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1.6.0
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10.12.0
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12.0.0
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certified/1.8.11-cert1
certified/1.8.11-cert10
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certified/1.8.11-cert6
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certified/1.8.15-cert1
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certified/1.8.28-cert1
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certified/1.8.28-cert2
certified/1.8.28-cert3
certified/1.8.28-cert4
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certified/1.8.6-cert1
certified/11.2-cert1
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certified/11.2-cert2
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certified/11.6-cert1
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certified/11.6-cert10
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certified/11.6-cert12
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certified/11.6-cert6
certified/11.6-cert7
certified/11.6-cert8
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certified/13.1-cert1
certified/13.1-cert1-rc1
certified/13.1-cert1-rc2
certified/13.1-cert1-rc3
certified/13.1-cert2
certified/13.1-cert3
${ noResults }
3310 Commits (d35c494df1bfc1cc44d7483d9750f31d0184d020)
Author | SHA1 | Message | Date |
---|---|---|---|
|
d35c494df1 |
Merge "res_pjsip/config_transport: Allow reloading transports." into 13
|
9 years ago |
|
6e70e8ccdb |
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 |
9 years ago |
|
ea9deff996 |
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 |
9 years ago |
|
aa637f0a91 |
Merge "res_pjsip_config_wizard: Add command to export primitive objects" into 13
|
9 years ago |
|
56561c386a |
Merge "res_pjproject: Add ability to map pjproject log levels to Asterisk log levels" into 13
|
9 years ago |
|
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 |
|
6b921f706d |
res_pjproject: Add ability to map pjproject log levels to Asterisk log levels
Warnings and errors in the pjproject libraries are generally handled by Asterisk. In many cases, Asterisk wouldn't even consider them to be warnings or errors so the messages emitted by pjproject directly are either superfluous or misleading. A good exampe of this are the level-0 errors pjproject emits when it can't open a TCP/TLS socket to a client to send an OPTIONS. We don't consider a failure to qualify a UDP client an "ERROR", why should a TCP/TLS client be treated any differently? A config file for res_pjproject has bene added (pjproject.conf) and a new log_mappings object allows mapping pjproject levels to Asterisk levels (or nothing). The defaults if no pjproject.conf file is found are the same as those that were hard-coded into res_pjproject initially: 0,1 = LOG_ERROR, 2 = LOG_WARNING, 3,4,5 = LOG_DEBUG<level> Change-Id: Iba7bb349c70397586889b8f45b8c3d6c6c8c3898 |
9 years ago |
|
f295088764 |
res_pjsip_outbound_publish: Fix processing 412 response
When Asterisk receives a 412 (Conditional Request Failed) response it has to recreate publish session. There is bug in res_pjsip_outbound_publish.c The function sip_outbound_publish_client_alloc is called with wrong object while processing 412 (Conditional Request Failed) response. This patch fixes it. ASTERISK-25229 #close Change-Id: I3b62f2debf6bb1e5817cde7b13ea39ef2bf14359 |
9 years ago |
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87ab65c557 |
res_odbc: Fix exports.in for missing symbols
res_odbc.exports.in was missing a few symbols. Changed to wildcards. Change-Id: Ieadd76df24e43ea92577f651d478a0f7b742c30c |
9 years ago |
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c0f3062031 |
res_statsd: Fix exports.in for missing symbols
res_statsd.export.in was missing the _va variations of the log functions causing Asterisk to crash in res_pjsip if OPTIONAL_API wasn't enabled. ASTERISK-25727 #close Reported-by: Gergely Dömsödi Change-Id: I395729f9f51bdd33c5ca757f5f96ebedad74077b |
9 years ago |
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5e848dae7b |
res_pjsip_config_wizard: Add command to export primitive objects
A new command (pjsip export config_wizard primitives) has been added that will export all the pjsip objects it created to the console or a file suitable for reuse in a pjsip.conf file. ASTERISK-24919 #close Reported-by: Ray Crumrine Change-Id: Ica2a5f494244b4f8345b0437b16d06aa0484452b |
9 years ago |
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34c64707d1 |
res_pjsip_caller_id: Fix segfault when replacing rpid or pai header
If the PJSIP_HEADER dialplan function adds a PAI or RPID header and send_rpid or send_pai is set, res_pjsip_caller_id attemps to retrieve, parse and modify the header added by the dialplan function. Since the header added by the dialplan function is generic string, there are no virtual functions to parse the uri and we get a segfault when we try. Since the modify, was really only an overwrite, we now just delete the old header if it was type PJSIP_H_OTHER and recreate it. This raises a question for another time though: What should happen with duplicate headers? Right now res_pjsip_header_funcs doesn't check for dups so if it's session supplement is loaded after res_pjsip_caller_id's (or any other module that adds headers), there'll be dups in the message. ASTERISK-25337 #close Change-Id: I5e296b52d30f106b822c0eb27c4c2b0e0f71c7fa |
9 years ago |
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9c28acd9db |
Merge "Fix creation race of contact_status structures." into 13
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9 years ago |
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ebe167f792 |
Fix creation race of contact_status structures.
It is possible when processing a SIP REGISTER request to have two threads end up creating contact_status structures in sorcery. contact_status is created using a "find or create" function. If two threads call into this at the same time, each thread will fail to find an existing contact_status, and so both will end up creating a new contact status. During testing, we would see sporadic failures because the PJSIP_CONTACT() dialplan function would operate on a different contact_status than what had been updated by res_pjsip/pjsip_options. The fix here is two-fold: 1) The "find or create" function for contact_status now has a lock around the entire operation. This way, if two threads attempt the operation simultaneously, the first to get there will create the object, and the second will find the object created by the first thread. 2) res_sorcery_memory has had its create callback updated so that it will not allow for objects with duplicate IDs to be created. Change-Id: I55b1460ff1eb0af0a3697b82d7c2bac9f6af5b97 |
9 years ago |
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1c4f2a920d |
res_pjsip_pubsub: Move where the subscription is stored to after initialized.
A problem arose when testing the AMI subscription listing actions where it was possible for a subscription that had not been fully initialized to be listed. This was problematic as the underlying listing code would crash. This change makes it so the subscription tree is fully set up before it is added to the list of subscriptions. This ensures that when the listing actions get the subscription it is valid. ASTERISK-25738 #close Change-Id: Iace2b13641c31bbcc0d43a39f99aba1f340c0f48 |
9 years ago |
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1cddd954f9 |
Merge "res_pjsip: Refactor load_module/unload_module" into 13
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9 years ago |
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953ba9da88 |
Merge "res_pjsip: Handle pjsip_dlg_create_uas deprecation" into 13
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9 years ago |
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b1b797e0e7 |
res_pjsip: Refactor load_module/unload_module
load_module was just too hairy with every step having to clean up all previous steps on failure. Some of the pjproject init calls have now been moved to a separate load_pjsip function and the unload_pjsip function was enhanced to clean up everything if an error happened at any stage of the load process. In the process, a bunch of missing pj_shutdowns, serializer_pool_shutdowns and ast_threadpool_shutdowns were also corrected. Change-Id: I5eec711b437c35b56605ed99537ebbb30463b302 |
9 years ago |
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8c15f309c2 |
Merge "Resources/res_phoneprov: fix memory leak and heap-use-after-free" into 13
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9 years ago |
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20e9792fbc |
Resources/res_phoneprov: fix memory leak and heap-use-after-free
* heap-use-after-free happens when we free "cfg" but then use "value" which refers to it * A memory leak occurs because in some cases it is not released "defaults" ASTERISK-25721 #close Reported by: Badalyan Vyacheslav Tested by: Badalyan Vyacheslav Change-Id: I3807d3f4726df6864430ec144cf6265d3f538469 |
9 years ago |
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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 |
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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 |
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c5e7d5c105 |
Merge "logging: Remove/fix some message annoyances" into 13
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9 years ago |
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6cac364284 |
Merge "res_stasis_device_state: Fix refcounting error." into 13
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9 years ago |
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014fc9ef65 |
Merge "res_xmpp: Does not connect in component mode" into 13
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9 years ago |
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23829b3253 |
res_stasis_device_state: Fix refcounting error.
Device state subscription lifetimes were governed by when the subscription was established and unsubscribed from. However, it is possible that at the time of unsubscription, there could be device state events still in flight. When those device state events occur, the device state callback could attempt to dereference a freed pointer. Crash. This change ensures that the lifetime of the device state subscription does not end until the underlying stasis subscription has confirmed that its final message has been sent. Change-Id: I25a0f1472894c1a562252fb7129671478e25e9b2 |
9 years ago |
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4e8e6d3922 |
res_rtp_asterisk: Allow ICE host candidates to be overriden
During ICE negotiation the IPs of the local interfaces are sent to the remote peer as host candidates. In many cases Asterisk is behind a static one-to-one NAT, so these host addresses will be internal IP addresses. To help in hiding the topology of the internal network, this patch adds the ability to override the host candidates by matching them against a user-defined list of replacements. Change-Id: I1c9541af97b83a4c690c8150d19bf7202c8bff1f |
9 years ago |
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2a6ee8caeb |
logging: Remove/fix some message annoyances
test_dlinklists doesn't need to NOTICE everyone that every macro worked. res_phoneprov doesn't need to VERBOSE everyone that a phoneprov extension or provider was registered. res_odbc was missing a newline at the end of one message. Change-Id: I6c06361518ef3711821795e535acd439782a995e |
9 years ago |
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32fc784284 |
res_sorcery_realtime: Fix regex regression.
A regression was introduced where searching for realtime PJSIP objects by regex by starting the regex with a leading "^" would cause no items to be returned. This was due to a change which attempted to drop the requirement for a leading "^" to be present due to how some CLI commands formulate their regexes. However, the change, rather than simply eliminating the requirement, caused any regexes that did begin with "^" to end up not returning the expected results. This change fixes the problem by inspecting the regex and formulating the realtime query differently depending on if it begins with "^". ASTERISK-25702 #close Reported by Nic Colledge Patches: realtime_retrieve_regex.patch submitted by Alexei Gradinari License #5691 Change-Id: I055df608a6e6a10732044fa737a9fe8dca602693 |
9 years ago |
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0405c31756 |
res_xmpp: Does not connect in component mode
The module res_xmpp does not accept usernames in the form used in component mode (XEP-0114). In component mode there is no @something in the name. In component mode the connection is now not dropped anymore. If the xmpp server sends out a "stream" tag before handshake is finished, the connection gets dropped in res_xmpp. Now this tag will be ignored and the connection will be established. After connecting there will be an exchange of presence states. This does not work as expected in component mode. The responsible function "xmpp_pak_presence" is left before the states get sent out. Sending presence states in component mode is now moved to the top of the function. ASTERISK-25735 #close Change-Id: I70e036f931c3124ebb2ad1e56f93ed35cfdd9d5c |
9 years ago |
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f6551868be |
Merge "res_odbc: Remove connection management" into 13
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9 years ago |
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8804d0973c |
build_system: Fix some warnings highlighted by clang
Fix some warnings found with clang. Change-Id: I5195b6189b148c2ee3ed4a19d015a6d4ef3e77bd |
9 years ago |
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109b0aff6b |
res/Makefile: Fix bug in "clean" target for ari
The "clean" target was attempting to clean res/ari from inside the res directory which doesn't remove anything. Removed the res/ prefix. Change-Id: Ib1a518d54efa81b9fd5a42742d43cc3767435bf6 |
9 years ago |
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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 |
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4cc784eb04 |
Merge "Stasis: Use custom structure when setting variables." into 13
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10 years ago |
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4a3275abb9 |
Stasis: Use custom structure when setting variables.
A recent change to queue channel variable setting to the Stasis control queue caused a regression. When setting channel variables, it is possible to give a NULL channel variable value in order to unset the variable (i.e. remove it from the channel variable list). The change introduced a call to ast_variable_new(), which is not tolerant of NULL channel variable values. This new change switches from using ast_variable to using a custom channel variable struct that is lighter weight and NULL value-tolerant. Change-Id: I784d7beaaa3c036ea936d103e7caf0bb1562162d |
10 years ago |
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8261bda1bf |
res_pjsip_pubsub: Prevent crash from AMI command on freed subscription.
A test recently uncovered that running an ill-timed AMI command to show inbound subscriptions could cause a crash since Asterisk will try to operate on a freed subscription. The fix for this is to remove the subscription tree from the list of subscriptions at the time that we are sending our final NOTIFY request out. This way, as the subscription is in the process of dying, it is inaccessible from AMI. Change-Id: Ic0239003d8d73e04c47c12dd2a7e23867e5b5b23 |
10 years ago |
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fcb6c1737d |
Merge "Stasis: Use control queue to prevent crash." into 13
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10 years ago |
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1003c2eb05 |
Stasis: Fix potential memory leak of control data.
When queuing tasks onto the Stasis control queue, you can pass an arbitrary data pointer and a function to free that data. All ARI commands that use the Stasis control queue made the assumption that the destructor function would be called in all paths, whether the task was queued successfully or not. However, this was not correct. If a task was queued onto a control structure that was already completed, the allocated data would not be freed properly. This patch corrects this by making sure that all return paths call the data destructor. Change-Id: Ibf06522094f8e5c4cce652537dc5d7222b1c4fcb |
10 years ago |
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eedd77fda0 |
Stasis: Use control queue to prevent crash.
A crash occurred when attempting to set a channel variable on a channel that had already been hung up. This is because there is a small window between when a control is grabbed and when the channel variable is set that the channel can be hung up. The fix here is to queue the setting of the channel variable onto the control queue. This way, the manipulation of the channel happens in a thread where it is safe to be done. In this change, I also noticed that the setting of bridge roles on channels was being done outside of the control queue, so I also changed those operations to be done in the control queue. ASTERISK-25709 #close Reported by Mark Michelson Change-Id: I2a0a4d51bce6fba6f1d9954e40935e42f366ea78 |
10 years ago |
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c3f4afe40c |
Merge "res/res_pjsip/presence_xml.c: Add missing 2nd call presence state case." into 13
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10 years ago |
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02035212de |
res/res_pjsip/presence_xml.c: Add missing 2nd call presence state case.
ASTERISK-25712 #close Reported by: Richard Mudgett Change-Id: I70634df24f8c6c3a2c66c45af61d021e4999253f |
10 years ago |
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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 |
10 years ago |
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a2928b6093 |
Merge "res_pjproject: Add module providing pjproject logging and utils" into 13
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10 years ago |
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d1113e0f56 |
Merge "pjsip_loging_refactor: Rename res_pjsip_log_forwarder to res_pjproject" into 13
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10 years ago |
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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 |
10 years ago |
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a0c79f3a4f |
pjsip_loging_refactor: Rename res_pjsip_log_forwarder to res_pjproject
Change-Id: I5387821f29e5caa0cba0b7d62b0fc0d341e7e20b |
10 years ago |
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935d641f3b |
Remove res/ari/* content during 'make clean'.
'make clean' and 'make distclean' can leave behind .o files in the res/ari/ directory. One observed consequence of this is that running Asterisk with MALLOC_DEBUG can cause Asterisk to crash immediately on startup sometimes. By ensuring that we are making a clean build, we can be sure that stale files are not being included in the build and causing problems when build options should have caused files to be re-built. ASTERISK-25683 #close Reported by yaron nahum Change-Id: I1f48baa904d2468eddeefb42ee68a56af7adc7b7 |
10 years ago |
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236896f391 |
Merge "pjsip: Add option global/regcontext" into 13
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10 years ago |
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f18ad96b77 |
Merge "res_pjsip_log_forwarder.c: Add CLI "pjsip show buildopts"." into 13
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10 years ago |