Re: Help ~~ , I`m New

[A-Lang Hsu - Asterisk/Linux/IT Consultant]

在 telnet 執行 asterisk -rx "sip show users" 檢查一下帳號密碼使否正確

將 X-Lite 的設定畫面貼上來

如果是遠端分機，先在本地端註冊看看

On Tue, Apr 2, 2013 at 10:36 PM, <regis...@gmail.com> wrote:

Hi All,

我是新手, 剛刷完 OSSLAB 的 NextPBX

請問一下，我用 X-Lite 設定完後會一直出現 "Registration error: 403 forbidden(Bad auth)"

我確定 IP/USER/PW 都正確, 可是就一直無法使用, 麻煩幫幫我@@

Global Settings:

----------------
UDP Bindaddress: 0.0.0.0:5060
TCP SIP Bindaddress: Disabled
TLS SIP Bindaddress: 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: No
Allow promisc. redir: No
Enable call counters: No
SIP domain support: No
Realm. auth: No
Our auth realm asterisk
Use domains as realms: No
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: No
User Agent: myATA
SDP Session Name: Asterisk PBX 1.8.17.0
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Legacy userfield parse: No
Caller ID: asterisk
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
Q.850 Reason header: No
Store SIP_CAUSE: No

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

Network Settings:
---------------------------
SIP address remapping: Disabled, no localnet list
Externhost: <none>
Externaddr: (null)
Externrefresh: 10

Global Signalling Settings:
---------------------------
Codecs: 0xe (gsm|ulaw|alaw)
Codec Order: ulaw:20,alaw:20,gsm:20
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: 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: No
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: No
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
Max forwards: 70

Default Settings:
-----------------
Allowed transports: UDP
Outbound transport: UDP
Context: default
Force rport: Yes
DTMF: rfc2833
Qualify: 0
Use ClientCode: No
Progress inband: Never
Language:
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: asterisk

----

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[regis...@gmail.com]

 Dear A.Lang ,

小弟不才, 執行了 show users 後即發現問題, 原來 Extention 就是 user name , 目前可接聽了感謝!!

另再請教一個問題, 我將 PBX 加到 DMZ

從外部連可正常註冊, 但互撥時都會出現 call failed : Service unavailable

有可能是什麼原因呢 ??

A.Lang(WebAdmin)於 2013年4月3日星期三UTC+8上午9時04分05秒寫道：

[A-Lang Hsu - Asterisk/Linux/IT Consultant]

請閱讀此篇

http://goo.gl/2tU4G