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Author	SHA1	Message	Date
George Joseph	0dc0356e39	pjproject_bundled: Add MALLOC_DEBUG capability ... pjproject_bundled will now use the asterisk memory debugging APIs if MALLOC_DEBUG is turned on in menuselect. Because this required stubs for the executable programs and the python bindings, some Makefile reorganization was needed to properly handle the dependencies. As a result, the makefile now individually makes each of the pjproject libraries separately instead of making them all in 1 shot. The only visible change is that there are separate status lines printed for each library instead oif 1 for all libs. Also, the making of the pjproject dependency files was eliminated. They're not needed for building unless you're actively modifying pjproject source files and it makes the build process faster. Finally, any issues with parallel builds should be resolved again making the build faster. Change-Id: Icc5e3d658fbfb00e0a46b44c66dcc2522d5171b0	9 years ago
Corey Farrell	dd873bcada	astobj2: Add backtrace to log_bad_ao2. ... * Compile __ast_assert_failed unconditionally. * Use __ast_assert_failed to log messages from log_bad_ao2 * Remove calls to ast_assert(0) that happen after log_bad_ao2 was run. Change-Id: I48f1af44b2718ad74a421ff75cb6397b924a9751	9 years ago
George Joseph	86550f9c17	alembic: Allow cdr, config and voicemail to exist in the same schema ... cdr, config and voicemail are all separate alembic trees. Because alembic's default is to use a table named 'alembic_version' to store the current tree revision, the 3 trees can't exist in the same schema without stepping on each other. Now each tree uses 'alembic_version_<tree_name>' as the version table. Each tree's env.py script now first checks for 'alembic_version'. If it finds it AND its revision is in the tree's history, the script renames it to 'alembic_version_<tree_name>'. Regardless, the script then continues with the migration using 'alembic_version_<tree_name>' and creates that table if it's not found. The result is that if an existing 'alembic_version' table was found but it didn't belong to this tree, it's left alone and 'alembic_version_<tree_name>' is used or created. WARNING: If multiple trees are using the same schema, they MUST NOT CRU or D any objects with names that might exist in the other trees. An example would be 'yesno_values' type. If two trees perform operations on it, one tree could pull it out from under the other. Thankfully we currently don't share any names among cdr, config and voicemail. NOTE: Since the env.py scripts in each tree were identical, a common env.py has been placed in the ast-db-manage directory and a symlink to it has been placed in each tree directory. ASTERISK-24311 #close Reported-by: Dafi Ni Change-Id: I4d593f000350deb5d21a14fa1e9bc3896844d898	9 years ago
Alexander Traud	f166681c12	chan_sip: Honor support of Symmetric Response (rport) for SIP requests. ... In the SIP channel driver chan_sip, the default is "auto_force_rport". When no NAT was detected, for example in case of IPv6, Asterisk uses the IP address from the headers within the SIP-REGISTER for subsequent SIP signaling. When the remote party specifies support for Symmetric Response (RFC 3581) via the parameter "rport", Asterisk should not extract the port from the SIP headers but reuse the port of the transport. This did not happen because of a typo. ASTERISK-26438 #close Change-Id: If6e7891848aaf96666dee5305695f7c6667cd5a6	9 years ago
Michael Walton	430f6e5388	audiohooks: Remove redundant codec translations when using audiohooks ... The main frame read and write handlers in main/channel.c don't use the optimum placement in the processing flow for calling audiohooks callbacks, as far as codec translation is concerned. This change places the audiohooks callback code: * After the channel read translation if the frame is not linear before the translation, thereby increasing the chance that the frame is linear as required by audiohooks * Before the channel write translation if the frame is linear at this point This prevents the audiohooks code from instantiating additional translation paths to/from linear where a linear frame format is already available, saving valuable CPU cycles ASTERISK-26419 Change-Id: I6edd5771f0740e758e7eb42558b953f046c01f8f	9 years ago
Joshua Colp	b0b106a473	Merge "Remove "format_ogg_opus: New format"" into 13	9 years ago
Kevin Harwell	2449d2877c	Remove "format_ogg_opus: New format" ... This reverts commit 40aa28131b. ASTERISK-26426 #close Change-Id: I81e55c3c512f1dd6f49896f0c6b97a07d74fd8f5	9 years ago
George Joseph	f0a2e628d6	download_externals: Fix issue with re-install ... Needed to ignore an xmlstarlet return code for optional element. Change-Id: I6a96f709b4b38c9a3f3dda4e8b07903787e16873 Reported-by: Dan Jenkins	9 years ago
George Joseph	1e8b3d00ce	Merge "codec_opus: Add download ability to menuselect" into 13	9 years ago
George Joseph	269ee510c9	Merge "codec_opus: Replace res_format_attr_opus with the one from codec_opus" into 13	9 years ago
George Joseph	0674f319b9	Merge "format_ogg_opus: New format" into 13	9 years ago
zuul	1497e29c4f	Merge "chan_sip: Resolve externhost not to IPv6; instead go for IPv4." into 13	9 years ago
George Joseph	5258c067ae	codec_opus: Add download ability to menuselect ... Updated codecs/codecs.xml to add codec_opus to the external download list. ASTERISK-26409 Change-Id: Ia07b36539f30e852125fb2b94147dc9774df31a4	9 years ago
George Joseph	a5af8709c8	codec_opus: Replace res_format_attr_opus with the one from codec_opus ... Preparation ASTERISK-26409 Change-Id: I9f20e7cce00c32464d9a180e81283d49d199d0a3 (cherry picked from commit 59f7662a93)	9 years ago
George Joseph	44c0c51cf1	format_ogg_opus: New format ... Add Ogg/Opus playback support. This uses libopusfile in order to be able to read .opus files and play them back. Writing/recording support is not present at this time. ASTERISK-26409 Change-Id: I8815d23345108d8ca7c0bd640f6a1ce6b4f56955	9 years ago
George Joseph	0ab443007b	build_tools: Add ability to download variants to download_externals ... Some external packages have multiple variants that apply to different builds of asterisk. The DPMA for instance has a "bundled" variant that needs to be downloaded if asterisk was configured with --with-pjproject-bundled. There are 2 ways to specify variants: If you need the user to make the decision about which variant to download, simply create multiple menuselect "member" entries like so... <member name="res_digium_phone" displayname="..snipped.."> <support_level>external</support_level> <depend>xmlstarlet</depend> <depend>bash</depend> <defaultenabled>no</defaultenabled> </member> <member name="res_digium_phone-bundled" displayname="..snipped.."> <support_level>external</support_level> <depend>xmlstarlet</depend> <depend>bash</depend> <defaultenabled>no</defaultenabled> </member> Note that the second entry has "-<variant>" appended to the name. You can then use the existing menuselect facilities to restrict which members to enable or disable. Youy probably don't want the user to enable multiple at the same time. If you want to hide the details of the variants, the better way to do it is to create 1 member with "variant" elements. <member name="res_digium_phone" displayname="..snipped.."> <support_level>external</support_level> <depend>xmlstarlet</depend> <depend>bash</depend> <defaultenabled>no</defaultenabled> <member_data> <downloader> <variants> <variant tag="bundled" condition='[[ "$PJPROJECT_BUNDLED" = "yes" ]]'/> </variants> </downloader> </member_data> </member> The condition must be a bash expression suitable for use with an "if" statement. Any environment variable can be used plus those available in makeopts. In this case, if asterisk was configured with --with-pjproject-bundled the bundled variant will be automatically downloaded. Otherwise the normal version will be downloaded. Change-Id: I4de23e06d4492b0a65e105c8369966547d0faa3e	9 years ago
zuul	9b0e6f9c86	Merge "channels/chan_pjsip: fix HANGUPCAUSE function bug." into 13	9 years ago
zuul	52e3c6c2e0	Merge "chan_sip: Address runaway when realtime peers subscribe to mailboxes" into 13	9 years ago
Aaron An	a0a17a8c6f	channels/chan_pjsip: fix HANGUPCAUSE function bug. ... HANGUPCAUSE not return 'SIP 200 Ok' when dialed channel answered. This patch change the call order of ast_queue_control_data and ast_queue_control in chan_pjsip_incoming_response. ASTERISK-26396 #close Reported by: AaronAn Tested by: AaronAn Change-Id: Ide2d31723d8d425961e985de7de625694580be61	9 years ago
Alexander Traud	0502675e5c	chan_sip: Resolve externhost not to IPv6; instead go for IPv4. ... For the channel driver chan_sip, you specify externhost=example.com in sip.conf when your Asterisk is behind a NAT and your IP address is assigned dynamically. Or stated differently: You do not have a static IP address to use "externaddr" directly. This NAT support is quite handy but just about IPv4. Previously, Asterisk resolved "externhost" to any IP version. When the first DNS answer resolved to an IPv6, Asterisk sent an IPv6 in SIP/SDP for origin (o=) and connection (c=). This happened in outgoing SIP-REGISTER and while answering SIP-INVITE. If the remote peer is IPv4-only, it might not handle o=/c= with an IPv6. This change makes sure, no IPv6 is resolved anymore for "externhost". ASTERISK-18232 #close Reported by: Jacek Kowalski Tested by: Alexander Traud patches: changes.patch submitted by Alessandro Crespi Change-Id: If68eedbeff65bd1c1d8a9ed921c02ba464b32dac	9 years ago
George Joseph	0056bcaebd	chan_sip: Address runaway when realtime peers subscribe to mailboxes ... Users upgrading from asterisk 13.5 to a later version and who use realtime with peers that have mailboxes were experiencing runaway situations that manifested as a continuous stream of taskprocessor congestion errors, memory leaks and an unresponsive chan_sip. A related issue was that setting rtcachefriends=no NEVER worked in asterisk 13 (since the move to stasis). In 13.5 and earlier, when a peer tried to register, all of the stasis threads would block and chan_sip would again become unresponsive. After 13.5, the runaway would happen. There were a number of causes... * mwi_event_cb was (indirectly) calling build_peer even though calls to mwi_event_cb are often caused by build_peer. * In an effort to prevent chan_sip from being unloaded while messages were still in flight, destroy_mailboxes was calling stasis_unsubscribe_and_join but in some cases waited forever for the final message. * add_peer_mailboxes wasn't properly marking the existing mailboxes on a peer as "keep" so build_peer would always delete them all. * add_peer_mwi_subs was unsubscribing existing mailbox subscriptions then just creating them again. All of this was causing a flood of subscribes and unsubscribes on multiple threads all for the same peer and mailbox. Fixes... * add_peer_mailboxes now marks mailboxes correctly and build_peer only deletes the ones that really are no longer needed by the peer. * add_peer_mwi_subs now only adds subscriptions marked as "new" instead of unsubscribing and resubscribing everything. It also adds the peer object's address to the mailbox instead of its name to the subscription userdata so mwi_event_cb doesn't have to call build_peer. With these changes, with rtcachefriends=yes (the most common setting), there are no leaks, locks, loops or crashes at shutdown. rtcachefriends=no still causes leaks but at least it doesn't lock, loop or crash. Since making rtcachefriends=no work wasnt in scope for this issue, further work will have to be deferred to a separate patch. Side fixes... * The ast_lock_track structure had a member named "thread" which gdb doesn't like since it conflicts with it's "thread" command. That member was renamed to "thread_id". ASTERISK-25468 #close Change-Id: I07519ef7f092629e1e844f855abd279d6475cdd0	9 years ago
zuul	7cce1a7817	Merge "core: Ensure presencestate subtype and message are NULL." into 13	9 years ago
Joshua Colp	d7587a67eb	Merge "res_odbc: Make pooling option deprecation notice more useful." into 13	9 years ago
zuul	94b705f9fb	Merge "cdr_mysql: fix UTC support" into 13	9 years ago
Joshua Colp	323aff3a09	core: Ensure presencestate subtype and message are NULL. ... When retrieving presence state information there is no guarantee that the subtype and message passed in are set to NULL. This change ensures they are. ASTERISK-26397 #close Change-Id: I61f8187972d5d8bbd7d6b7f4daa4f4f7e8237b23	9 years ago
zuul	81bb672861	Merge "logger: Fix default console settings." into 13	9 years ago
Joshua Colp	10c180760c	res_odbc: Make pooling option deprecation notice more useful. ... This changes the notice for the deprecation of the old pooling options to point to the new option for doing pooling. This gives a clearer direction as to what to look into. ASTERISK-26389 #close Change-Id: I2ca9cdfdcd75aec170a7db9d5ff69a4cd25b7c10	9 years ago
zuul	5cb3fc5d67	Merge "core: Fix LOW_MEMORY missing symbol ast_pbx_uuid_get." into 13	9 years ago
Tzafrir Cohen	42cc267016	cdr_mysql: fix UTC support ... * Make 'cdrzone=UTC' work properly. * Fix the documentation of cdr_mysql.conf: it's cdrzone and not timezone ASTERISK-26359 #close Change-Id: I2a6f67b71bbbe77cac31a34d0bbfb1d67c933778	9 years ago
Joshua Colp	f16ab19292	odbc: Remove options that are no longer applicable. ... The pooling, shared_connection, limit, and idlecheck options are no longer used in res_odbc. ASTERISK-26389 Change-Id: I2fde7b467d01f9d1c82cc0a339bb4f7e1dd6bbe6	9 years ago
zuul	a6b05e6371	Merge "asterisk.c: Non-root users also get the astcanary after core restart." into 13	9 years ago
Corey Farrell	c9ce299b64	core: Fix LOW_MEMORY missing symbol ast_pbx_uuid_get. ... Move the function outside the conditional block that excludes LOW_MEMORY. ASTERISK-26273 #close Change-Id: Ic290fa128222c410c3531107e30efacabc8493b4	9 years ago
Joshua Colp	77fafd1534	Merge "sd_notify (systemd status notifications) support" into 13	9 years ago
zuul	177557bc00	Merge "res_pjsip_multihomed: Change Contact port to listening port." into 13	9 years ago
Corey Farrell	610eb4c189	logger: Fix default console settings. ... When logger.conf is missing or invalid we should be printing notices, warnings and errors to the console. The logmask was incorrectly calculated. Change-Id: Ibaa9465a8682854bc1a5e9ba07079bea1bfb6bb3	9 years ago
Tzafrir Cohen	36092ee3a0	sd_notify (systemd status notifications) support ... sd_notify() is used to notify systemd of changes to the status of the process. This allows the systemd daemon to know when the process finished loading (and thus only start another program after Asterisk has finished loading). To use this, use a systemd unit with 'Type=notify' for Asterisk. This commit also adds the function ast_sd_notify(), a wrapper around sd_notify that does nothing if not built with systemd support. Also adds support for libsystemd detection in the configure script. Change-Id: Ied6a59dafd5ef331c5c7ae8f3ccd2dfc94be7811 (cherry picked from commit 07b95f7c65)	9 years ago
Walter Doekes	9372d32100	asterisk.c: Non-root users also get the astcanary after core restart. ... Without this change, a 'core restart' would kill the astcanary forever if you're not running as root. Both with and without this patch, the scheduling priority was still SCHED_RR after restart. Additionally, the astcanary is now spawned if you start with high priority and Asterisk doesn't get a chance to lower it. For example through: `chrt -r 10 sudo -u asterisk asterisk -c` Also reap killed astcanary processes on core restart. ASTERISK-26352 #close Change-Id: Iacb49f26491a0717084ad46ed96b0bea5f627a55	9 years ago
zuul	34461b89ac	Merge "asterisk.c: When astcanary dies on linux, reset priority on all threads." into 13	9 years ago
zuul	0bd4398b8a	Merge "Fix showing of swap details when sysinfo() is available" into 13	9 years ago
zuul	9383beb073	Merge "res_config_odbc.c: Fix buffer size limitation creating invalid SQL." into 13	9 years ago
Walter Doekes	e96448e991	asterisk.c: When astcanary dies on linux, reset priority on all threads. ... Previously only the canary checking thread itself had its priority set to SCHED_OTHER. Now all threads are traversed and adjusted. ASTERISK-19867 #close Reported by: Xavier Hienne Change-Id: Ie0dd02a3ec42f66a78303e9c1aac28f7ed9aae39	9 years ago
Timo Teräs	01884a7af6	Fix showing of swap details when sysinfo() is available ... If sysinfo() is available, but not sysctl() or swapctl() the printing code for swap buffer sizes is incorrectly omitted. The above condition happens with musl c-library. Fix #if rule to consider defined(HAVE_SYSINFO). And also remove the redundant || defined(HAVE_SYSCTL) which was incorrectly there to start with. Now swap information is displayed only if an actual libc function to get it is available. This also fixes warnings previously seen with musl libc: [CC] asterisk.c -> asterisk.o asterisk.c: In function 'handle_show_sysinfo': asterisk.c:773:6: warning: variable 'totalswap' set but not used [-Wunused-but-set-variable] int totalswap = 0; ^~~~~~~~~ asterisk.c:770:11: warning: variable 'freeswap' set but not used [-Wunused-but-set-variable] uint64_t freeswap = 0; ^~~~~~~~ Change-Id: I1fb21dad8f27e416c60f138c6f2bff03fb626eca	9 years ago
Richard Mudgett	cdbad152c7	res_config_odbc.c: Fix buffer size limitation creating invalid SQL. ... Creating ODBC SQL queries resulted in queries too large to fit into the supplied buffer. The resulting truncated buffer contained an invalid SQL query. * Made SQL query generation code use a thread storage buffer that can increase in size as needed. * Fixed bad multi-line warning messages. ASTERISK-26263 #close Reported by: Jeppe Ryskov Larsen Change-Id: I23f3cdd43c2dac80bed3ded4dd77d18cb17f21ae	9 years ago
Joshua Colp	449719be00	res_pjsip_multihomed: Change Contact port to listening port. ... The res_pjsip_multihomed module determines what interface and transport a request is going out on and updates the SIP message accordingly with the address information. This currently incorrectly updates the Contact header for connectionful protocols to the ephemeral connection port, instead of the bound address for the listening socket which can actually accept the connection back. If the remote side attempts to connect back on the epehemeral port it will fail. This change makes it so the port is updated to the bound port on connectionful protocols and is maintained on UDP (as there can be multiple of those). ASTERISK-26374 #close Change-Id: I50f8dab65b9f75117d73ba5f6bbcf6c9871854ab	9 years ago
George Joseph	4d64b176eb	pjproject_bundled: Prevent SERVFAIL from marking name server bad ... A name server that returns "Server Failure" is indicating only that the server couldn't process that particular request. We should NOT assume that the name server is incapable of serving other requests. Here's the scenario we've been encountering... * 2 local name servers configured in resolv.conf. * An OPTIONS request causes a request for A and AAAA records to go out to both nameservers. * The A responses both come back successfully resolved. * Because of an issue at some upstream nameserver, the AAAA responses for that particular query come back as "SERVFAIL" from both local name servers. * Both local servers are marked as bad and no further queries can be sent until the 60 second ttl expires. Only previously cached results can be used. * In this case, 60 seconds is just enough time for another OPTIONS request to go out to the same host so the cycle repeats. We could set the bad ttl really low but that also affects REFUSED and NOTAUTH which probably DO signal a real server issue. Besides, even a really low bad ttl would be an issue on a pbx. Although we use our own resolver in 14 and master and don't have this issue there, Teluu has merged this patch upstream so it's appropriate to cherry-pick to 14 and master to keep pjproject consistent. Change-Id: Ie03ba902288e274aff23f9b9bb2786e1e8be09e0	9 years ago
zuul	11d05fc369	Merge "res_pjsip_transport_management: Convert time in log message to seconds." into 13	9 years ago
zuul	1ddaa825ec	Merge "chan_sip: Fix session timeout on retransmit of non-UDP packets" into 13	9 years ago
zuul	f0baa12538	Merge "rtp: Preserve timestamps on video frames." into 13	9 years ago
zuul	139d86c8ab	Merge "sip_to_pjsip.py: Map legacy_useroption_parsing." into 13	9 years ago
Joshua Colp	1cac856e17	rtp: Preserve timestamps on video frames. ... Currently when receiving video over RTP we store only a calculated samples on the frame. When starting the video it can take some time for this calculation to actually yield a value as it requires constant changing timestamps. As well if a video frame passes over multiple RTP packets this calculation will fail as the timestamp is the same as the previous RTP packet and the number of samples calculated will be 0. This change preserves the timestamp on the frame and allows it to pass through the core. When sending the video this timestamp is used instead of a new one being calculated. ASTERISK-26367 #close Change-Id: Iba8179fb5c14c9443aee4baf670d2185da3ecfbd	9 years ago