3271 Commits (01e58943882969b02cbe3482a873e21f3284cb4c)

Author	SHA1	Message	Date
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
Dade Brandon	136c537695	res_http_websocket.c: prevent avoidable disconnections caused by write errors ... Updated ast_websocket_write to encode the entire frame in to one write operation, to ensure that we don't end up with a situation where the websocket header has been sent, while the body can not be written. Previous to August's patch in commit b9bd3c14, certain network conditions could cause the header to be written, and then the sub-sequent body to fail - which would cause the next successful write to contain a new header, and a new body (resulting in the peer receiving two headers - the second of which would be read as part of the body for the first header). This was patched to have both write operations individually fail by closing the websocket. In a case available to the submitter of this patch, the same body which would consistently fail to write, would succeed if written at the same time as the header. This update merges the two operations in to one, adds debug messages indicating the reason for a websocket connection being closed during a write operation, and clarifies some variable names for code legibility. Change-Id: I4db7a586af1c7a57184c31d3d55bf146f1a40598	10 years ago
Richard Mudgett	36097a185d	Fix sscanf() format string type mismatch. ... ASTERISK-25615 Reported by: George Joseph Change-Id: Ieff35307254ca193f3d473cff2e396ca57c7ce0b	10 years ago
George Joseph	5b867fa904	pjsip/config_transport: Check pjproject version at runtime for async ops ... pjproject < 2.5.0 will segfault on a tls transport if async_operations is greater than 1. A runtime version check has been added to throw an error if the version is < 2.5.0 and async_operations > 1. To assist in the check, a new api "ast_compare_versions" was added to utils which compares 2 major.minor.patch.extra version strings. ASTERISK-25615 #close Change-Id: I8e88bb49cbcfbca88d9de705496d6f6a8c938a98 Reported-by: George Joseph Tested-by: George Joseph	10 years ago
George Joseph	21962dad93	res_pjsip: Add existence and readablity checks for tls related files ... Both transport and endpoint now check for the existence and readability of tls certificate and key files before passing them on to pjproject. This will cause the object to not load rather than waiting for pjproject to discover that there's a problem when a session is attempted. NOTE: chan_sip also uses ast_rtp_dtls_cfg_parse but it's located in build_peer which is gigantic and I didn't want to disturb it. Error messages will emit but it won't interrupt chan_sip loading. ASTERISK-25618 #close Change-Id: Ie43f2c1d653ac1fda6a6f6faecb7c2ebadaf47c9 Reported-by: George Joseph Tested-by: George Joseph	10 years ago
George Joseph	ed9134282e	res_pjsip: Update logging to show contact->uri in messages ... An earlier commit changed the id of dynamic contacts to contain a hash instead of the uri. This patch updates status change logging to show the aor/uri instead of the id. This required adding the aor id to contact and contact_status and adding uri to contact_status. The aor id gets added to contact and contact_status in their allocators and the uri gets added to contact_status in pjsip_options when the contact_status is created or updated. ASTERISK-25598 #close Reported-by: George Joseph Tested-by: George Joseph Change-Id: I56cbec1d2ddbe8461367dd8b6da8a6f47f6fe511	10 years ago
Jonathan Rose	eadad24b59	Unset BRIDGEPEER when leaving a bridge ... Currently if a channel is transferred out of a bridge, the BRIDGEPEER variable (also BRIDGEPVTCALLID) remain set even once the channel is out of the bridge. This patch removes these variables when leaving the bridge. ASTERISK-25600 #close Reported by: Mark Michelson Change-Id: I753ead2fffbfc65427ed4e9244c7066610e546da	10 years ago
Matt Jordan	5d80a6e714	Merge "main: Slight refactor of main. Improve color situation." into 13	10 years ago
Walter Doekes	b2787876d6	main: Slight refactor of main. Improve color situation. ... Several issues are addressed here: - main() is large, and half of it is only used if we're not rasterisk; fixed by spliting up the daemon part into a separate function. - Call ast_term_init from rasterisk as well. - Remove duplicate code reading/writing asterisk history file. - Attempt to tackle background color issues and color changes that occur. Tested by starting asterisk -c until the colors stopped changing at odd locations. ASTERISK-25585 #close Change-Id: Ib641a0964c59ef9fe6f59efa8ccb481a9580c52f	10 years ago
David M. Lee	59881fbb99	Fixed some typos ... Fixes some minor typos in the CHANGES file, plus an embarrasing typo in the StatsD API. Change-Id: I9ca4858c64a4a07d2643b81baa64baebb27a4eb7	10 years ago
Joshua Colp	3f85a1be5a	Merge "translate: Provide translation modules the result of SDP negotiation." into 13	10 years ago
Alexander Traud	0b508789ab	translate: Provide translation modules the result of SDP negotiation. ... Previously, a trancoding module did not have access to the joint but cached format. Therefore, the module did not have access to the attributes negotiated via SDP (line fmtp). Now, a translation module receives the joint format. ASTERISK-25545 #close Change-Id: Id6878a989b50573298dab115d3371ea369e1a718	10 years ago
Matt Jordan	3354b325c6	res_statsd: Add functions that support variable arguments ... Often, the metric names of statistics we are generating for StatsD have some dynamic component to them. This can be the name of a particular resource, or some internal status label in Asterisk. With the current set of functions, callers of the statsd API must first build the metric name themselves, then pass this to the API functions. This results in a large amount of boilerplate code and usage of either fixed length static buffers or dynamic memory allocation, neither of which is desireable. This patch adds two new functions to the StatsD API that support a printf style format specifier for constructing the metric name. A dynamic string, allocated in threadstorage, is used to build the metric name. This eases the burden on users of the StatsD API. Change-Id: If533c72d1afa26d807508ea48b4d8c7b32f414ea	10 years ago
tcambron	1e0040b88f	StatsD: Add res_statsd compatibility ... Added a new api to res_statsd.c to allow it to receive a character pointer for the value argument. This allows for a '+' and a '-' to easily be sent with the value. ASTERISK-25419 Reported By: Ashley Sanders Change-Id: Id6bb53600943d27347d2bcae26c0bd5643567611	10 years ago
Mark Michelson	4f43b85c92	Taskprocessors: Increase high-water mark ... In practical tests, we have seen certain taskprocessors, specifically Stasis subscription taskprocessors, cross the recently-added high-water mark and emit a warning. This high-water mark warning is only intended to be emitted when things have tanked on the system and things are heading south quickly. In the practical tests, the Stasis taskprocessors sometimes had a max depth of 180 tasks in them, and Asterisk wasn't in any danger at all. As such, this ups the high-water mark to 500 tasks instead. It also redefines the SIP threadpool request denial number to be a multiple of the taskprocessor high-water mark. Change-Id: Ic8d3e9497452fecd768ac427bb6f58aa616eebce	10 years ago
Mark Michelson	2f9cb7d62b	res_pjsip: Deny requests when threadpool queue is backed up. ... We have observed situations where the SIP threadpool may become deadlocked. However, because incoming traffic is still arriving, the SIP threadpool's queue can continue to grow, eventually running the system out of memory. This change makes it so that incoming traffic gets rejected with a 503 response if the queue is backed up too much. Change-Id: I4e736d48a2ba79fd1f8056c0dcd330e38e6a3816	10 years ago
Alexander Traud	1bff400df7	ast_format_cap_get_names: To display all formats, the buffer was increased. ... ASTERISK-25533 #close Change-Id: Ie1a9d1a6511b3f1a56b93d04475fbf8a4e40010a	10 years ago
Corey Farrell	d098d00424	Fix cli display of build options. ... A previous commit reduced the AST_BUILDOPTS compiler define to only include options that affected ABI. This included some options that were previously displayed by cli "core show settings". This change corrects the CLI display while still restricting buildopts.h to ABI effecting options only. ASTERISK-25434 #close Reported by: Rusty Newton Change-Id: Id07af6bedd1d7d325878023e403fbd9d3607e325	10 years ago
Joshua Colp	ba566b0f0f	Merge "res_pjsip: Add "like" processing to pjsip list and show commands" into 13	10 years ago
Matt Jordan	4d576fe6d9	Merge "res_pjsip_pubsub: Solidify lifetime and ownership of objects." into 13	10 years ago
George Joseph	162acd45f7	res_pjsip: Add "like" processing to pjsip list and show commands ... Add the ability to filter output from pjsip list and show commands using the "like" predicate like chan_sip. For endpoints, aors, auths, registrations, identifyies and transports, the modification was a simple change of an ast_sorcery_retrieve_by_fields call to ast_sorcery_retrieve_by_regex. For channels and contacts a little more work had to be done because neither of those objects are true sorcery objects. That was just removing the non-matching object from the final container. Of course, a little extra plumbing in the common pjsip_cli code was needed to parse the "like" and pass the regex to the get_container callbacks. Some of the get_container code in res_pjsip_endpoint_identifier was also refactored for simplicity. ASTERISK-25477 #close Reported by: Bryant Zimmerman Tested by: George Joseph Change-Id: I646d9326b778aac26bb3e2bcd7fa1346d24434f1	10 years ago
Mark Michelson	b3cc2bd7df	res_pjsip_pubsub: Solidify lifetime and ownership of objects. ... There have been crashes and general instability seen in the pubsub code, so this patch introduces three changes to increase the stability. First, the ownership model for subscriptions has been modified. Due to RLS, subscriptions are stored in memory as a tree structure. Prior to my patch, the PJSIP subscription was the owner of the subscription tree. When the PJSIP subscription told us that it was terminating, we started destroying the subscription tree along with all of the individual leaf subscriptions that belong to the tree. The problem with this model is that the two actors in play here, the PJSIP subscription and the individual leaf subscriptions, need to have joint ownership of the subscription tree. So now, the PJSIP subscription and the individual leaf subscriptions each have a reference to the subscription tree. This way, we will not actually free memory until no players are left that care. The PJSIP subscription is a bigger stakeholder, in that if the PJSIP subscription's reference to the subscription tree is removed, the subscription tree instructs the leaf subscriptions to shut down and drop their references to the subscription tree when possible. The individual leaf subscriptions, upon being told to shut down, can drop their stasis subscriptions or whatever they use to learn of new state, and then drop their reference to the subscription tree once they are ready to die. Second, the lifetime of a PJSIP subscription's reference to our subscription tree has been altered. As I learned from doing a deep dive, the PJSIP evsub code can tell Asterisk multiple times that the subscription has been terminated, and not all of these times are especially helpful. I have altered the message flow that we use for SIP subscriptions such that we will always drop the PJSIP subscription's reference to the subscription tree when we send the NOTIFY that terminates a SIP subscription. This also means that we will now queue NOTIFY requests to be sent after responding to incoming SUBSCRIBEs so that we can have predictable state changes from the PJSIP evsub code. Third, the synchronization of operations has been improved. PJSIP can call into our code from a serializer thread (e.g. upon receiving an incoming request) or from the monitor thread (e.g. when a subscription times out). Because of this, there is the possibility of competing threads stepping on each other. PJSIP attempts to do some synchronization on its own by always keeping the dialog lock held when it calls into us. However, since we end up pushing tasks into the serializer, the result was that serialized operations were not grabbing the dialog lock and could, as a result, step on something that was being attempted by a different thread. Now we ensure that serialized operations grab the dialog lock, then check for extenuating circumstances, then proceed with their operation if they can. Change-Id: Iff2990c40178dad9cc5f6a5c7f76932ec644b2e5	10 years ago
Joshua Colp	f2725c8b77	res_pjsip: Move URI validation to use time. ... In a realtime based system with a limited number of threadpool threads it is possible for a deadlock to occur. This happens when permanent endpoint state is updated, which will cause database queries to be done. These queries may result in URI validation being done which is done synchronously using a PJSIP thread. If all PJSIP threads are in use processing traffic they themselves may be blocked waiting to get the permanent endpoint container lock when identifying an endpoint. This change moves URI validation to occur at use time instead of configuration time. While this comes at a cost of not seeing a problem until you use it it does solve the underlying deadlock problem. ASTERISK-25486 #close Change-Id: I2d7d167af987d23b3e8199e4a68f3359eba4c76a	10 years ago
Matt Jordan	4c9f613309	ARI: Add the ability to subscribe to all events ... This patch adds the ability to subscribe to all events. There are two possible ways to accomplish this: (1) On initial WebSocket connection. This patch adds a new query parameter, 'subscribeAll'. If present and True, Asterisk will subscribe the applications to all ARI events. (2) Via the applications resource. When subscribing in this manner, an ARI client should merely specify a blank resource name, i.e., 'channels:' instead of 'channels:12354'. This will subscribe the application to all resources of the 'channels' type. ASTERISK-24870 #close Change-Id: I4a943b4db24442cf28bc64b24bfd541249790ad6	11 years ago
Mark Michelson	9d1f176e29	res_pjsip: Copy default_from_user to avoid crash. ... The default_from_user retrieval function was pulling the default_from_user from the global configuration struct in an unsafe way. If using a database as a backend configuration store, the global configuration struct is short-lived, so grabbing a pointer from it results in referencing freed memory. The fix here is to copy the default_from_user value out of the global configuration struct. Thanks go to John Hardin for discovering this problem and proposing the patch on which this fix is based. ASTERISK-25390 #close Reported by Mark Michelson Change-Id: I6b96067a495c1259da768f4012d44e03e7c6148c	11 years ago
Mark Michelson	ac62928d6b	res_pjsip: Change default from user value. ... When Asterisk sends an outbound SIP request, if there is no direct reason to place a specific value for the username in the From header, Asterisk would generate a UUID. For example, this would happen when sending outbound OPTIONS requests when qualifying or when sending outbound INVITE requests when originating (if no explicit caller ID were provided). The issue is that some SIP providers reject these sorts of requests with a "Name too long" error response. This patch aims to fix this by changing the default outbound username in From headers to "asterisk". This value can be overridden by changing the default_from_user option in the global options if desired. ASTERISK-25377 #close Reported by Mark Michelson Change-Id: I6a4d34a56ff73ff4f661b0075aeba5461b7f3190	11 years ago
Joshua Colp	1b1561f4c8	res_pjsip_sdp_rtp: Fix multiple keepalive scheduled items. ... The keepalive support in res_pjsip_sdp_rtp currently assumes that a stream will only be negotiated once. This is false. If the stream is replaced and later added back it can be negotiated again causing multiple keepalive scheduled items to exist. This change explicitly deletes the existing keepalive scheduled item before adding the new one. The res_pjsip_sdp_rtp module also does not stop RTP keepalives or timeout timer if the stream has been replaced. This change adds a callback to the session media interface to allow a media stream to be stopped without the resources being destroyed. This allows the scheduled items and RTP to be stopped when the stream no longer exists. ASTERISK-25356 #close Change-Id: Ibe6a7cc0927c87326fd5f1c0d4ad889dbfbea1de	11 years ago
Joshua Colp	2a4eee0cd9	res_pjsip: Add common ast_sip_get_host_ip API. ... Modules commonly used the pj_gethostip function for retrieving the IP address of the host. This function does not cache the result and may result in a DNS lookup occurring, or additional work. If the DNS server is unreachable or network issues arise this can cause the pj_gethostip function to block for a period of time. This change adds an ast_sip_get_host_ip and ast_sip_get_host_ip_string function which does the same thing but caches the host IP address at module load time. This results in no additional work being done each time the local host IP address is needed. ASTERISK-25342 #close Change-Id: I3205deb679b01fa5ac05a94b623bfd620a2abe1e	11 years ago
Matt Jordan	a0f451c35e	main/format: Add an API call for retrieving format attributes ... Some codecs that may be a third party library to Asterisk need to have knowledge of the format attributes that were negotiated. Unfortunately, when the great format migration of Asterisk 13 occurred, that ability was lost. This patch adds an API call, ast_format_attribute_get, to the core format API, along with updates to the unit test to check the new API call. A new callback is also now available for format attribute modules, such that they can provide the format attribute values they manage. Note that the API returns a void *. This is done as the format attribute modules themselves may store format attributes in any particular manner they like. Care should be taken by consumers of the API to check the return value before casting and dereferencing. Consumers will obviously need to have a priori knowledge of the type of the format attribute as well. Change-Id: Ieec76883dfb46ecd7aff3dc81a52c81f4dc1b9e3	11 years ago
Scott Emidy	df9ce36366	ARI: Retrieve existing log channels ... An http request can be sent to get the existing Asterisk logs. The command "curl -v -u user:pass -X GET 'http://localhost:8088 /ari/asterisk/logging'" can be run in the terminal to access the newly implemented functionality. * Retrieve all existing log channels ASTERISK-25252 Change-Id: I7bb08b93e3b938c991f3f56cc5d188654768a808	11 years ago
Scott Emidy	e9f1bc08cb	ARI: Creating log channels ... An http request can be sent to create a log channel in Asterisk. The command "curl -v -u user:pass -X POST 'http://localhost:088/ari/asterisk/logging/mylog? configuration=notice,warning'" can be run in the terminal to access the newly implemented functionality for ARI. * Ability to create log channels using ARI ASTERISK-25252 Change-Id: I9a20e5c75716dfbb6b62fd3474faf55be20bd782	11 years ago
Scott Emidy	78364132ce	ARI: Deleting log channels ... An http request can be sent to delete a log channel in Asterisk. The command "curl -v -u user:pass -X DELETE 'http://localhost:8088 /ari/asterisk/logging/mylog'" can be run in the terminal to access the newly implemented functionally for ARI. * Able to delete log channels using ARI ASTERISK-25252 Change-Id: Id6eeb54ebcc511595f0418d586ff55914bc3aae6	11 years ago
Mark Duncan	aed068844c	res/res_rtp_asterisk: Add ECDH support ... This will add ECDH support to Asterisk. It will detect auto ECDH support in OpenSSL (1.0.2b and above) during ./configure. If this is available, it will use it, otherwise it will fall back to prime256v1 (this behavior is consistent with other projects such as Apache and nginx). This fixes WebRTC being broken in Firefox 38+ due to Firefox now only supporting ciphers with perfect forward secrecy. ASTERISK-25265 #close Change-Id: I8c13b33a2a79c0bde2e69e4ba6afa5ab9351465b	11 years ago
Joshua Colp	20ee33e22e	Merge topic 'misc_rtp_tweaks' into 13 ... * changes: rtp_engine.h: No sense allowing payload types larger than RFC allows. rtp_engine.c: Minor tweaks. rtp_engine.h: Misc comment fixes. chan_sip.c: Tweak glue->update_peer() parameter nil value.	11 years ago
Benjamin Ford	1ae762634c	ARI: Rotate log channels. ... An http request can be sent to rotate a specified log channel. If the channel does not exist, an error response will be returned. The command "curl -v -u user:pass -X PUT 'http://localhost:8088 /ari/asterisk/logging/logChannelName/rotate'" can be run in the terminal to access this new functionality. * Added the ability to rotate log files through ARI ASTERISK-25252 Change-Id: Iaefa21cbbc1b29effb33004ee3d89c977e76ab01	11 years ago
Richard Mudgett	89b21fd9a3	rtp_engine.h: No sense allowing payload types larger than RFC allows. ... * Tweaked add_static_payload() to not use magic numbers. Change-Id: I1719ff0f6d3ce537a91572501eae5bcd912a420b	11 years ago
Richard Mudgett	e20f435b60	rtp_engine.h: Misc comment fixes. ... Change-Id: If98139264d5d97427b4685ecbdc54518f725bc43	11 years ago