search for: function_pjsip_send_session_refresh

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2019 May 24
2
Is there a way to make asterisk send a INVITE in-dialog to re-establish the audio
We are working with an Avaya switch. We send them a REFER. If the transfer is successful, everything is great. If it fails (busy), they send an INVITE in-dialog with a media attribute of inactive. After that, they send a 486 busy. The problem is Avaya basically put the call on hold so audio is not active. The Avaya rep is indicating we need to send in dialog invite to get the call audio back?