asterisk y voipbuster: Forbidden para llamadas "salientes"

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Jaume

unread,
Nov 13, 2007, 7:55:41 PM11/13/07
to asterisk-es
Hola,

Estoy usando Asterisk 1.4.13 y tengo configurados dos proveedores via
SIP, adiptel y voipbuster. Con adiptel no tengo ningun problema, pero
con voipbuster, cada vez que intento hacer una llamada recibo un
Forbidden. El log exacto es:

[Nov 13 04:15:56] WARNING[27316] chan_sip.c: Received response:
"Forbidden" from '"wifiphone"
<sip:my_...@sip.voipbuster.com>;tag=as3fdc4519'
[Nov 13 04:15:56] NOTICE[29548] cdr.c: CDR on channel 'SIP/
voipbuster-081ec4b8' not posted

Creo que no es un problema de la cuenta ni de autenticacion pq usando
un softphone (twinkle) directamente conectado a voipbuster puedo
realizar llamadas.

Tengo trazas de WireShark con la llamada desde asterisk y directamente
desde twinkle, pero no consigo ver diferencia aparte del orden de los
headers SIP. Si creeis que pueden ser de interes los posteo.

Cualquier idea es bienvenida! Llevo unas semanas luchando con este
tema ;)

Mi config es la siguiente:

sip.conf
[voipbuster]
type=friend
disallow=all
allow=ulaw
allow=alaw
host=sip.voipbuster.com
username=my_user
fromuser=my_user
;reguser=my_user
password=my_password
fromdomain=sip.voipbuster.com
canreinvite=no
insecure=very
qualify=no
nat=yes
context=voipbuster
useragent="Twinkle/1.0"

extensions.conf:
[internal]
include => voipbuster_out
[voipbuster_out]
exten => _00349NNXXXXXX,1,Dial(SIP/${EXTEN}@voipbuster,30,t)
exten => _X.,1,Dial(SIP/${EXTEN}@voipbuster,30,t)

Iñaki Baz Castillo

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Nov 14, 2007, 6:14:40 AM11/14/07
to aster...@googlegroups.com
El Wednesday 14 November 2007 01:55:41 Jaume escribió:
> Hola,
>
> Estoy usando Asterisk 1.4.13 y tengo configurados dos proveedores via
> SIP, adiptel y voipbuster. Con adiptel no tengo ningun problema, pero
> con voipbuster, cada vez que intento hacer una llamada recibo un
> Forbidden. El log exacto es:
>
> [Nov 13 04:15:56] WARNING[27316] chan_sip.c: Received response:
> "Forbidden" from '"wifiphone"
> <sip:my_...@sip.voipbuster.com>;tag=as3fdc4519'
> [Nov 13 04:15:56] NOTICE[29548] cdr.c: CDR on channel 'SIP/
> voipbuster-081ec4b8' not posted
>
> Creo que no es un problema de la cuenta ni de autenticacion pq usando
> un softphone (twinkle) directamente conectado a voipbuster puedo
> realizar llamadas.
>
> Tengo trazas de WireShark con la llamada desde asterisk y directamente
> desde twinkle, pero no consigo ver diferencia aparte del orden de los
> headers SIP. Si creeis que pueden ser de interes los posteo.

Postealos a ver.

--
Iñaki Baz Castillo
i...@in.ilimit.es

paco gil

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Nov 14, 2007, 7:31:02 AM11/14/07
to aster...@googlegroups.com
que te pone el agent-user de asterisk?? que te pone el agent-user de twinkle???

saludos,

Abel Molina Landrián

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Nov 14, 2007, 10:20:33 AM11/14/07
to aster...@googlegroups.com
Yo ya he comprobado que no llego a los clones de betamax. No sé si será
ONO o si serán los de Betamax pero lo cierto es que no llego.

Usa el nmap y lanza al UDP 5060 de alguno de los clones.

Salu2


Iñaki Baz Castillo escribió:

Jaume

unread,
Nov 15, 2007, 5:07:27 AM11/15/07
to asterisk-es
Gracias por las respuestas!

Paco: Con las opciones useragent en el contexto, el header useragent
de twinkle y el de asterisk coinciden.
Iñaki: Aqui van las trazas. No sabia como postearlas, asi que las
"pego" en texto plano.

La traza con twinkle en la conexion que funciona:
No. Time Source Destination
Protocol Info
3 21.794854 192.168.1.2 194.221.62.198 SIP/
SDP Request: INVITE sip:003493...@sip.voipbuster.com, with
session description

Frame 3 (874 bytes on wire, 874 bytes captured)
Ethernet II, Src: AsustekC_ae:a0:7a (00:0c:6e:ae:a0:7a), Dst: D-
Link_21:e6:ed (00:19:5b:21:e6:ed)
Internet Protocol, Src: 192.168.1.2 (192.168.1.2), Dst: 194.221.62.198
(194.221.62.198)
User Datagram Protocol, Src Port: sip-tls (5061), Dst Port: sip (5060)
Session Initiation Protocol
Request-Line: INVITE sip:003493...@sip.voipbuster.com SIP/2.0
Message Header
Via: SIP/2.0/UDP 192.168.1.2:5061;rport;branch=z9hG4bKfavcahhf
Max-Forwards: 70
To: <sip:003493...@sip.voipbuster.com>
From: "Jaume" <sip:xo...@sip.voipbuster.com>;tag=bvfqr
Call-ID: uegyqjy...@192.168.1.2
CSeq: 794 INVITE
Contact: <sip:xo...@192.168.1.2:5061>
Content-Type: application/sdp
Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
Supported: replaces,norefersub,100rel
User-Agent: Twinkle/1.1
Content-Length: 304
Message body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): xomix 766119170 1855509643
IN IP4 192.168.1.2
Session Name (s): -
Connection Information (c): IN IP4 192.168.1.2
Time Description, active time (t): 0 0
Media Description, name and address (m): audio 8000 RTP/
AVP 98 97 8 0 3 101
Media Attribute (a): rtpmap:98 speex/16000
Media Attribute (a): rtpmap:97 speex/8000
Media Attribute (a): rtpmap:8 PCMA/8000
Media Attribute (a): rtpmap:0 PCMU/8000
Media Attribute (a): rtpmap:3 GSM/8000
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute (a): fmtp:101 0-15
Media Attribute (a): ptime:20

No. Time Source Destination
Protocol Info
4 22.049665 194.221.62.198 192.168.1.2
SIP Status: 401 Unauthorized

Frame 4 (551 bytes on wire, 551 bytes captured)
Ethernet II, Src: D-Link_21:e6:ed (00:19:5b:21:e6:ed), Dst:
AsustekC_ae:a0:7a (00:0c:6e:ae:a0:7a)
Internet Protocol, Src: 194.221.62.198 (194.221.62.198), Dst:
192.168.1.2 (192.168.1.2)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip-tls (5061)
Session Initiation Protocol
Status-Line: SIP/2.0 401 Unauthorized
Message Header
Via: SIP/2.0/UDP 192.168.1.2:5061;branch=z9hG4bKfavcahhf;rport
From: "Jaume" <sip:xo...@sip.voipbuster.com>;tag=bvfqr
To: <sip:003493...@sip.voipbuster.com>
Contact: sip:003493...@194.221.62.198:5060
Call-ID: uegyqjy...@192.168.1.2
CSeq: 794 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
WWW-Authenticate: Digest
realm="sip.voipbuster.com",nonce="4058156265",algorithm=MD5
Content-Length: 0

No. Time Source Destination
Protocol Info
5 22.051260 192.168.1.2 194.221.62.198
SIP Request: ACK sip:003493...@sip.voipbuster.com

Frame 5 (373 bytes on wire, 373 bytes captured)
Ethernet II, Src: AsustekC_ae:a0:7a (00:0c:6e:ae:a0:7a), Dst: D-
Link_21:e6:ed (00:19:5b:21:e6:ed)
Internet Protocol, Src: 192.168.1.2 (192.168.1.2), Dst: 194.221.62.198
(194.221.62.198)
User Datagram Protocol, Src Port: sip-tls (5061), Dst Port: sip (5060)
Session Initiation Protocol
Request-Line: ACK sip:003493...@sip.voipbuster.com SIP/2.0
Message Header
Via: SIP/2.0/UDP 192.168.1.2:5061;rport;branch=z9hG4bKfavcahhf
Max-Forwards: 70
To: <sip:003493...@sip.voipbuster.com>
From: "Jaume" <sip:xo...@sip.voipbuster.com>;tag=bvfqr
Call-ID: uegyqjy...@192.168.1.2
CSeq: 794 ACK
User-Agent: Twinkle/1.1
Content-Length: 0

No. Time Source Destination
Protocol Info
6 22.052256 192.168.1.2 194.221.62.198 SIP/
SDP Request: INVITE sip:003493...@sip.voipbuster.com, with
session description

Frame 6 (1061 bytes on wire, 1061 bytes captured)
Ethernet II, Src: AsustekC_ae:a0:7a (00:0c:6e:ae:a0:7a), Dst: D-
Link_21:e6:ed (00:19:5b:21:e6:ed)
Internet Protocol, Src: 192.168.1.2 (192.168.1.2), Dst: 194.221.62.198
(194.221.62.198)
User Datagram Protocol, Src Port: sip-tls (5061), Dst Port: sip (5060)
Session Initiation Protocol
Request-Line: INVITE sip:003493...@sip.voipbuster.com SIP/2.0
Message Header
Via: SIP/2.0/UDP 192.168.1.2:5061;rport;branch=z9hG4bKyekvfuvi
Max-Forwards: 70
To: <sip:003493...@sip.voipbuster.com>
From: "Jaume" <sip:xo...@sip.voipbuster.com>;tag=bvfqr
Call-ID: uegyqjy...@192.168.1.2
CSeq: 795 INVITE
Contact: <sip:xo...@192.168.1.2:5061>
Content-Type: application/sdp
Authorization: Digest
username="xomix",realm="sip.voipbuster.com",nonce="4058156265",uri="sip:
003493...@sip.voipbuster.com",response="7b696e880780b43b9212afbe08cb246f",algorithm=MD5
Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
Supported: replaces,norefersub,100rel
User-Agent: Twinkle/1.1
Content-Length: 304
Message body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): xomix 766119170 1855509643
IN IP4 192.168.1.2
Session Name (s): -
Connection Information (c): IN IP4 192.168.1.2
Time Description, active time (t): 0 0
Media Description, name and address (m): audio 8000 RTP/
AVP 98 97 8 0 3 101
Media Attribute (a): rtpmap:98 speex/16000
Media Attribute (a): rtpmap:97 speex/8000
Media Attribute (a): rtpmap:8 PCMA/8000
Media Attribute (a): rtpmap:0 PCMU/8000
Media Attribute (a): rtpmap:3 GSM/8000
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute (a): fmtp:101 0-15
Media Attribute (a): ptime:20

No. Time Source Destination
Protocol Info
7 22.321127 194.221.62.198 192.168.1.2
SIP Status: 100 Trying

Frame 7 (459 bytes on wire, 459 bytes captured)
Ethernet II, Src: D-Link_21:e6:ed (00:19:5b:21:e6:ed), Dst:
AsustekC_ae:a0:7a (00:0c:6e:ae:a0:7a)
Internet Protocol, Src: 194.221.62.198 (194.221.62.198), Dst:
192.168.1.2 (192.168.1.2)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip-tls (5061)
Session Initiation Protocol
Status-Line: SIP/2.0 100 Trying
Message Header
Via: SIP/2.0/UDP 192.168.1.2:5061;branch=z9hG4bKyekvfuvi;rport
From: "Jaume" <sip:xo...@sip.voipbuster.com>;tag=bvfqr
To: <sip:003493...@sip.voipbuster.com>
Contact: sip:003493...@194.221.62.198:5060
Call-ID: uegyqjy...@192.168.1.2
CSeq: 795 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Length: 0

No. Time Source Destination
Protocol Info
8 22.340817 194.221.62.198 192.168.1.2 SIP/
SDP Status: 183 Session progress, with session description

Frame 8 (737 bytes on wire, 737 bytes captured)
Ethernet II, Src: D-Link_21:e6:ed (00:19:5b:21:e6:ed), Dst:
AsustekC_ae:a0:7a (00:0c:6e:ae:a0:7a)
Internet Protocol, Src: 194.221.62.198 (194.221.62.198), Dst:
192.168.1.2 (192.168.1.2)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip-tls (5061)
Session Initiation Protocol
Status-Line: SIP/2.0 183 Session progress
Message Header
Via: SIP/2.0/UDP 192.168.1.2:5061;branch=z9hG4bKyekvfuvi;rport
From: "Jaume" <sip:xo...@sip.voipbuster.com>;tag=bvfqr
To: <sip:
003493...@sip.voipbuster.com>;tag=cb1710accb2b10ac470cb5282bad78
Contact: sip:003493...@194.221.62.198:5060
Call-ID: uegyqjy...@192.168.1.2
CSeq: 795 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Type: application/sdp
Content-Length: 200
Message body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): xomix 1194936404 1194936404
IN IP4 80.239.235.162
Session Name (s): SIP Call
Connection Information (c): IN IP4 80.239.235.162
Time Description, active time (t): 0 0
Media Description, name and address (m): audio 57364 RTP/
AVP 8 101
Media Attribute (a): rtpmap:8 PCMA/8000
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute (a): ptime:20

No. Time Source Destination
Protocol Info
9 48.752499 194.221.62.198 192.168.1.2 SIP/
SDP Status: 200 Ok, with session description

Frame 9 (723 bytes on wire, 723 bytes captured)
Ethernet II, Src: D-Link_21:e6:ed (00:19:5b:21:e6:ed), Dst:
AsustekC_ae:a0:7a (00:0c:6e:ae:a0:7a)
Internet Protocol, Src: 194.221.62.198 (194.221.62.198), Dst:
192.168.1.2 (192.168.1.2)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip-tls (5061)
Session Initiation Protocol
Status-Line: SIP/2.0 200 Ok
Message Header
Via: SIP/2.0/UDP 192.168.1.2:5061;branch=z9hG4bKyekvfuvi;rport
From: "Jaume" <sip:xo...@sip.voipbuster.com>;tag=bvfqr
To: <sip:
003493...@sip.voipbuster.com>;tag=cb1710accb2b10ac470cb5282bad78
Contact: sip:003493...@194.221.62.198:5060
Call-ID: uegyqjy...@192.168.1.2
CSeq: 795 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Type: application/sdp
Content-Length: 200
Message body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): xomix 1194936430 1194936430
IN IP4 80.239.235.162
Session Name (s): SIP Call
Connection Information (c): IN IP4 80.239.235.162
Time Description, active time (t): 0 0
Media Description, name and address (m): audio 57364 RTP/
AVP 8 101
Media Attribute (a): rtpmap:8 PCMA/8000
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute (a): ptime:20

No. Time Source Destination
Protocol Info
10 48.794612 192.168.1.2 194.221.62.198
SIP Request: ACK sip:003493...@194.221.62.198:5060

Frame 10 (596 bytes on wire, 596 bytes captured)
Ethernet II, Src: AsustekC_ae:a0:7a (00:0c:6e:ae:a0:7a), Dst: D-
Link_21:e6:ed (00:19:5b:21:e6:ed)
Internet Protocol, Src: 192.168.1.2 (192.168.1.2), Dst: 194.221.62.198
(194.221.62.198)
User Datagram Protocol, Src Port: sip-tls (5061), Dst Port: sip (5060)
Session Initiation Protocol
Request-Line: ACK sip:003493...@194.221.62.198:5060 SIP/2.0
Message Header
Via: SIP/2.0/UDP 192.168.1.2:5061;rport;branch=z9hG4bKpnnuorel
Max-Forwards: 70
To: <sip:
003493...@sip.voipbuster.com>;tag=cb1710accb2b10ac470cb5282bad78
From: "Jaume" <sip:xo...@sip.voipbuster.com>;tag=bvfqr
Call-ID: uegyqjy...@192.168.1.2
CSeq: 795 ACK
Authorization: Digest
username="xomix",realm="sip.voipbuster.com",nonce="4058156265",uri="sip:
003493...@sip.voipbuster.com",response="7b696e880780b43b9212afbe08cb246f",algorithm=MD5
User-Agent: Twinkle/1.1
Content-Length: 0

No. Time Source Destination
Protocol Info
11 53.593752 192.168.1.2 194.221.62.198
SIP Request: BYE sip:003493...@194.221.62.198:5060

Frame 11 (409 bytes on wire, 409 bytes captured)
Ethernet II, Src: AsustekC_ae:a0:7a (00:0c:6e:ae:a0:7a), Dst: D-
Link_21:e6:ed (00:19:5b:21:e6:ed)
Internet Protocol, Src: 192.168.1.2 (192.168.1.2), Dst: 194.221.62.198
(194.221.62.198)
User Datagram Protocol, Src Port: sip-tls (5061), Dst Port: sip (5060)
Session Initiation Protocol
Request-Line: BYE sip:003493...@194.221.62.198:5060 SIP/2.0
Message Header
Via: SIP/2.0/UDP 192.168.1.2:5061;rport;branch=z9hG4bKyggpihvc
Max-Forwards: 70
To: <sip:
003493...@sip.voipbuster.com>;tag=cb1710accb2b10ac470cb5282bad78
From: "Jaume" <sip:xo...@sip.voipbuster.com>;tag=bvfqr
Call-ID: uegyqjy...@192.168.1.2
CSeq: 796 BYE
User-Agent: Twinkle/1.1
Content-Length: 0

No. Time Source Destination
Protocol Info
12 53.835465 194.221.62.198 192.168.1.2
SIP Status: 200 Ok

Frame 12 (487 bytes on wire, 487 bytes captured)
Ethernet II, Src: D-Link_21:e6:ed (00:19:5b:21:e6:ed), Dst:
AsustekC_ae:a0:7a (00:0c:6e:ae:a0:7a)
Internet Protocol, Src: 194.221.62.198 (194.221.62.198), Dst:
192.168.1.2 (192.168.1.2)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip-tls (5061)
Session Initiation Protocol
Status-Line: SIP/2.0 200 Ok
Message Header
Via: SIP/2.0/UDP 192.168.1.2:5061;branch=z9hG4bKyggpihvc;rport
From: "Jaume" <sip:xo...@sip.voipbuster.com>;tag=bvfqr
To: <sip:
003493...@sip.voipbuster.com>;tag=cb1710accb2b10ac470cb5282bad78
Contact: sip:003493...@194.221.62.198:5060
Call-ID: uegyqjy...@192.168.1.2
CSeq: 796 BYE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Length: 0




La traza con asterisk en la conexion "Forbiden":

No. Time Source Destination
Protocol Info
10 19.751072 192.168.1.2 194.120.0.198 SIP/
SDP Request: INVITE sip:003493...@sip.voipbuster.com, with
session description

Frame 10 (865 bytes on wire, 865 bytes captured)
Ethernet II, Src: AsustekC_ae:a0:7a (00:0c:6e:ae:a0:7a), Dst: D-
Link_21:e6:ed (00:19:5b:21:e6:ed)
Internet Protocol, Src: 192.168.1.2 (192.168.1.2), Dst: 194.120.0.198
(194.120.0.198)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
Request-Line: INVITE sip:003493...@sip.voipbuster.com SIP/2.0
Message Header
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK5ae8e630;rport
From: "jaume" <sip:xo...@sip.voipbuster.com>;tag=as7ca3e295
To: <sip:003493...@sip.voipbuster.com>
Contact: <sip:xo...@192.168.1.2>
Call-ID: 597d505706f6565a...@sip.voipbuster.com
CSeq: 102 INVITE
User-Agent: "Twinkle/1.0"
Max-Forwards: 70
Date: Tue, 13 Nov 2007 06:49:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 262
Message body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): root 25425 25425 IN IP4
192.168.1.2
Session Name (s): session
Connection Information (c): IN IP4 192.168.1.2
Time Description, active time (t): 0 0
Media Description, name and address (m): audio 12936 RTP/
AVP 0 8 101
Media Attribute (a): rtpmap:0 PCMU/8000
Media Attribute (a): rtpmap:8 PCMA/8000
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute (a): fmtp:101 0-16
Media Attribute (a): silenceSupp:off - - - -
Media Attribute (a): ptime:20
Media Attribute (a): sendrecv

No. Time Source Destination
Protocol Info
11 20.010324 194.120.0.198 192.168.1.2
SIP Status: 401 Unauthorized

Frame 11 (579 bytes on wire, 579 bytes captured)
Ethernet II, Src: D-Link_21:e6:ed (00:19:5b:21:e6:ed), Dst:
AsustekC_ae:a0:7a (00:0c:6e:ae:a0:7a)
Internet Protocol, Src: 194.120.0.198 (194.120.0.198), Dst:
192.168.1.2 (192.168.1.2)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
Status-Line: SIP/2.0 401 Unauthorized
Message Header
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK5ae8e630;rport
From: "jaume" <sip:xo...@sip.voipbuster.com>;tag=as7ca3e295
To: <sip:003493...@sip.voipbuster.com>
Contact: sip:003493...@194.120.0.198:5060
Call-ID: 597d505706f6565a...@sip.voipbuster.com
CSeq: 102 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
WWW-Authenticate: Digest
realm="sip.voipbuster.com",nonce="1223832829",algorithm=MD5
Content-Length: 0

No. Time Source Destination
Protocol Info
12 20.010647 192.168.1.2 194.120.0.198
SIP Request: ACK sip:003493...@sip.voipbuster.com

Frame 12 (438 bytes on wire, 438 bytes captured)
Ethernet II, Src: AsustekC_ae:a0:7a (00:0c:6e:ae:a0:7a), Dst: D-
Link_21:e6:ed (00:19:5b:21:e6:ed)
Internet Protocol, Src: 192.168.1.2 (192.168.1.2), Dst: 194.120.0.198
(194.120.0.198)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
Request-Line: ACK sip:003493...@sip.voipbuster.com SIP/2.0
Message Header
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK5ae8e630;rport
From: "jaume" <sip:xo...@sip.voipbuster.com>;tag=as7ca3e295
To: <sip:003493...@sip.voipbuster.com>
Contact: <sip:xo...@192.168.1.2>
Call-ID: 597d505706f6565a...@sip.voipbuster.com
CSeq: 102 ACK
User-Agent: "Twinkle/1.0"
Max-Forwards: 70
Content-Length: 0

No. Time Source Destination
Protocol Info
13 20.010869 192.168.1.2 194.120.0.198 SIP/
SDP Request: INVITE sip:003493...@sip.voipbuster.com, with
session description

Frame 13 (1068 bytes on wire, 1068 bytes captured)
Ethernet II, Src: AsustekC_ae:a0:7a (00:0c:6e:ae:a0:7a), Dst: D-
Link_21:e6:ed (00:19:5b:21:e6:ed)
Internet Protocol, Src: 192.168.1.2 (192.168.1.2), Dst: 194.120.0.198
(194.120.0.198)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
Request-Line: INVITE sip:003493...@sip.voipbuster.com SIP/2.0
Message Header
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK6be0a48e;rport
From: "jaume" <sip:xo...@sip.voipbuster.com>;tag=as7ca3e295
To: <sip:003493...@sip.voipbuster.com>
Contact: <sip:xo...@192.168.1.2>
Call-ID: 597d505706f6565a...@sip.voipbuster.com
CSeq: 103 INVITE
User-Agent: "Twinkle/1.0"
Max-Forwards: 70
Authorization: Digest username="xomix",
realm="sip.voipbuster.com", algorithm=MD5, uri="sip:
003493...@sip.voipbuster.com", nonce="1223832829",
response="c12a5e37584897c74794d3f006e419e9", opaque=""
Date: Tue, 13 Nov 2007 06:49:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 262
Message body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): root 25425 25426 IN IP4
192.168.1.2
Session Name (s): session
Connection Information (c): IN IP4 192.168.1.2
Time Description, active time (t): 0 0
Media Description, name and address (m): audio 12936 RTP/
AVP 0 8 101
Media Attribute (a): rtpmap:0 PCMU/8000
Media Attribute (a): rtpmap:8 PCMA/8000
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute (a): fmtp:101 0-16
Media Attribute (a): silenceSupp:off - - - -
Media Attribute (a): ptime:20
Media Attribute (a): sendrecv

No. Time Source Destination
Protocol Info
14 20.287183 194.120.0.198 192.168.1.2
SIP Status: 100 Trying

Frame 14 (487 bytes on wire, 487 bytes captured)
Ethernet II, Src: D-Link_21:e6:ed (00:19:5b:21:e6:ed), Dst:
AsustekC_ae:a0:7a (00:0c:6e:ae:a0:7a)
Internet Protocol, Src: 194.120.0.198 (194.120.0.198), Dst:
192.168.1.2 (192.168.1.2)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
Status-Line: SIP/2.0 100 Trying
Message Header
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK6be0a48e;rport
From: "jaume" <sip:xo...@sip.voipbuster.com>;tag=as7ca3e295
To: <sip:003493...@sip.voipbuster.com>
Contact: sip:003493...@194.120.0.198:5060
Call-ID: 597d505706f6565a...@sip.voipbuster.com
CSeq: 103 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Length: 0

No. Time Source Destination
Protocol Info
15 20.290316 194.120.0.198 192.168.1.2
SIP Status: 403 Forbidden

Frame 15 (490 bytes on wire, 490 bytes captured)
Ethernet II, Src: D-Link_21:e6:ed (00:19:5b:21:e6:ed), Dst:
AsustekC_ae:a0:7a (00:0c:6e:ae:a0:7a)
Internet Protocol, Src: 194.120.0.198 (194.120.0.198), Dst:
192.168.1.2 (192.168.1.2)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
Status-Line: SIP/2.0 403 Forbidden
Message Header
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK6be0a48e;rport
From: "jaume" <sip:xo...@sip.voipbuster.com>;tag=as7ca3e295
To: <sip:003493...@sip.voipbuster.com>
Contact: sip:003493...@194.120.0.198:5060
Call-ID: 597d505706f6565a...@sip.voipbuster.com
CSeq: 103 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Length: 0

No. Time Source Destination
Protocol Info
16 20.290432 192.168.1.2 194.120.0.198
SIP Request: ACK sip:003493...@sip.voipbuster.com

Frame 16 (438 bytes on wire, 438 bytes captured)
Ethernet II, Src: AsustekC_ae:a0:7a (00:0c:6e:ae:a0:7a), Dst: D-
Link_21:e6:ed (00:19:5b:21:e6:ed)
Internet Protocol, Src: 192.168.1.2 (192.168.1.2), Dst: 194.120.0.198
(194.120.0.198)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
Request-Line: ACK sip:003493...@sip.voipbuster.com SIP/2.0
Message Header
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK6be0a48e;rport
From: "jaume" <sip:xo...@sip.voipbuster.com>;tag=as7ca3e295
To: <sip:003493...@sip.voipbuster.com>
Contact: <sip:xo...@192.168.1.2>
Call-ID: 597d505706f6565a...@sip.voipbuster.com
CSeq: 103 ACK
User-Agent: "Twinkle/1.0"
Max-Forwards: 70
Content-Length: 0


On 14 nov, 09:31, "paco gil" <pag...@gmail.com> wrote:
> que te pone el agent-user de asterisk?? que te pone el agent-user de
> twinkle???
>
> saludos,
>
> On Nov 14, 2007 12:14 PM, Iñaki Baz Castillo <i...@in.ilimit.es> wrote:
>
>
>
> > El Wednesday 14 November 2007 01:55:41 Jaume escribió:
> > > Hola,
>
> > > Estoy usando Asterisk 1.4.13 y tengo configurados dos proveedores via
> > > SIP, adiptel y voipbuster. Con adiptel no tengo ningun problema, pero
> > > con voipbuster, cada vez que intento hacer una llamada recibo un
> > > Forbidden. El log exacto es:
>
> > > [Nov 13 04:15:56] WARNING[27316] chan_sip.c: Received response:
> > > "Forbidden" from '"wifiphone"
> > > <sip:my_u...@sip.voipbuster.com>;tag=as3fdc4519'

paco gil

unread,
Nov 15, 2007, 5:29:59 AM11/15/07
to aster...@googlegroups.com
mira te digo lo del user-agent, porque me ha pasado con un proveedor... utilizaba el x-lite y todo bien, pero si utilizaba el asterisk, no me dejaba llamar (en este caso me salia una locucion del proveedor indicando que no tenía permisos o algo así.)

total que les envie un correo y me dijeron que en el campo "useragent", donde antes ponía "asterisk" de forma predeterminada, ahora debía poner "Digest/1.1.0". Hasta que no cambié eso, no me dejo llamar con dicho proveedor. A lo mejor voipbuster hace algo así....

On Nov 15, 2007 11:07 AM, Jaume <txo...@gmail.com> wrote:

Gracias por las respuestas!

Paco: Con las opciones useragent en el contexto, el header useragent
de twinkle y el de asterisk coinciden.
Iñaki: Aqui van las trazas. No sabia como postearlas, asi que las
"pego" en texto plano.

La traza con twinkle en la conexion que funciona:
No.     Time        Source                Destination
Protocol Info
     3 21.794854   192.168.1.2           194.221.62.198        SIP/
SDP  Request: INVITE sip:003493...@sip.voipbuster.com , with
Internet Protocol, Src: 194.221.62.198 ( 194.221.62.198), Dst:
Internet Protocol, Src: 192.168.1.2 ( 192.168.1.2), Dst: 194.221.62.198
Internet Protocol, Src: 192.168.1.2 ( 192.168.1.2), Dst: 194.120.0.198
SDP  Request: INVITE sip:003493...@sip.voipbuster.com , with
Internet Protocol, Src: 192.168.1.2 ( 192.168.1.2), Dst: 194.120.0.198

Iñaki Baz Castillo

unread,
Nov 15, 2007, 5:43:18 AM11/15/07
to aster...@googlegroups.com
El Thursday 15 November 2007 11:07:27 Jaume escribió:
> Gracias por las respuestas!
>
> Paco: Con las opciones useragent en el contexto, el header useragent
> de twinkle y el de asterisk coinciden.
> Iñaki: Aqui van las trazas. No sabia como postearlas, asi que las
> "pego" en texto plano.

Hola, ¿te importaría volver a capturarlas como te voy a indicar? (es por
costumbre mía, me es más cómodo):

Instala ngrep y ejecuta:

ngrep -d any -P ' ' -W byline -T -t " " port 5060

Y luego haz una llamada desde Twinkle.

Y luego desde Asterisk.

Jaume

unread,
Nov 15, 2007, 9:59:13 PM11/15/07
to asterisk-es
Claro que no me importa! Al contrario, muchas gracias por tu interes!

Aqui estan:

Twinkle (funciona):
interface: eth0 (192.168.1.0/255.255.255.0)
filter: (ip or ip6) and ( port 5061 )
match:

#
U 2007/11/15 23:50:02.819485 192.168.1.2:5061 -> 194.221.62.198:5060
INVITE sip:003493...@sip.voipbuster.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5061;rport;branch=z9hG4bKkuwfcnmy
Max-Forwards: 70
To: <sip:003493...@sip.voipbuster.com>
From: "Jaume" <sip:xo...@sip.voipbuster.com>;tag=ipxie
Call-ID: crhjzcr...@192.168.1.2
CSeq: 4 INVITE
Contact: <sip:xo...@192.168.1.2:5061>
Content-Type: application/sdp
Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
Supported: replaces,norefersub,100rel
User-Agent: Twinkle/1.1
Content-Length: 304

v=0
o=xomix 1434298066 314242592 IN IP4 192.168.1.2
s=-
c=IN IP4 192.168.1.2
t=0 0
m=audio 8000 RTP/AVP 98 97 8 0 3 101
a=rtpmap:98 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

#
U 2007/11/15 23:50:03.086181 194.221.62.198:5060 -> 192.168.1.2:5061
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.2:5061;branch=z9hG4bKkuwfcnmy;rport
From: "Jaume" <sip:xo...@sip.voipbuster.com>;tag=ipxie
Call-ID: crhjzcr...@192.168.1.2
CSeq: 4 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
WWW-Authenticate: Digest
realm="sip.voipbuster.com",nonce="1473847079",algorithm=MD5
Content-Length: 0


#
U 2007/11/15 23:50:03.088417 192.168.1.2:5061 -> 194.221.62.198:5060
ACK sip:003493...@sip.voipbuster.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5061;rport;branch=z9hG4bKkuwfcnmy
Max-Forwards: 70
To: <sip:003493...@sip.voipbuster.com>
From: "Jaume" <sip:xo...@sip.voipbuster.com>;tag=ipxie
Call-ID: crhjzcr...@192.168.1.2
CSeq: 4 ACK
User-Agent: Twinkle/1.1
Content-Length: 0


#
U 2007/11/15 23:50:03.089795 192.168.1.2:5061 -> 194.221.62.198:5060
INVITE sip:003493...@sip.voipbuster.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5061;rport;branch=z9hG4bKwdtfykdg
Max-Forwards: 70
To: <sip:003493...@sip.voipbuster.com>
From: "Jaume" <sip:xo...@sip.voipbuster.com>;tag=ipxie
Call-ID: crhjzcr...@192.168.1.2
CSeq: 5 INVITE
Contact: <sip:xo...@192.168.1.2:5061>
Content-Type: application/sdp
Authorization: Digest
username="xomix",realm="sip.voipbuster.com",nonce="1473847079",uri="sip:
003493...@sip.voipbuster.com",response="4052b529771ff674924db5743417cc74",algorithm=MD5
Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
Supported: replaces,norefersub,100rel
User-Agent: Twinkle/1.1
Content-Length: 304

v=0
o=xomix 1434298066 314242592 IN IP4 192.168.1.2
s=-
c=IN IP4 192.168.1.2
t=0 0
m=audio 8000 RTP/AVP 98 97 8 0 3 101
a=rtpmap:98 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

#
U 2007/11/15 23:50:03.360142 194.221.62.198:5060 -> 192.168.1.2:5061
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.2:5061;branch=z9hG4bKwdtfykdg;rport
From: "Jaume" <sip:xo...@sip.voipbuster.com>;tag=ipxie
Call-ID: crhjzcr...@192.168.1.2
CSeq: 5 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Length: 0


#
U 2007/11/15 23:50:03.409210 194.221.62.198:5060 -> 192.168.1.2:5061
SIP/2.0 183 Session progress
Via: SIP/2.0/UDP 192.168.1.2:5061;branch=z9hG4bKwdtfykdg;rport
From: "Jaume" <sip:xo...@sip.voipbuster.com>;tag=ipxie
To: <sip:
003493...@sip.voipbuster.com>;tag=c91710acc92b10ac470ca8322fdbf0
Contact: sip:003493...@194.221.62.198:5060
Call-ID: crhjzcr...@192.168.1.2
CSeq: 5 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Type: application/sdp
Content-Length: 198

v=0
o=xomix 1195181403 1195181403 IN IP4 194.120.0.164
s=SIP Call
c=IN IP4 194.120.0.164
t=0 0
m=audio 41804 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20

#
U 2007/11/15 23:50:09.819972 192.168.1.2:5061 -> 194.221.62.198:5060
CANCEL sip:003493...@sip.voipbuster.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5061;rport;branch=z9hG4bKwdtfykdg
Max-Forwards: 70
To: <sip:003493...@sip.voipbuster.com>
From: "Jaume" <sip:xo...@sip.voipbuster.com>;tag=ipxie
Call-ID: crhjzcr...@192.168.1.2
CSeq: 5 CANCEL
User-Agent: Twinkle/1.1
Content-Length: 0


#
U 2007/11/15 23:50:10.067357 194.221.62.198:5060 -> 192.168.1.2:5061
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.1.2:5061;branch=z9hG4bKwdtfykdg;rport
From: "Jaume" <sip:xo...@sip.voipbuster.com>;tag=ipxie
Call-ID: crhjzcr...@192.168.1.2
CSeq: 5 CANCEL
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Length: 0


#
U 2007/11/15 23:50:10.070524 194.221.62.198:5060 -> 192.168.1.2:5061
SIP/2.0 487 Request terminated
Via: SIP/2.0/UDP 192.168.1.2:5061;branch=z9hG4bKwdtfykdg;rport
From: "Jaume" <sip:xo...@sip.voipbuster.com>;tag=ipxie
Call-ID: crhjzcr...@192.168.1.2
CSeq: 5 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Length: 0


#
U 2007/11/15 23:50:10.072616 192.168.1.2:5061 -> 194.221.62.198:5060
ACK sip:003493...@sip.voipbuster.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5061;rport;branch=z9hG4bKwdtfykdg
Max-Forwards: 70
To: <sip:003493...@sip.voipbuster.com>
From: "Jaume" <sip:xo...@sip.voipbuster.com>;tag=ipxie
Call-ID: crhjzcr...@192.168.1.2
CSeq: 5 ACK
Authorization: Digest
username="xomix",realm="sip.voipbuster.com",nonce="1473847079",uri="sip:
003493...@sip.voipbuster.com",response="4052b529771ff674924db5743417cc74",algorithm=MD5
User-Agent: Twinkle/1.1
Content-Length: 0


exit
10 received, 0 dropped


Asterisk - Forbidden:

interface: eth0 (192.168.1.0/255.255.255.0)
filter: (ip or ip6) and ( port 5060 )
match:
#
U 2007/11/15 23:53:00.461220 192.168.1.2:5060 -> 194.221.62.198:5060
INVITE sip:003493...@sip.voipbuster.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK74e04e65;rport
From: "jaume" <sip:xo...@sip.voipbuster.com>;tag=as6c7b1f27
Call-ID: 425768ab7754bde2...@sip.voipbuster.com
CSeq: 102 INVITE
User-Agent: "Twinkle/1.0"
Max-Forwards: 70
Date: Fri, 16 Nov 2007 02:53:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 6332 6332 IN IP4 192.168.1.2
s=session
c=IN IP4 192.168.1.2
t=0 0
m=audio 16016 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

#
U 2007/11/15 23:53:00.729574 194.221.62.198:5060 -> 192.168.1.2:5060
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK74e04e65;rport
From: "jaume" <sip:xo...@sip.voipbuster.com>;tag=as6c7b1f27
Call-ID: 425768ab7754bde2...@sip.voipbuster.com
CSeq: 102 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
WWW-Authenticate: Digest
realm="sip.voipbuster.com",nonce="1474024719",algorithm=MD5
Content-Length: 0


#
U 2007/11/15 23:53:00.730140 192.168.1.2:5060 -> 194.221.62.198:5060
ACK sip:003493...@sip.voipbuster.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK74e04e65;rport
From: "jaume" <sip:xo...@sip.voipbuster.com>;tag=as6c7b1f27
Call-ID: 425768ab7754bde2...@sip.voipbuster.com
CSeq: 102 ACK
User-Agent: "Twinkle/1.0"
Max-Forwards: 70
Content-Length: 0


#
U 2007/11/15 23:53:00.730345 192.168.1.2:5060 -> 194.221.62.198:5060
INVITE sip:003493...@sip.voipbuster.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK7bb59136;rport
From: "jaume" <sip:xo...@sip.voipbuster.com>;tag=as6c7b1f27
Call-ID: 425768ab7754bde2...@sip.voipbuster.com
CSeq: 103 INVITE
User-Agent: "Twinkle/1.0"
Max-Forwards: 70
Authorization: Digest username="xomix", realm="sip.voipbuster.com",
algorithm=MD5, uri="sip:003493...@sip.voipbuster.com",
nonce="1474024719", response="66834b9ade02979cbcd6fa9849dc8c8c",
opaque=""
Date: Fri, 16 Nov 2007 02:53:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 6332 6333 IN IP4 192.168.1.2
s=session
c=IN IP4 192.168.1.2
t=0 0
m=audio 16016 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

#
U 2007/11/15 23:53:01.007714 194.221.62.198:5060 -> 192.168.1.2:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK7bb59136;rport
From: "jaume" <sip:xo...@sip.voipbuster.com>;tag=as6c7b1f27
Call-ID: 425768ab7754bde2...@sip.voipbuster.com
CSeq: 103 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Length: 0


#
U 2007/11/15 23:53:01.011040 194.221.62.198:5060 -> 192.168.1.2:5060
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK7bb59136;rport
From: "jaume" <sip:xo...@sip.voipbuster.com>;tag=as6c7b1f27
Call-ID: 425768ab7754bde2...@sip.voipbuster.com
CSeq: 103 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Length: 0


#
U 2007/11/15 23:53:01.011204 192.168.1.2:5060 -> 194.221.62.198:5060
ACK sip:003493...@sip.voipbuster.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK7bb59136;rport
From: "jaume" <sip:xo...@sip.voipbuster.com>;tag=as6c7b1f27
Call-ID: 425768ab7754bde2...@sip.voipbuster.com
CSeq: 103 ACK
User-Agent: "Twinkle/1.0"
Max-Forwards: 70
Content-Length: 0


exit
7 received, 0 dropped

Jaume

unread,
Nov 15, 2007, 10:19:11 PM11/15/07
to asterisk-es
Creo que ya he contestado pero no veo mi post, asi que lo vuelvo a
postear...
Gracias Iñaki, claro que no me importa!

Aqui van las trazas. Para twinkle capturo el puerto 5061, para
asterisk 5060:

Twinkle - OK:

interface: eth0 (192.168.1.0/255.255.255.0)
filter: (ip or ip6) and ( port 5061 )
match:

#
U 2007/11/15 23:50:02.819485 192.168.1.2:5061 -> 194.221.62.198:5060
INVITE sip:003493...@sip.voipbuster.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5061;rport;branch=z9hG4bKkuwfcnmy
Max-Forwards: 70
To: <sip:003493...@sip.voipbuster.com>
From: "Jaume" <sip:xo...@sip.voipbuster.com>;tag=ipxie
Call-ID: crhjzcr...@192.168.1.2
CSeq: 4 INVITE
Contact: <sip:xo...@192.168.1.2:5061>
Content-Type: application/sdp
Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
Supported: replaces,norefersub,100rel
User-Agent: Twinkle/1.1
Content-Length: 304

v=0
o=xomix 1434298066 314242592 IN IP4 192.168.1.2
s=-
c=IN IP4 192.168.1.2
t=0 0
m=audio 8000 RTP/AVP 98 97 8 0 3 101
a=rtpmap:98 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

#
U 2007/11/15 23:50:03.086181 194.221.62.198:5060 -> 192.168.1.2:5061
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.2:5061;branch=z9hG4bKkuwfcnmy;rport
From: "Jaume" <sip:xo...@sip.voipbuster.com>;tag=ipxie
Call-ID: crhjzcr...@192.168.1.2
CSeq: 4 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
WWW-Authenticate: Digest
realm="sip.voipbuster.com",nonce="1473847079",algorithm=MD5
Content-Length: 0


#
U 2007/11/15 23:50:03.088417 192.168.1.2:5061 -> 194.221.62.198:5060
ACK sip:003493...@sip.voipbuster.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5061;rport;branch=z9hG4bKkuwfcnmy
Max-Forwards: 70
To: <sip:003493...@sip.voipbuster.com>
From: "Jaume" <sip:xo...@sip.voipbuster.com>;tag=ipxie
Call-ID: crhjzcr...@192.168.1.2
CSeq: 4 ACK
User-Agent: Twinkle/1.1
Content-Length: 0


#
U 2007/11/15 23:50:03.089795 192.168.1.2:5061 -> 194.221.62.198:5060
INVITE sip:003493...@sip.voipbuster.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5061;rport;branch=z9hG4bKwdtfykdg
Max-Forwards: 70
To: <sip:003493...@sip.voipbuster.com>
From: "Jaume" <sip:xo...@sip.voipbuster.com>;tag=ipxie
Call-ID: crhjzcr...@192.168.1.2
CSeq: 5 INVITE
Contact: <sip:xo...@192.168.1.2:5061>
Content-Type: application/sdp
Authorization: Digest
username="xomix",realm="sip.voipbuster.com",nonce="1473847079",uri="sip:
003493...@sip.voipbuster.com",response="4052b529771ff674924db5743417cc74",algorithm=MD5
Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
Supported: replaces,norefersub,100rel
User-Agent: Twinkle/1.1
Content-Length: 304

v=0
o=xomix 1434298066 314242592 IN IP4 192.168.1.2
s=-
c=IN IP4 192.168.1.2
t=0 0
m=audio 8000 RTP/AVP 98 97 8 0 3 101
a=rtpmap:98 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

#
U 2007/11/15 23:50:03.360142 194.221.62.198:5060 -> 192.168.1.2:5061
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.2:5061;branch=z9hG4bKwdtfykdg;rport
From: "Jaume" <sip:xo...@sip.voipbuster.com>;tag=ipxie
Call-ID: crhjzcr...@192.168.1.2
CSeq: 5 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Length: 0


#
U 2007/11/15 23:50:03.409210 194.221.62.198:5060 -> 192.168.1.2:5061
SIP/2.0 183 Session progress
Via: SIP/2.0/UDP 192.168.1.2:5061;branch=z9hG4bKwdtfykdg;rport
From: "Jaume" <sip:xo...@sip.voipbuster.com>;tag=ipxie
To: <sip:
003493...@sip.voipbuster.com>;tag=c91710acc92b10ac470ca8322fdbf0
Contact: sip:003493...@194.221.62.198:5060
Call-ID: crhjzcr...@192.168.1.2
CSeq: 5 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Type: application/sdp
Content-Length: 198

v=0
o=xomix 1195181403 1195181403 IN IP4 194.120.0.164
s=SIP Call
c=IN IP4 194.120.0.164
t=0 0
m=audio 41804 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20

#
U 2007/11/15 23:50:09.819972 192.168.1.2:5061 -> 194.221.62.198:5060
CANCEL sip:003493...@sip.voipbuster.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5061;rport;branch=z9hG4bKwdtfykdg
Max-Forwards: 70
To: <sip:003493...@sip.voipbuster.com>
From: "Jaume" <sip:xo...@sip.voipbuster.com>;tag=ipxie
Call-ID: crhjzcr...@192.168.1.2
CSeq: 5 CANCEL
User-Agent: Twinkle/1.1
Content-Length: 0


#
U 2007/11/15 23:50:10.067357 194.221.62.198:5060 -> 192.168.1.2:5061
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.1.2:5061;branch=z9hG4bKwdtfykdg;rport
From: "Jaume" <sip:xo...@sip.voipbuster.com>;tag=ipxie
Call-ID: crhjzcr...@192.168.1.2
CSeq: 5 CANCEL
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Length: 0


#
U 2007/11/15 23:50:10.070524 194.221.62.198:5060 -> 192.168.1.2:5061
SIP/2.0 487 Request terminated
Via: SIP/2.0/UDP 192.168.1.2:5061;branch=z9hG4bKwdtfykdg;rport
From: "Jaume" <sip:xo...@sip.voipbuster.com>;tag=ipxie
Call-ID: crhjzcr...@192.168.1.2
CSeq: 5 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Length: 0


#
U 2007/11/15 23:50:10.072616 192.168.1.2:5061 -> 194.221.62.198:5060
ACK sip:003493...@sip.voipbuster.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5061;rport;branch=z9hG4bKwdtfykdg
Max-Forwards: 70
To: <sip:003493...@sip.voipbuster.com>
From: "Jaume" <sip:xo...@sip.voipbuster.com>;tag=ipxie
Call-ID: crhjzcr...@192.168.1.2
CSeq: 5 ACK
Authorization: Digest
username="xomix",realm="sip.voipbuster.com",nonce="1473847079",uri="sip:
003493...@sip.voipbuster.com",response="4052b529771ff674924db5743417cc74",algorithm=MD5
User-Agent: Twinkle/1.1
Content-Length: 0


exit
10 received, 0 dropped


Asterisk - Forbidden:

interface: eth0 (192.168.1.0/255.255.255.0)
filter: (ip or ip6) and ( port 5060 )
match:
#
U 2007/11/15 23:53:00.461220 192.168.1.2:5060 -> 194.221.62.198:5060
INVITE sip:003493...@sip.voipbuster.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK74e04e65;rport
From: "jaume" <sip:xo...@sip.voipbuster.com>;tag=as6c7b1f27
Call-ID: 425768ab7754bde2...@sip.voipbuster.com
CSeq: 102 INVITE
User-Agent: "Twinkle/1.0"
Max-Forwards: 70
Date: Fri, 16 Nov 2007 02:53:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 6332 6332 IN IP4 192.168.1.2
s=session
c=IN IP4 192.168.1.2
t=0 0
m=audio 16016 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

#
U 2007/11/15 23:53:00.729574 194.221.62.198:5060 -> 192.168.1.2:5060
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK74e04e65;rport
From: "jaume" <sip:xo...@sip.voipbuster.com>;tag=as6c7b1f27
Call-ID: 425768ab7754bde2...@sip.voipbuster.com
CSeq: 102 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
WWW-Authenticate: Digest
realm="sip.voipbuster.com",nonce="1474024719",algorithm=MD5
Content-Length: 0


#
U 2007/11/15 23:53:00.730140 192.168.1.2:5060 -> 194.221.62.198:5060
ACK sip:003493...@sip.voipbuster.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK74e04e65;rport
From: "jaume" <sip:xo...@sip.voipbuster.com>;tag=as6c7b1f27
Call-ID: 425768ab7754bde2...@sip.voipbuster.com
CSeq: 102 ACK
User-Agent: "Twinkle/1.0"
Max-Forwards: 70
Content-Length: 0


#
U 2007/11/15 23:53:00.730345 192.168.1.2:5060 -> 194.221.62.198:5060
INVITE sip:003493...@sip.voipbuster.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK7bb59136;rport
From: "jaume" <sip:xo...@sip.voipbuster.com>;tag=as6c7b1f27
Call-ID: 425768ab7754bde2...@sip.voipbuster.com
CSeq: 103 INVITE
User-Agent: "Twinkle/1.0"
Max-Forwards: 70
Authorization: Digest username="xomix", realm="sip.voipbuster.com",
algorithm=MD5, uri="sip:003493...@sip.voipbuster.com",
nonce="1474024719", response="66834b9ade02979cbcd6fa9849dc8c8c",
opaque=""
Date: Fri, 16 Nov 2007 02:53:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 6332 6333 IN IP4 192.168.1.2
s=session
c=IN IP4 192.168.1.2
t=0 0
m=audio 16016 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

#
U 2007/11/15 23:53:01.007714 194.221.62.198:5060 -> 192.168.1.2:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK7bb59136;rport
From: "jaume" <sip:xo...@sip.voipbuster.com>;tag=as6c7b1f27
Call-ID: 425768ab7754bde2...@sip.voipbuster.com
CSeq: 103 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Length: 0


#
U 2007/11/15 23:53:01.011040 194.221.62.198:5060 -> 192.168.1.2:5060
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK7bb59136;rport
From: "jaume" <sip:xo...@sip.voipbuster.com>;tag=as6c7b1f27
Call-ID: 425768ab7754bde2...@sip.voipbuster.com
CSeq: 103 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Length: 0


#
U 2007/11/15 23:53:01.011204 192.168.1.2:5060 -> 194.221.62.198:5060
ACK sip:003493...@sip.voipbuster.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK7bb59136;rport
From: "jaume" <sip:xo...@sip.voipbuster.com>;tag=as6c7b1f27
Call-ID: 425768ab7754bde2...@sip.voipbuster.com
CSeq: 103 ACK
User-Agent: "Twinkle/1.0"
Max-Forwards: 70
Content-Length: 0


exit
7 received, 0 dropped

Iñaki Baz Castillo

unread,
Nov 16, 2007, 4:17:57 AM11/16/07
to aster...@googlegroups.com
Tontería, pero prueba a poner el User-Agent en Asterisk sin comillas:

User-Agent: "Twinkle/1.0" -> User-Agent: Twinkle/1.0

(supongo que valdrá si quitas las comillas en el parámetro en sip.conf.

Jaume

unread,
Nov 16, 2007, 10:13:37 PM11/16/07
to asterisk-es
He quitado las comillas y sigo con el Forbidden...

Hay un header que veo distinto, y es el Call-Id. Twinkle lo deja con
@192.168.1.2 mientras que * lo pone @sip.voipbuster.com. Que es el
header call-id? No parece mi identificacion de llamada, sino algo
propio de la sesion... Como puedo modificarlo en *?

La otra diferencia, que no creo que afecte, es el orden de los headers
SIP.

Iñaki Baz Castillo

unread,
Nov 19, 2007, 5:39:21 AM11/19/07
to aster...@googlegroups.com
El Saturday 17 November 2007 04:13:37 Jaume escribió:
> He quitado las comillas y sigo con el Forbidden...
>
> Hay un header que veo distinto, y es el Call-Id. Twinkle lo deja con
> @192.168.1.2 mientras que * lo pone @sip.voipbuster.com. Que es el
> header call-id? No parece mi identificacion de llamada, sino algo
> propio de la sesion... Como puedo modificarlo en *?

El Call-ID es una cabecera que crea quien inicia el diálogo, y cada UAC lo
hace como quiere (el carácter @ es perfectamente válido). Apuesto 100% a que
no tiene nada que vercon eso.


> La otra diferencia, que no creo que afecte, es el orden de los headers
> SIP.

No debería en absoluto.


¿Están ambos (Twinkle y Asterisk) tras el mismo NAT y sin usar STUN ni nada?
¿tienes configurado externip y esas cosas en Asterisk?

Jaume

unread,
Nov 23, 2007, 5:59:50 PM11/23/07
to asterisk-es
Pues por el NAT, en realidad es en la misma maquina. O sea que si,
iguales condiciones (NAT "de puertos" con una sola IP publica
dinamica).

Y en asterisk no tengo ninguna configuracion especial para NAT. A que
te refieres con externip y eso? externip con Ip dinamica?
A notar que con adiptel, ningun problema, solo con voipbuster.

batik

unread,
Nov 28, 2007, 9:50:52 AM11/28/07
to asterisk-es
Hola, podias enviar la configuracion que utilizas con adiptel, a mi me
interesa bastante trabajar con este proveedor.

Saludos

On 14 nov, 01:55, Jaume <txo...@gmail.com> wrote:
> Hola,
>
> Estoy usando Asterisk 1.4.13 y tengo configurados dos proveedores via
> SIP, adiptel y voipbuster. Con adiptel no tengo ningun problema, pero
> con voipbuster, cada vez que intento hacer una llamada recibo un
> Forbidden. El log exacto es:
>
> [Nov 13 04:15:56] WARNING[27316] chan_sip.c: Received response:
> "Forbidden" from '"wifiphone"
> <sip:my_u...@sip.voipbuster.com>;tag=as3fdc4519'

Jaume

unread,
Jan 2, 2008, 11:52:23 PM1/2/08
to asterisk-es
Hola,

Con mucho retraso (disculpa...) te envio la configuracion que uso para
adiptel. Yo solo la uso para recibir llamadas de un DID.

sip.conf:

[general]
context=default
register => 515xxxxxx:pass...@sip2.adiptel.com

[adiptel]
type=friend
username=515xxxxxx
password=_tu-password_
host=sip2.adiptel.com
fromuser=515xxxxxx
fromdomain=sip2.adiptel.com
canreinvite=no
insecure=very
qualify=no
nat=yes
context=adiptel

extensions.conf:
[internal]
include => adiptel_out

[adiptel]
exten => s,1,Answer()
exten => s,n,Dial(SIP/tc,25,r)
exten => s,n,Hangup()

[adiptel_out]
exten => _515XXXXXX,1,Dial(SIP/${EXTEN}@adiptel)
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