Hii, We have Issue about calls hangup 40s -50s using asterisk 13.16.0. I dont know wheres something wrong about it. call hangup to extension random. Thanks
There is insufficient information. You almost certainly need to provide at least verbose logs, and probably more, but you haen’t even told us if this is circuit switched, and if so analogue or digital, or VoIP, and in that case, SIP (in which case whether chan_sip or chan_pjsip), IAX, H.323, Skinny, etc.