裝邊個 asterisk server 版本

ckleea

最近有網友遇上安裝 asterisk 的問題，唔知大家用緊邊個 asterisk 版本，同邊個linux server版本？你們的配備是怎樣呢？

我的是 asterisk 1.8 ，自已 download and compile from SVN trunk
Linux 是 Centos 5.6 16 bits
沒有Analog card 接 PSTN
用 SPA3000 and OBi110做橋接PSTN
還有 VOIP account 如 2b account 做FXO

你地是怎樣？

現在我們用的 embedded IPPBX are using mostly 1.4.xx asterisk

亞星

我啱啱安裝咗 AsteriskNow 1.6 再用 yum 上 asterisk 1.8 仲有好多野唔識要同各位師兄師組請教

ckleea

I thought you directly install asterisk 1.8. Am I correct?

亞星

我先裝了 AsteriskNow 1.6 再用 yum remove asterisk16-core, 之後再用 yum install asterisk 1.8

ckleea

I believe you have not remove the old asterisk completely.

亞星

Linux 我真係乜都唔識裝到已經好開心

ckleea

不用擔心，慢慢試。另外可參考

http://www.howtoforge.com

有很多裝server tutorial

Qnewbie

Installed ClearOS5.1+Ast1.6+SPA3000/3102 for a few months. Currently there is not Linux machine with Asterisk except IP01 with FXO port. Just for simple voice communications with different ATAs. The main problem is missing network in mainland China

freewilly

I have Dockstar + 8GB usb + Debian + Asterisk 1.8.4(with gtalk) + 3x OBi110. I've installed both ObiOn and Sipdroid on my Android phones, Obion seems working well in terms of call quality. The lack of integration of Android phone dialer, phone book and bluetooth are huge minus for Obion.
On the other hand, Sipdroid did very well.

Asterisk is working fine except google voice incoming calls.

I have to press "1" to accept the call from gtalk on Asterisk. Can't find any solution for that.

Set google voice on Obi's SP1, SP2 is connected to Asterisk.
Set SP1.X_InboundCallRoute=SP2 won't dial out, Asterisk says " == Using SIP RTP CoS mark 5" without any exension dial out.
Don't want set SP1.X_InboundCallRoute=SP2(1234) which will dial out.