Archives : August-2017
Let me provide the details first:* Asterisk 1.8.32 on CentOS behind the NAT firewall* Two (2) SIP trunks with canreinvite=no and directmedia=noIf a call comes from either trunk and is bridged to a local extension there is never a problem with aud..
folks.I have a couple of questions regarding RTP.The background of my inquiry is that I have packet captures of SIP and RTP traffic on an Asterisk and Broadworks SIP trunk and the RTP many times has a time stamp that rewinds by 480 using g.711u. ..
since al long time I have used UNIQUEID for identify calls in my dialplan, statistics…Now I have had an problem, after I have checked log file I found out following:calls same time ( hours:seconds) came in.CallID, DID, channel name (00003cf9 to 00003c..