Im trying to figure out how to commit some code for its review, but when applying Git appear some conflicts I dont know how..
Cant find a way to control the dtmf mode on a per session basis with pjsip, used to use SIPDtmfMode from the dialplan with chan_sip. Any hints on how t..
While trying to use direct_media Im seeing RTP payload mismatches after succesful reinvites.Initial INVITE from endpoint A to asterisk has rfc4733 DMTFm=audio 35648 RTP/AVP 9 8 111 96a=rtpmap:96 telephone-event/8000a=fmtp:96 0-16From asterisk to upstr..
With pjsip (asterisk 13.14.1) I see the problem that an anonymous from header gets user=phone appendend to the URI if user_eq_phone=yes is specified:On the incoming leg:From: anonymous ;tag=Q5zBj7BMnvI6Fe6O2866fox3ZHmn-smt Get transformed to From: Anonym..
Im trying to implement direct_media between multiple peers and an uplink provider, all of whom have direct_media=yes configures.For originating calls to the uplink provider direct_media=yes works like expected. SIP flows through asterisk, rtp doesntS..