Displaying 2 results from an estimated 2 matches for "ast_sip_session_media".
2016 Nov 22
2
Regression in 13.13.0-RC1
...es_rtp_asterisk.c:2665 int
ast_rtp_new(struct ast_rtp_instance *, struct ast_sched_context *,
struct ast_sockaddr *, void *): Failed to create a new socket for RTP
instance '0x805647c30'
[Nov 22 10:49:26] ERROR[101105]: res_pjsip_sdp_rtp.c:184 int
create_rtp(struct ast_sip_session *, struct ast_sip_session_media *):
Unable to create RTP instance using RTP engine 'asterisk'
Please note that with asterisk 13.12.2 everything works fine. No other
change was made to the system.
Is this a known issue being worked on? Should I file a bug report?
I am the maintainer of the FreeBSD Asterisk port in the p...
2015 Jan 17
1
Fwd: Asterisk pjsip auto dtmf mode
...ip_sdp_rtp.c (I added session_media to get_codecs and used it
in order to update dtmf settings on rtp instance when telephone-event is
not included in the sdp):
150:
static void get_codecs(struct ast_sip_session *session, const struct
pjmedia_sdp_media *stream, struct ast_rtp_codecs *codecs, struct
ast_sip_session_media *session_media)
159:
char fmt_param[256];
int tel_event = 0;
177:
ast_copy_pj_str(name, &rtpmap->enc_name, sizeof(name));
if (strcmp(name,"telephone-event") == 0) {
tel_event++;
}
202:
}...