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Asterisk won't does as extension_custom.conf says...


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Programming language suggestion for web development...


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Freepbx settings storage...


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how to write application to freepbx...


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Asterisk & freePBX...


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Playback file in Call Waiting on Asterisk...


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Asterisk and AMI Originate extension is not busy...


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Astarisk Call File_Can wait or get response to start playback?...


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Cannot make outbound & inbound calls using asterisk 13 (FreePBX) and Twilio...


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Transfer call to custom extension to pause recording and dial externally...


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incoming call to the server from WAN drop after some sconds...


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Asterisk ringgroup get call after registration...


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channel originate, how to do call from a local channel? (call intercom and send dtmf)...


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Cannot connect to voip server from client device?...


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Vtiger +FreePBX : Call Status is not updated after call hang-up...


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Freepbx: Add Extension Programmatically...


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FreePBX call recording - increase volume on one stream...


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Asterisk / Freepbx does not set CallerID to calling party when queue contains a cell phone...


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Cant install PIAF, issue with 'stopping ntp services'...


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FreePBX add a new SIP extension...


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freepbx and java server interation...


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FreePBX call macros before any dial...


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How to make a trunk in Asterisk 13 FreePBX to call to other SIP networks, a voip peering?...


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FreePBX showing only Welcome Page. Internal Pages are not working...


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Getting "Mute State" in ARI...


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Send SMS from PHP script with FreePBX Bootstrap...


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