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To use ATA as a VOIP gateway?

本帖最後由 Qnewbie 於 2010-8-10 20:00 編輯

We all know that SPA can be used as PSTN gateway for Asterisk.

Situation:
Some VSPs allow only one account at one IP(say ET2x3) . Hence, if you want to use two accounts simultaneously, you have to register another account at another place.

Question:
If we do not use the PSTN line(simply not connected), could SPA VOIP2 used as voip gateway for Asterisk? If yes, how to config?

Assume asterisk(A) is located at A, SPA(S) is linked at location B and SPA(S) should register to VSP provider C.

This issue has been discussed in the HKEPC forum. Using a telephine line connects FXS and FXO of SPA3000, then you may have VoIP Gateway between to two different VoIP accounts.

Using the following logical thinking:

FXS port: Current/Voltage sourcing port
FXO port: Current/Voltage sinking port

Examples of FXS port or device
1) Fixed telephone line from telephone company
2) Extension line from PBX system
3) FXS port of SPA3000

Examples of FXO port or device
1) Ordinary telephone set
2) CO port (connecting fixed line from telephone company) of PBX system
3) FXO port of SPA3000

Leitany had reported the setup procedures for this issue in HKEPC. If you have time, pleaea take a look at the VoIP thread in HKEPC.


YH

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回復 2# 角色


It is not exactly the solution I prefer. Direct connection between FXO and FXS port just connects VOIP2 to VOIP1. However, the FXS port connects to a phone and is served as an extension to Asterisk.

I think one possible solution is the Gateway function.

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Theorectically yes, I think.  

Instead of using VoIP2 in PSTN Tab, you may turn the direction to VoIP1 in Line 1 Tab in where we have another 4 gateway accounts from Gateway1 to Gateway4.

I haven't tried to test but I think the logic is same.

I am successfully using the similar method to initiate PSTN call thru remote VoIP2 of SPA3102 via HTTP Digest.  The trick is to register a new trunk in Asterisk Server as usual with your desired ET2x3 in one of the 4 Gateway accounts.

In my case, the trunk is not really registered but it can still be connected. YHFung also successfull makes use of this HTTP Digest method to use other remote PSTN land line.

Of course, in your case, I understand that you just want to use VoIP Gateway instead of PSTN Gateway.  So that why I ask you to try the 4 extra VoIP Gateways in Line 1 instead of PSTN Line.

You may refer to tamtsin C-hing's manual on page 6 of his Admin Guide to set up the Gateway first and then establish a trunk in your Asterisk Server to achieve it.  Hope you can succeed and please share if you can manage to do it.

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Use HTTP digest is the best.

現在我用的方法是

經asterisk , 用 HTTP digest,而且可以因應唔同 users, 設定唔同 dial plan through different authentization ,還可以設定唔同 outgoing dialing rule

而 PSTN dial into the SPA3000, 我就用 Pin

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非常同意。

以前用一般方法設定PSTN-To-VoIP 或 VoIP-To-PSTN 時,經常接不到所要的線路,特別是加了PIN之後,問題好大機會是出自DTMF的辨識成功率非常低。但用了HTTP Digest 的方法之後,接入接出都很好。

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For the use of HTTP digest:
It is under the PSTN tag, VoIP Users and Passwords (HTTP Authentication). It is not under the Line1, Gateway Accounts.

It sounds a bit confusing

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Sorry for the confusion.  

I've just taken HTTP Digest as an example to explain how we can establish a trunk in Asterisk to use the PSTN landline in remote ATA without using the regular subscriber account in PSTN Tab.

I think you can do similar settings by registering ET263 in one of the 4 Gateway Accounts in Line1 first.  Then, establish a trunk as usual in Asterisk to dial out via these gateway accounts in the remote end to reach the target party.

The point is that we don't use the Regular Subcriber Accounts in Line1 to initiate calls.  We use the extra 4 GATEWAY Accounts.  This is the unique function on SPA3000 or SPA3102.

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請問各位C-hing,有冇教學文件可以下載呢?
因為set極都set唔到

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