Whenever and wherever someone call your sip account, your HK mobile phone (91234567) will ring. So, your sip account is ported into your mobile phone. You dont need to tell your mobile number to your friends but your friends can still find you via your sip account.
If you want to call any HK phone numbers instead of a fixed number, replace the above line to: Voice Services > SP1 Service
X_InboundCallroute: {LI1(xxxxxxxx)}
Once someone dial your sip account, he will hear the second dial tone. At this time, he may enter a HK phone number, e.g. 91234567, 21234567 etc.
This is VOIP to PSTN without using AA.作者: bubblestar 時間: 2011-3-17 16:56
本帖最後由 bubblestar 於 2011-3-17 17:42 編輯
Thanks! uncle7. This is another good way to implement a flexible call forwading and VoIP2PSTN usage.
For security reason, we'd better add some measure to avoid 3rd party from dialing-in using our PSTN for making further paid calls, such as IDD.作者: uncle7 時間: 2011-3-17 16:57
备用帖子xxx。作者: uncle7 時間: 2011-3-17 16:57
备用帖子xxx。作者: uncle7 時間: 2011-3-17 16:58
备用帖子xxx。作者: uncle7 時間: 2011-3-17 16:58
备用帖子xxx。作者: uncle7 時間: 2011-3-17 17:01
本帖最後由 uncle7 於 2011-3-17 17:33 編輯
Thanks! uncle7. This is another good way to implement a flexible call forwading and VoIP2PSTN usage ...
bubblestar 發表於 2011-3-17 16:56
This is just the simplest setting to make VOIP to PTSN work. Of course, we should do some fine tuning to filter off unwanted PSTN numbers and/or add a password.作者: 亞星 時間: 2011-3-17 17:07
uncle7 帥兄 Thanks for sharing 作者: bubblestar 時間: 2011-3-17 17:09
Totally agree with uncle7's idea. More exchange of different ideas, scenarios, sharing and interaction will definitely benefit all VoIP lovers here.作者: 亞星 時間: 2011-3-17 22:20
試左好多次出現一個問題, 有時 VOIP 轉左去 PSTN 聽到 dial tone 但打唔出電話, 拔電 reset 過都係唔得, 你地有無試過咁既情況 作者: bubblestar 時間: 2011-3-17 22:24