OPENWRT上成功運行asterisk with gui 打出打入cm phone

321

首先你需要一台能刷openwrt 的路由器,而路由器的flash最小也得8M,否則不夠空間刷asterisk(套件需要3.x MB的空間,安裝前請確認有足夠的空間)
首先,ssh入router,打以下指令:
opkg update  (取得最新的軟件清單)
opkg install asterisk11-gui (也可以是asterisk18-gui)
等待安裝完後,動啟路由器就可以入到asterisk-gui界面了
以下是我用tplink 741n 改8mb flash, 64mb ram後運行asterisk gui的截圖,有什麼不懂的地方可以在此貼發問

下載 (304.65 KB)

2014-1-13 11:06

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其實32mb ram已經足夠運行openwrt + asterisk with GUI, 開機只用了25MB ram.
這應該是最便宜的asterisk方案了

321

附上sip trunk參數

下載 (51.62 KB)

2014-1-13 11:17

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現在不能打出了,我打91000出去,會忙音. 下面的log請大家幫忙分柝一下問題出在哪裡

== Using SIP RTP CoS mark 5
    -- Executing [91000@DLPN_DialPlan1:1] Macro("SIP/6080-0000000c", "trunkdial-

failover-0.3,SIP/trunk_1/1000,,trunk_1,") in new stack
    -- Executing [s@macro-trunkdial-failover-0.3:1] GotoIf("SIP/6080-0000000c", "0?1

-fmsetcid,1") in new stack
    -- Executing [s@macro-trunkdial-failover-0.3:2] GotoIf("SIP/6080-0000000c", "0?1

-setgbobname,1") in new stack
    -- Executing [s@macro-trunkdial-failover-0.3:3] Set("SIP/6080-0000000c",

"CALLERID(num)=") in new stack
    -- Executing [s@macro-trunkdial-failover-0.3:4] Set("SIP/6080-0000000c",

"CALLERID(all)=") in new stack
    -- Executing [s@macro-trunkdial-failover-0.3:5] GotoIf("SIP/6080-0000000c", "0?1

-dial,1") in new stack
    -- Executing [s@macro-trunkdial-failover-0.3:6] Set("SIP/6080-0000000c",

"CALLERID(all)=") in new stack
    -- Executing [s@macro-trunkdial-failover-0.3:7] Set("SIP/6080-0000000c",

"CALLERID(all)=") in new stack
    -- Executing [s@macro-trunkdial-failover-0.3:8] Goto("SIP/6080-0000000c", "1-

dial,1") in new stack
    -- Goto (macro-trunkdial-failover-0.3,1-dial,1)
    -- Executing [1-dial@macro-trunkdial-failover-0.3:1] Dial("SIP/6080-0000000c",

"SIP/trunk_1/1000") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/trunk_1/1000
    -- Got SIP response 503 "Service Unavailable" back from 202.0.179.3:5060
    -- SIP/trunk_1-0000000d is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [1-dial@macro-trunkdial-failover-0.3:2] GotoIf("SIP/6080-0000000c",

"0 > 0 ?1-CONGESTION,1:1-out,1") in new stack
    -- Goto (macro-trunkdial-failover-0.3,1-out,1)
    -- Executing [1-out@macro-trunkdial-failover-0.3:1] Hangup("SIP/6080-0000000c",

"") in new stack
  == Spawn extension (macro-trunkdial-failover-0.3, 1-out, 1) exited non-zero on

'SIP/6080-0000000c' in macro 'trunkdial-failover-0.3'
  == Spawn extension (DLPN_DialPlan1, 91000, 1) exited non-zero on 'SIP/6080-

0000000c'
OpenWrt*CLI>

321

似乎這裡非常冷清

角色

没有看到嘛！应该要很多settings才可以拨出，详细看其他有关帖子。

321

没有看到嘛！应该要很多settings才可以拨出，详细看其他有关帖子。
角色 發表於 2014-1-14 20:05

已經按照了你的帖子去做，還是無法打出打入。
相同的設置下，centos下編譯的asterisk11 with gui可以完全正常打出打入。
難道openwrt的asterisk11還需要按照其他插件才能打出？
另外我openwrt asterisk11可以trunk到我舊的asterisk server with cmphone,並且可以打出電話

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Inernal_timming=yes
Quality=no，也試過，依然無法打入打出

裝上了asterisk11-res-timing-pthread這個軟件包後,就可以打出打入了.

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== Using SIP RTP CoS mark 5
    -- Executing [91000@DLPN_DialPlan1:1] Macro("SIP/6000-00000000", "trunkdial-failover-0.3,SIP/trunk_1/1000,,trunk_1,") in new stack
    -- Executing [s@macro-trunkdial-failover-0.3:1] GotoIf("SIP/6000-00000000", "0?1-fmsetcid,1") in new stack
    -- Executing [s@macro-trunkdial-failover-0.3:2] GotoIf("SIP/6000-00000000", "0?1-setgbobname,1") in new stack
    -- Executing [s@macro-trunkdial-failover-0.3:3] Set("SIP/6000-00000000", "CALLERID(num)=") in new stack
    -- Executing [s@macro-trunkdial-failover-0.3:4] Set("SIP/6000-00000000", "CALLERID(all)=") in new stack
    -- Executing [s@macro-trunkdial-failover-0.3:5] GotoIf("SIP/6000-00000000", "0?1-dial,1") in new stack
    -- Executing [s@macro-trunkdial-failover-0.3:6] Set("SIP/6000-00000000", "CALLERID(all)=") in new stack
    -- Executing [s@macro-trunkdial-failover-0.3:7] Set("SIP/6000-00000000", "CALLERID(all)=") in new stack
    -- Executing [s@macro-trunkdial-failover-0.3:8] Goto("SIP/6000-00000000", "1-dial,1") in new stack
    -- Goto (macro-trunkdial-failover-0.3,1-dial,1)
    -- Executing [1-dial@macro-trunkdial-failover-0.3:1] Dial("SIP/6000-00000000", "SIP/trunk_1/1000") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/trunk_1/1000
    -- Got SIP response 503 "Service Unavailable" back from 202.0.179.3:5060
    -- SIP/trunk_1-00000001 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [1-dial@macro-trunkdial-failover-0.3:2] GotoIf("SIP/6000-00000000", "0 > 0 ?1-CONGESTION,1:1-out,1") in new stack
    -- Goto (macro-trunkdial-failover-0.3,1-out,1)
    -- Executing [1-out@macro-trunkdial-failover-0.3:1] Hangup("SIP/6000-00000000", "") in new stack
  == Spawn extension (macro-trunkdial-failover-0.3, 1-out, 1) exited non-zero on 'SIP/6000-00000000' in macro 'trunkdial-failover-0.3'
  == Spawn extension (DLPN_DialPlan1, 91000, 1) exited non-zero on 'SIP/6000-00000000'

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其中
-- Called SIP/trunk_1/1000
    -- Got SIP response 503 "Service Unavailable" back from 202.0.179.3:5060
    -- SIP/trunk_1-00000001 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)

為什麼comnet phone會feedback busy?

321

終於成功打入打出.我把所有asterisk11的package裝好了就能打出打入了

角色

回復 11# 321

恭喜你。

321

回復 12# 角色

謝謝你對我學習asterisk的幫助。
現在我的路由asterisk server已經取代了我用了兩年的asterisk PC server.
現在運作我十分滿意。
之前無法打出打入，是因為我沒有安裝 asterisk11-res-timing-pthread這個套件，並非是設定上的問題。
我稍後會重新整理，再把openwrt上安裝asterisk gui及設定cmphone出簡單的教程。
目前這裡好像還沒有兄弟在路由器上跑asterisk吧？希望有興趣的朋友可以一起討論一下這個話題