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CM Phone不能在Asterisk注册成功的原因是(切底解决!!!)

本帖最後由 角色 於 2012-10-2 02:01 編輯

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注册不成功主要是在sip.conf没有加上
[general]
pedantic=yes

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下面是一般的Authorized问题时候怎样出。

CM Phone 的UAS出的nonce是特别长,一般是8位,但是它超过30。
  1. On 8/15/07, Stanisław Pitucha <stanis at zimbra-1.gradwell.net> wrote:
  2. >
  3. > ----- "Rizwan Hisham" <rizwanhasham at gmail.com> wrote:
  4. > > WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
  5. > nonce="584760da"
  6. >
  7. > > Authorization: Digest username="bernart48", realm="asterisk",
  8. > algorithm=MD5, uri="sip:bernart48 at 64.182.161.2:9060", nonce="584760da",
  9. > response="948d3923bf2df47eca17c572713af2c7", opaque=""
  10. >
  11. > > What i dont know, and would very much like to know, is what is the
  12. > > purpose of this parameter in sip packets?
  13. >
  14. > It's kind of challenge algorithm. What you see in "response" is not
  15. > MD5(password), but MD5('password', 'realm', ..., 'nonce'). Nonce is
  16. > generated by server so that you don't get the same hash for for every
  17. > authorization by that user. It prevents someone who can see only one way
  18. > communication from breaking your sip session + makes breaking hash a little
  19. > bit harder.
  20. > Nonce should be unique per authorization.
  21. > If nonce wasn't used you could reuse the same response in next connection
  22. > even if you don't know the real password.
  23. >
複製代碼

回復 37# 角色

非常感謝

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你可以参照我下面的帖子:

http://www.telecom-cafe.com/forum/viewthread.php?tid=5261

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回復 35# 角色

明白, 謝謝

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估计你所指No,可能是No Authentication,要自己写script,然后做cron job。

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回復 33# 角色

Xpenology, Digium GUI, Synology 套件

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本帖最後由 角色 於 2013-12-7 18:31 編輯

你用什么GUI?Asterisk-GUI吗?还用你是用什么东西起Asterisk Server呢?你自己compiled?

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回復 31# 角色

我而家係重啟asterisk 或嚮trunk裡面 edit > save(其實無改嘢) > apply 就註冊番

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是否可以sip reload可以搞定呢?

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請教: 基本上每朝瞓醒都發現comnet 狀態為 "No", 要手動再註冊先打到電話, 請問要點set? 謝謝
2013-12-06_154201.png

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請問角色兄,你的第一版的:
注册不成功主要是在sip.conf没有加上
[general]
pedantic=yes

我的是Elastix, 找不到在哪里設這個選項呢,我不是太敢手動改.conf.

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回復 27# bubblestar

bubblestar师兄,谢谢你的信息,问题已经已经切底解决。

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本帖最後由 bubblestar 於 2012-10-1 21:37 編輯

If qualify is set to "yes" then, by the looks of it, Asterisk will use the information about round trip time to decide whether or not to bother registering. If the SIP OPTIONS packet doesn't receive a response, it assumes the server is unreachable and probably doesn't bother trying to register. Switching off "qualify" is obviously necessary with such providers.

It's also generally a good idea to have qualify=no for softphones and maybe some hardphones. the OPTIONS packets can cause problems with them.

Purpose of qualify=yes
On the other hand, one of the main benefits of qualify=yes is to detect network problems with peers.

We send a lot of calls via a service provider using SIP but we have qualify-yes set so that if it becomes unreachable the dial fails
immediatly without having to wait for a timeout which enables us to
seamlessly failover to an ISDN or other connection.

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本帖最後由 角色 於 2012-10-1 21:34 編輯

同一个Network,Router,UAS从Asterisk换成Zoiper就马上可以打出打入。他们之间有什么区别呢?

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本帖最後由 角色 於 2012-10-1 20:00 編輯

关于NAT and Trunk,下面的帖子写的最清楚。

http://www.informaticapressapoch ... -sip-to-rtp-part-5/

http://blog.lithiumblue.com/2007 ... sip-calls-from.html

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