bubblestar

回復 323# 角色


   
You may refer to the 7th floor about ckleea 's guide for compilation.

http://www.telecom-cafe.com/foru ... d=2963&pid=8014

bubblestar

是否在同一LAN 之內?  X-lite 所在電腦有沒有FIREWALL 檔住唔俾入?

bubblestar

Thanks YH.  The screen captures give us a clear picture of the new build and let us know what can be chosen for the upgrade.

bubblestar

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

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。

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.

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 ?!

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.

bubblestar

回復 360# yiucsw


    Why don't you use the build-in Incoming Calling Rules to set up your PSTN call?

    I don't understand how you come up with this extension context as we won't use exten=_X.,1,Hangup in incoming PSTN call.

bubblestar

The code quoted by ckleea C-Hing is under the context [demo] in extensions.conf and extensions_custom.conf.

bubblestar

By default, the option in mailbox of Analog Trunk is blank.  please double check from GUI if it is still empty.

bubblestar

Like this one.

下載 (45.15 KB)

2011-2-2 21:22

bubblestar

When an incoming call come from PSTN, it will be forwarded to one of your internal extensions, for example, extension 6001.  If your Ext. 6001's voicemail is activated, please try to disable it and see what the result is.

If all the above trial is in vain, I suggest you to roll back to previous version like SVN432 or SVN437.

bubblestar

回復 385# Qnewbie


   
The default settings in Analog Trunks are for the USA.  I think it is a bit different from European's requirement so that why you might need to fine tune for adapting itself to your own territory.

BTW, IP01 has an option in CID signalling for DTMF (Denmark, Sweden and Holland).

bubblestar

What a co-incidence!  I tried to compile several times all this afternoon but stuck at inatech with bad response 404.

So disappointing!