Displaying 2 results from an estimated 2 matches for "ast_copy_pj_str".
2017 Jun 18
2
asterisk 13.16. - sigseg during negotiation
...kom doesn't support it (port and ip
addresses are set to 0).
On completing the negotiation after 200 ok SDP and ACK from fax client,
asterisk crashes. Stack trace is attached!
Regards,
Michael
-------------- next part --------------
Program terminated with signal 11, Segmentation fault.
#0 ast_copy_pj_str (dest=0x7fb9f5901100 "x\277\001<h\025\220", <incomplete sequence \365>, src=0x20, size=1025) at res_pjsip.c:4147
#1 0x00007fb9f0b02334 in negotiate_incoming_sdp_stream (session=0x7fba3c031200, session_media=<value optimized out>, sdp=<value optimized out>, stream=<...
2015 Jan 17
1
Fwd: Asterisk pjsip auto dtmf mode
...phone-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:
}
if (tel_event==0) {
ast_rtp_instance_dtmf_mode_set(session_media->rtp,
AST_RTP_DTMF_MO...