角色

没有，是免费版的。

角色

亞星

想問問如果重新由 ATCOM firmware 上 Switchfin 之後係咪唔可以用 GUI restore Switchfin 之前 backup 出黎既 config 0架?

bubblestar

轉了Switchfin 後，可以restore 以前在前版本Switchfin 所 backup 的 config 的。

ckleea

回復 346# 角色

你有冇用IAX? 免費版得兩個 accounts，唔夠用 (試機用)

bubblestar

剛剛由SVN437 upgrade 至SVN479，發覺SVN437之後的版本，無法完全restore到以前的configuration settings。  例如，Dial Plans，Outgoing Dialing Rules 及 Incoming Dialing Rules，需要重新建立。至於user extensions 就沒有問題。

原因可能就是開發者已把extensions.conf 作為主軸轉到custom.conf 及 extensions_custom.conf 了。現在你在GUI的 Admin/File Editor 入面已經找不到extensions.conf 這一項目了。
但是在WinSCP 入面看，仍可發現它的踪影。不過，它裡面只有以下兩個項目:

#include custom_conf
#include extensions_custom.conf

當你重新建立Dial Plan 之後，又會發現 extensins.conf 和 extensions_custom.conf 的內容會變成一模一樣的。之後上面的include 項目便消失了。而GUI裡Admin/File Editor 仍不會有extensions.conf 這一項的。

因此，我猜想extensions_custom.conf 已經代替了extensions.conf 了。而又由於這兩個extensions files 都是自動generate 出來的。以後自己寫的Dial Plan，我估是要放在custom.conf 入面行。

稍後試一試再証實一下。 暫時係咁多先。

角色

回復  角色

你有冇用IAX? 免費版得兩個 accounts，唔夠用 (試機用)
ckleea 發表於 2011-2-1 13:08

可以用两台PC，同时安装Zoiper，那么我就可以有4个IAX了。

角色

bubblestar

回復 350# bubblestar


   
Oh my! the new build can only restore incoming settings of IPTEL.  Outgoing Dialing Rule is missing.

Not comfortable with all these changing and will turn back to SVN437 or SVN432 which are the most stable and familiar ones that I have been using.  Also, I don't want to debug and get headache in the coming Lunar New Year holiday.

yiucsw

FXO trunk - password? or disable incoming call.
I want to disable incoming call, or wait for few minute before answer the call. where can it set?

ckleea

回復 353# yiucsw
If you do not have a catchall setting of the FXO context (or even without a context for FXO analog trunk, it will not answer

bubblestar

回復 353# yiucsw


   
Haha!  Are you sure the caller will be patient enough to wait for a few minutes before answering the call ?!

yiucsw

sorry when user call in, the switchfin boardcast an message ...asterisk is an open source... iax extension, how can i cancel the message. and wait for 20 min before answer. I think it may be voicemail system.

ckleea

回復 356# yiucsw


Not sure what you mean. In asterisk console, you can see the incoming call string but not answer if no DP context to handle. Therefore, the call will then drop.
Do you want to pass the incoming call to voicemail after certain period of ring?

yiucsw

i believe it is some setup issue. there is no problem for outgoing call, but for incoming call from FXO, an voice file is play....thank you for installing asterisk open source pbx.....
As i do not need incoming call, can you tell me how to disable the voice message.

i try to add the following:
exten=s,1,wait(1000)
but no effect.

bubblestar

回復 358# yiucsw


   
In IP01 GUI environment, if you don't want to forward incoming calls to voicemail after a certain period of time, please uncheck the option "Enable Voice for this user" when establishing your user extension.

If you want to do it in APL, you may write your incoming call context like this:

[pstn-incoming]
exten => s,1,Answer(500)                       ; the 500 inside the bracket represent 0.5 seconds after answer, the call will go to 6001
exten => s,n,Dial(SIP/6001,30,r)             ; if nobody answer after ringing 30 seconds time, the next line of hangup call is executed
exten => s,n,Hangup()

In that case, the unanswered call will not be transfered to voicemail.

yiucsw

I put the following in
file extensions_custom.conf section [DID_trunk_1_default]
exten=_X.,1,Hangup
Still there is no effect.
When call from outside to PSTN, it direct go to Voice message"thank you for install asterisk PBX....."