8558 Commits (18f5f5fc99a8106c1856d13aeeeeafa2b13ef033)

Author	SHA1	Message	Date
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
George Joseph	216cc44e8e	Merge "pjsip: replace 180 by 183 if SDP negotiation has completed"	6 years ago
Guido Falsi	db535439f2	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	466a17964f	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	7 years ago
George Joseph	c5c953c1f1	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	7 years ago
Kevin Harwell	ff0d0ac23a	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	7 years ago
Salah Ahmed	5009d6d97a	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	7 years ago
Ben Ford	dd1cc7791c	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	7 years ago
Alexei Gradinari	f236377ce9	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	7 years ago
Sean Bright	1499640da9	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	7 years ago
Kevin Harwell	8f5a6ed52d	Merge "chan_dahdi: Add logical group at DAHDIChannel event and CHANNEL function"	7 years ago
cirillor	0d6d51b175	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	7 years ago
Torrey Searle	4661c08549	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	7 years ago
Sean Bright	2473b791b9	Replace calls to strtok() with strtok_r() ... strtok() uses a static buffer, making it not thread safe. Also add a #define to cause a compile failure if strtok is used. Change-Id: Icce265153e1e65adafa8849334438ab6d190e541	7 years ago
sungtae kim	fb651756c7	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	7 years ago
Giuseppe Sucameli	0bde3751a0	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
Joshua Colp	50ac85cb40	stasis: Segment channel snapshot to reduce creation cost. ... When a channel snapshot was created it used to be done from scratch, copying all data (many strings). This incurs a cost when doing so. This change segments the channel snapshot into different components which can be reused if unchanged from the previous snapshot creation, reducing the cost. In normal cases this results in some pointers being copied with reference count being bumped, some integers being set, and a string or two copied. The other benefit is that it is now possible to determine if a channel snapshot update is redundant and thus stop it before a message is published to stasis. The specific segments in the channel snapshot were split up based on whether they are changed together, how often they are changed, and their general grouping. In practice only 1 (or 0) of the segments actually get changed in normal operation. Invalidation is done by setting a flag on the channel when the segment source is changed, forcing creation of a new segment when the channel snapshot is created. ASTERISK-28119 Change-Id: I5d7ef3df963a88ac47bc187d73c5225c315f8423	7 years ago
Joshua Colp	d0ccbb3377	stasis: Use an implementation specific channel snapshot cache. ... Channels no longer use the Stasis cache for channel snapshots. Instead they are stored in a hash table in stasis_channels which reduces the number of Stasis messages created and allows better storage. As a result the following APIs are no longer available since the stasis cache is no longer used: ast_channel_topic_cached() ast_channel_topic_all_cached() The ast_channel_cache_all() and ast_channel_cache_by_name() functions now return an ao2_container of ast_channel_snapshots rather than a container of stasis_messages therefore you can't (and don't need to) call stasis_cache functions on it. The ast_channel_topic_all() function now returns a normal topic not a cached one so you can't use stasis cache functions on it either. The ast_channel_snapshot_type() stasis message now has the ast_channel_snapshot_update structure as it's data. It contains the last snapshot and the new one. ast_channel_snapshot_get_latest() still returns the latest snapshot. The latest snapshot is now stored on the channel itself to eliminate cache hits when Stasis messages that have the snapshot as a payload are created. ASTERISK-28102 Change-Id: I9334febff60a82d7c39703e49059fa3a68825786	7 years ago
Corey Farrell	021ce938ca	astobj2: Remove legacy ao2_container_alloc routine. ... Replace usage of ao2_container_alloc with ao2_container_alloc_hash or ao2_container_alloc_list. Remove ao2_container_alloc macro. Change-Id: I0907d78bc66efc775672df37c8faad00f2f6c088	7 years ago
Joshua Colp	614c8a60d7	Merge "stasis: Add internal filtering of messages."	7 years ago
Alexei Gradinari	fa048183aa	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	3077ad0c24	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
Jasper Hafkenscheid	2cf5079205	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
George Joseph	26810197c7	Merge "pjsip: new endpoint's options to control Connected Line updates"	7 years ago
Alexei Gradinari	eee935983b	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