search for: res_pjsip_sdp_rtp

...tbound_registration.so res_pjsip_path.so res_pjsip_pidf_body_generator.so res_pjsip_pidf_digium_body_supplement.so res_pjsip_pidf_eyebeam_body_supplement.so res_pjsip_publish_asterisk.so res_pjsip_pubsub.so res_pjsip_refer.so res_pjsip_registrar_expire.so res_pjsip_registrar.so res_pjsip_rfc3326.so res_pjsip_sdp_rtp.so res_pjsip_send_to_voicemail.so res_pjsip_session.so res_pjsip_sips_contact.so res_pjsip.so res_pjsip_t38.so res_pjsip_transport_management.so res_pjsip_transport_websocket.so res_pjsip_xpidf_body_generator.so Asterisk CLI : [Aug 14 12:02:32] WARNING[20712]: loader.c:599 load_dynamic_module:...

...RTP socket: Protocol not supported [Nov 22 10:49:26] 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 f...

...e this problem. Asterisk 13.7 (chan_pjsip) & Zyxel Keenetic Plus DECT, firmware v2.06(AAGJ.9)C1 Outbound call from Zyxel Keenetic (pjsip endpoint) to PSTN (pjsip trunk). Call using early media (183 Session in progress) and rtp_timeout=10. After 10 seconds: [2016-12-16 13:53:15] NOTICE[6654] res_pjsip_sdp_rtp.c: Disconnecting channel 'PJSIP/xxx-0000027b' for lack of RTP activity in 10 seconds SIP dump is attached. According to [1] before called user agent send OK or ACK there is one way SDP. In sip dump (attached) i didn't find such SIP packets. Whether that call was canceled due to RTP...

...d appreciate if someone would look at what I did and see if I didn't do stupid things. If you think this is something you would like to add to one of the next releases I am willing to help - add the 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_se...

...Daniele Pallastrelli) * ASTERISK-26311 - [patch] rtp_engine: Allow more than 32 dynamic payload types. (Reported by Alexander Traud) * ASTERISK-26549 - app_dial: When PickupChan() is used some channels may have incorrect device state (Reported by Joshua Colp) * ASTERISK-26541 - res_pjsip_sdp_rtp: Restrict number of formats to maximum (Reported by Joshua Colp) * ASTERISK-25070 - Fix FTBFS on Hurd (Reported by Gabriele Giacone) * ASTERISK-26476 - chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings" (Reported by Sergey...

...nd_publish.c (Reported by Matt Krokosz) * ASTERISK-25070 - Fix FTBFS on Hurd (Reported by Gabriele Giacone) * ASTERISK-26476 - chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings" (Reported by Sergey Grachev) * ASTERISK-26541 - res_pjsip_sdp_rtp: Restrict number of formats to maximum (Reported by Joshua Colp) * ASTERISK-26537 - AMI: NewConnectedLine event is not documented (Reported by Etienne Lessard) * ASTERISK-26526 - [UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy (Reported by Badalian Vyachesla...

...(Reported by Robin Leffmann) * ASTERISK-28664 - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by Pascal Cadotte Michaud) * ASTERISK-28604 - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0 (Reported by George Joseph) * ASTERISK-28659 - res_pjsip_sdp_rtp: Bundle includes non-existent media stream if codecs create additional streams and offer does not have them (Reported by nappsoft) * ASTERISK-28660 - res_fax: wrap Asterisk initiated negotiation with config option (Reported by Kevin Harwell) * ASTERISK-28636 - app_ch...

...(Reported by Robin Leffmann) * ASTERISK-28664 - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by Pascal Cadotte Michaud) * ASTERISK-28604 - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0 (Reported by George Joseph) * ASTERISK-28659 - res_pjsip_sdp_rtp: Bundle includes non-existent media stream if codecs create additional streams and offer does not have them (Reported by nappsoft) * ASTERISK-28660 - res_fax: wrap Asterisk initiated negotiation with config option (Reported by Kevin Harwell) * ASTERISK-28636 - app_ch...

...(Reported by Jean Aunis - Prescom) * ASTERISK-29097 - res_pjsip_config_wizard: Crash when freeing string when failing to add extension (Reported by Vieri) * ASTERISK-26424 - app_voicemail: Undocumented behavior from VMSayName (Reported by Eric Smith) * ASTERISK-29051 - res_pjsip_sdp_rtp: Does not set correct values on RTP instance when "auto" DTMF is used (Reported by Sebastian Damm) * ASTERISK-28311 - dsp: ast_dsp_silence_noise_with_energy wrong judgment of frame format (Reported by ���������) * ASTERISK-24329 - Music On Hold announcement...

...y ASTERISK-28317 (Reported by abelbeck) * ASTERISK-26006 - Show offending IP for TLS setup failures in logs (Reported by Oleksandr Natalenko) * ASTERISK-28444 - chan_pjsip: Peer IP for SSL handshake errors not logged (Reported by Bernhard Schmidt) * ASTERISK-28460 - res_pjsip_sdp_rtp: Fix ICE candidate leak with specific usage (Reported by Joshua C. Colp) * ASTERISK-28018 - IP Fragmentation happening instead of DTLS fragmentation on handshake server hello certificate (Reported by vijay kumar) * ASTERISK-25371 - Crash in hangup at chan_pjsip.c:174...

...n cause a segfault in a T.38 reINVITE (Reported by Francesco Castellano) Bugs fixed in this release: ----------------------------------- * ASTERISK-28457 - [patch] Fix crash in chan_dahdi on 32-bit systems caused by ASTERISK-28317 (Reported by abelbeck) * ASTERISK-28458 - res_pjsip_sdp_rtp: Remove unused variable (Reported by Michael Maier) * ASTERISK-26006 - Show offending IP for TLS setup failures in logs (Reported by Oleksandr Natalenko) * ASTERISK-28444 - chan_pjsip: Peer IP for SSL handshake errors not logged (Reported by Bernhard Schmidt) * A...

...ed behavior from VMSayName (Reported by Eric Smith) * ASTERISK-29091 - Crash when ast_translator_build_path fails (Reported by Jasper van der Neut) * ASTERISK-29099 - res_musiconhold: Realtime MOH only loads a single entry (Reported by laszlovl) * ASTERISK-29051 - res_pjsip_sdp_rtp: Does not set correct values on RTP instance when "auto" DTMF is used (Reported by Sebastian Damm) * ASTERISK-28311 - dsp: ast_dsp_silence_noise_with_energy wrong judgment of frame format (Reported by ���������) * ASTERISK-29085 - func_curl: Segmentation fa...

...jsip_outbound_authenticator_digest.so noload => res_pjsip_outbound_registration.so load => res_pjsip_pidf.so load => res_pjsip_pubsub.so noload => res_pjsip_refer.so noload => res_pjsip_registrar_expire.so noload => res_pjsip_registrar.so load => res_pjsip_rfc3326.so load => res_pjsip_sdp_rtp.so load => res_pjsip_session.so noload => res_pjsip_t38.so noload => res_pjsip_transport_websocket.so load => res_pjsip_acl.so I need to identify the caller, and since there is no way to see the IP address of the caller in the dial plan, at least I need to force Asterisk to match it to...

...c Plus DECT, firmware >> v2.06(AAGJ.9)C1 >> >> Outbound call from Zyxel Keenetic (pjsip endpoint) to PSTN (pjsip >> trunk). >> Call using early media (183 Session in progress) and rtp_timeout=10. >> After 10 seconds: [2016-12-16 13:53:15] NOTICE[6654] >> res_pjsip_sdp_rtp.c: Disconnecting channel 'PJSIP/xxx-0000027b' for >> lack of RTP activity in 10 seconds >> >> SIP dump is attached. >> >> According to [1] before called user agent send OK or ACK there is one >> way SDP. >> In sip dump (attached) i didn't find...

...ing when failing to add extension (Reported by Vieri) * ASTERISK-26424 - app_voicemail: Undocumented behavior from VMSayName (Reported by Eric Smith) * ASTERISK-29099 - res_musiconhold: Realtime MOH only loads a single entry (Reported by laszlovl) * ASTERISK-29051 - res_pjsip_sdp_rtp: Does not set correct values on RTP instance when "auto" DTMF is used (Reported by Sebastian Damm) * ASTERISK-29091 - Crash when ast_translator_build_path fails (Reported by Jasper van der Neut) * ASTERISK-28311 - dsp: ast_dsp_silence_noise_with_energy wrong...

...VMSayName (Reported by Eric Smith) * ASTERISK-29097 - res_pjsip_config_wizard: Crash when freeing string when failing to add extension (Reported by Vieri) * ASTERISK-29091 - Crash when ast_translator_build_path fails (Reported by Jasper van der Neut) * ASTERISK-29051 - res_pjsip_sdp_rtp: Does not set correct values on RTP instance when "auto" DTMF is used (Reported by Sebastian Damm) * ASTERISK-29099 - res_musiconhold: Realtime MOH only loads a single entry (Reported by laszlovl) * ASTERISK-28311 - dsp: ast_dsp_silence_noise_with_energy wr...

...RTCP packet sending may be incorrect (Reported by Joshua C. Colp) Improvements made in this release: ----------------------------------- * ASTERISK-28401 - app_confbridge: Add *_all remb behavior variants (Reported by Joshua C. Colp) * ASTERISK-28400 - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for transport-cc (Reported by Joshua C. Colp) * ASTERISK-28363 - Millisecond-resolution call stats including PDD in channel variables (Reported by Antoni Goldstein) * ASTERISK-20207 - Asterisk should clear out any .lock files in the voice mail directory...

...preventing early media playback (Reported by Matt Jordan) * ASTERISK-24178 - [patch]fromdomainport used even if not set (Reported by Elazar Broad) * ASTERISK-22252 - res_musiconhold cleanup - REF_DEBUG reload warnings and ref leaks (Reported by Walter Doekes) * ASTERISK-23994 - res_pjsip_sdp_rtp: owner address in SDP may not be fully qualified domainname (Reported by Private Name) * ASTERISK-24147 - ARI: channel hangup crashes asterisk process (Reported by Edvin Vidmar) * ASTERISK-23997 - chan_sip: port incorrectly incremented for RTCP ICE candidates in SDP answer (Repo...

...preventing early media playback (Reported by Matt Jordan) * ASTERISK-24178 - [patch]fromdomainport used even if not set (Reported by Elazar Broad) * ASTERISK-22252 - res_musiconhold cleanup - REF_DEBUG reload warnings and ref leaks (Reported by Walter Doekes) * ASTERISK-23994 - res_pjsip_sdp_rtp: owner address in SDP may not be fully qualified domainname (Reported by Private Name) * ASTERISK-24147 - ARI: channel hangup crashes asterisk process (Reported by Edvin Vidmar) * ASTERISK-23997 - chan_sip: port incorrectly incremented for RTCP ICE candidates in SDP answer (Repo...

...orted by Joshua C. Colp) * ASTERISK-30367 - pbx: Fix outdated channel snapshots with pbx_exec (Reported by N A) * ASTERISK-28767 - chan_pjsip: Caller ID not used when checking for extension, callerid supplement executed too late (Reported by Oleg) * ASTERISK-30350 - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when moh_passthrough has call on hold (Reported by Benjamin Keith Ford) * ASTERISK-30240 - app voicemail odbc build error with gcc 11.1 (Reported by Michael Bradeen) * ASTERISK-30100 - res_pjsip: Path is ignored on INVITE to endpoi...