3307 Commits (601602f44b72c20a265d39f217eeda3d335712e9)

Author	SHA1	Message	Date
Alexei Gradinari	c4426f1035	res_pjsip: disable multi domain to improve realtime performace ... This patch added new global pjsip option 'disable_multi_domain'. Disabling Multi Domain can improve Realtime performance by reducing number of database requests. ASTERISK-25930 #close Change-Id: I2e7160f3aae68475d52742107949a799aa2c7dc7	9 years ago
Richard Mudgett	8dd79720e6	bridge: Hold off more than one imparting channel at a time. ... An earlier patch blocked the ast_bridge_impart() call until the channel either entered the target bridge or it failed. Unfortuantely, if the target bridge is stasis and the imprted channel is not a stasis channel, stasis bounces the channel out of the bridge to come back into the bridge as a proper stasis channel. When the channel is bounced out, that released the block on ast_bridge_impart() to continue. If the impart was a result of a transfer, then it became a race to see if the swap channel would get hung up before the imparted channel could come back into the stasis bridge. If the imparted channel won then everything is fine. If the swap channel gets hung up first then the transfer will fail because the swap channel is leaving the bridge. * Allow a chain of ast_bridge_impart()'s to happen before any are unblocked to prevent the race condition described above. When the channel finally joins the bridge or completely fails to join the bridge then the ast_bridge_impart() instances are unblocked. ASTERISK-25947 Reported by: Richard Mudgett ASTERISK-24649 Reported by: John Bigelow ASTERISK-24782 Reported by: John Bigelow Change-Id: I8fef369171f295f580024ab4971e95c799d0dde1	9 years ago
zuul	dcebcaa3da	Merge "build: Add configure check for proto field of PJSIP TLS transport setting." into 13	9 years ago
Joshua Colp	677a65fcbb	build: Add configure check for proto field of PJSIP TLS transport setting. ... Older versions of PJSIP do not have the proto field on the TLS transport setting structure. This change adds a configure check so even if it is not present we will still be able to build. Change-Id: Ibf3f47befb91ed1b8194bf63888baa6fee05aba9	9 years ago
Walter Doekes	336cae73cc	app_chanspy: Fix occasional deadlock with ChanSpy and Local channels. ... Channel masquerading had a conflict with autochannel locking. When locking autochannel->channel, the channel is fetched from the autochannel and then locked. During the fetch, the autochannel -- which has no locks itself -- can be modified by someone who owns the channel lock. That means that the value of autochan->channel cannot be trusted until you hold the lock. In practice, this caused problems with Local channels getting masqueraded away while the ChanSpy attempted to get info from that channel. The old channel which was about to get removed got locked, but the new (replaced) channel got unlocked (no-op). Because the replaced channel was now locked (and would never get unlocked), it couldn't get removed from the channel list in a timely manner, and would now cause deadlocks when iterating over the channel list. This change checks the autochannel after locking the channel for changes to the autochannel. If the channel had been changed, the lock is reobtained on the new channel. In theory it seems possible that after this fix, the lock attempt on the old (wrong) channel can be on an already destroyed lock, maybe causing a crash. But that hasn't been observed in the wild and is harder induce than the current deadlock. Thanks go to Filip Frank for suggesting a fix similar to this and especially to IRC user hexanol for pointing out why this deadlock was possible and testing this fix. And to Richard for catching my rookie while loop mistake ;) ASTERISK-25321 #close Change-Id: I293ae0014e531cd0e675c3f02d1d118a98683def	9 years ago
zuul	9263ea0b13	Merge "res_pjsip_caller_id: Anonymize 'From' when caller id presentation is prohibited" into 13	9 years ago
George Joseph	530cff5f5f	res_pjsip: Strip spaces from items parsed from comma-separated lists ... Configurations like "aors = a, b, c" were either ignoring everything after "a" or trying to look up " b". Same for mailboxes, ciphers, contacts and a few others. To fix, all the strsep(&copy, ",") calls have been wrapped in ast_strip. To facilitate this, ast_strip, ast_skip_blanks and ast_skip_nonblanks were updated to handle null pointers. In some cases, an ast_strlen_zero() test was added to skip consecutive commas. There was also an attempt to ast_free an ast_strdupa'd string in ast_sip_for_each_aor which was causing a SEGV. I removed it. Although this issue was reported for realtime, the issue was in the res_pjsip modules so all config mechanisms were affected. ASTERISK-25829 #close Reported-by: Mateusz Kowalski Change-Id: I0b22a2cf22a7c1c50d4ecacbfa540155bec0e7a2	9 years ago
George Joseph	27f32cd0a6	res_pjsip_caller_id: Anonymize 'From' when caller id presentation is prohibited ... Per RFC3325, the 'From' header is now anonymized on outgoing calls when caller id presentation is prohibited. TID = trust_id_outbound PRO = Set(CALLERID(pres)=prohib) USR = endpoint/from_user DOM = endpoint/from_domain PAI = YES(privacy=off), NO(not sent), PRI(privacy=full) (assumes send_pai=yes) Conditions |Result --------------------|---------------------------------------------------- TID PRO USR DOM |PAI FROM --------------------|---------------------------------------------------- Y Y abc def.ghi |PRI "Anonymous" <sip:abc@def.ghi> Y Y abc |PRI "Anonymous" <sip:abc@anonymous.invalid> Y Y def.ghi |PRI "Anonymous" <sip:anonymous@def.ghi> Y Y |PRI "Anonymous" <sip:anonymous@anonymous.invalid> Y N abc def.ghi |YES <sip:abc@def.ghi> Y N abc |YES <sip:abc@<ip_address>> Y N def.ghi |YES "Caller Name" <sip:<caller_exten>@def.ghi> Y N |YES "Caller Name" <sip:<caller_exten>@<ip_address>> N Y abc def.ghi |NO "Anonymous" <sip:abc@def.ghi> N Y abc |NO "Anonymous" <sip:abc@anonymous.invalid> N Y def.ghi |NO "Anonymous" <sip:anonymous@def.ghi> N Y |NO "Anonymous" <sip:anonymous@anonymous.invalid> N N abc def.ghi |YES <sip:abc@def.ghi> N N abc |YES <sip:abc@<ip_address>> N N def.ghi |YES "Caller Name" <sip:<caller_exten>@def.ghi> N N |YES "Caller Name" <sip:<caller_exten>@<ip_address>> ASTERISK-25791 #close Reported-by: Anthony Messina Change-Id: I2c82a5ca1413c2c00fb62ea95b0ae8e97af54dc9	9 years ago
Kevin Harwell	40d9e9e238	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	9 years ago
George Joseph	b59956a875	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	9 years ago
Richard Mudgett	221422be50	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	9 years ago
Richard Mudgett	0a5bc64491	channel api: Create is_t38_active accessor functions. ... ASTERISK-25582 Change-Id: I69451920b122de7ee18d15bb231c80ea7067a22b	9 years ago
Richard Mudgett	513638a5f4	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	9 years ago
zuul	d35c494df1	Merge "res_pjsip/config_transport: Allow reloading transports." into 13	9 years ago
Richard Mudgett	e7a6abbbd3	rtp_engine.h: Remove extraneous semicolons. ... Change-Id: Ib462633d396fa941379dfef648dcd2245e350084	9 years ago
George Joseph	d2a1457e0b	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	9 years ago
zuul	953ba9da88	Merge "res_pjsip: Handle pjsip_dlg_create_uas deprecation" into 13	9 years ago
Joshua Colp	249d80f120	Merge "res_pjsip: Fix infinite recursion when loading transports from realtime" into 13	9 years ago
George Joseph	c1bf014ea0	res_pjsip: Handle pjsip_dlg_create_uas deprecation ... Pjproject has deprecated pjsip_dlg_create_uas in 2.5 and replaced it with pjsip_dlg_create_uas_and_inc_lock which, as the name implies, automatically increments the lock on the returned dialog. To account for this, configure.ac now detects the presence of pjsip_dlg_create_uas_and_inc_lock and res_pjsip.c has an #ifdef HAVE_PJSIP_DLG_CREATE_UAS_AND_INC_LOCK to decide whether to use the original call or the new one. If the new one was used, the ref count is decremented before returning. ASTERISK-25751 #close Reported-by Josh Colp Change-Id: I1be776b94761df03bd0693bc7795a75682615ca8	9 years ago
Corey Farrell	93e8ed0154	Simplify and fix conditional in FD_SET. ... FD_SET contains a conditional statement to protect against buffer overruns. The statement was overly complicated and prevented use of the last array element of ast_fdset. We now just verify the fd is less than ast_FDMAX. Change-Id: I41895c0b497b052aef5bf49d75c817c48b326f40	9 years ago
George Joseph	2451d4e455	res_pjsip: Fix infinite recursion when loading transports from realtime ... Attempting to load a transport from realtime was forcing asterisk into an infinite recursion loop. The first thing transport_apply did was to do a sorcery retrieve by id for an existing transport of the same name. For files, this just returns the previous object from res_sorcery_config's internal container, if any. For realtime, the res_sourcery_realtime driver looks in the database and finds the existing row but now it has to rehydrate it into a sorcery object which means calling... transport_apply. And so it goes. The main issue with loading from realtime (apart from the loop) was that transport stores structures and pointers directly in the ast_sip_transport structure instead of the separate ast_transport_state structure. This patch separates those items into the ast_sip_transport_state structure. The pattern is roughly the same as res_pjsip_outbound_registration. Although all current usages of ast_sip_transport and ast_sip_transport_state were modified to use the new ast_sip_get_transport_state API, the original items are left in ast_sip_transport and kept updated to maintain ABI compatability for third-party modules. They are marked as deprecated and noted that they're now in ast_sip_transport_state. ASTERISK-25606 #close Reported-by: Martin Moučka Change-Id: Ic7a836ea8e786e8def51fe3f8cce855ea54f5f19	9 years ago
Mark Michelson	6a799cd78f	Check for OpenSSL defines before trying to use them. ... The SSL_OP_NO_TLSv1_1 and SSL_OP_NO_TLSv1_2 defines did not exist prior to OpenSSL version 1.0.1. A recent commit attempts to, by default, set these options, which can cause problems on systems with older OpenSSL installations. This commit adds a configure script check for those defines and will not attempt to make use of those if they do not exist. We will print a warning urging the user to upgrade their OpenSSL installation if those defines are not present. Change-Id: I6a2eb9a43fd0738b404d8f6f2cf4b5c22d9d752d	9 years ago
Joshua Colp	f8acadde2c	AST-2016-001 http: Provide greater control of TLS and set modern defaults. ... This change exposes the configuration of various aspects of the TLS support and sets the default to the modern standards. The TLS cipher is now set to the best values according to the Mozilla OpSec team, different TLS versions can now be disabled, and the cipher order can be forced to be that of the server instead of the client. ASTERISK-24972 #close Change-Id: I0a10f2883f7559af5e48dee0901251dbf30d45b8	9 years ago
Mark Michelson	65bd4fcc3f	res_odbc: Remove connection management ... Asterisk by default will create a single database connection and share it among all threads that attempt to access the database. In previous versions of Asterisk, this was tolerable, because the most used channel driver, chan_sip, mostly accessed the database from a single thread. With PJSIP, however, many threads may be attempting to perform database operations, and there is the potential for many more database accesses, meaning the concurrency is a horrible bottleneck if only one connection is shared. Asterisk has a connection pooling facility built into it, but the implementation has flaws. For one, there is a strict limit on the number of simultaneous connections that could be made to the database. Anything beyond the maximum would result in a failed operation. Attempting to predict what the maximum should be is nearly impossible even for someone intimately familiar with Asterisk's threading model. In addition, use of transactions in the dialplan can cause some severe bugs if connection pooling is enabled. This commit seeks to fix the concurrency problem by removing all connection management code from Asterisk and leaving that to the underlying unixODBC code instead. Now, Asterisk does not share a single connection, nor does it try to maintain a connection pool. Instead, all Asterisk ever does is request a connection from unixODBC and allow unixODBC to either allocate those connections or retrieve them from a pool. Doing this has a bit of a ripple effect. For one, since connections are not long-lived objects, several of the safeguards that previously existed have been removed. We don't have to worry about trying to use a connection that has gone stale. In every case, when we request a connection, it has just been made and we don't need to perform any sanity checks to be sure it's still active. Another major player affected by this change is transactions. Transactions and their respective connections were so tightly coupled that it was almost pornographic. This code change moves transaction-related code to its own file separate from the core ODBC functionality. This way, the core of ODBC does not even have to know that transactions exist. In making this large change, I had to look at a lot of code and understand it. When making this change, I discovered several places where the behavior is definitely not ideal, but it seemed outside the scope of this change to be fixing it. Instead, any place where I saw some sort of room for improvement has had a XXX comment added explaining what could be altered to improve it. Change-Id: I37a84def5ea4ddf93868ce8105f39de078297fbf	9 years ago
Richard Mudgett	f87c3275cc	res_pjsip: Add CLI "pjsip dump endpt [details]" ... Dump the res_pjsip endpt internals. In non-developer mode we will not document or make easily accessible the "details" option even though it is still available. The user has to know it exists to use it. Presumably they would also be aware of the potential crash warning below. Warning: PJPROJECT documents that the function used by this CLI command may cause a crash when asking for details because it tries to access all active memory pools. Change-Id: If2d98a3641c9873364d1daaad971376311aef3cb	9 years ago
George Joseph	137fe5ae01	res_pjproject: Add module providing pjproject logging and utils ... res_pjsip_log_forwarder has been renamed to res_pjproject and enhanced as follows: As a follow-on to the recent 'Add CLI "pjsip show buildopts"' patch, a new ast_pjproject_get_buildopt function has been added. It allows the caller to get the value of one of the buildopts. The initial use case is retrieving the runtime value of PJ_MAX_HOSTNAME to insure we don't send a hostname greater than pjproject can handle. Since it can differ between the version of pjproject that Asterisk was compiled against and the version of pjproject that Asterisk is running against, we can't use the PJ_MAX_HOSTNAME macro directly in Asterisk source code. Change-Id: Iab6e82fec3d7cf00c1cf6185c42be3e7569dee1e	9 years ago
Kevin Harwell	660fedecb7	bridge_basic: don't cache xferfailsound during an attended transfer ... The xferfailsound was read from the channel at the beginning of the transfer, and that value is "cached" for the duration of the transfer. Therefore, changing the xferfailsound on the channel using the FEATURE() dialplan function does nothing once the transfer is under way. This makes it so the transfer code instead gets the xferfailsound configuration options from the channel when it is actually going to be used. This patch also fixes a potential memory leak of the props object as well as making sure the condition variable gets initialized before being destroyed. ASTERISK-25696 #close Change-Id: Ic726b0f54ef588bd9c9c67f4b0e4d787934f85e4	10 years ago
Joshua Colp	236896f391	Merge "pjsip: Add option global/regcontext" into 13	10 years ago
Joshua Colp	092c0db493	Merge "pjsip_sdp_rtp: Add option endpoint/bind_rtp_to_media_address" into 13	10 years ago
Joshua Colp	b91dfcd1fb	Merge "res_pjsip: Create human friendly serializer names." into 13	10 years ago
Joshua Colp	56d1162dc2	Merge topic 'update_taskprocessor_commands' into 13 ... * changes: Sorcery: Create human friendly serializer names. Stasis: Create human friendly taskprocessor/serializer names. taskprocessor.c: New API for human friendly taskprocessor names. taskprocessor.c: Sort CLI "core show taskprocessors" output.	10 years ago
George Joseph	219c204a41	pjsip_sdp_rtp: Add option endpoint/bind_rtp_to_media_address ... On a system with multiple ip addresses in the same subnet, if a transport is bound to a specific ip address and endpoint/media_address is set, the SIP/SDP will have the correct address in all fields but the rtp stream MAY still originate from one of the other ip addresses, most probably the "primary" ip address. This happens because res_pjsip_sdp_rtp/create_rtp always calls ast_instance_new with the "all" ip address (0.0.0.0 or ::). The new option causes res_pjsip_sdp_rtp/create_rtp to call ast_rtp_instance_new with the endpoint's media_address (if specified) instead of the "all" address. This causes the packets to originate from the specified address. ASTERISK-25632 ASTERISK-25637 Reported-by: Olivier Krief Reported-by: Dan Journo Change-Id: I3dfaa079e54ba7fb7c4fd1f5f7bd9509bbf8bd88	10 years ago
Daniel Journo	22801a06ee	pjsip: Add option global/regcontext ... Added new global option (regcontext) to pjsip. When set, Asterisk will dynamically create and destroy a NoOp priority 1 extension for a given endpoint who registers or unregisters with us. ASTERISK-25670 #close Reported-by: Daniel Journo Change-Id: Ib1530c5b45340625805c057f8ff1fb240a43ea62	10 years ago
Richard Mudgett	cf8e7a580b	res_pjsip: Create human friendly serializer names. ... PJSIP name formats: pjsip/aor/<aor>-<seq> -- registrar thread pool serializer pjsip/default-<seq> -- default thread pool serializer pjsip/messaging -- messaging thread pool serializer pjsip/outreg/<registration>-<seq> -- outbound registration thread pool serializer pjsip/pubsub/<endpoint>-<seq> -- pubsub thread pool serializer pjsip/refer/<endpoint>-<seq> -- REFER thread pool serializer pjsip/session/<endpoint>-<seq> -- session thread pool serializer pjsip/websocket-<seq> -- websocket thread pool serializer Change-Id: Iff9df8da3ddae1132cb2ef65f64df0c465c5e084	10 years ago
Richard Mudgett	ec1f1c6742	taskprocessor.c: New API for human friendly taskprocessor names. ... * Add new API call to get a sequence number for use in human friendly taskprocessor names. * Add new API call to create a taskprocessor name in a given buffer and append a sequence number. Change-Id: Iac458f05b45232315ed64aa31b1df05b875537a9	10 years ago
Diederik de Groot	4285dee778	include/asterisk/time.h: Renamed global declaration:tv ... Renamed global declaration:tv to dummy_tv_var_for_types, which would oltherwise cause 'shadow' warnings when 'tv' was declared as a local variable elsewhere. Added comment to note that dummy_tv_var_for_types is never really exported and only used as a place holder. ASTERISK-25627 #close Change-Id: I9a6e17995006584f3627efe8988e3f8aa0f5dc28	10 years ago
Joshua Colp	b1ee692568	Merge topic 'pbx-split' into 13 ... * changes: main/pbx: Move hangup handler routines to pbx_hangup_handler.c. main/pbx: Move dialplan application management routines to pbx_app.c. main/pbx: Move switch routines to pbx_switch.c.	10 years ago
George Joseph	881dc862e0	asterisk.h: Add ASTERISK_REGISTER_FILE macro ... The 11/13 branches and master use 2 different file version macros. 11/13 uses ASTERISK_FILE_VERSION but master uses ASTERISK_REGISTER_FILE. This means a new file added to 11/13 can't just be cherry-picked to master because the macro has to be changed. To make cherry-picking possible, ASTERISK_REGISTER_FILE was added to asterisk.h as a simple alias for ASTERISK_FILE_VERSION(__FILE__, NULL) The "$Revision$" tag doesn't do anything since Asterisk moved to git so just passing NULL as the verison works fine. asterisk.h was also annotated to deprecate ASTERISK_FILE_VERSION and suggest using ASTERISK_REGISTER_FILE for all new files. Finally, 2 recent file additions, pbx_builtins.c and pbx_functions.c, were modified to use the new macro to make sure it actually worked. 'core show file version' showed the correct output. Change-Id: I5867ed898818d26ee49bb6e5c7d4c1a45d4789a5	10 years ago
Matt Jordan	9016e51c03	Merge "main/pbx: Move variable routines to pbx_variables.c." into 13	10 years ago
George Joseph	d228b62fd4	stasis_cache_pattern: Backport to 13 ... Somehow stasis_cache_pattern got out of sync between 13 and master and it was causing duplicate channel message issues in 13 when related to a specific endpoint. I.E. from statsd, 'endpoints.PJSIP.1174.channels 0|g' was being emitted twice. Backporting stasis_cache_pattern from master to 13 solved the issue and running the unit and testsuite tests confirmed that no new ones were created. ASTERISK-25317 #close Change-Id: Ia8707462f62d15eed14541c37f332a7bbbceb548	10 years ago
Corey Farrell	e462f0063f	main/pbx: Move hangup handler routines to pbx_hangup_handler.c. ... This is the sixth patch in a series meant to reduce the bulk of pbx.c. This moves hangup handler management functions to their own source. Change-Id: Ib25a75aa57fc7d5c4294479e5cc46775912fb104	10 years ago
Corey Farrell	ab191d124c	main/pbx: Move dialplan application management routines to pbx_app.c. ... This is the sixth patch in a series meant to reduce the bulk of pbx.c. This moves dialplan application management functions to their own source. Change-Id: I444c10fb90a3cdf9f3047605d6a8aad49c22c44c	10 years ago
Corey Farrell	09a9b93896	main/pbx: Move switch routines to pbx_switch.c. ... This is the fifth patch in a series meant to reduce the bulk of pbx.c. This moves ast_switch functions to their own source. Change-Id: Ic2592a18a5c4d8a3c2dcf9786c9a6f650a8c628e	10 years ago
George Joseph	4ec85a9f07	voicemail: Move app_voicemail / res_mwi_external conflict to runtime ... The menuselect conflict between app_voicemail and res_mwi_external makes it hard to package 1 version of Asterisk. There no actual build dependencies between the 2 so moving this check to runtime seems like a better solution. The ast_vm_register and ast_vm_greeter_register functions in app.c were modified to return AST_MODULE_LOAD_DECLINE instead of -1 if there is already a voicemail module registered. The modules' load_module functions were then modified to return DECLINE instead of -1 to the loader. Since -1 is interpreted by the loader as AST_MODULE_LOAD_FAILURE, the modules were incorrectly causing Asterisk to stop so this needed to be cleaned up anyway. Now you can build both and use modules.conf to decide which voicemail implementation to load. The default menuselect options still build app_voicemail and not res_mwi_external but if both ARE built, res_mwi_external will load first and become the voicemail provider unless modules.conf rules prevent it. This is noted in CHANGES. Change-Id: I7d98d4e8a3b87b8df9e51c2608f0da6ddfb89247	10 years ago
Corey Farrell	7fdcfd7724	main/pbx: Move variable routines to pbx_variables.c. ... This is the third patch in a series meant to reduce the bulk of pbx.c. This moves channel and global variable routines to their own source. Change-Id: Ibe8fb4647db11598591d443a99e3f99200a56bc6	10 years ago
Corey Farrell	2ffade4574	main/pbx: Move custom function routines to pbx_functions.c. ... This is the second patch in a series meant to reduce the bulk of pbx.c. This moves custom function management routines to their own source. Change-Id: I34a6190282f781cdbbd3ce9d3adeac3c3805e177	10 years ago
Matt Jordan	bc7c882326	Merge "main/pbx: Move pbx_builtin dialplan applications to pbx_builtins.c" into 13	10 years ago
George Joseph	20b8474f20	main/pbx: Move pbx_builtin dialplan applications to pbx_builtins.c ... We joked about splitting pbx.c into multiple files but this first step was fairly easy. All of the pbx_builtin dialplan applications have been moved into pbx_builtins.c and a new pbx_private.h file was added. load_pbx_builtins() is called by asterisk.c just after load_pbx(). A few functions were renamed and are cross-exposed between the 2 source files. Change-Id: I87066be3dbf7f5822942ac1449d98cc43fc7561a	10 years ago
Joshua Colp	a68467d293	Merge "res_http_websocket.c: prevent avoidable disconnections caused by write errors" into 13	10 years ago
George Joseph	3a1c4885be	endpoint/stasis: Eliminate duplicate events on endpoint status change ... When an endpoint is created, its messages are forwarded to both the tech endpoint topic and the all endpoints topic. This is done so that various parties interested in endpoint messages can subscribe to just the tech endpoint and receive all messages associated with that particular technology, as opposed to subscribing to the all endpoints topic. Unfortunately, when the tech endpoint is created, it also forwards all of its messages to the all topic. This results in duplicate messages whenever an endpoint publishes its messages. This patch resolves the duplicate message issue by creating a new function for Stasis caching topics, stasis_cp_sink_create. In most respects, this acts as a normal caching topic, save that it no longer forwards messages it receives to the all endpoints topic. This allows it to act as an aggregation "sink", while preserving the necessary caching behaviour. ASTERISK-25137 #close Reported-by: Vitezslav Novy ASTERISK-25116 #close Reported-by: George Joseph <george.joseph@fairview5.com> Tested-by: George Joseph <george.joseph@fairview5.com> Change-Id: Ie47784adfb973ab0063e59fc18f390d7dd26d17b	10 years ago