asterisk

Commit Graph

Author	SHA1	Message	Date
Friendly Automation	2a8f759374	Merge "chan_sip: voice frames are no longer transmitted after emitting a COLP"	6 years ago
Friendly Automation	c3cf0e330c	Merge "chan_sip: in case of tcp/tls, be less annoying about tx errors."	6 years ago
Jaco Kroon	365d007eb6	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
Jean Aunis	9c9296c635	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
Frederic LE FOLL	7624cbb155	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
Joshua Colp	cd3a2a478f	Merge "core: Improve MALLOC_DEBUG for frames."	6 years ago
Ben Ford	4a1cadeadb	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	a68299f508	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	bdd785d31c	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
Corey Farrell	8a1f30af04	core: Improve MALLOC_DEBUG for frames. * Pass caller information to frame allocation functions. * Disable caching as it interfers with MALLOC_DEBUG reporting. * Stop using ast_calloc_cache. Change-Id: Id343cd80a3db941d2daefde2a060750fea8cd260	6 years ago
Salah Ahmed	ddb0091da5	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	c03f50c1c8	chan_pjsip: Prevent segfault when running PlayDTMF on hungup channel ASTERISK-28086 #close Change-Id: Ib3baadc89b9f0477a6f25a63861433812368c5ea	6 years ago
Torrey Searle	b43cdc7f1e	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
Friendly Automation	2205e25073	Merge "core: Add H.265/HEVC passthrough support"	6 years ago
Friendly Automation	0630dc3e49	Merge "chan_pjsip: Relock correct channel during "fax" redirect."	6 years ago
Florian Floimair	c18983207d	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
Guido Falsi	4072e219f7	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
Joshua Colp	c358da472e	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
Sean Bright	32ce6e9a06	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
Friendly Automation	bf63dcab6c	Merge "chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up"	6 years ago
Joshua Colp	a009dd1c1c	Merge "chan_sip: Update links referenced in deprecation notice"	6 years ago
Frederic LE FOLL	41b67f150e	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	5fb9b23105	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	3863ab9af9	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	d3cfab159c	Merge "chan_unistim: Fix code, causing all incoming DTMF sent back to asterisk"	6 years ago
Igor Goncharovsky	1d06a1efb3	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	649003821e	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	19045db392	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	3656c42cb0	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
Francesco Castellano	8438d19b81	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	6 years ago
Kevin Harwell	9637e1dfdc	MWI: Update modules that subscribe to MWI to use new API calls The MWI core recently got some new API calls that make tracking MWI state lifetime more reliable. This patch updates those modules that subscribe to specific MWI topics to use the new API. Specifically, these modules now subscribe to both MWI topics and MWI state. ASTERISK-28442 Change-Id: I32bef880b647246823dbccdf44a98d384fcabfbd	6 years ago
Chris-Savinovich	0e669712e2	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	6 years ago
Dan Cropp	e52fbae00f	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	6 years ago
George Joseph	13e89d372b	sig_pri: Address gcc9 issues A few more format truncation issues addressed. Change-Id: I047f373169caaca0eec4889d3c0e5e10f130017a	6 years ago
George Joseph	92d4ec2906	chan_dahdi: Address gcc9 issues Fixed format-truncation issues in chan_dahdi.c and sig_analog.c. Since they're related to fields provided by dahdi-tools we can't change the buffer sizes so we're just checking the return from snprintf and printing an errior if we overflow. Change-Id: Idc1f3c1565b88a7d145332a0196074b5832864e5	6 years ago
agupta	d2f7b22640	chan_pjsip.c: Check for channel and session to not be NULL in hangup We have seen some rare case of segmentation fault in hangup function and we could notice that channel pointer was NULL. Debug log shows that there is a 200 OK answer and SIP timeout at the same time. It looks that while the SIP session was being destroyed due to timeout call hangup due to answer event lead to race condition and channel is being destroyed from two different places. The check ensures we check it not to be NULL before freeing it. ASTERISK-25371 Change-Id: I19f6566830640625e08f7b87bfe15758ad33a778	6 years ago

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8582 Commits (2a8f75937423f5a9aa14da948d8016fe8805227b)