asterisk

Commit Graph

Author	SHA1	Message	Date
Joshua C. Colp	5e24379cf2	chan_sip: Send 403 when ACL fails. Change-Id: I0910c79196f2b7c7e5ad6f1db95e83800ac737a2	5 years ago
Michael Neuhauser	580e260ff8	chan_psip, res_pjsip_sdp_rtp: ignore rtptimeout if direct-media is active Do not hang up a PJSIP channel on RTP timeout if that channel is in a direct-media bridge. Also reset the time of the last received RTP packet when direct-media ends (wait full rtp_timeout period before checking first time after audio came back to Asterisk). ASTERISK-28774 Reported-by: Michael Neuhauser Change-Id: I8b62012be7685849e8fb2b1c5dd39d35313ca2d1	5 years ago
Sean Bright	31c6a20862	chan_vpb: Fix 'catching polymorphic type ... by value' error Fixes the following compile error: chan_vpb.cc:2688:26: error: catching polymorphic type ‘class std::exception’ by value Change-Id: Ic87bc357d72427d77626735c83200fd278a7a649	5 years ago
Paulo Vicentini	8645a294ca	chan_pjsip: Check audio frame when remote SSRC changes. If the SSRC of a received RTP packet differed from the previous SSRC an SSRC change control frame would be queued ahead of the media frame. In the case of audio this would result in the format of the audio frame not being checked, and if it differed or was not allowed then it could cause the call to drop due to failure to set up a translation path. The chan_pjsip module will now no longer assume the first frame will be the audio frame and instead goes through the complete list to find it. ASTERISK-28759 Change-Id: I6d854cc523f343e299a615636fc65bdbd5f809ec	5 years ago
Walter Doekes	a5ec14970d	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	5 years ago
Friendly Automation	5192a045f6	Merge "chan_sip: Always process updated SDP on media source change" into 17	5 years ago
Walter Doekes	be6eec7b25	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	5 years ago
Sean Bright	ace97a3786	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	5 years ago
Joshua Colp	73cd259895	Merge "chan_dahdi: Change 999999 to INT_MAX to better reflect "no timeout"" into 17	5 years ago
Andrew Siplas	1c5116f842	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	5 years ago
Sean Bright	7c137ebbd4	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	5 years ago
George Joseph	015cdc9f64	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	5 years ago
Jean Aunis	811c7bdabe	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	5 years ago
Jaco Kroon	32ce8fa046	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	3e73893e53	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	665a94cb76	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
Frederic LE FOLL	d3dd4c5459	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	8bc6fa0fbd	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	40acd7d198	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: I3e05980dd4694577be6d39be2c21a5736bae3c6f	6 years ago
lvl	71f86e78b6	chan_pjsip: Prevent segfault when running PlayDTMF on hungup channel ASTERISK-28086 #close Change-Id: Ib3baadc89b9f0477a6f25a63861433812368c5ea	6 years ago
Torrey Searle	55b760d762	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	d2087607b6	Merge "core: Add H.265/HEVC passthrough support" into 17	6 years ago
Joshua Colp	69e8a44039	Merge "chan_pjsip: Relock correct channel during "fax" redirect." into 17	6 years ago
Florian Floimair	d7a3e4f5cf	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
Joshua Colp	4d1baa3ae8	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	5ff2f7a016	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	2fa296e7d4	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
Joshua Colp	377f56bdc0	Merge "chan_sip: Update links referenced in deprecation notice" into 17	6 years ago
Frederic LE FOLL	9e67c92502	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	6 years ago
George Joseph	c060ef7dee	chan_sip: Update links referenced in deprecation notice The links in the deprecation notice were the shortened variety but it makes better sense to show the unshortened links as they're more descriptive. I.E. wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip rather than wiki.asterisk.org/wiki/x/tAHOAQ Change-Id: If2da5d5243e2d4a6f193b15691d23e7e5a7c57a9	6 years ago
Igor Goncharovsky	056ddf76ce	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	6 years ago
George Joseph	2cd5b84af8	Merge "chan_unistim: Fix code, causing all incoming DTMF sent back to asterisk" into 17	6 years ago
Igor Goncharovsky	95b3c9839d	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	6 years ago
Igor Goncharovsky	e7f1f7eefd	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	6 years ago
George Joseph	581ffdf6a9	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	6 years ago
Kevin Harwell	a305f2fdcb	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	6 years ago
Friendly Automation	7646f4196b	Merge "openr2(6/6): Set hangup cause"	6 years ago
George Joseph	96a2ce1a0d	Merge "openr2(5/6): added cli command -- mfcr2 destroy link <index>"	6 years ago
George Joseph	64b6d0fc28	Merge "openr2(4/6): added new cli command -- mfcr2 show links"	6 years ago
Friendly Automation	d24d94beba	Merge "openr2(3/6): Convert r2links to standard Asterisk AST_LIST*"	6 years ago
George Joseph	8c3ed46829	Merge "openr2(2/6): Stop polling channels when DAHDI returns -ENODEV (e.g: plug-out)"	6 years ago
George Joseph	497308b5d9	Merge "openr2(1/6): bugfix in configuration saving"	6 years ago
George Joseph	799c4cf494	Merge "chan_pjsip: Transmit REFER waits for the REFER result setting TRANSFERSTATUS"	6 years ago
Leonid Fainshtein	098797628e	openr2(6/6): Set hangup cause Change-Id: I94dc38920e6e77cc73062648f62fdd613d0d1452 Signed-off-by: Oron Peled <oron.peled@xorcom.com>	6 years ago
Tzafrir Cohen	f67094503d	openr2(5/6): added cli command -- mfcr2 destroy link <index> Change-Id: I452d6a853bcd8c6e194455b19e5e017713e9c0fe Signed-off-by: Oron Peled <oron.peled@xorcom.com>	6 years ago
Tzafrir Cohen	64bf3e3e82	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>	6 years ago
Tzafrir Cohen	f61adf2cf5	openr2(3/6): Convert r2links to standard Asterisk AST_LIST* Change-Id: Ibcb2401515a58782a1488c0b9efbed201c3f3a17 Signed-off-by: Oron Peled <oron.peled@xorcom.com>	6 years ago
Tzafrir Cohen	97d2549bb1	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>	6 years ago
Tzafrir Cohen	2f0a8e12f9	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>	6 years ago
Kevin Harwell	857ee76f4b	Merge "MWI: Update modules that subscribe to MWI to use new API calls"	6 years ago

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8589 Commits (1b01a6abfd1f323c9043fc9c65f4ab038296f77c)