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回復 114# 雯雯


   

我給你看看我全部 sip.conf 的整體設定,以方便你做對比,可能當中你我之間的分別,就是導致你打不出的原因。 你在你的 File Editor 之下便會找到 sip.conf 內容。

請查看 PM

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本帖最後由 bubblestar 於 2012-9-30 18:33 編輯

回復 114# 雯雯


   
請確定你在user.conf 裡的cmphone 設定中,qualify 一項要設定成 No (My post in #106 above).  因為qualify default = yes = 2 秒的話,可能在偏遠的地方或NAT 因不同的回應速度太慢下回應binding request,Asterisk 便會無法打出電話的。解決辦法一係加大qualify 參數,例如3000-5000 (3-5秒),一係就qualify = No,索性不做binding request,我選擇後者,一直沒有問題。當然這是要因Provider 而定的。cmphone 便要這樣做了。
你在大陸這麼遠,更可能需要做此設定。

詳細解說:  http://www.voip-info.org/wiki/view/Asterisk+sip+qualify

This feature may also be used to keep a UDP session open to a device that is located behind a network address translator (NAT). By sending the OPTIONS request, the UDP port binding in the NAT (on the outside address of the NAT/firewall device) is maintained by sending traffic through it. If the binding were to expire, there would be no way for Asterisk to initiate a call to the SIP device. This can be used in conjunction with the nat=yes setting.

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