[LARGO] No puedo llamar desde una extension registrada

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RazaMetaL | Only The Good Die Young

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Feb 22, 2008, 12:51:19 PM2/22/08
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Hola,

Acabo de instalar asterisk 1.4.8 en un debian etch y tengo un problema
al cual no le encuentro solución :(

El lio es que registro mi extensión pero al marcar se queda unos
segundos mudo y luego me da tono de ocupado, en el CLI nunca apace que
se ha ejecutado Dial:

Mi extension es 10003 y la ip 123.123.123.123 (por seguridad he
modificado las direcciones ip verdaderas):


SIP*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
vm_2000 132.132.132.1 5060
Unmonitored
1002 (Unspecified) D N 0
UNKNOWN
10001 (Unspecified) D N 0
UNKNOWN
10003/10003 123.123.123.123 D 5060 OK
(140 ms)
10002/10002 123.123.123.122 D N 21402
Unmonitored
3343374/3343374 (Unspecified) D N 0
Unmonitored

SIP*CLI> sip show peer 10003
SIP*CLI>

* Name : 10003
Secret : <Set>
MD5Secret : <Not set>
Context : test
Subscr.Cont. : <Not set>
Language :
Accountcode : 10003
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Mailbox :
VM Extension : *97
LastMsgsSent : 32767/65535
Call limit : 0
Dynamic : Yes
Callerid : "" <10003>
MaxCallBR : 384 kbps
Expire : 3514
Insecure : no
Nat : RFC3581
ACL : No
T38 pt UDPTL : Yes
CanReinvite : No
PromiscRedir : No
User=Phone : No
Video Support: Yes
Trust RPID : No
Send RPID : No
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : rfc2833
LastMsg : 0
ToHost :
Addr->IP : 123.123.123.123 Port 5060
Defaddr->IP : 0.0.0.0 Port 5060
Def. Username: 10003
SIP Options : (none)
Codecs : 0x1c050f (g723|gsm|ulaw|alaw|g729|ilbc|h261|h263|h263p)
Codec Order : (g729:20,g723:30,ilbc:30,gsm:20,ulaw:20,alaw:20)
Auto-Framing: No
Status : OK (134 ms)
Useragent :
Reg. Contact : sip:10...@123.123.123.123:5060;user=phone;transport=udp


La configuración en sip.con :

[10003]
type=friend
username=10003
accountcode=10003
regexten=10003
callerid=10003
;amaflags=billing
secret=10003
nat=no
dtmfmode=RFC2833
qualify=yes
canreinvite=no
disallow=all
allow=h261
allow=h263
allow=h263p
allow=g729
allow=g723
allow=ilbc
allow=gsm
allow=ulaw
allow=alaw
host=dynamic
context=test
regseconds=0
cancallforward=yes

La configuración de vm_2000 (el trunk):

[vm_2000]
host=132.132.132.1
type=peer
insecure=very
disallow=all
allow=g729

El contexto test de extension.conf :
SIP*CLI> dialplan show test
[ Context 'test' created by 'pbx_config' ]
'_1X.' => 1. Dial(SIP/vm_2000/${EXTEN}|30|r) [pbx_config]
2. Hangup() [pbx_config]

-= 1 extension (2 priorities) in 1 context. =-

El debug cuando marco 14163503000 :

SIP*CLI> sip set debug peer 10003
SIP Debugging Enabled for IP: 123.123.123.123:5060
SIP*CLI>
<--- SIP read from 123.123.123.123:5060 --->
INVITE sip:14163...@0.0.0.0;user=phone SIP/2.0
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK97e2712f8676dbe8
From: Guillermo Salas M <sip:10...@0.0.0.0;user=phone>;tag=2810720510
To: <sip:14163...@0.0.0.0;user=phone>
Call-ID: 24354...@123.123.123.123
CSeq: 1 INVITE
Contact: Guillermo Salas M
<sip:10...@123.123.123.123:5060;user=phone;transport=udp>
User-Agent: Cisco-CP7905/8.0.0-060111A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE
Supported: replaces, 100rel
Proxy-Authorization: Digest
username="10003",realm="asterisk",nonce="79462de2",uri="sip:14163...@0.0.0.0",response="2e905a8598ccdee267eb3b1e4a2809aa"
Expires: 300
Content-Length: 279
Content-Type: application/sdp

v=0
o=10003 62502 62502 IN IP4 123.123.123.123
s=Cisco 7905 SIP Call
c=IN IP4 123.123.123.123
t=0 0
m=audio 16384 RTP/AVP 18 8 0 101
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<------------->
--- (14 headers 12 lines) ---
Sending to 123.123.123.123 : 5060 (no NAT)
Using INVITE request as basis request - 24354...@123.123.123.123
SIP*CLI>
<--- Reliably Transmitting (no NAT) to 123.123.123.123:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
123.123.123.123:5060;branch=z9hG4bK97e2712f8676dbe8;received=123.123.123.123
From: Guillermo Salas M <sip:10...@0.0.0.0;user=phone>;tag=2810720510
To: <sip:14163...@0.0.0.0;user=phone>;tag=as4da09b27
Call-ID: 24354...@123.123.123.123
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="31225f9a"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '24354...@123.123.123.123' in
32000 ms (Method: INVITE)
Found user '10003'
SIP*CLI>
<--- SIP read from 123.123.123.123:5060 --->
INVITE sip:14163...@132.132.132.132;user=phone SIP/2.0
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK97e2712f8676dbe8
From: Guillermo Salas M <sip:10...@132.132.132.132;user=phone>;tag=2810720510
To: <sip:14163...@132.132.132.132;user=phone>
Call-ID: 24354...@123.123.123.123
CSeq: 1 INVITE
Contact: Guillermo Salas M
<sip:10...@123.123.123.123:5060;user=phone;transport=udp>
User-Agent: Cisco-CP7905/8.0.0-060111A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE
Supported: replaces, 100rel
Proxy-Authorization: Digest
username="10003",realm="asterisk",nonce="31225f9a",uri="sip:14163...@0.0.0.0",response="7b8e448c209e8f57584ba42244e1f49e"
Expires: 300
Content-Length: 279
Content-Type: application/sdp

v=0
o=10003 62552 62552 IN IP4 123.123.123.123
s=Cisco 7905 SIP Call
c=IN IP4 123.123.123.123
t=0 0
m=audio 16384 RTP/AVP 18 8 0 101
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<------------->
--- (14 headers 12 lines) ---
Ignoring this INVITE request
Retransmitting #1 (no NAT) to 123.123.123.123:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
123.123.123.123:5060;branch=z9hG4bK97e2712f8676dbe8;received=123.123.123.123
From: Guillermo Salas M <sip:10...@0.0.0.0;user=phone>;tag=2810720510
To: <sip:14163...@0.0.0.0;user=phone>;tag=as4da09b27
Call-ID: 24354...@123.123.123.123
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="31225f9a"
Content-Length: 0


---
SIP*CLI>
<--- SIP read from 123.123.123.123:5060 --->
INVITE sip:14163...@132.132.132.132;user=phone SIP/2.0
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK97e2712f8676dbe8
From: Guillermo Salas M <sip:10...@132.132.132.132;user=phone>;tag=2810720510
To: <sip:14163...@132.132.132.132;user=phone>
Call-ID: 24354...@123.123.123.123
CSeq: 1 INVITE
Contact: Guillermo Salas M
<sip:10...@123.123.123.123:5060;user=phone;transport=udp>
User-Agent: Cisco-CP7905/8.0.0-060111A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE
Supported: replaces, 100rel
Proxy-Authorization: Digest
username="10003",realm="asterisk",nonce="31225f9a",uri="sip:14163...@0.0.0.0",response="7b8e448c209e8f57584ba42244e1f49e"
Expires: 300
Content-Length: 279
Content-Type: application/sdp

v=0
o=10003 62652 62652 IN IP4 123.123.123.123
s=Cisco 7905 SIP Call
c=IN IP4 123.123.123.123
t=0 0
m=audio 16384 RTP/AVP 18 8 0 101
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<------------->
--- (14 headers 12 lines) ---
Ignoring this INVITE request
Retransmitting #2 (no NAT) to 123.123.123.123:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
123.123.123.123:5060;branch=z9hG4bK97e2712f8676dbe8;received=123.123.123.123
From: Guillermo Salas M <sip:10...@0.0.0.0;user=phone>;tag=2810720510
To: <sip:14163...@0.0.0.0;user=phone>;tag=as4da09b27
Call-ID: 24354...@123.123.123.123
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="31225f9a"
Content-Length: 0


---
Retransmitting #3 (no NAT) to 123.123.123.123:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
123.123.123.123:5060;branch=z9hG4bK97e2712f8676dbe8;received=123.123.123.123
From: Guillermo Salas M <sip:10...@0.0.0.0;user=phone>;tag=2810720510
To: <sip:14163...@0.0.0.0;user=phone>;tag=as4da09b27
Call-ID: 24354...@123.123.123.123
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="31225f9a"
Content-Length: 0


---
Retransmitting #4 (no NAT) to 123.123.123.123:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
123.123.123.123:5060;branch=z9hG4bK97e2712f8676dbe8;received=123.123.123.123
From: Guillermo Salas M <sip:10...@0.0.0.0;user=phone>;tag=2810720510
To: <sip:14163...@0.0.0.0;user=phone>;tag=as4da09b27
Call-ID: 24354...@123.123.123.123
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="31225f9a"
Content-Length: 0


---
Retransmitting #5 (no NAT) to 123.123.123.123:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
123.123.123.123:5060;branch=z9hG4bK97e2712f8676dbe8;received=123.123.123.123
From: Guillermo Salas M <sip:10...@0.0.0.0;user=phone>;tag=2810720510
To: <sip:14163...@0.0.0.0;user=phone>;tag=as4da09b27
Call-ID: 24354...@123.123.123.123
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="31225f9a"
Content-Length: 0


---
Retransmitting #6 (no NAT) to 123.123.123.123:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
123.123.123.123:5060;branch=z9hG4bK97e2712f8676dbe8;received=123.123.123.123
From: Guillermo Salas M <sip:10...@0.0.0.0;user=phone>;tag=2810720510
To: <sip:14163...@0.0.0.0;user=phone>;tag=as4da09b27
Call-ID: 24354...@123.123.123.123
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="31225f9a"
Content-Length: 0


La configuración sip:

Global Settings:
----------------
SIP Port: 5060
Bindaddress: 0.0.0.0
Videosupport: Yes
AutoCreatePeer: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: Yes
Promsic. redir: No
SIP domain support: No
Call to non-local dom.: Yes
URI user is phone no: No
Our auth realm asterisk
Realm. auth: No
Always auth rejects: No
Call limit peers only: Yes
Direct RTP setup: No
User Agent: Asterisk PBX
MWI checking interval: 10 secs
Reg. context: (not set)
Caller ID: Unknown
From: Domain:
Record SIP history: Off
Call Events: Off
IP ToS SIP: CS3
IP ToS RTP audio: EF
IP ToS RTP video: AF41
T38 fax pt UDPTL: Yes
RFC2833 Compensation: No
SIP realtime: Disabled

Global Signalling Settings:
---------------------------
Codecs: 0x3c050f
(g723|gsm|ulaw|alaw|g729|ilbc|h261|h263|h263p|h264)
Codec Order: ilbc:30,g729:20,g723:30,gsm:20,alaw:20,ulaw:20
T1 minimum: 100
Relax DTMF: No
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 0 (Disabled)
RTP Hold Timeout: 0 (Disabled)
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: Yes
Pedantic SIP support: No
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Notify ringing state: Yes
Notify hold state: Yes
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No

Default Settings:
-----------------
Context: test
Nat: RFC3581
DTMF: rfc2833
Qualify: 0
Use ClientCode: No
Progress inband: Never
Language: (Defaults to English)
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: *97


A alguien se le ocurre cual sería la causa de mi problema? Es la
primera vez que me ocurre esto :(


Saludos,


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Linux User: 255902

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Ramses II

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Feb 23, 2008, 6:35:11 AM2/23/08
to aster...@googlegroups.com
¿Has forzado codec's en ambas puntas?.


Saludos,

Ramses

> -----Mensaje original-----
> De: aster...@googlegroups.com
> [mailto:aster...@googlegroups.com] En nombre de RazaMetaL
> | Only The Good Die Young
> Enviado el: viernes, 22 de febrero de 2008 18:51
> Para: aster...@googlegroups.com
> Asunto: [Asterisk-ES] [LARGO] No puedo llamar desde una
> extension registrada

> 1635...@0.0.0.0",response="2e905a8598ccdee267eb3b1e4a2809aa"

> 1635...@0.0.0.0",response="7b8e448c209e8f57584ba42244e1f49e"

> 1635...@0.0.0.0",response="7b8e448c209e8f57584ba42244e1f49e"

RazaMetaL | Only The Good Die Young

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Feb 23, 2008, 9:28:45 PM2/23/08
to aster...@googlegroups.com
On Sat, Feb 23, 2008 at 6:35 AM, Ramses II <ramses....@gmail.com> wrote:
>
> ¿Has forzado codec's en ambas puntas?.
>

No. voy a probarlo y les aviso si funciona.

Adrià Vidal

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Feb 24, 2008, 5:04:59 AM2/24/08
to aster...@googlegroups.com
Creo que el Cisco que tienes al otro lado no es autentifica correctamente, prueba a ponerlo sin pw a ver si la llamada se cursa, y para no liar más las cosas mejor que le quites la r de las opciones de Dial que solo puede que confundirte....

adria vidal
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Adrià Vidal
adria...@gmail.com
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