31618 Commits (76876300748045d89e77dce3a14a7c68ff888ab2)

 

All Branches   

Search

Author	SHA1	Message	Date
George Joseph	361295f192	Merge "channel.c: Address stack overflow in does_id_conflict()" into 16	7 years ago
Kevin Harwell	755febe497	rtp_engine: rtcp_report_to_json can overflow the ssrc integer value ... When writing an RTCP report to json the code attempts to pack the "ssrc" and "source_ssrc" unsigned integer values as a signed int value type. This of course means if the ssrc's unsigned value is greater than that which can fit into a signed integer value it gets converted to a negative number. Subsequently, the negative value goes out in the json report. This patch now packs the value as a json_int_t, which is the widest integer type available on a given system. This should make it so the value no longer overflows. Note, this was caught by two failing tests hep/rtcp-receiver/ and hep/rtcp-sender. Change-Id: I2af275286ee5e795b79f0c3d450d9e4b28e958b0	7 years ago
George Joseph	ba91c73f8e	app_voicemail: Fix stack overrun in append_mailbox ... The append_mailbox function wasn't calculating the correct length to pass to ast_alloca and it wasn't handling the case where context might be empty. Found by the Address Sanitizer. Change-Id: I7eb51c7bd18a7a8dbdba261462a95cc69e84f161	7 years ago
George Joseph	0a1a96d331	channel.c: Address stack overflow in does_id_conflict() ... does_id_conflict() was passing a pointer to an int to a callback that expected a pointer to a size_t. Found by the Address Sanitizer. Change-Id: I0ff542067eef63a14a60301654d65d34fe2ad503	7 years ago
Corey Farrell	7a7b21f3a0	res_rtp_asterisk: Fix crash on ast_rtp_new failure. ... ast_rtp_new free'd rtp upon failure, but rtp_engine.c would also call the destroy callback. Remove call to ast_free from ast_rtp_new, leave it to rtp_engine.c to initiate the full cleanup. Add error detection for the ssrc_mapping vector initialization. In rtp_allocate_transport set rtp->s = -1 in the failure path where we close that FD to ensure we don't try closing it twice. ASTERISK-27854 #close Change-Id: Ie02aecbb46228ca804e24b19cec2bb6f7b94e451	7 years ago
Sean Bright	880905e7eb	res_rtp_asterisk: Reset all settings on module reload ... 'rtpchecksums' and 'rtcpinterval' are not being reset to their defaults if they are not present in the updated configuration file. Change-Id: I1162e40199314d46cf3225d5e1271c4c81176670	7 years ago
George Joseph	fa1b836374	app_voicemail: Cleanup mailbox topic and cache ... app_voicemail wasn't properly cleaning up the stasis cache or the mwi topic pool when the module was unloaded or when a user was deleted as a result of a reload. This resulted in leaks in both areas. * app_voicemail now calls ast_delete_mwi_state_full when it frees a user structure and ast_delete_mwi_state_full in turn now calls the new stasis_topic_pool_delete_topic function to clear the topic from the pool. Change-Id: Ide23144a4a810e7e0faad5a8e988d15947965df8	7 years ago
George Joseph	dec6ebd9e1	Merge "stasis: No need to keep a stasis type ref in a stasis msg or cache object." into 16	7 years ago
Sean Bright	339bf0cf7b	AST-2018-009: Fix crash processing websocket HTTP Upgrade requests ... The HTTP request processing in res_http_websocket allocates additional space on the stack for various headers received during an Upgrade request. An attacker could send a specially crafted request that causes this code to overflow the stack, resulting in a crash. * No longer allocate memory from the stack in a loop to parse the header values. NOTE: There is a slight API change when using the passed in strings as is. We now require the passed in strings to no longer have leading or trailing whitespace. This isn't a problem as the only callers have already done this before passing the strings to the affected function. ASTERISK-28013 #close Change-Id: Ia564825a8a95e085fd17e658cb777fe1afa8091a	7 years ago
George Joseph	1a9c69d729	stasis: Add function to delete topic from pool ... There's been a long standing leak when using topic pools. The topics in the pool get cleaned up when the last pool reference is released but you can't remove a topic specifically. If you reloaded app_voicemail for instance, and mailboxes went away, their topics were left in the pool. * Added stasis_topic_pool_delete_topic() so modules can clean up topics from pools. * Registered the topic pool containers so it can be examined from the CLI when AO2_DEBUG is enabled. They'll be named "<topic_pool_name>-pool". Change-Id: Ib7957951ee5c9b9b4482af7b9b4349112d62bc25	7 years ago
Joshua Colp	84b60aeb70	Merge "stasis_cache: Stop caching stasis subscription change messages" into 16	7 years ago
hajekd	8811ab1803	chan_sip.c: chan_sip unstable with TLS after asterisk start or reloads ... Fixes random asterisk crash on start or reload with TLS phones. ASTERISK-28034 #close Reported-by: David Hajek Change-Id: I2a859f97dc80c348e2fa56e918214ee29521c4ac	7 years ago
Joshua Colp	36898351e5	Merge "pjproject: Update initial 2.8 patches to apply cleanly." into 16	7 years ago
Joshua Colp	2f38bcdfc0	res_remb_modifier: Add module for controlling REMB from CLI. ... This adds a module which registers a CLI command that can set the REMB bitrate value for REMB as it enters or exits Asterisk. This allows you to ignore what Asterisk or a client produces and is useful for demonstrations. This does not generate REMB frames, however, but just modifies them as they flow to or from a channel. Change-Id: Ib089427c46a4a36d645cecfe02406adb38c17bec	7 years ago
Joshua Colp	e313ec3a7e	Merge "alembic: fix suppress_q850_reason_headers column name" into 16	7 years ago
Joshua Colp	3c1a8be11e	Merge "app_voicemail: Remove need to subscribe to stasis" into 16	7 years ago
Richard Mudgett	f6695249a5	stasis: No need to keep a stasis type ref in a stasis msg or cache object. ... Stasis message types are global ao2 objects and we make stasis messages and cache entries hold references to them. Since there are currently situations where cache objects are never deleted, the reference count on the types can exceed 100000 and generate a FRACK assertion message. The stasis message cache could conceivably also have that many messages legitimately on large systems. The only down side to not holding the message type ref in the stasis message is it only makes a crash either at shutdown or when manually unloading a busy module slightly more likely. However, this is more exposing a pre-existing stasis shutdown ordering issue than a problem with not holding a message type ref in stasis messages. * Made stasis messages and cache entries no longer hold a ref to the message type. Change-Id: Ibaa28efa8d8ad3836f0c65957192424c7f561707	7 years ago
Richard Mudgett	c008c27c85	pjproject: Update initial 2.8 patches to apply cleanly. ... ASTERISK-28059 Change-Id: I027472f2753391646dde594a709a75f14422db93	7 years ago
Joshua Colp	7912ad9bf7	Merge "res_pjsip_session: Don't add declined stream if one does not exist." into 16	7 years ago
George Joseph	40082f3e00	Merge "pjproject: Upgrade to 2.8." into 16	7 years ago
Richard Mudgett	192f71b7de	stasis_message.c: Don't create immutable stasis objects with locks. ... * Create the stasis message object without a lock as it is immutable. * Create the stasis message type object without a lock as it is immutable. * Creating the stasis message type could crash if the passed in type name is NULL and REF_DEBUG is enabled. Added missing NULL check when passing the ao2 object tag string. Change-Id: I28763c58bb9f0b427c11971d0103bf94055e7b32	7 years ago
Joshua Colp	60258b4ec1	pjproject: Upgrade to 2.8. ... This change brings in PJSIP 2.8, removes all the patches that were merged upstream, and makes a minor change to support a breaking change that was done. ASTERISK-28059 Change-Id: I5097772b11b0f95c3c1f52df6400158666f0a189	7 years ago
Florian Floimair	3e48c34f14	alembic: fix suppress_q850_reason_headers column name ... In the original commit introducing the feature the column in the alembic script was called 'suppress_q850_reason_header'. In the code however the option is called 'suppress_q850_reason_headers' (trailing 's'). This leads to errors when ARI push configuration is used. Change-Id: Ie84808adbca6fcc9136556e4f5d741adbef5d14f	7 years ago
George Joseph	29115e2384	app_voicemail: Remove need to subscribe to stasis ... app_voicemail was using the stasis cache to build and maintain a list of mailboxes that had subscribers. It then used this list to determine if a mailbox should be polled for new messages if polling was enabled. For this to work, stasis had to cache every subscription and unsubscription to the mailbox which caused a lot of overhead, both cpu and memory related. Since polling is only required when changes are being made to mailboxes outside of app_voicemail and since the number of mailboxes that don't have any subscribers is likely to be very low, all mailboxes are now polled instead of just the ones with subscribers. This paves the way for disabling the caching of stasis subscription change messages. Also fixed cleanup in some of the unit tests that not only left test users in the users list but also caused segfaults if the tests were run more than once. ASTERISK-27121 Change-Id: I5cceb737246949f9782955c64425b8bd25a9e9ee	7 years ago
Joshua Colp	6e79e6b097	res_pjsip_session: Don't add declined stream if one does not exist. ... Given a scenario where a session refresh was done with a removed stream we would always add a removed stream to the outgoing SDP even if one did not already exist. This change makes it so that a removed stream is only placed into the SDP if one already exists. ASTERISK-28047 Change-Id: Ibb97d21cdeb87a8acae0c720861b0ff255708442	7 years ago
Sean Bright	b0a0b975c5	autoconf: Check for srtp_get_version_string() before using it ... Change-Id: Id2a916ff9448706090e72ff2c7fb3f5ba24a05df	7 years ago
George Joseph	0107e1aa5a	Merge "res_srtp.c: Show linked version of libsrtp on module init" into 16	7 years ago
Jenkins2	4300410c9a	Merge "res_pjsip: Log IPv6 addresses correctly" into 16	7 years ago
George Joseph	4a309839eb	CI: Fix typo in testsuite git checkout ... Change-Id: I30024515e5b00a5044fd39fbff27d818f016b719	7 years ago
Sean Bright	55ca51af21	res_srtp.c: Show linked version of libsrtp on module init ... Change-Id: Ib0a645d6985de5757cc4399ed2524b2d02c4f342	7 years ago
Sean Bright	887a315e17	res_pjsip: Log IPv6 addresses correctly ... Both pjsip_tx_data.tp_info.dst_name and pjsip_rx_data.pkt_info.src_name store IPv6 addresses without enclosing brackets. This causes some log output to be confusing because it is difficult to separate the IPv6 address from a port specification. * Use pj_sockaddr_print() along with pjsip_tx_data.tp_info.dst_addr and pjsip_rx_data.pkt_info.src_addr where possible for consistent IPv6 output. * When a pj_sockaddr is not available, explicitly wrap IPv6 addresses in brackets. * When assigning pjsip_rx_data.pkt_info.src_name ourselves, make sure to also set pjsip_rx_data.pkt_info.src_addr. Change-Id: I5cfe997ced7883862a12b9c7d8551d76ae02fcf8	7 years ago
George Joseph	3f9544c1f5	CI: Use proper credentials for Security testsuite checkout ... Can't do anonymous http checkout from Security-testsuite. Need to use same credentials as the gerrit review checkout. Change-Id: I87af68c995cb8926f5e87f9af245600d76984f05	7 years ago
George Joseph	349355f1f1	Merge "res_musiconhold.c: Restart MOH if previous hold just reached end-of-file" into 16	7 years ago
George Joseph	17d6d9e1e7	stasis_cache: Stop caching stasis subscription change messages ... Since app_voicemail no longer uses the cache to maintain its state there is no longer a need to cache these messages. ASTERISK-27121 Change-Id: I321c708505f5ad8d00e1b0afc4c27dc2ac12ecb4	7 years ago
George Joseph	06d51a0408	Merge "optional_api: Remove unused nonoptreq fields" into 16	7 years ago
George Joseph	9db82309d5	Merge "CI: Use .gitreview to default BRANCH_NAME." into 16	7 years ago
Jenkins2	39829f0a78	Merge "res_pjproject: Fix sockaddr conversion routines for non-bundled PJSIP" into 16	7 years ago
Corey Farrell	5842741689	CI: Use .gitreview to default BRANCH_NAME. ... This ensures that binary modules are avoided in the master branch even if BRANCH_NAME is not set. Change-Id: I79162d2063f22fa9d6b31fde4827ace2dd5bf0da	7 years ago
Walter Doekes	78453e65fd	optional_api: Remove unused nonoptreq fields ... As they're not actively used, they only grow stale. The moduleinfo field itself is kept in Asterisk 13/15 for ABI compatibility. ASTERISK-28046 #close Change-Id: I8df66a7007f807840414bb348511a8c14c05a9fc	7 years ago
Joshua Colp	7ed02b4925	Merge "manager: Set AMI event "Newexten" to the EVENT_FLAG_DIALPLAN class" into 16	7 years ago
lvl	f4bffe2326	manager: Set AMI event "Newexten" to the EVENT_FLAG_DIALPLAN class ... The documentation already specified EVENT_FLAG_DIALPLAN for this event, but the implementation was using EVENT_FLAG_CALL. Using EVENT_FLAG_DIALPLAN allows AMI clients to opt out of receiving this highly verbose event. ASTERISK-28033 Change-Id: I45b3119f30e4dbc17b49831f2b1a4f2c1beadafe	7 years ago
Sean Bright	e5739c494c	res_pjproject: Fix sockaddr conversion routines for non-bundled PJSIP ... The bundled version of pjproject has a patch for Solaris compatability that changes the definition of various socket structures which we need to account for when compiling against a non-bundled version. ASTERISK-28049 #close Change-Id: Ia1ea47c433fc2d915115193ee889a752373925f0	7 years ago
Corey Farrell	ecb3b23b07	Build System: Resolve conflict between DESTDIR and bundled jansson. ... If Asterisk is built using a DESTDIR this will cause the bundled jansson to be installed to an unexpected location and we will fail to find it. Change-Id: Id033e2813261e0d45232383d44c6391122169548	7 years ago
Frederic LE FOLL	ccfd2e0f5d	res_musiconhold.c: Restart MOH if previous hold just reached end-of-file ... On MOH activation, moh_files_readframe() is called while the current stream attached to the channel is NULL and it calls ast_moh_files_next() immediately. However, it won't call ast_moh_files_next() again if sample reading fails. The failure may occur because res_musiconhold retains the last sample reading position in the channel data and MOH during the previous hold/retrieve just reached EOF. Obviously, a bit of bad luck is required here. * Restructured moh_files_readframe() to try a second time to start MOH if there was no stream setup and the saved position was at EOF. Also added comments describing what is going on for each step. ASTERISK-28029 Change-Id: I1508cf2c094f8feca22d6f76deaa9fdfa9944860	7 years ago
Jenkins2	c1a2c84361	Merge "core: Don't stop generators when writing RTCP frames." into 16	7 years ago
Joshua Colp	3c52cc32f1	Merge "stasis_cache: Prune stasis_subscription_change messages" into 16	7 years ago
Joshua Colp	6344cceed2	Merge "app_queue: Update realtime queuemembers after wait_a_bit(), not before" into 16	7 years ago
Joshua Colp	af6a3d02e1	core: Don't stop generators when writing RTCP frames. ... Generators provide such functionality as tone generation or silence generation. RTCP frames provide RTCP information and should not stop generators from operating. ASTERISK-28005 Change-Id: Ieadada07b068a7aa426e8763f1b73a18e1ac34a9	7 years ago
lvl	034a3d8b86	app_queue: Update realtime queuemembers after wait_a_bit(), not before ... This ensures the most up-to-date information is used for the next call attempt. ASTERISK-28032 Change-Id: I02fc17c6ffb50bb60ea97c2d2e6023e8061815ce	7 years ago
Sean Bright	3134fd95a9	res_pjproject: Add utility functions to convert between socket structures ... Currently, to convert from a pj_sockaddr to an ast_sockaddr, the address needs to be rendered to a string and then parsed into the correct structure. This also involves a call to getaddrinfo(3). The same is true for the inverse operation. Instead, because we know the internal structure of both ast_sockaddr and pj_sockaddr, we can translate directly between the two without the need for an intermediate string. Change-Id: If0fc4bba9643f755604c6ffbb0d7cc46020bc761	7 years ago