我是用這個Compile Asterisk 1.8 的

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16^#

發表於 2011-1-10 23:06 | 只看該作者

I have just migrated my original asterisk to the new D525 ATOM. Just few recompilations of dahdi modules as I am now default to use xen kernel

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15^#

發表於 2011-1-9 21:53 | 只看該作者

Maybe I get you wrong.

OK.  As far as I know,if we use GUI to create extensions, it is NOT necessary to put those extensions in SIP.conf since they will be included in USERS.conf.  As I am not using GUI for Asterisk 1.8, all my extensions are put inside SIP.conf and do not need to touch users.conf.

Basically, we should handle extensions for IAX in the same fashion.  However, I notice from the Internet that we might face some strange problems if we check both IAX and SIP in GUI.  Hence, it is suggested that we'd better create IAX and SIP extensions separately to avoid confusion.

Those information indicates that GUI (at least Asterisk's own version) is really not the best choice at this moment.  Elastix, FreePBX, Trixbox and PiaF outperform in this respect.

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14^#

發表於 2011-1-9 17:54 | 只看該作者

回復 13# bubblestar

My 6000 is assigned to the Siemens IP phone. Works without problem.
Only I did not put any extension setting in SIP.conf.

My point is since we need to add the extensions setting in sip.conf and iax.conf. What is the point of using GUI to create users/extensions? Only you can see them under GUI.

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13^#

發表於 2011-1-9 10:32 | 只看該作者

I, myself, do not use SIP/6000. As mentioned before, extension 6000 is predefined to some other purpose, probably for auto-attendant.

I think we can adjust or fix to use extension 6000 again.

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12^#

發表於 2011-1-9 05:57 | 只看該作者

回復 7# bubblestar

So, do you set up your extension under GUI or just paste the codes into sip.conf? I believe I can't use SIP/6000 is due to the setup under GUI. The information is in users.conf not sip.conf.

Then what is the purpose of GUI?

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11^#

發表於 2011-1-7 15:18 | 只看該作者

本帖最後由 bubblestar 於 2011-1-7 15:19 編輯

回復 bubblestar

I have no problem but from time to time, if I use Dial (SIP/{exten}), someti ...
ckleea 發表於 2011-1-7 13:53

Double check if you have added $ in front of {EXTEN}, i.e., SIP/${EXTEN}

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10^#

發表於 2011-1-7 13:53 | 只看該作者

回復 9# bubblestar

I have no problem but from time to time, if I use Dial (SIP/{exten}), sometimes it does not work

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9^#

發表於 2011-1-7 12:18 | 只看該作者

回復 6# ckleea

It seems Local/Channel is a new feature introduced in Asterisk 1.8.

http://ofps.oreilly.com/titles/9780596517342/ch10.html#id3174771

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8^#

發表於 2011-1-5 22:47 | 只看該作者

用了此功能，偷懶的朋友也可以變成看似忙碌的員工了。只要在Google Calendar 設定好時間，咁樣便經常會有電話找大家。真亦假時，假亦真，老板們真係多得呢個功能唔少啦!

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7^#

發表於 2011-1-5 22:42 | 只看該作者

In my case, SIP/exten is one of the real registered extension in the sip.conf.

I have not tried Local/exten, so no ideas about that. Is "Local" one of your context in extension.conf or sip.conf?
I think the point is your designated extension must be reachable and can get the incoming call through ringing.

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6^#

發表於 2011-1-5 22:19 | 只看該作者

One naive question. I am unable to use SIP/exten. Instead I can use Local/exten. Do I need to register every extension in the sip.conf. A

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5^#

發表於 2011-1-5 15:06 | 只看該作者

本帖最後由 bubblestar 於 2011-1-5 15:20 編輯

成功利用Google Calendar 在 Asterisk 1.8 之下使用 Wake up Call.

先在網上的Google Calendar 隨便設定一個約會或什麼東西也可以。記著要剔取Alarm 或 Reminder，當時間一到，Asterisk 1.8 Server 即刻會致電給你作為提示，其中一個用途可當作Wake up call，貪睡的朋友或在公司蛇王的朋友，到時到候它會提你起身啦

/etc/asterisk/calendar.conf

[myGoogleCal]
type = caldav
url = https://www.google.com/calendar/ ... e@gmail.com/events/
user = your_username@gmail.com
secret = your_top_secret_password
refresh = 15
timeframe = 60
autoreminder = 10
channel = SIP/6688
waittime = 30
app = Playback
appdata = this-is-yr-wakeup-call ; 這裡如果選擇播放音樂檔應該沒有問題。