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Author	SHA1	Message	Date
George Joseph	195100e770	loader: Retry dlopen when loading fails ... Although we use the RTLD_LAZY flag when calling dlopen the first time on a module, this only defers resolution for function calls. Pointer references to functions are determined at link time so dlopen expects them to be there. Since we don't cross-module link, pointers to functions in other modules won't be available and dlopen will fail. Doing a "hardened" build also causes problems because it typically sets "-z now" on the ld command line which overrides RTLD_LAZY at run time. If the failing module isn't a GLOBAL_SYMBOLS module, then dlopen will be called again after all the GLOBAL_SYMBOLS modules have been loaded and they'll eventually resolve. If the calling module IS a GLOBAL_SYMBOLS module itself and a third module depends on it, then there's an issue because the second time through the dlopen loop, GLOBAL_SYMBOLS modules aren't given any special treatment and since the order in which dlopen is called isn't deterministic, the dependent may again be tried before the module it needs is loaded. Simple solution: Save modules that fail load_resource because of a dlopen error in a list and retry them immediately after the first pass. Keep retrying until the failed list is empty or we reach a #defined max retries. Error messages are suppressed until the final pass which also gets rid of those confusing error messages about module failures that are later corrected. Change-Id: Iddae1d97cd2f00b94e61662447432765755f64bb	10 years ago
zuul	6705f308cb	Merge "res_pjsip_dtmf_info: NULL terminate the message body."	10 years ago
Kevin Harwell	15c5743ac1	bridge.c: Crash during attended transfer when missing a local channel half ... It's possible for the transferer channel to get hung up early during the attended transfer process. For instance, a phone may send a "bye" immediately upon receiving a sip notify that contains a sip frag 100 (I'm looking at you Jitsi). When this occurs a race begins between the transferer being hung up and completion of the transfer code. If the channel hangs up too early during a transfer involving stasis bridging for instance, then when the created local channel goes to look up its swap channel (and associated datastore) it can't find it (since it is no longer in the bridge) thus it fails to enter the stasis application. Consequently, the created local channel(s) hang up as well. If the timing is just right then the bridging code attempts to add the message link with missing local channel(s). Hence the crash. Unfortunately, there is no great way to solve the problem of the unexpected "bye". While we can't guarantee we won't receive an early hangup, and in this case still fail to enter the stasis application, we can make it so asterisk does not crash. This patch does just that by locking the local channel structure, checking that the local channel's peer has not been lost, and then continuing. This keeps the local channel's peer from being ripped out from underneath it by the local/unreal hangup code while attempting to set the stasis message link. ASTERISK-25771 Change-Id: Ie6d6061e34c7c95f07116fffac9a09e5d225c880	10 years ago
Kevin Harwell	0d2ccbca62	res_pjsip_refer.c: Delay sending the initial SIP Notify with frag 100 ... During the transfer process, some phones (okay it was the Jitsi softphone, but maybe others are out there) send a "bye" immediately after receiving a SIP Notify. When a "bye" is received early for some types of transfers the transferer channel may no longer be available during late stage transfer processing. For instance, during an attended transfer involving stasis bridging at one point the created local channel looks for an associated swap channel in order to retrieve the stasis application name. If the transferer has hung up then the local channel will fail to find it. The local channel then has no way to know which stasis app to enter, so it fails and hangs up as well. Thus the transfer does not complete as expected. This patch delays the sending of the initial notify in order to give the transfer process enough time to gather the necessary data for a successful transfer. ASTERISK-25771 Change-Id: I09cfc9a5d6ed4c007bc70625e0972b470393bf16	10 years ago
zuul	7023055def	Merge "build-system: Allow building with static pjproject"	10 years ago
Joshua Colp	6af7fc4c37	res_pjsip_dtmf_info: NULL terminate the message body. ... PJSIP does not ensure that when printing the message body the buffer will be NULL terminated. This is problematic when searching for the signal and duration values of the DTMF. This change ensures the buffer is always NULL terminated. Change-Id: I52653a1a60c93092d06af31a27408d569cc98968	10 years ago
Joshua Colp	b78ec68c39	Merge "func_callerid.c: Update REDIRECTING reason documentation."	10 years ago
Joshua Colp	d7fe2becdd	Merge "SIP diversion: Fix REDIRECTING(reason) value inconsistencies."	10 years ago
Joshua Colp	8140d7a8ef	Merge "res_pjsip_send_to_voicemail.c: Allow either quoted or not send_to_vm reason."	10 years ago
zuul	71427f1454	Merge "res_pjsip_send_to_voicemail.c: Fix off-nominal double channel unref."	10 years ago
zuul	53f84a4670	Merge "CHAOS: cleanup possible null vars on msg alloc failure"	10 years ago
George Joseph	b8b7c2e428	alembic: Fix downgrade and tweak for sqlite ... Downgrade had a few issues. First there was an errant 'update' statement in add_auto_dtmf_mode that looks like it was a copy/paste error. Second, we weren't cleaning up the ENUMs so subsequent upgrades on postgres failed because the types already existed. For sqlite... sqlite doesn't support ALTER or DROP COLUMN directly. Fortunately alembic batch_operations takes care of this for us if we use it so the alter and drops were converted to use batch operations. Here's an example downgrade: with op.batch_alter_table('ps_endpoints') as batch_op: batch_op.drop_column('tos_audio') batch_op.drop_column('tos_video') batch_op.add_column(sa.Column('tos_audio', yesno_values)) batch_op.add_column(sa.Column('tos_video', yesno_values)) batch_op.drop_column('cos_audio') batch_op.drop_column('cos_video') batch_op.add_column(sa.Column('cos_audio', yesno_values)) batch_op.add_column(sa.Column('cos_video', yesno_values)) with op.batch_alter_table('ps_transports') as batch_op: batch_op.drop_column('tos') batch_op.add_column(sa.Column('tos', yesno_values)) # Can't cast integers to YESNO_VALUES, so dropping and adding is required batch_op.drop_column('cos') batch_op.add_column(sa.Column('cos', yesno_values)) Upgrades from base to head and downgrades from head to base were tested repeatedly for postgresql, mysql/mariadb, and sqlite3. Change-Id: I862b0739eb3fd45ec3412dcc13c2340e1b7baef8	10 years ago
George Joseph	7b71bca8a4	config_transport: Fix objects returned by ast_sip_get_transport_states ... ast_sip_get_transport_states was returning a container of internal_state objects instead of ast_sip_transport_state objects. This was causing transport lookups to fail, most noticably in res_pjsip_nat, which couldn't find the correct external addresses. This was causing contacts to go out with internal ip addresses. ASTERISK-25830 #close Reported-by: Sean Bright Change-Id: I1aee6a2fd46c42e8dd0af72498d17de459ac750e	10 years ago
Scott Griepentrog	0a3f0e85ac	CHAOS: cleanup possible null vars on msg alloc failure ... In message.c, if msg_alloc fails to init the string field, vars may be null, so use a null tolerant cleanup. In res_pjsip_messaging.c, if msg_data_create fails, mdata will be null, so use a null tolerant cleanup. ASTERISK-25323 Change-Id: Ic2d55c2c3750d5616e2a05ea92a19c717507ff56	10 years ago
Scott Griepentrog	60aa871be3	CHAOS: prevent crash on failed strdup ... This patch avoids crashing on a null pointer if the strdup() allocation fails. ASTERISK-25323 Change-Id: I3f67434820ba53b53663efd6cbb42749f4f6c0f5	10 years ago
Richard Mudgett	0bdbf0d882	func_callerid.c: Update REDIRECTING reason documentation. ... Change-Id: I6e8d39b0711110a4bceafa652e58b30465e28386	10 years ago
Richard Mudgett	25de01f301	SIP diversion: Fix REDIRECTING(reason) value inconsistencies. ... Previous chan_sip behavior: Before this patch chan_sip would always strip any quotes from an incoming reason and pass that value up as the REDIRECTING(reason). For an outgoing reason value, chan_sip would check the value against known values and quote any it didn't recognize. Incoming 480 response message reason text was just assigned to the REDIRECTING(reason). Previous chan_pjsip behavior: Before this patch chan_pjsip would always pass the incoming reason value up as the REDIRECTING(reason). For an outgoing reason value, chan_pjsip would send the reason value as passed down. With this patch: Both channel drivers match incoming reason values with values documented by REDIRECTING(reason) and values documented by RFC5806 regardless of whether they are quoted or not. RFC5806 values are mapped to the equivalent REDIRECTING(reason) documented value and is set in REDIRECTING(reason). e.g., an incoming RFC5806 'unconditional' value or a quoted string version ('"unconditional"') is converted to REDIRECTING(reason)'s 'cfu' value. The user's dialplan only needs to deal with 'cfu' instead of any of the aliases. The incoming 480 response reason text supported by chan_sip checks for known reason values and if not matched then puts quotes around the reason string and assigns that to REDIRECTING(reason). Both channel drivers send outgoing known REDIRECTING(reason) values as the unquoted RFC5806 equivalent. User custom values are either sent as is or with added quotes if SIP doesn't allow a character within the value as part of a RFC3261 Section 25.1 token. Note that there are still limitations on what characters can be put in a custom user value. e.g., embedding quotes in the middle of the reason string is silly and just going to cause you grief. * Setting a REDIRECTING(reason) value now recognizes RFC5806 aliases. e.g., Setting REDIRECTING(reason) to 'unconditional' is converted to the 'cfu' value. * Added missing malloc() NULL return check in res_pjsip_diversion.c set_redirecting_reason(). * Fixed potential read from a stale pointer in res_pjsip_diversion.c add_diversion_header(). The reason string needed to be copied into the tdata memory pool to ensure that the string would always be available. Otherwise, if the reason string returned by reason_code_to_str() was a user's reason string then the string could be freed later by another thread. Change-Id: Ifba83d23a195a9f64d55b9c681d2e62476b68a87	10 years ago
Richard Mudgett	8c8ef4efb0	res_pjsip_send_to_voicemail.c: Allow either quoted or not send_to_vm reason. ... Change-Id: Id6350b3c7d4ec8df7ec89863566645e2b0f441fd	10 years ago
Richard Mudgett	75ec137e91	res_pjsip_send_to_voicemail.c: Fix off-nominal double channel unref. ... * Fix double unref of other_party channel in off nominal path. * This is unlikely to be a real problem. However, for safety, in handle_incoming_request() keep the datastore ref with the other_party channel ref until we are finished with the other_party channel. Change-Id: I78f22547bf0bb99fb20814ceab75952bd857f821	10 years ago
George Joseph	3173e91bab	build-system: Allow building with static pjproject ... Background here: http://lists.digium.com/pipermail/asterisk-dev/2016-January/075266.html From CHANGES: * To help insure that Asterisk is compiled and run with the same known version of pjproject, a new option (--with-pjproject-bundled) has been added to ./configure. When specified, the version of pjproject specified in third-party/versions.mak will be downloaded and configured. When you make Asterisk, the build process will also automatically build pjproject and Asterisk will be statically linked to it. Once a particular version of pjproject is configured and built, it won't be configured or built again unless you run a 'make distclean'. To facilitate testing, when 'make install' is run, the pjsua and pjsystest utilities and the pjproject python bindings will be installed in ASTDATADIR/third-party/pjproject. The default behavior remains building with the shared pjproject installation, if any. Building: All you have to do is include the --with-pjproject-bundled option on the ./configure command line (and remove any existing --with-pjproject option if specified). Everything else is automatic. Behind the scenes: The top-level Makefile was modified to include 'third-party' in the list of MOD_SUBDIRS. The third-party directory was created to contain any third party packages that may be needed in the future. Its Makefile automatically iterates over any subdirectories passing on targets. The third-party/pjproject directory was created to house the pjproject source distribution. Its Makefile contains targets to download, patch configure, generate dependencies, compile libs, apps and python bindings, sanitized build.mak and generate a symbols list. When bootstrap.sh is run, it automatically includes the configure.m4 file in third-party/pjproject. This file has a macro to download and conifgure pjproject and get and set PJPROJECT_INCLUDE, PJPROJECT_DIR and PJPROJECT_BUNDLED. It also tests for the capabilities like PJ_TRANSACTION_GRP_LOCK by parsing preprocessor output as opposed to trying to compile. Of course, bootstrap.sh is only run once and the configure file is incldued in the patch. When configure is run with the new options, the macro in configure.m4 triggers the download, patch, conifgure and tests. No compilation is performed at this time. The downloaded tarball is cached in /tmp so it doesn't get downloaded again on a distclean. When make is run in the top-level Asterisk source directory, it will automatically descend all the subdirectories in third_party just as it does for addons, apps, etc. The top-level Makefile makes sure that the 'third-party' is built before 'main' so that dependencies from the other directories are built first. When main does build, a new shared library (libasteriskpj) is created that links statically to the pjproject .a files and exports all their symbols. The asterisk binary links to that, just as it does with libasteriskssl. When Asterisk is installed, the pjsua and pjsystest apps, and the pjproject python bindings are installed in ASTDATADIR/third-party/pjproject. This will facilitate testing, including running the testsuite which will be updated to check that directory for the pjsua module ahead of the system python library. Modules should continue to depend on pjproject if they use pjproject APIs directly. They should not care about the implementation. No changes to any res_pjsip modules were made. Change-Id: Ia7a60c28c2e9ba9537c5570f933c1ebcb20a3103	10 years ago
Joshua Colp	d0b26c3133	Merge "chan_sip.c: Fix T.38 issues caused by leaving a bridge."	10 years ago
Joshua Colp	91f8763452	Merge "res_pjsip_t38.c: Back out part of an earlier fix attempt."	10 years ago
Joshua Colp	b075549286	Merge "bridge core: Add owed T.38 terminate when channel leaves a bridge."	10 years ago
Joshua Colp	aadc58a1e7	Merge "channel api: Create is_t38_active accessor functions."	10 years ago
Joshua Colp	916cc68585	Merge "bridge_channel: Don't settle owed events on an optimization."	10 years ago
Joshua Colp	806e4a664f	Merge "channel.c: Route all control frames to a channel through the same code."	10 years ago
zuul	c47d15fda7	Merge "res_pjsip_mwi: Turn some NOTICEs and WARNINGs into debug 1s."	10 years ago
Richard Mudgett	2dae4a1ccf	chan_sip.c: Fix T.38 issues caused by leaving a bridge. ... chan_sip could not handle AST_T38_TERMINATED frames being sent to it when the channel left the bridge. The action resulted in overlapping outgoing reINVITEs. The testsuite tests/fax/sip/directmedia_reinvite_t38 was not happy. * Force T.38 to be remembered as locally bridged. Now when the channel leaves the native RTP bridge after T.38, the channel remembers that it has already reINVITEed the media back to Asterisk. It just needs to terminate T.38 when the AST_T38_TERMINATED arrives. * Prevent redundant AST_T38_TERMINATED from causing problems. Redundant AST_T38_TERMINATED frames could cause overlapping outgoing reINVITEs if they happen before the T.38 state changes to disabled. Now the T.38 state is set to disabled before the reINVITE is sent. ASTERISK-25582 #close Change-Id: I53f5c6ce7d90b3f322a942af1a9bcab6d967b7ce	10 years ago
Richard Mudgett	bf29a4e2e6	res_pjsip_t38.c: Back out part of an earlier fix attempt. ... This backs out item 4 of the 4875e5ac32 commit. Item 4 added the t38_bye_supplement. Unfortunately, the frame that it puts into the bridge may or may not be processed by the time the bridged peer is kicked out of the bridge. If it is processed then all is well. However, if it is not processed then that channel is stuck in fax mode until it hangs up or maybe if it joins another bridge for T.38 faxing. ASTERISK-25582 Change-Id: Ib20a03ecadf1bf8a0dcadfadf6c2f2e60919a9f7	10 years ago
Richard Mudgett	c7d45b84f9	bridge core: Add owed T.38 terminate when channel leaves a bridge. ... The channel is now going to get T.38 terminated when it leaves the bridging system and the bridged peers are going to get T.38 terminated as well. ASTERISK-25582 Change-Id: I77a9205979910210e3068e1ddff400dbf35c4ca7	10 years ago
Richard Mudgett	0e296563d7	channel api: Create is_t38_active accessor functions. ... ASTERISK-25582 Change-Id: I69451920b122de7ee18d15bb231c80ea7067a22b	10 years ago
Richard Mudgett	86f7336c91	bridge_channel: Don't settle owed events on an optimization. ... Local channel optimization could cause DTMF digits to be duplicated. Pending DTMF end events would be posted to a bridge when the local channel optimizes out and is replaced by the channel further down the chain. When the real digit ends, the channel would get another DTMF end posted to the bridge. A -- LocalA;1/n -- LocalA;2/n -- LocalB;1 -- LocalB;2 -- B 1) LocalA has the /n flag to prevent optimization. 2) B is sending DTMF to A through the local channel chain. 3) When LocalB optimizes out it can move B to the position of LocalB;1 4) Without this patch, when B swaps with LocalB;1 then LocalB;1 would settle an owed DTMF end to the bridge toward LocalA;2. 5) When B finally ends its DTMF it sends the DTMF end down the chain. 6) Without this patch, A would hear the DTMF digit end when LocalB optimizes out and when B ends the original digit. ASTERISK-25582 Change-Id: I1bbd28b8b399c0fb54985a5747f330a4cd2aa251	10 years ago
Richard Mudgett	128c96456c	channel.c: Route all control frames to a channel through the same code. ... Frame hooks can conceivably return a control frame in exchange for an audio frame inside ast_write(). Those returned control frames were not handled quite the same as if they were sent to ast_indicate(). Now it doesn't matter if you use ast_write() to send an AST_FRAME_CONTROL to a channel or ast_indicate(). ASTERISK-25582 Change-Id: I5775f41421aca2b510128198e9b827bf9169629b	10 years ago
George Joseph	4422905218	sorcery: Refactor create, update and delete to better deal with caches ... The ast_sorcery_create, update and delete function have been refactored to better deal with caches and errors. The action is now called on all non-caching wizards first. If ANY succeed, the action is called on all caching wizards and the observers are notified. This way we don't put something in the cache (or update or delete) before knowing the action was performed in at least 1 backend and we only call the observers once even if there were multiple writable backends. ast_sorcery_create was never adding to caches in the first place which was preventing contacts from getting added to a memory_cache when they were created. In turn this was causing memory_cache to emit errors if the contact was deleted before being retrieved (which would have populated the cache). ASTERISK-25811 #close Reported-by: Ross Beer Change-Id: Id5596ce691685a79886e57b0865888458d6e7b46	10 years ago
George Joseph	acf329a3c7	res_pjsip_mwi: Turn some NOTICEs and WARNINGs into debug 1s. ... There are a few cases where we're emitting notices or warnings for things that really need neither, like a client retrying to subscribe to mwi when they're not conifgured for it. They get a 404 so there's no need for non-debug messages. Change-Id: I05e38a7ff6c2f2521146f4be6a79731b9864e61f	10 years ago
Joshua Colp	62d98b5a7f	Merge "res_pjsip/config_transport: Allow reloading transports."	10 years ago
Joshua Colp	a3c33a99c5	Merge "res_sorcery_memory_cache: Fix SEGV in some CLI commands"	10 years ago
zuul	170941990b	Merge "chan_sip: Optionally supply fromuser/fromdomain in SIP dial string."	10 years ago
George Joseph	7e3e1ddf7e	res_sorcery_memory_cache: Fix SEGV in some CLI commands ... A few of the CLI commands weren't checking for enough arguments and were SEGVing. Change-Id: Ie6494132ad2fe54b4f014bcdc112a37c36a9b413	10 years ago
zuul	ae8dbcf8b8	Merge "chan_sip.c: Suppress T.38 SDP c= line if addr is the same."	10 years ago
zuul	d6b2c1d98e	Merge "res_config_sqlite3: Fix crashes when reading peers from sqlite3 tables"	10 years ago
zuul	6f080f15d6	Merge "rtp_engine.h: Remove extraneous semicolons."	10 years ago
zuul	062857ece3	Merge "res_pjsip_config_wizard: Add command to export primitive objects"	10 years ago
Richard Mudgett	803a2fc2d5	rtp_engine.h: Remove extraneous semicolons. ... Change-Id: Ib462633d396fa941379dfef648dcd2245e350084	10 years ago
Richard Mudgett	886ee09471	chan_sip.c: Suppress T.38 SDP c= line if addr is the same. ... Use the correct comparison function since we only care if the address without the port is the same. Change-Id: Ibf6c485f843a1be6dee58a47b33d81a7a8cbe3b0	10 years ago
Joshua Colp	def3fb4634	Merge "res_pjproject: Add ability to map pjproject log levels to Asterisk log levels"	10 years ago
Christof Lauber	b7970cabfa	res_config_sqlite3: Fix crashes when reading peers from sqlite3 tables ... Introduced realloaction of ast_str buf in sqlite3_escape functions in case the returned buffer from threadstorage was actually too small. Change-Id: I3c5eb43aaade93ee457943daddc651781954c445	10 years ago
George Joseph	ba8adb4ce3	res_pjsip/config_transport: Allow reloading transports. ... The 'reload' mechanism actually involves closing the underlying socket and calling the appropriate udp, tcp or tls start functions again. Only outbound_registration, pubsub and session needed work to reset the transport before sending requests to insure that the pjsip transport didn't get pulled out from under them. In my testing, no calls were dropped when a transport was changed for any of the 3 transport types even if ip addresses or ports were changed. To be on the safe side however, a new transport option was added (allow_reload) which defaults to 'no'. Unless it's explicitly set to 'yes' for a transport, changes to that transport will be ignored on a reload of res_pjsip. This should preserve the current behavior. Change-Id: I5e759850e25958117d4c02f62ceb7244d7ec9edf	10 years ago
Walter Doekes	c00082329e	chan_sip: Optionally supply fromuser/fromdomain in SIP dial string. ... Previously you could add [!dnid] to the SIP dial string to alter the To: header. This change allows you to alter the From header as well. SIP dial string extra options now look like this: [![touser[@todomain]][![fromuser][@fromdomain]]] INCOMPATIBLE CHANGE: If you were using an exclamation mark in your To: header, that is no longer possible. ASTERISK-25803 #close Change-Id: I2457e9ba7a89eb1da22084bab5a4d4328e189db7	10 years ago
zuul	6fc57b3e1f	Merge "Fix failing threadpool_auto_increment test."	10 years ago