asterisk

Commit Graph

Author	SHA1	Message	Date
Joshua Colp	16790d6029	Merge "chan_pjsip: Relock correct channel during "fax" redirect." into 13	6 years ago
Joshua Colp	a97bd89f84	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	ae70c2a428	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	622e487cd4	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
Frederic LE FOLL	2b563368df	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	e6f74e94f6	Merge "chan_unistim: Fix clang warning: variable sized type not at end of a struct" into 13	6 years ago
Igor Goncharovsky	8e07fcc30d	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
Friendly Automation	8b3cee9068	Merge "chan_unistim: Fix code, causing all incoming DTMF sent back to asterisk" into 13	6 years ago
Igor Goncharovsky	7f1a574ce7	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	b003c9b3ae	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
Kevin Harwell	f145b58542	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
Leonid Fainshtein	8e09fbb4e9	openr2(6/6): Set hangup cause Change-Id: I94dc38920e6e77cc73062648f62fdd613d0d1452 Signed-off-by: Oron Peled <oron.peled@xorcom.com>	6 years ago
Tzafrir Cohen	fdaefbb20a	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	effca9917d	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	b3de2c0e1e	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	4b779a59e4	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
George Joseph	0cded803ec	Merge "openr2(1/6): bugfix in configuration saving" into 13	6 years ago
Francesco Castellano	1318a3a2b7	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
Tzafrir Cohen	d8bf4b1608	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
Chris-Savinovich	e206a54d59	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
George Joseph	79087b6aeb	sig_pri: Address gcc9 issues A few more format truncation issues addressed. Change-Id: I047f373169caaca0eec4889d3c0e5e10f130017a	6 years ago
George Joseph	d0f01af913	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	67841b8f55	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
Friendly Automation	e4d91ab4b2	Merge "pjsip: replace 180 by 183 if SDP negotiation has completed" into 13	6 years ago
Guido Falsi	ac4921c373	chan_dahdi: add missing include. After some definitions have been moved to asterisk/mwi.h the files channels/chan_dahdi.h channels/sig_pri.c are missing this new include. ASTERISK-28427 #close Change-Id: Ia8cc595eeda653324643f40dcd9799d4c3f0ac91	6 years ago
Alexei Gradinari	595d60846a	pjsip: replace 180 by 183 if SDP negotiation has completed The caller endpoint hears dead silence if a callee replies 180 (without SDP) and the caller already received 183 (with SDP). It happens because Asterisk sends 180 (WITH SDP) to the caller, there are not incoming RTP packets from the callee and Asterisk does not generate inband ringing, so there are not any outgoing RTP packets to the caller. This patch replaces 180 by 183 if SDP negotiation has completed, as if the caller endpoint is configured with "inband_progress=yes". In this case Asterisk will generate inband ringing untill Asterisk receive incoming RTP packets from the callee. ASTERISK-27994 #close Change-Id: I7450b751083ec30d68d6abffe922215a15ae5a73	6 years ago
George Joseph	4337895aee	Fixes for GCC 9 Various fixes for issues caught by gcc 9. Mostly snprintf trying to copy to a buffer potentially too small. ASTERISK-28412 Change-Id: I9e85a60f3c81d46df16cfdd1c329ce63432cf32e	6 years ago
Kevin Harwell	4ea20c9c85	mwi core: Move core MWI functionality into its own files There is enough MWI functionality to warrant it having its own 'c' and header files. This patch moves all current core MWI data structures, and functions into the following files: main/mwi.h main/mwi.c Note, code was simply moved, and not modified. However, this patch is also in preparation for core MWI changes, and additions to come. Change-Id: I9dde8bfae1e7ec254fa63166e090f77e4d3097e0	6 years ago
George Joseph	1a4fbaa52e	Merge "chan_pjsip: DTMF Mode auto_info fallback lead to both inband and info" into 13	6 years ago
Salah Ahmed	a9a0303544	chan_pjsip: DTMF Mode auto_info fallback lead to both inband and info When the dtmf_mode on an endpoint is configured as "auto_info" Asterisk will produce an inband DTMF tone alongside an INFO message when sending DTMF. ASTERISK-28371 Change-Id: I1380b82f006e110a1b83fbb50c9873edd13a5d9a	6 years ago
Ben Ford	4853fc2218	build: Fix compiler warnings/errors. The compiler complained about a couple of variables that weren't initialized but were being used. Initializing them to NULL resolves the warnings/errors. ASTERISK-28362 #close Change-Id: I6243afc5459b416edff6bbf571b0489f6b852e4b	6 years ago
Alexei Gradinari	6e20e071a9	pjsip: restrict function PJSIP_PARSE_URI to parse only SIP/SIPS URIs The next usage of PJSIP_PARSE_URI will crash asterisk ${PJSIP_PARSE_URI(tel:+1234567890,host)} or ${PJSIP_PARSE_URI(192.168.1.1:5060,host)} The function pjsip_parse_uri successfully parses then, but returns struct pjsip_other_uri *. This patch restricts parsing only SIP/SIPS URIs. Change-Id: I16f255c2b86a80a67e9f9604b94b129a381dd25e	6 years ago
Sean Bright	53aa750839	chan_sip: Ensure 'qualifygap' isn't negative Passing negative intervals to the scheduler rips a hole in the space-time continuum. ASTERISK-25792 #close Reported by: Paul Sandys Change-Id: Ie706f21cee05f76ffb6f7d89e9c867930ee7bcd7	6 years ago
Kevin Harwell	9717d1672c	Merge "chan_dahdi: Add logical group at DAHDIChannel event and CHANNEL function" into 13	6 years ago
Friendly Automation	4f6daa0aff	Merge "chan_pjsip: add a flag to ignore 183 responses if no SDP present" into 13	6 years ago
Torrey Searle	cbc704c5ec	chan_pjsip: add a flag to ignore 183 responses if no SDP present chan_sip will always ignore 183 responses that do not contain SDP however, chan_pjsip will currently always translate it into a 183 with SDP. This new flag allows chan_pjsip to have the same behavior as chan_sip. ASTERISK-28322 #close Change-Id: If81cfaa17c11b6ac703e3d71696f259d86c6be4a	6 years ago
Sean Bright	1cb6466268	Replace calls to strtok() with strtok_r() strtok() uses a static buffer, making it not thread safe. Change-Id: Icce265153e1e65adafa8849334438ab6d190e541	6 years ago
cirillor	5065f31fca	chan_dahdi: Add logical group at DAHDIChannel event and CHANNEL function Add logical group at DAHDIChannel event and create "dahdi_group" at CHANNEL function. ASTERISK-28317 Change-Id: Ic1f834cd53982a9707a9748395ee746d6575086a	6 years ago
sungtae kim	f6689547ae	chan_pjsip: Changed to continued after invalid media for pjsip show channelstats Currently, the pjsip show channelstats cli does not show channel's stats after hits the invalid channel info. This makes hard to use this cli. Changed to keep iterate after hits the invalid channel info. ASTERISK-28292 Change-Id: I3efdff1c9e1b1efd3c971fb82ae77aa133a6f43c	6 years ago
Giuseppe Sucameli	f4afd097af	chan_sip: Fix leak using contact ACL Free old peer's contactacl before overwrite it within build_peer. ASTERISK-28194 Change-Id: Ie580db6494e50cee0e2a44b38e568e34116ff54c	7 years ago
Corey Farrell	0a9904e1c6	astobj2: Eliminate usage of legacy ao2_container_alloc routine. Replace usage of ao2_container_alloc with ao2_container_alloc_hash or ao2_container_alloc_list. ao2_container_alloc is now restricted to modules only and is being removed from Asterisk 17. Change-Id: I0907d78bc66efc775672df37c8faad00f2f6c088	7 years ago
Joshua Colp	b7b581f209	Merge "stasis: Add internal filtering of messages." into 13	7 years ago
Alexei Gradinari	b6d0fbda9d	pjsip: New function PJSIP_PARSE_URI to parse URI and return part of URI New dialplan function PJSIP_PARSE_URI added to parse an URI and return a specified part of the URI. This is useful when need to get part of the URI instead of cutting it using a CUT function. For example to get 'user' part of Remote URI ${PJSIP_PARSE_URI(${CHANNEL(pjsip,remote_uri)},user)} ASTERISK-28144 #close Change-Id: I5d828fb87f6803b6c1152bb7b44835f027bb9d5a	7 years ago
Joshua Colp	d748ed4147	stasis: Add internal filtering of messages. This change adds the ability for subscriptions to indicate which message types they are interested in accepting. By doing so the filtering is done before being dispatched to the subscriber, reducing the amount of work that has to be done. This is optional and if a subscriber does not add message types they wish to accept and set the subscription to selective filtering the previous behavior is preserved and they receive all messages. There is also the ability to explicitly force the reception of all messages for cases such as AMI or ARI where a large number of messages are expected that are then generically converted into a different format. ASTERISK-28103 Change-Id: I99bee23895baa0a117985d51683f7963b77aa190	7 years ago
George Joseph	8b965c386a	Merge "chan_sip: Attempt ast_do_pickup in handle_invite_replaces" into 13	7 years ago
Jasper Hafkenscheid	cf193d53ad	chan_sip: Attempt ast_do_pickup in handle_invite_replaces When a call pickup is performed using and invite with replaces header the ast_do_pickup method is attempted and a PICKUP stasis message is sent. ASTERISK-28081 #close Reported-by: Luit van Drongelen Change-Id: Ieb1442027a3ce6ae55faca47bc095e53972f947a	7 years ago
Alexei Gradinari	bfe3821800	pjsip: new endpoint's options to control Connected Line updates This patch adds new options 'trust_connected_line' and 'send_connected_line' to the endpoint. The option 'trust_connected_line' is to control if connected line updates are accepted from this endpoint. The option 'send_connected_line' is to control if connected line updates can be sent to this endpoint. The default value is 'yes' for both options. Change-Id: I16af967815efd904597ec2f033337e4333d097cd	7 years ago
Corey Farrell	54a1fbe428	astobj2: Eliminate usage of legacy container allocation macros. These macros have been documented as legacy for a long time but are still used in new code because they exist. Remove all references to: * ao2_container_alloc_options * ao2_t_container_alloc_options * ao2_t_container_alloc These macro's are still available for use but only in modules. Only ao2_container_alloc remains due to it's use in over 100 places. Change-Id: I1a26258b5bf3deb081aaeed11a0baa175c933c7a	7 years ago
Corey Farrell	c4b979a841	chan_sip: Tell module loader that chan_sip is extended support. Change-Id: I33508c134b1be888b8884f5dcfee19087634e415	7 years ago
pk16208	84c574bb8b	chan_sip: SipNotify on Chan_Sip vi AMI behave different to CLI With tls and udp enabled asterisk generates a warning about sending message via udp instead of tls. sip notify command via cli works as expected and without warning. asterisk has to set the connection information accordingly to connection and not on presumption ASTERISK-28057 #close Change-Id: Ib43315aba1f2c14ba077b52d8c5b00be0006656e	7 years ago

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8463 Commits (c881d363190c1994e3f4ba5bb75f45444baf00ed)