Dear All, I have the below issue: I created an extension(5678) under extensions_custom.conf to record voice messages and playback the voice as you can see below: [custom-recordme]
exten => 5678,1,Wait(2) exten => 5678,2,Record(/tmp/asterisk-recording:g729) exten => 5678,3,Wait(2) exten => 5678,4,Playback(/tmp/asterisk-recording) exten => 5678,5,Wait(2) exten => 5678,6,Hangup 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... Can you please give me and step by step the configuration that i should do in order to accomplish this task? Regards
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