content-Length: 0


---
12 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.123.102:5060:
OPTIONS sip:1...@192.168.123.102:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.123.65:5060;branch=z9hG4bK5f45c99b;rport
From: "asterisk" <sip:aste...@192.168.123.65>;tag=as63551439
To: <sip:1...@192.168.123.102:5060;user=phone>
Contact: <sip:aste...@192.168.123.65>
Call-ID: 63e27c573ff837cf...@192.168.123.65
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 07 Jan 2007 16:23:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
    -- Registered SIP '102' at 192.168.123.102 port 5060 expires 600
Transmitting (no NAT) to 192.168.123.102:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.123.102:5060;branch=z9hG4bK3286474214203158208;received=192.168.123.102
From: <sip:102@F1CONNECTING ;user=phone>;tag=c0a80101-84df
To: <sip:102@F1CONNECTING;user=phone>;tag=as363c0e19
Call-ID: 84d3-c0a...@192.168.123.102
CSeq: 4 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 600
Contact: <sip:1...@192.168.123.102:5060;user=phone>;expires=600
Date: Sun, 07 Jan 2007 16:23:26 GMT
Content-Length: 0


---
Scheduling destruction of call '84d3-c0a...@192.168.123.102' in 15000 ms
Obelisk*CLI>
<-- SIP read from 192.168.123.102:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.123.65:5060;branch=z9hG4bK5f45c99b;rport
From: "asterisk"< sip:aste...@192.168.123.65>;tag=as63551439
To: <sip:1...@192.168.123.102:5060;user=phone>;tag=c0a80101-850a
Call-ID: 63e27c573ff837cf...@192.168.123.65
CSeq: 102 OPTIONS
Contact: <sip:1...@192.168.123.102:5060;user=phone>
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Supported: timer, replaces
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 271

v=0
o=102 34058 34058 IN IP4 192.168.123.102
s=-
c=IN IP4 192.168.123.102
t=0 0
m=audio 41000 RTP/AVP 8 0 18 4 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=sendrecv

--- (12 headers 13 lines) ---
Destroying call '63e27c573ff837cf...@192.168.123.65'
Destroying call '84d3-c0a...@192.168.123.102'
    -- Remote UNIX connection
    -- Remote UNIX connection
    -- Remote UNIX connection disconnected
    -- Remote UNIX connection disconnected
    -- Remote UNIX connection
    -- Remote UNIX connection disconnected
12 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.123.102:5060:
OPTIONS sip:1...@192.168.123.102:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.123.65:5060;branch=z9hG4bK5113e5eb;rport
From: "asterisk" <sip:aste...@192.168.123.65 >;tag=as1f0b7981
To: <sip:1...@192.168.123.102:5060;user=phone>
Contact: <sip:aste...@192.168.123.65>
Call-ID: 3107ca876f250650...@192.168.123.65
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 07 Jan 2007 16:24:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Obelisk*CLI>
<-- SIP read from 192.168.123.102:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.123.65:5060;branch=z9hG4bK5113e5eb;rport
From: "asterisk"<sip:aste...@192.168.123.65>;tag=as1f0b7981
To: <sip:1...@192.168.123.102:5060;user=phone>;tag=c0a80101-fa3f
Call-ID: 3107ca876f250650...@192.168.123.65
CSeq: 102 OPTIONS
Contact: <sip:1...@192.168.123.102:5060;user=phone>
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Supported: timer, replaces
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 271

v=0
o=102 64063 64063 IN IP4 192.168.123.102
s=-
c=IN IP4 192.168.123.102
t=0 0
m=audio 41000 RTP/AVP 8 0 18 4 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=sendrecv

--- (12 headers 13 lines) ---
Destroying call '3107ca876f250650...@192.168.123.65'

 

Jan  7 17:32:27 DEBUG[4119] chan_sip.c: Stopping retransmission on '3935a4e776426eea...@192.168.123.65' of Request 102: Match Found
Jan  7 17:33:27 DEBUG[4119] chan_sip.c: Stopping retransmission on '743df1194860ef84...@192.168.123.65' of Request 102: Match Found
Jan  7 17:33:42 DEBUG[4119] chan_sip.c: Auto destroying call '84d3-c0a...@192.168.123.102'
Jan  7 17:34:27 DEBUG[4119] chan_sip.c: Stopping retransmission on '17cbf5a730fb748a...@192.168.123.65' of Request 102: Match Found


 
2007/1/4, Elio Rojano <hel...@gmail.com>:

Sergio Aparicio

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Jan 7, 2007, 11:57:05 AM1/7/07
to aster...@googlegroups.com
Acabo de leer esta web sobre el protocolo sip:
 
 
 
Ya entiendo por que se registran y por que primero da 401, pero sigo sin entender por que aparece el Request 102: Match Found cuando aparentemente el telefono se registra bien.

Via: SIP/2.0/UDP 192.168.123.102:5060;branch=z9hG4bK9142030814803754864;received= 192.168.123.102

From: <sip:102@F1CONNECTING;user=phone>;tag=c0a80101-84d3
To: <sip:102@F1CONNECTING;user=phone>

Call-ID: 84d3-c0a...@192.168.123.102
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: < sip:1...@192.168.123.65>
Content-Length: 0


---
    -- Unregistered SIP '102'
Transmitting (no NAT) to 192.168.123.102:5060:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.123.102:5060;branch=z9hG4bK9142030814803754864;received= 192.168.123.102


From: <sip:102@F1CONNECTING;user=phone>;tag=c0a80101-84d3
To: <sip:102@F1CONNECTING;user=phone>;tag=as363c0e19
Call-ID: 84d3-c0a...@192.168.123.102

Via: SIP/2.0/UDP 192.168.123.102:5060;branch=z9hG4bK3286474214203158208;received= 192.168.123.102

From: <sip:102@F1CONNECTING;user=phone>;tag=c0a80101-84df
To: <sip:102@F1CONNECTING;user=phone>

Sergio Aparicio

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Jan 7, 2007, 3:46:58 PM1/7/07
to aster...@googlegroups.com
Por si a alguien le ocurre:
 
   estos mensajes en el debug: "Jan  7 17:33:27 DEBUG[4119] chan_sip.c: Stopping retransmission on '743df1194860ef84...@192.168.123.65' of Request 102: Match Found "
 
 
   Es porque qualify esta activado entonces Asterisk (mas bien el  chan_sip.c) hace ack para comprobar la latencia, este mensaje indicaria que la comprobación ha ido ok.
 
  La funcion que lo hace es: static int __sip_ack (struct sip_pvt *p, int seqno, int resp, int sipmethod) y la salida del LOG_DEBUG es:
 

ast_log(
LOG_DEBUG, 
"Stopping retransmission on '%s' of %s %d: Match %s\n"
, p->
callid, resp ? "Response" : 
"Request", 
seqno, res ? "Not Found" : "Found"
);
 

 
               Saludos / Sam
 
 

seqno
 
2007/1/7, Sergio Aparicio <sergio....@gmail.com>:
2007/1/4, Elio Rojano <hel...@gmail.com>:



Saludos / Sam



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Saludos / Sam
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