<div>Dear All,</div>
<div>I have the below issue:</div>
<div> </div>
<div>I created an extension(5678) under extensions_custom.conf to record voice messages and playback the voice as you can see below:</div>
<div>[custom-recordme]<br><br>exten => 5678,1,Wait(2)<br>exten => 5678,2,Record(/tmp/asterisk-recording:g729)<br>exten => 5678,3,Wait(2)<br>exten => 5678,4,Playback(/tmp/asterisk-recording)<br>exten => 5678,5,Wait(2)<br>
exten => 5678,6,Hangup</div>
<div> </div>
<div>When dialing this extension from another extension registered on the same asterisk server everything works fine...The issue begins if I try to make a call from an OpenSer server....The SIP authentication did not work...</div>
<div> </div>
<div>Can you please give me and step by step the configuration that i should do in order to accomplish this task?</div>
<div> </div>
<div>Regards</div>