Обратите внимание, что directrtpsetup не работает с nat, и если sdp предлагает DIFFERENT для обоих одноранговых узлов.
требуют также как DirectMedia и директивы directrtpsetup и звездочку версии выше 11. *
;directmedia=yes ; Asterisk by default tries to redirect the
; RTP media stream to go directly from
; the caller to the callee. Some devices do not
; support this (especially if one of them is behind a NAT).
; The default setting is YES. If you have all clients
; behind a NAT, or for some other reason want Asterisk to
; stay in the audio path, you may want to turn this off.
; This setting also affect direct RTP
; at call setup (a new feature in 1.4 - setting up the
; call directly between the endpoints instead of sending
; a re-INVITE).
; Additionally this option does not disable all reINVITE operations.
; It only controls Asterisk generating reINVITEs for the specific
; purpose of setting up a direct media path. If a reINVITE is
; needed to switch a media stream to inactive (when placed on
; hold) or to T.38, it will still be done, regardless of this
; setting. Note that direct T.38 is not supported.
;directmedia=nonat ; An additional option is to allow media path redirection
; (reinvite) but only when the peer where the media is being
; sent is known to not be behind a NAT (as the RTP core can
; determine it based on the apparent IP address the media
; arrives from).
;directmedia=update ; Yet a third option... use UPDATE for media path redirection,
; instead of INVITE. This can be combined with 'nonat', as
; 'directmedia=update,nonat'. It implies 'yes'.
;directmedia=outgoing ; When sending directmedia reinvites, do not send an immediate
; reinvite on an incoming call leg. This option is useful when
; peered with another SIP user agent that is known to send
; immediate direct media reinvites upon call establishment. Setting
; the option in this situation helps to prevent potential glares.
; Setting this option implies 'yes'.
.
;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
; the call directly with media peer-2-peer without re-invites.
; Will not work for video and cases where the callee sends
; RTP payloads and fmtp headers in the 200 OK that does not match the
; callers INVITE. This will also fail if directmedia is enabled when
; the device is actually behind NAT.
Пожалуйста, напишите sip.conf [general], sip configs для двух конечных точек и раздел абонентской группы, на котором вы набираете номер. – MichelV69