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handling Incoming PSTN call with AsteriskTechnical support, how-to guides, troubleshooting, and general assistance, from beginner to seasoned pro, this is where to discuss Asterisk, the most powerful open source PBX. |
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| Hi, hope you guys can help me with this. I am planning to setup an asterisk server which will look like this: PSTNlandline --> AsteriskServer -->internet We only have 1 pstn line so do i need an rj11 splitter to split the landline into 2 lines? One for the ASTERISK SERVER and one for our ANALOG PHONE. my question is about the Asterisk way of handling Ordinary Inbound PSTN calls. which of these is possible? 1.) PSTN caller will ring the ANALOG PHONE. if nobody picks up the phone in a configurable no. of seconds, ASTERISK's FXO port will pickup and will ask for a PIN 2.) PSTN caller will be given a configurable amount of time (maybe 2secs) to press a PIN to create a VOIP call. if no PIN is entered, the ANALOG phone will ring and it will be an Ordinary Inbound PSTN call or do you know anyother way for asterisk to handle PSTN calls that will be transparent to the PSTN Caller. thanks |
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| The best way to integrate your PSTN Line into your Asterisk server is with a device like the Sipura SPA3000. You should configure the SPA-3000 as two extensions of your Asterisk by using the Voxilla Asterisk/SPA-3000 configuration wizard. You then need to configure your extensions.conf file to include the behaviors you described for receiving a phone call. For example, upon detecting an incoming call on the PSTN Line of the SPA Asterisk would ring the extension associated with the SPA's Line 1 for a number of rings, and then pass the call to the Asterisk's IVR function. The IVR function would then play an announcement like "Please press the extension you wish or push 0 for an operator." If the caller presses 0 he will be transferred to ring the phone once again. If, however, he knows the "extension" that points to the DISA function, he will receive a Dial Tone for entering a DISA PIN and will then be permitted appropriate access to the functions of the Asterisk. All of these functions are described in detail at www.asterisk.org. An important benefit of using a Sipura SPA3000 for this function is that if the Asterisk fails or your power fails the SPA will manually connect your analog phone to your PSTN Line as a safety backup.
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