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Comnet phone in Asterisk忽然間不能用了

本帖最後由 321 於 2015-5-27 00:07 編輯

原本一直可以用,但是幾天前忽然不能打出打入

=========================================================================
Connected to Asterisk 11.15.0 currently running on 1043ND (pid = 2015)
  == Using SIP RTP CoS mark 5
    -- Executing [91000@DLPN_DialPlan1:1] Macro("SIP/6080-00000006", "trunkdial-failover-0.3,SIP/trunk_1/1000,,trunk_1,") in new stack
    -- Executing [s@macro-trunkdial-failover-0.3:1] GotoIf("SIP/6080-00000006", "0?1-fmsetcid,1") in new stack
    -- Executing [s@macro-trunkdial-failover-0.3:2] GotoIf("SIP/6080-00000006", "0?1-setgbobname,1") in new stack
    -- Executing [s@macro-trunkdial-failover-0.3:3] Set("SIP/6080-00000006", "CALLERID(num)=") in new stack
    -- Executing [s@macro-trunkdial-failover-0.3:4] Set("SIP/6080-00000006", "CALLERID(all)=") in new stack
    -- Executing [s@macro-trunkdial-failover-0.3:5] GotoIf("SIP/6080-00000006", "0?1-dial,1") in new stack
    -- Executing [s@macro-trunkdial-failover-0.3:6] Set("SIP/6080-00000006", "CALLERID(all)=") in new stack
    -- Executing [s@macro-trunkdial-failover-0.3:7] Set("SIP/6080-00000006", "CALLERID(all)=") in new stack
    -- Executing [s@macro-trunkdial-failover-0.3:8] Goto("SIP/6080-00000006", "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/6080-00000006", "SIP/trunk_1/1000") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/trunk_1/1000
[May 26 03:11:26] WARNING[2411][C-00000018]: chan_sip.c:23037 handle_response_invite: Received response: "Forbidden" from '"asterisk" <sip:asterisk@x.x.x.x:5060>;tag=as655e53b4'
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [1-dial@macro-trunkdial-failover-0.3:2] GotoIf("SIP/6080-00000006", "0 > 0 ?1-CHANUNAVAIL,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/6080-00000006", "") in new stack
  == Spawn extension (macro-trunkdial-failover-0.3, 1-out, 1) exited non-zero on 'SIP/6080-00000006' in macro 'trunkdial-failover-0.3'
  == Spawn extension (DLPN_DialPlan1, 91000, 1) exited non-zero on 'SIP/6080-00000006'

另外,好像幾天前comnet做了系統升級,不知道是否和此有關?
有沒有人的comnet phone有影響?
321

本帖最後由 321 於 2015-5-26 11:51 編輯

看了其它貼子,很多人也是這幾天不能用asterisk with comnet phone,但大多數都是能接不能打出,我連打入也不能,出現忙音.
以下是打入時的log
[May 26 03:23:25] NOTICE[2423][C-0000000b]: chan_sip.c:25662 handle_request_invite: Call from '852xxxxxxx' (202.0.179.3:5060) to extension 's' rejected because extension not found in context 'DID_trunk_1'.
321

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補充一下,本人是用openwrt 的asterisk server
321

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應該是coment改了incoming call的SIP INVITE format. 原來SIP INVITE header內有各位的電話號碼,但而家變 ...
tpenguin 發表於 2015-5-26 12:56


謝謝你的信息,我回去試試.

但是大家是否仍未解決打出問題?
本人comnet 到6月30日會約滿,正打算取消掉comnet用28元的2B電話,不知道這個28元2B是否可以放在asterisk上使用?
321

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回復  321


   可以放在Freepbx中使用的
bigtoodog 發表於 2015-5-27 13:04

那麼如何解決的的asterisk不能打入的問題?
321

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回復  bigtoodog


    今日下午整番好,之前係有問題的
99BB 發表於 2015-5-27 17:41


而家asterisk上恢恢正常??
我仍未能打出打入
321

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openwrt上的asterisk11似乎和一般的asterisk有不同, config file的定義也不一樣.
321

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把fromdomain=huawei.com
改成 fromdomain=202.0.179.3 後,能成功打出電話, 但仍然無法打入電話.
試上樓上的freepbx的做法,仍無法打入電話
321

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應該是coment改了incoming call的SIP INVITE format. 原來SIP INVITE header內有各位的電話號碼,但而家變 ...
tpenguin 發表於 2015-5-26 12:56


問題果然是正如你所說的,現在問題已經解決,可以正常打出打入.

請問一下 exten => s,1,NoOP(Called from Comnet) 這句是有什麼用?
321

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