my sip.conf [kamailio_ext1] type=friend host = my.superprovider.com port = 5068 canreinvite = no insecure = invite,port transport=udp trustrpid=yes context = incoming videosupport=no directmedia=no dtlsenable = no tlsenable=no disallow=all allow=alaw allow=opus allow=ulaw
connection goes great. SIP session have all packets So astersik send INVITE -> to trunk then <- TRYING <- RINGING <- OK ->ACK and at the end some of callers send BYE And then Goes OK and ACK full SIP session. Signaling is Ok. At CDR I see ANSWERED Wheb Call Unanswered I see INVITE -> to trunk then <- TRYING <- RINGING some of callers CANCEL OK ACK So There is full session too At CDR I see No Answered session (that is write) And after that I see empty record file 2014-09-22 18:40 GMT+04:00 A J Stiles <asterisk_l...@earthshod.co.uk>: > **** THIS IS NOT WHERE YOUR REPLY BELONGS **** > > On Monday 22 Sep 2014, Yuriy Gorlichenko wrote: > > 2014-09-22 12:12 GMT+04:00 A J Stiles <asterisk_l...@earthshod.co.uk>: > > > On Monday 22 Sep 2014, Yuriy Gorlichenko wrote: > > > > Hello I have an issue wit MixMonitor. I need to record only answered > > > > calls, > > > > so I set "b" option for this but calls still recording even call no > > > > answered My asterisk version 12.5.1, at my other servers with older > > > > versions of asterisk (11.8 for example) MixMonitor works fine. > > > > > > What technology are you using for your outgoing calls? SIP trunk, IAX > > > trunk, > > > ISDN, mobile or analogue phone lines? > > > > SIP trunks > > Well, SIP certainly allows for full supervisory information (analogue > doesn't, and all calls are deemed answered if the exchange line was > available). What have you got in your sip.conf ? And what does your SIP > trunk provider have to say on the matter? (It wouldn't be totally unknown > for > a dodgy telco to provide not-entirely-truthful supe.) > > -- > AJS > > Note: Originating address only accepts e-mail from list! If replying off- > list, change address to asterisk1list at earthshod dot co dot uk . > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
