search for: ast_sip_sess

...:49:26] WARNING[101105]: res_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...

...additional dtmf mode ... I based my development on 13.1.0. The following are my changes: In res/res_pjsip_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 (strcm...

...://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.2.1 resolves an issue reported by the community and would have not been possible without your participation. Thank you! The following is the issue resolved in this release: * --- pjsip: resolve compatibility problem with ast_sip_session (Closes issue ASTERISK-24941. Reported by Matt Jordan) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.2.1 Thank you for your continued support of Asterisk!

...://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.3.1 resolves an issue reported by the community and would have not been possible without your participation. Thank you! The following is the issue resolved in this release: * --- pjsip: resolve compatibility problem with ast_sip_session (Closes issue ASTERISK-24941. Reported by Matt Jordan) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.3.1 Thank you for your continued support of Asterisk!

...://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.2.1 resolves an issue reported by the community and would have not been possible without your participation. Thank you! The following is the issue resolved in this release: * --- pjsip: resolve compatibility problem with ast_sip_session (Closes issue ASTERISK-24941. Reported by Matt Jordan) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.2.1 Thank you for your continued support of Asterisk!

...://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.3.1 resolves an issue reported by the community and would have not been possible without your participation. Thank you! The following is the issue resolved in this release: * --- pjsip: resolve compatibility problem with ast_sip_session (Closes issue ASTERISK-24941. Reported by Matt Jordan) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.3.1 Thank you for your continued support of Asterisk!

...session_inv_on_rx_offer (inv=0x7fba3c0504e8, offer=0x7fba3c0adfb8) at res_pjsip_session.c:3009 3009 if (handle_incoming_sdp(session, offer)) { (gdb) list 3004 static void session_inv_on_rx_offer(pjsip_inv_session *inv, const pjmedia_sdp_session *offer) 3005 { 3006 struct ast_sip_session *session = inv->mod_data[session_module.id]; 3007 pjmedia_sdp_session *answer; 3008 3009 if (handle_incoming_sdp(session, offer)) { 3010 return; 3011 } 3012 3013 if ((answer = create_local_sdp(inv, session, offer))) {