8580 Commits (fc1d93cf97e6d059584b1feb5d075d22ead31669)

Author	SHA1	Message	Date
Walter Doekes	34ce90a9e7	chan_sip: Return 503 if we're out of RTP ports ... If you're for some reason out of RTP ports, chan_sip would previously responde to an INVITE with a 403, which will fail the call. Now, it returns a 503, allowing the device/proxy to retry the call on a different machine. ASTERISK-28718 Change-Id: I968dcf6c1e30ecddcce397dcda36db727c83ca90	6 years ago
Friendly Automation	de3b98fcc4	Merge "chan_sip: Always process updated SDP on media source change" into 16	6 years ago
Walter Doekes	ad9a40a539	chan_sip: Always process updated SDP on media source change ... Fixes no-audio issues when the media source is changed and strictrtp is enabled (default). If the peer media source changes, the SDP session version also changes. If it is lower than the one we had stored, chan_sip would ignore it. This changeset keeps track of the remote media origin identifier, comparing that as well. If it changes, the session version needn't be higher for us to accept the SDP. Common scenario where this would've caused problems: a separate media gateway that informs the caller about premium rates before handing off the call to the final destination. (An alternative fix would be to set ignoresdpversion=yes on the peer.) ASTERISK-28686 Change-Id: I88fdbc5aeb777b583e7738c084254c482a7776ee	6 years ago
Sean Bright	1532da3008	chan_pjsip: Ignore RTP that we haven't negotiated ... If chan_pjsip receives an RTP packet whose payload differs from the channel's native format, and asymmetric_rtp_codec is disabled (the default), Asterisk will switch the channel's native format to match that of the incoming packet without regard to the negotiated payloads. We now check that the received frame is in a format we have negotiated before switching payloads which results in these packets being dropped instead of causing the session to terminate. ASTERISK-28139 #close Reported by: Paul Brooks Change-Id: Icc3b85cee1772026cee5dc1b68459bf9431c14a3	6 years ago
Friendly Automation	567d572b98	Merge "chan_dahdi: Change 999999 to INT_MAX to better reflect "no timeout"" into 16	6 years ago
Andrew Siplas	9895e94dba	chan_dahdi: Change 999999 to INT_MAX to better reflect "no timeout" ... The no-entry timeout set to 999999 == 16⅔ minutes, change to INT_MAX to match behavior of "no timeout" defined in comment. ASTERISK-28702 #close Change-Id: I4ea015986e061374385dba247b272f7aac60bf11	6 years ago
Sean Bright	31027f33db	chan_sip.c: Stop handling continuation lines after reading headers ... lws2sws() does not stop trying to handle header continuation lines even after all headers have been found. This is problematic if the first character of a SIP message body is a space or tab character, so we update to recognize the end of the message header. ASTERISK-28693 #close Reported by: Frank Matano Change-Id: Idec8fa58545cd3fd898cbe0075d76c223f8d33df	6 years ago
George Joseph	64692a3c72	sig_pri: Fix deadlock caused by sig_pri_queue_hangup ... The change to add setting hangupsource to sig_pri_queue_hangup() made in https://gerrit.asterisk.org/c/asterisk/+/12857 casued deadlocks when a hangup request was received from the core at the same time a hanguprequest was received from the remote end via the D channel. Although the PRI's channel private structure was being unlocked before setting the hangupsource, the PRI's own lock was still being held during the process. If channel actions were also coming from the core, a deadlock on the PRI could result. This deadlock could then escalate to the entire DAHDI subsystem via DAHDI's global interface list lock, especially if someone used the PRI CLI commands. Fix: * We now unlock the PRI as well as the PRI's channel private structure before setting the hangupsource, then relock both afterwards. ASTERISK-28605 Reported by: Dirk Wendland Change-Id: Id74aaa5d4e3746063dbe9deed188eb65193cb9c9	6 years ago
Jean Aunis	82a870c8c7	chan_sip: voice frames are no longer transmitted after emitting a COLP ... The SIP transaction state was reset when emitting an UPDATE or a re-INVITE related to a COLP, preventing RTP packets to be emitted. ASTERISK-28647 Change-Id: Ie7a30fa7a97f711e7ba6cc17f221a0993d48bd8b	6 years ago
Friendly Automation	7172230cd0	Merge "chan_sip: in case of tcp/tls, be less annoying about tx errors." into 16	6 years ago
Jaco Kroon	055737d645	chan_sip: in case of tcp/tls, be less annoying about tx errors. ... chan_sip.c:3782 __sip_xmit: sip_xmit of 0x7f1478069230 (len 600) to 213.150.203.60:1492 returned -2: Interrupted system call returned -2 implies this wasn't actually an OS error, so errno makes no sense either. Internal error was already logged higher up, and -2 generally means that either there isn't a valid connection available, or the pipe notification failed, and that is already correctly logged. ASTERISK-28651 #close Change-Id: I46eb82924beeff9dfd86fa6c7eb87d2651b950f2 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	6 years ago
Frederic LE FOLL	31173f6586	chan_sip+native_bridge_rtp: no directmedia for ptime other than default ptime. ... During capabilities selection (joint capabilities of us and peer, configured capability for this peer, or general configured capabilities), if sip_new() does not keep framing information, then directmedia activation will fail for any framing different from default framing. ASTERISK-28637 Change-Id: I99257502788653c2816fc991cac7946453082466	6 years ago
Ben Ford	8cdaa93e65	chan_sip.c: Prevent address change on unauthenticated SIP request. ... If the name of a peer is known and a SIP request is sent using that peer's name, the address of the peer will change even if the request fails the authentication challenge. This means that an endpoint can be altered and even rendered unusuable, even if it was in a working state previously. This can only occur when the nat option is set to the default, or auto_force_rport. This change checks the result of authentication first to ensure it is successful before setting the address and the nat option. ASTERISK-28589 #close Change-Id: I581c5ed1da60ca89f590bd70872de2b660de02df	6 years ago
George Joseph	a1eb0d7355	Merge "chan_dahdi: PRI span status may stay "Down, Active" after a short alarm" into 16	6 years ago
Frederic LE FOLL	faf353e931	chan_dahdi: PRI span status may stay "Down, Active" after a short alarm ... Upon a short PRI disconnection, libpri may maintain Q.921 layer 'up' and may thus not send PRI_EVENT_DCHAN_DOWN / PRI_EVENT_DCHAN_UP events. If pri_event_alarm() clears DCHAN_UP status bit upon alarm detection and no Q.921 reconnection sequence occurs, chan_dahdi will keep seeing span status "Down" at the end of alarm. This patch modifies pri_event_alarm() in order to keep DCHAN_UP bit unchanged. libpri will send a PRI_EVENT_DCHAN_DOWN event if it detects a disconnection of Q.921 layer and this will clear DCHAN_UP if required. ASTERISK-28615 Change-Id: Ibe27df4971fd4c82cc6850020bce4a8b2692c996	6 years ago
Kevin Harwell	30c0af7257	various files - fix some alerts raised by lgtm code analysis ... This patch fixes several issues reported by the lgtm code analysis tool: https://lgtm.com/projects/g/asterisk/asterisk Not all reported issues were addressed in this patch. This patch mostly fixes confirmed reported errors, potential problematic code points, and a few other "low hanging" warnings or recommendations found in core supported modules. These include, but are not limited to the following: * innapropriate stack allocation in loops * buffer overflows * variable declaration "hiding" another variable declaration * comparisons results that are always the same * ambiguously signed bit-field members * missing header guards Change-Id: Id4a881686605d26c94ab5409bc70fcc21efacc25	6 years ago
Salah Ahmed	c73aaa2760	Crash during "pjsip show channelstats" execution ... During execution "pjsip show channelstats" cli command by an external module asterisk crashed. It seems this is a separate thread running to fetch and print rtp stats. The crash happened on the ao2_lock method, just before it going to read the rtp stats on a rtp instance. According to gdb backtrace log, it seems the session media was already cleaned up at that moment. ASTERISK-28578 Change-Id: Ib7c7f39996d942984394782296bb99aae67862e5	6 years ago
Friendly Automation	983bebb680	Merge "chan_pjsip: Prevent segfault when running PlayDTMF on hungup channel" into 16	6 years ago
lvl	19565e0b90	chan_pjsip: Prevent segfault when running PlayDTMF on hungup channel ... ASTERISK-28086 #close Change-Id: Ib3baadc89b9f0477a6f25a63861433812368c5ea	6 years ago
Torrey Searle	9a933c3adc	channel/chan_pjsip: add dialplan function for music on hold ... Add a new dialplan function PJSIP_MOH_PASSTHROUGH that allows the on-hold behavior to be controlled on a per-call basis ASTERISK-28542 #close Change-Id: Iebe905b2ad6dbaa87ab330267147180b05a3c3a8	6 years ago
Joshua Colp	120f2fb5f9	Merge "core: Add H.265/HEVC passthrough support" into 16	6 years ago
Friendly Automation	f9f17f3bfe	Merge "chan_pjsip: Relock correct channel during "fax" redirect." into 16	6 years ago
Joshua Colp	49e1346185	chan_pjsip: Relock correct channel during "fax" redirect. ... When fax detection occurs on an outbound PJSIP channel the redirect operation will result in a masquerade occurring and the underlying channel on the session changing. The code incorrectly relocked the new channel instead of the old channel when returning. This resulted in the new channel being locked indefinitely. The code now always acts on the expected channel. ASTERISK-28538 Change-Id: I2b2e60d07e74383ae7e90d752c036c4b02d6b3a3	6 years ago
Guido Falsi	448b8c9bc2	chan_dahdi: Fix build with clang/llvm ... On FreeBSD using the clang/llvm compiler build fails to build due to the switch statement argument being a non integer type expression. Switch to an if/else if/else construct to sidestep the issue. ASTERISK-28536 #close Change-Id: Idf4a82cc1e94580a2d017fe9e351c226f23e20c8	6 years ago
Sean Bright	518b6bfb5c	channels: Allow updating variable value ... When modifying an already defined variable in some channel drivers they add a new variable with the same name to the list, but that value is never used, only the first one found. Introduce ast_variable_list_replace() and use it where appropriate. ASTERISK-23756 #close Patches: setvar-multiplie.patch submitted by Michael Goryainov Change-Id: Ie1897a96c82b8945e752733612ee963686f32839	6 years ago
Florian Floimair	f85631cf82	core: Add H.265/HEVC passthrough support ... This change adds H.265/HEVC as a known codec and creates a cached "h265" media format for use. Note that RFC 7798 section 7.2 also describes additional SDP parameters. Handling of these is not yet supported. ASTERISK-28512 Change-Id: I26d262cc4110b4f7e99348a3ddc53bad0d2cd1f2	6 years ago
Friendly Automation	ce5029a4db	Merge "chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up" into 16	6 years ago
Frederic LE FOLL	c6b17b5212	chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up ... When the remote ISDN party ends an ISDN call on a PRI link (DISCONNECT), CHANNEL(hangupsource) information is not available. chan_dahdi already contains an ast_set_hangupsource() in __dahdi_exception() function but it seems that ISDN message processing does not use this part of code. Two other channel modules associate ast_queue_hangup() and ast_set_hangupsource() functions calls: - chan_pjsip in chan_pjsip_session_end() function, - chan_sip in sip_queue_hangup_cause() function. chan_iax2 separates them, in iax2_queue_hangup()/iax2_destroy() and set_hangup_source_and_cause(). Thus, I propose to add ast_set_hangupsource() beside ast_queue_hangup() in sig_pri_queue_hangup(), like chan_pjsip and chan_sip already do. ASTERISK-28525 Change-Id: I0f588a4bcf15ccd0648fd69830d1b801c3f21b7c	7 years ago
Igor Goncharovsky	92261d60c8	chan_unistim: Fix clang warning: variable sized type not at end of a struct ... On reading information about initial client packet unistim use dirty implementation of destination ip address retrieval. This fix uses CMSG_*(..) to get ip address and make clang compile without warning. ASTERISK-25592 #close Reported-by: Alexander Traud Change-Id: Ic1fd34c2c2bcc951da65bf62e3f7a8adff8351b1	7 years ago
George Joseph	cc1b57a51d	Merge "chan_unistim: Fix code, causing all incoming DTMF sent back to asterisk" into 16	7 years ago
Igor Goncharovsky	78d00c277c	chan_unistim: Fix code, causing all incoming DTMF sent back to asterisk ... Current implementation of ast_channel_tech send_digit_begin hook uses same function for tone playback as key press handler. This cause every incoming dtmf send back to asterisk. In case of two unistim phones connected to each other, it'll cause indefinite DTMF loop. Fix add separate function for dtmf tone phone play. Change-Id: I5795db468df552f0c89c7576b6b3858b26c4eab4	7 years ago
Igor Goncharovsky	821b7561f8	chan_unistim: Fix RTP port byte order for big-endian arch ... This patch fixes one-way oudio that users expirienced on big-endian architechtires. RTP port number bytes was stored in improper order and phone sent RTP to wrong RTP port. Reported-by: Andrey Ionov Change-Id: I9a9ca7f26e31a67bbbceff12923baa10dfb8a3be	7 years ago
George Joseph	c00a010fe8	chan_rtp: Accept hostname as well as ip address as destination ... The UnicastRTP channel driver provided by chan_rtp now accepts "<hostname>:<port>" as an alternative to "<ip_address>:<port>" in the destination. The first AAAA (preferred) or A record resolved will be used as the destination. The lookup is synchronous so beware of possible dialplan delays if you specify a hostname. Change-Id: Ie6f95b983a8792bf0dacc64c7953a41032dba677	7 years ago
Kevin Harwell	6bb14150c4	various modules: json integer overflow ... There were still a few places in the code that could overflow when "packing" a json object with a value outside the base type integer's range. For instance: unsigned int value = INT_MAX + 1 ast_json_pack("{s: i}", value); would result in a negative number being "packed". In those situations this patch alters those values to a ast_json_int_t, which widens the value up to a long or long long. ASTERISK-28480 Change-Id: Ied530780d83e6f1772adba0e28d8938ef30c49a1	7 years ago
George Joseph	85dcc699fa	Merge "openr2(6/6): Set hangup cause" into 16	7 years ago
George Joseph	d2a2131447	Merge "openr2(5/6): added cli command -- mfcr2 destroy link <index>" into 16	7 years ago
Friendly Automation	f7c30bc93d	Merge "openr2(4/6): added new cli command -- mfcr2 show links" into 16	7 years ago
Friendly Automation	24219c8d1d	Merge "openr2(3/6): Convert r2links to standard Asterisk AST_LIST*" into 16	7 years ago
George Joseph	2b407cb531	Merge "openr2(2/6): Stop polling channels when DAHDI returns -ENODEV (e.g: plug-out)" into 16	7 years ago
Friendly Automation	6e33772432	Merge "openr2(1/6): bugfix in configuration saving" into 16	7 years ago
Leonid Fainshtein	3814faf848	openr2(6/6): Set hangup cause ... Change-Id: I94dc38920e6e77cc73062648f62fdd613d0d1452 Signed-off-by: Oron Peled <oron.peled@xorcom.com>	7 years ago
Tzafrir Cohen	06515707df	openr2(5/6): added cli command -- mfcr2 destroy link <index> ... Change-Id: I452d6a853bcd8c6e194455b19e5e017713e9c0fe Signed-off-by: Oron Peled <oron.peled@xorcom.com>	7 years ago
Tzafrir Cohen	93a093f6c4	openr2(4/6): added new cli command -- mfcr2 show links ... * This command show the MFC/R2 links Change-Id: I213822e1b7ef9c05bd89a2ba62df8e0856ce9f84 Signed-off-by: Oron Peled <oron.peled@xorcom.com>	7 years ago
Tzafrir Cohen	a45cfefb77	openr2(3/6): Convert r2links to standard Asterisk AST_LIST* ... Change-Id: Ibcb2401515a58782a1488c0b9efbed201c3f3a17 Signed-off-by: Oron Peled <oron.peled@xorcom.com>	7 years ago
Tzafrir Cohen	ec6e88592a	openr2(2/6): Stop polling channels when DAHDI returns -ENODEV (e.g: plug-out) ... Otherwise, OpenR2 threads go crazy and consume almost all CPU resources Change-Id: I10a41f617613fe7399c5bdced5c64a2751173f28 Signed-off-by: Oron Peled <oron.peled@xorcom.com>	7 years ago
Tzafrir Cohen	7b6df814a6	openr2(1/6): bugfix in configuration saving ... Details: - The memcpy() call copied part of "dahdi_conf" and not "dahdi_conf.mfcr2" - As a result, the memcmp() in dahdi_r2_get_link() always fails - This cause dahdi_r2_get_link() to create new link for every channel (instead of a new link for every ~30 channels) - With the fix, far less links are generated -- so we use far less threads Change-Id: I7259dd6272f5e46e8a6c7f5bf3e8c2ec01b8c132 Signed-off-by: Oron Peled <oron.peled@xorcom.com>	7 years ago
Dan Cropp	f4896703b9	chan_pjsip: Transmit REFER waits for the REFER result setting TRANSFERSTATUS ... Previously, when a Transfer (REFER) was performed, chan_pjsip would set the TRANSFERSTATUS to SUCCESS when the REFER was queued up. This did not reflect a successful/unsuccessful transfer the way chan_sip did. Added a callback module to process the refer subscription information. Now depends on res_pjsip_pubsub so call transfer progress can be monitored and reported ASTERISK-26968 #close Reported-by: Dan Cropp Change-Id: If6c27c757c66f71e8b75e3fe49da53ebe62395dc	7 years ago
Francesco Castellano	6c59df17a5	chan_sip: Handle invalid SDP answer to T.38 re-invite ... The chan_sip module performs a T.38 re-invite using a single media stream of udptl, and expects the SDP answer to be the same. If an SDP answer is received instead that contains an additional media stream with no joint codec a crash will occur as the code assumes that at least one joint codec will exist in this scenario. This change removes this assumption. ASTERISK-28465 Change-Id: I8b02845b53344c6babe867a3f0a5231045c7ac87	7 years ago
Chris-Savinovich	da1db4f842	chan_dahdi.c: crash in chan_dahdi ... Fixes a crash in chan_dahdi occurring on 32-bit systems. A previous patch introduced a variable of type unassigned long long which is 64-bits. Casting it as 'ast_json_int_t' along with JSON type 'I' makes it work with 32-bit systems. ASTERISK-28457 Change-Id: I9cef6b5f2d826fc5c93f2f6a1c997c4e3e6c93fe	7 years ago
George Joseph	31d755e805	sig_pri: Address gcc9 issues ... A few more format truncation issues addressed. Change-Id: I047f373169caaca0eec4889d3c0e5e10f130017a	7 years ago