Problemas con registro de extensiones , iax y sip

wysiwyg

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#1
Buenas,
He actualizado las ultimas versiones a FreePBX 2.9.0beta2.1 y resulta que ahora las extensiones que tenia creadas funcionan pero si creo una nueva , bien sea iax o sip, no hay manera de que se registren. Supongo que algun parametro se me esta pasando por alto y no hay manera...

HP ProLiant DL120 G5 Xeon E3110 3 GHz
1 x Intel Xeon E3110 / 3 GHz (DualCore)
1 GB (instalados) / 8 GB (máx.) DDR2 SDRAM ECC 800 MHz PC2-6400
1 x 250 GB estándar Serial ATA150
DVD RW

Distribución Linux: Redhat CentOS release 5.5 (Final)
Version Asterisk: Asterisk 1.6.2.13, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Versión Elastix: 2.0.1 (64 bits)
Freepbx Versión: 2.8.0.4 (ultima estable que funcionaba todo)

Quisiera que me aclarasen si existe algún problema con la nueva beta o que es lo que pudiese estar fallando....

dolar*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
2999/2999 192.168.0.25 D N A 5061 OK (29 ms)
3001/3001 192.168.0.25 D N A 5060 OK (26 ms)
3002/3002 192.168.0.26 D N A 5060 OK (309 ms)
3003/3003 192.168.0.7 D N A 10011 OK (31 ms)
3004/3004 192.168.0.7 D N A 5061 OK (35 ms)
3005/3005 192.168.0.24 D N A 5061 OK (18 ms)
3007/3007 192.168.0.20 D N A 1025 OK (24 ms)
3008/3008 87.223.235.165 D N 5060 OK (223 ms)
3009/3009 83.56.151.7 D N 5060 OK (161 ms)
3010/3010 192.168.0.7 D N A 10005 OK (29 ms)
3011/3011 192.168.0.7 D N A 10012 OK (35 ms)
3012/3012 192.168.0.7 D N A 10013 OK (26 ms)
3013/3013 192.168.0.7 D N A 10002 OK (38 ms)
3014/3014 192.168.0.20 D N A 1024 OK (38 ms)
3015/3015 192.168.0.7 D N A 10760 OK (17 ms)
3016/3016 192.168.0.24 D N A 5060 OK (18 ms)
4000 (Unspecified) D N A 5060 UNKNOWN


Parametros Globales sip settings


Global Settings:
----------------
UDP SIP Port: 5060
UDP Bindaddress: 0.0.0.0
TCP SIP Port: Disabled
TLS SIP Port: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: No
Match Auth Username: No
Allow unknown access: No
Allow subscriptions: Yes
Allow overlap dialing: Yes
Allow promsic. redir: No
Enable call counters: No
SIP domain support: Yes
Realm. auth: No
Our auth realm asterisk
Call to non-local dom.: No
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: No
User Agent: FPBX-2.9.0beta2(1.6.2.13)
SDP Session Name: Asterisk PBX 1.6.2.13
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Caller ID: Unknown
From: Domain:
Record SIP history: Off
Call Events: Off
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: -1
SIP realtime: Disabled
Qualify Freq : 60000 ms

Network QoS Settings:
---------------------------
IP ToS SIP: CS3
IP ToS RTP audio: EF
IP ToS RTP video: AF41
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: No
Jitterbuffer forced: No
Jitterbuffer max size: -1
Jitterbuffer resync: -1
Jitterbuffer impl:
Jitterbuffer log: No

Network Settings:
---------------------------
SIP address remapping: Enabled using externhost
Externhost: 87.111.66.50
Externip: 87.111.66.50:5060
Externrefresh: 10
Internal IP: 127.0.0.1:5060
Localnet: 192.168.0.0/255.255.255.0
STUN server: 0.0.0.0:0

Global Signalling Settings:
---------------------------
Codecs: 0x10e (gsm|ulaw|alaw|g729)
Codec Order: ulaw:20,gsm:20,alaw:20,g729:20
Relax DTMF: No
RFC2833 Compensation: No
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 30
RTP Hold Timeout: 300
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
Include CID: No
Notify hold state: Yes
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy: <not set>
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes

Default Settings:
-----------------
Allowed transports: UDP
Outbound transport: UDP
Context: from-sip-external
Nat: Always
DTMF: rfc2833
Qualify: 0
Use ClientCode: No
Progress inband: Never
Language: es
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: *97
Forward Detected Loops: Yes



Parametros de la extension 4000 que no conecta...


dolar*CLI> sip show peer 4000


* Name : 4000
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : from-internal
Subscr.Cont. : <Not set>
Language : es
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Mailbox : 4000@device
VM Extension : *97
LastMsgsSent : 32767/65535
Call limit : 2147483647
Dynamic : Yes
Callerid : "device" <4000>
MaxCallBR : 384 kbps
Expire : -1
Insecure : no
Nat : Always
ACL : Yes
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: -1
DirectMedia : No
PromiscRedir : No
User=Phone : No
Video Support: No
Text Support : No
Ign SDP ver : No
Trust RPID : Yes
Send RPID : No
Subscriptions: Yes
Overlap dial : Yes
Forward Loop : Yes
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost :
Addr->IP : (Unspecified) Port 5060
Defaddr->IP : 0.0.0.0 Port 5060
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username:
SIP Options : (none)
Codecs : 0x10e (gsm|ulaw|alaw|g729)
Codec Order : (ulaw:20,gsm:20,alaw:20,g729:20)
Auto-Framing : No
100 on REG : No
Status : UNKNOWN
Useragent :
Reg. Contact :
Qualify Freq : 60000 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
Parkinglot :


Tenia una extension 3000 anterior en iax y se ha conectado perfectamente antes de actualizar el sistema.... no entiendo....




Gracias de antemano...
 

asepulveda

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#2
He actualizado las ultimas versiones a FreePBX 2.9.0beta2.1
Espero que no lo hayas hecho en un servidor en producción, es una version Beta lo que sucede pueden ser miles de cosas, a mi parecer es algo de permisos
 

wysiwyg

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#3
Lamentablemente, si asi es, es un servidor en produccion y he sido tan iluso de actualizar la version beta.....

Hoy mismo me ha llegado otra version de la beta, supongo que ahora empezarán a enviar versiones hasta que esté solucionado.....

Gracias por tu interes.
Un cordial saludo.
jose.
 

asepulveda

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#4
Entonces supongo que si te dieran una droga experimental para curar la gripa la tomarias no?

De por si uno nunca debe actualizar un sistema en producción aun y cuando los releases sean estables sin antes probarlo en un conmutador de testing, un BETA jamas se pone en uno en producción, error grave
 

fmvillares

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#5
de todas formas algo te salio mal porqu8e yo uso la beta ya que soy tester de freepbx y anda de pelos pisando la freepbx 2.8 de elastix no rompe nada ams que la embedded...todo anda bien y con asterisk 1.8.y 1.6
 

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