Ans: Asterisk 是由社群所開發的套件,為了有效改善套件,使用群除了免費使用外,還能將所發現的 bug 送交給開發社群來解決,而如何有效提供 bug 訊息,官方有提供教學,請參閱 Getting a Backtrace。
Ans: 有兩種方法
sed -i 's/ast_verb(4, "ast_get_srv: SRV lookup for/ast_verb(6, "ast_get_srv: SRV lookup for/' main/srv.c sed -i 's/ast_verb(4, "doing dnsmgr_lookup for/ast_verb(6, "doing dnsmgr_lookup for/' main/dnsmgr.c
Ans: 假設撥入的 context 是
[incoming] exten => 22168434,1,Answer() exten => 22168434,n,Background(demo-thanks) exten => _101,1,Playback(vm-nobodyavail) exten => _101,n,Hangup() exten => i,1,Playback(pbx-invalid) exten => i,2,Goto(s,1) exten => t,1,Playback(vm-goodbye) exten => t,2,Hangup()
問題:電話撥入後,聽到 demo-thanks,但不會等到 timeout 就立即被掛斷,CLI Logs
== Using SIP RTP CoS
-- Executing [22168434@incoming:1] Answer("SIP/202.101.42.32-09ce0e30", "") in new stack
-- Executing [22168434@incoming:2] BackGround("SIP/202.101.42.32-09ce0e30", "demo-thanks") in new stack
-- <SIP/202.101.42.32-09ce0e30> Playing 'demo-thanks.gsm' (language 'en')
-- Auto fallthrough, channel 'SIP/202.101.42.32-09ce0e30' status is 'UNKNOWN'
解決:在 extensions.conf 加上
[general] ... autofallthrough=no
或者,修改 dialplan
... exten => 22168434,n,Background(demo-thanks) exten => 22168434,n,WaitExten() <===== here exten => _101,1,Playback(vm-nobodyavail) ...
Ans: 若沒有使用 H323 協定,可以將它關閉。編輯 modules.conf,增加這一行:
noload => chan_ooh323.so
重啟 Asterisk 服務。
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