Asterisk 16.0, PJSIPFor the first caller to a conference, I want to dial out and bridge that conference to a new PJSIP external call.For the next callers, I just want them to join the local Asterisk conference.After the last caller leaves the conferen..

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I am currently evaluating asterisk 16. I have noticed an issue using application playback. The beginning and the end of the audio file are missing. If I use answer and wait(1) before playback, the beginning is correct. I am using chan_sip, if this..

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