58 Commits (master)

Author	SHA1	Message	Date
Luz Paz	d65af637ac	docs: Fix various typos in channels/ ... Found via `codespell -q 3 -S "./CREDITS,*.po" -L abd,asent,atleast,cachable,childrens,contentn,crypted,dne,durationm,enew,exten,inout,leapyear,mye,nd,oclock,offsetp,ot,parm,parms,preceeding,pris,ptd,requestor,re-use,re-used,re-uses,ser,siz,slanguage,slin,thirdparty,varn,varns,ues`	2 months ago
Holger Hans Peter Freyther	9fc631d028	ari/pjsip: Make it possible to control transfers through ARI ... Introduce a ChannelTransfer event and the ability to notify progress to ARI. Implement emitting this event from the PJSIP channel instead of handling the transfer in Asterisk when configured. Introduce a dialplan function to the PJSIP channel to switch between the "core" and "ari-only" behavior. UserNote: Call transfers on the PJSIP channel can now be controlled by ARI. This can be enabled by using the PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.	2 months ago
Sean Bright	82bef2ed90	docs: Indent <since> tags. ... Also updates the 'since' of applications/functions that existed before XML documentation was introduced (1.6.2.0).	3 months ago
George Joseph	198b01c536	docs: Add version information to application and function XML elements ... * Do a git blame on the embedded XML application or function element. * From the commit hash, grab the summary line. * Do a git log --grep <summary> to find the cherry-pick commits in all branches that match. * Do a git patch-id to ensure the commits are all related and didn't get a false match on the summary. * Do a git tag --contains <commit> to find the tags that contain each commit. * Weed out all tags not ..0. * Sort and discard any .0.0 and following tags where the commit appeared in an earlier branch. * The result is a single tag for each branch where the application or function was defined. The applications and functions defined in the following files were done by hand because the XML was extracted from the C source file relatively recently. * channels/pjsip/dialplan_functions_doc.xml * main/logger_doc.xml * main/manager_doc.xml * res/res_geolocation/geoloc_doc.xml * res/res_stir_shaken/stir_shaken_doc.xml	3 months ago
George Joseph	114fa12f05	docs: Various XML fixes ... * channels/pjsip/dialplan_functions_doc.xml: Added xmlns:xi to docs element. * main/bucket.c: Removed XML completely since the "bucket" and "file" objects are internal only with no config file. * main/named_acl.c: Fixed the configFile element name. It was "named_acl.conf" and should have been "acl.conf" * res/res_geolocation/geoloc_doc.xml: Added xmlns:xi to docs element. * res/res_http_media_cache.c: Fixed the configFile element name. It was "http_media_cache.conf" and should have been "res_http_media_cache.conf".	3 months ago
Sean Bright	88abf2207b	dialplan_functions_doc.xml: Document PJSIP_MEDIA_OFFER's `media` argument. ... Resolves: #1023	3 months ago
Sean Bright	468601fd13	manager: Add `<since>` tags for all AMI actions.	3 months ago
Mike Bradeen	b2455b2732	res_pjsip_notify: add dialplan application ... Add dialplan application PJSIPNOTIFY to send either pre-configured NOTIFY messages from pjsip_notify.conf or with headers defined in dialplan. Also adds the ability to send pre-configured NOTIFY commands to a channel via the CLI. Resolves: #799 UserNote: A new dialplan application PJSIPNotify is now available which can send SIP NOTIFY requests from the dialplan. The pjsip send notify CLI command has also been enhanced to allow sending NOTIFY messages to a specific channel. Syntax: pjsip send notify <option> channel <channel>	9 months ago
Sean Bright	25686a57da	pjsip: Add PJSIP_PARSE_URI_FROM dialplan function. ... Various SIP headers permit a URI to be prefaced with a `display-name` production that can include characters (like commas and parentheses) that are problematic for Asterisk's dialplan parser and, specifically in the case of this patch, the PJSIP_PARSE_URI function. This patch introduces a new function - `PJSIP_PARSE_URI_FROM` - that behaves identically to `PJSIP_PARSE_URI` except that the first argument is now a variable name and not a literal URI. Fixes #756	11 months ago
George Joseph	29a273618d	pjsip show channelstats: Prevent possible segfault when faxing ... Under rare circumstances, it's possible for the original audio session in the active_media_state default_session to be corrupted instead of removed when switching to the t38/image media session during fax negotiation. This can cause a segfault when a "pjsip show channelstats" attempts to print that audio media session's rtp statistics. In these cases, the active_media_state topology is correctly showing only a single t38/image stream so we now check that there's an audio stream in the topology before attempting to use the audio media session to get the rtp statistics. Resolves: #592	1 year ago
Naveen Albert	ac71e40042	logger: Fix linking regression. ... Commit 008731b0a4 caused a regression by resulting in logger.xml being compiled and linked into the asterisk binary in lieu of logger.c on certain platforms if Asterisk was compiled in dev mode. To fix this, we ensure the file has a unique name without the extension. Most existing .xml files have been named differently from any .c files in the same directory or did not pose this issue. channels/pjsip/dialplan_functions.xml does not pose this issue but is also being renamed to adhere to this policy. Resolves: #539	1 year ago
George Joseph	f309ffad3d	chan_pjsip: Add PJSIPHangup dialplan app and manager action ... See UserNote below. Exposed the existing Hangup AMI action in manager.c so we can use all of it's channel search and AMI protocol handling without duplicating that code in dialplan_functions.c. Added a lookup function to res_pjsip.c that takes in the string represenation of the pjsip_status_code enum and returns the actual status code. I.E. ast_sip_str2rc("DECLINE") returns 603. This allows the caller to specify PJSIPHangup(decline) in the dialplan, just like Hangup(call_rejected). Also extracted the XML documentation to its own file since it was almost as large as the code itself. UserNote: A new dialplan app PJSIPHangup and AMI action allows you to hang up an unanswered incoming PJSIP call with a specific SIP response code in the 400 -> 699 range.	1 year ago
George Joseph	7dc8773178	res_rtp_asterisk: Asterisk Media Experience Score (MES) ... ----------------- This commit reinstates MES with some casting fixes to the functions in time.h that convert between doubles and timeval structures. The casting issues were causing incorrect timestamps to be calculated which caused transcoding from/to G722 to produce bad or no audio. ASTERISK-30391 ----------------- This module has been updated to provide additional quality statistics in the form of an Asterisk Media Experience Score. The score is avilable using the same mechanisms you'd use to retrieve jitter, loss, and rtt statistics. For more information about the score and how to retrieve it, see https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score * Updated chan_pjsip to set quality channel variables when a call ends. * Updated channels/pjsip/dialplan_functions.c to add the ability to retrieve the MES along with the existing rtcp stats when using the CHANNEL dialplan function. * Added the ast_debug_rtp_is_allowed and ast_debug_rtcp_is_allowed checks for debugging purposes. * Added several function to time.h for manipulating time-in-samples and times represented as double seconds. * Updated rtp_engine.c to pass through the MES when stats are requested. Also debug output that dumps the stats when an rtp instance is destroyed. * Updated res_rtp_asterisk.c to implement the calculation of the MES. In the process, also had to update the calculation of jitter. Many debugging statements were also changed to be more informative. * Added a unit test for internal testing. The test should not be run during normal operation and is disabled by default. Change-Id: I4fce265965e68c3fdfeca55e614371ee69c65038	2 years ago
George Joseph	3a3d6c7dcb	Revert "res_rtp_asterisk: Asterisk Media Experience Score (MES)" ... This reverts commit e66c5da145. Reason for revert: Issue when transcoding to/from g722 Change-Id: I12853c5b1d3a77f5b9200f41908fd238a17159dc	2 years ago
George Joseph	e66c5da145	res_rtp_asterisk: Asterisk Media Experience Score (MES) ... This module has been updated to provide additional quality statistics in the form of an Asterisk Media Experience Score. The score is avilable using the same mechanisms you'd use to retrieve jitter, loss, and rtt statistics. For more information about the score and how to retrieve it, see https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score * Updated chan_pjsip to set quality channel variables when a call ends. * Updated channels/pjsip/dialplan_functions.c to add the ability to retrieve the MES along with the existing rtcp stats when using the CHANNEL dialplan function. * Added the ast_debug_rtp_is_allowed and ast_debug_rtcp_is_allowed checks for debugging purposes. * Added several function to time.h for manipulating time-in-samples and times represented as double seconds. * Updated rtp_engine.c to pass through the MES when stats are requested. Also debug output that dumps the stats when an rtp instance is destroyed. * Updated res_rtp_asterisk.c to implement the calculation of the MES. In the process, also had to update the calculation of jitter. Many debugging statements were also changed to be more informative. * Added a unit test for internal testing. The test should not be run during normal operation and is disabled by default. ASTERISK-30280 Change-Id: I458cb9a311e8e5dc1db769b8babbcf2e093f107a	2 years ago
Naveen Albert	0c70d497bc	documentation: Adds versioning information. ... Adds version information for applications, functions, and manager events/actions. This is not completely exhaustive by any means but covers most new things added that have release versioning information in the issue tracker. ASTERISK-29940 #close Change-Id: I506401e93c799715dbbe97c0a8ba18af2bf5e131	3 years ago
Kevin Harwell	3c4a1722b6	chan_pjsip: disallow PJSIP_SEND_SESSION_REFRESH pre-answer execution ... This patch makes it so if the PJSIP_SEND_SESSION_REFRESH dialplan function is called on a channel prior to answering a warning is issued and the function returns unsuccessful. ASTERISK-28878 #close Change-Id: I053f767d10cf3b2b898fa9e3e7c35ff07e23c9bb	5 years ago
Kevin Harwell	4eba6b9eb2	PJSIP_MEDIA_OFFER: override configuration on refresh ... When using the PSJIP_MEDIA_OFFER dialplan function it was not overriding an endpoint's configured codecs on refresh unless they had a shared codec between the two. This patch makes it so whatever is set using PJSIP_MEDIA_OFFER is used when creating the SDP for a refresh no matter what. ASTERISK-28878 #close Change-Id: I0f7dc86fd0fb607c308e6f98ede303c54d1eacb6	5 years ago
Joshua C. Colp	1c5e68580a	stream: Enforce formats immutability and ensure formats exist. ... Some places in Asterisk did not treat the formats on a stream as immutable when they are. The ast_stream_get_formats function is now const to enforce this and parts of Asterisk have been updated to take this into account. Some violations of this were also fixed along the way. An additional minor tweak is that streams are now allocated with an empty format capabilities structure removing the need in various places to check that one is present on the stream. ASTERISK-28846 Change-Id: I32f29715330db4ff48edd6f1f359090458a9bfbe	5 years ago
Salah Ahmed	ddb0091da5	Crash during "pjsip show channelstats" execution ... During execution "pjsip show channelstats" cli command by an external module asterisk crashed. It seems this is a separate thread running to fetch and print rtp stats. The crash happened on the ao2_lock method, just before it going to read the rtp stats on a rtp instance. According to gdb backtrace log, it seems the session media was already cleaned up at that moment. ASTERISK-28578 Change-Id: I3e05980dd4694577be6d39be2c21a5736bae3c6f	6 years ago
Torrey Searle	b43cdc7f1e	channel/chan_pjsip: add dialplan function for music on hold ... Add a new dialplan function PJSIP_MOH_PASSTHROUGH that allows the on-hold behavior to be controlled on a per-call basis ASTERISK-28542 #close Change-Id: Iebe905b2ad6dbaa87ab330267147180b05a3c3a8	6 years ago
Alexei Gradinari	f236377ce9	pjsip: restrict function PJSIP_PARSE_URI to parse only SIP/SIPS URIs ... The next usage of PJSIP_PARSE_URI will crash asterisk ${PJSIP_PARSE_URI(tel:+1234567890,host)} or ${PJSIP_PARSE_URI(192.168.1.1:5060,host)} The function pjsip_parse_uri successfully parses then, but returns struct pjsip_other_uri *. This patch restricts parsing only SIP/SIPS URIs. Change-Id: I16f255c2b86a80a67e9f9604b94b129a381dd25e	6 years ago
sungtae kim	fb651756c7	chan_pjsip: Changed to continued after invalid media for pjsip show channelstats ... Currently, the pjsip show channelstats cli does not show channel's stats after hits the invalid channel info. This makes hard to use this cli. Changed to keep iterate after hits the invalid channel info. ASTERISK-28292 Change-Id: I3efdff1c9e1b1efd3c971fb82ae77aa133a6f43c	6 years ago
Joshua Colp	50ac85cb40	stasis: Segment channel snapshot to reduce creation cost. ... When a channel snapshot was created it used to be done from scratch, copying all data (many strings). This incurs a cost when doing so. This change segments the channel snapshot into different components which can be reused if unchanged from the previous snapshot creation, reducing the cost. In normal cases this results in some pointers being copied with reference count being bumped, some integers being set, and a string or two copied. The other benefit is that it is now possible to determine if a channel snapshot update is redundant and thus stop it before a message is published to stasis. The specific segments in the channel snapshot were split up based on whether they are changed together, how often they are changed, and their general grouping. In practice only 1 (or 0) of the segments actually get changed in normal operation. Invalidation is done by setting a flag on the channel when the segment source is changed, forcing creation of a new segment when the channel snapshot is created. ASTERISK-28119 Change-Id: I5d7ef3df963a88ac47bc187d73c5225c315f8423	7 years ago
Joshua Colp	d0ccbb3377	stasis: Use an implementation specific channel snapshot cache. ... Channels no longer use the Stasis cache for channel snapshots. Instead they are stored in a hash table in stasis_channels which reduces the number of Stasis messages created and allows better storage. As a result the following APIs are no longer available since the stasis cache is no longer used: ast_channel_topic_cached() ast_channel_topic_all_cached() The ast_channel_cache_all() and ast_channel_cache_by_name() functions now return an ao2_container of ast_channel_snapshots rather than a container of stasis_messages therefore you can't (and don't need to) call stasis_cache functions on it. The ast_channel_topic_all() function now returns a normal topic not a cached one so you can't use stasis cache functions on it either. The ast_channel_snapshot_type() stasis message now has the ast_channel_snapshot_update structure as it's data. It contains the last snapshot and the new one. ast_channel_snapshot_get_latest() still returns the latest snapshot. The latest snapshot is now stored on the channel itself to eliminate cache hits when Stasis messages that have the snapshot as a payload are created. ASTERISK-28102 Change-Id: I9334febff60a82d7c39703e49059fa3a68825786	7 years ago
Alexei Gradinari	fa048183aa	pjsip: New function PJSIP_PARSE_URI to parse URI and return part of URI ... New dialplan function PJSIP_PARSE_URI added to parse an URI and return a specified part of the URI. This is useful when need to get part of the URI instead of cutting it using a CUT function. For example to get 'user' part of Remote URI ${PJSIP_PARSE_URI(${CHANNEL(pjsip,remote_uri)},user)} ASTERISK-28144 #close Change-Id: I5d828fb87f6803b6c1152bb7b44835f027bb9d5a	7 years ago
Salah Ahmed	a90177cd63	dialplan_functions: wrong srtp use status report of a dialplan function ... If asterisk offer an endpoint with SRTP and that endpoint respond with non srtp, in that case channel(rtp,secure,audio) reply wrong status. Why delete flag AST_SRTP_CRYPTO_OFFER_OK while check identical remote_key: Currently this flag has being set redundantly. In either case identical or different remote_key this flag has being set. So if we don't set it while we receive identical remote_key or non SRTP SDP response then we can take decision of srtp use by using that flag. ASTERISK-27999 Change-Id: I29dc2843cf4e5ae2604301cb4ff258f1822dc2d7	7 years ago
Richard Mudgett	237d341bbd	res_pjsip.c: Split ast_sip_push_task_synchronous() to fit expectations. ... ast_sip_push_task_synchronous() did not necessarily execute the passed in task under the specified serializer. If the current thread is any registered pjsip thread then it would execute the task immediately instead of under the specified serializer. Reentrancy issues could result if the task does not execute with the right serializer. The original reason ast_sip_push_task_synchronous() checked to see if the current thread was a registered pjsip thread was because of a deadlock with masquerades and the channel technology's fixup callback (ASTERISK_22936). A subsequent masquerade deadlock fix (ASTERISK_24356) involving call pickups avoided the original deadlock situation entirely. The PJSIP channel technology's fixup callback no longer needed to call ast_sip_push_task_synchronous(). However, there are a few places where this unexpected behavior is still required to avoid deadlocks. The pjsip monitor thread executes callbacks that do calls to ast_sip_push_task_synchronous() that would deadlock if the task were actually pushed to the specified serializer. I ran into one dealing with the pubsub subscriptions where an ao2 destructor called ast_sip_push_task_synchronous(). * Split ast_sip_push_task_synchronous() into ast_sip_push_task_wait_servant() and ast_sip_push_task_wait_serializer(). ast_sip_push_task_wait_servant() has the old behavior of ast_sip_push_task_synchronous(). ast_sip_push_task_wait_serializer() has the new behavior where the task is always executed by the specified serializer or a picked serializer if one is not passed in. Both functions behave the same if the current thread is not a SIP servant. * Redirected ast_sip_push_task_synchronous() to ast_sip_push_task_wait_servant() to preserve API for released branches. ASTERISK_26806 Change-Id: Id040fa42c0e5972f4c8deef380921461d213b9f3	7 years ago
Richard Mudgett	22810fc635	chan_pjsip/res_pjsip: Add CHANNEL(pjsip,request_uri) ... This patch does three things associated with the initial incoming INVITE request URI. 1) Add access to the full initial incoming INVITE request URI. 2) We were not setting DNID on incoming PJSIP channels. The DNID is the user portion of the initial incoming INVITE Request-URI. The value is accessed by reading CALLERID(dnid). 3) Fix CHANNEL(pjsip,target_uri) documentation. * The initial incoming INVITE request URI is now available using CHANNEL(pjsip,request_uri). * Set the DNID on PJSIP channel creation so CALLERID(dnid) can return the initial incoming INVITE request URI user portion. * CHANNEL(pjsip,target_uri) now correctly documents that the target URI is the contact URI. * Refactored print_escaped_uri() out of channel_read_pjsip() to handle pjsip_uri_print() error condition when the buffer is too small. ASTERISK-27478 Change-Id: I512e60d1f162395c946451becb37af3333337b33	7 years ago
Andre Nazario	71be8d5bbe	chan_pjsip: Add tag info in CHANNEL function ... Create local_tag and remote_tag in CHANNEL info to get tag from From and To headers of a SIP dialog. ASTERISK-27220 Change-Id: I59b16c4b928896fcbde02ad88f0e98922b15d524	8 years ago
Torrey Searle	33a648d4c6	res/res_pjsip_session: allow SDP answer to be regenerated ... If an SDP answer hasn't been sent yet, it's legal to change it. This is required for PJSIP_DTMF_MODE to work correctly, and can also have use in the future for updating codecs too. ASTERISK-27209 #close Change-Id: Idbbfb7cb3f72fbd96c94d10d93540f69bd51e7a1	8 years ago
Torrey Searle	65c560894d	chan_pjsip: add a new function PJSIP_DTMF_MODE ... This function is a replica of SIPDtmfMode, allowing the DTMF mode of a PJSIP call to be modified on a per-call basis ASTERISK-27085 #close Change-Id: I20eef5da3e5d1d3e58b304416bc79683f87e7612	8 years ago
Mark Michelson	45df25a579	chan_pjsip: Add support for multiple streams of the same type. ... The stream topology (list of streams and order) is now stored with the configured PJSIP endpoints and used during the negotiation process. Media negotiation state information has been changed to be stored in a separate object. Two of these objects exist at any one time on a session. The active media state information is what was previously negotiated and the pending media state information is what the media state will become if negotiation succeeds. Streams and other state information is stored in this object using the index (or position) of each individual stream for easy lookup. The ability for a media type handler to specify a callback for writing has been added as well as the ability to add file descriptors with a callback which is invoked when data is available to be read on them. This allows media logic to live outside of the chan_pjsip module. Direct media has been changed so that only the first audio and video stream are directly connected. In the future once the RTP engine glue API has been updated to know about streams each individual stream can be directly connected as appropriate. Media negotiation itself will currently answer all the provided streams on an offer within configured limits and on an offer will use the topology created as a result of the disallow/allow codec lines. If a stream has been removed or declined we will now mark it as such within the resulting SDP. Applications can now also request that the stream topology change. If we are told to do so we will limit any provided formats to the ones configured on the endpoint and send a re-invite with the new topology. Two new configuration options have also been added to PJSIP endpoints: max_audio_streams: determines the maximum number of audio streams to offer/accept from an endpoint. Defaults to 1. max_video_streams: determines the maximum number of video streams to offer/accept from an endpoint. Defaults to 1. ASTERISK-27076 Change-Id: I8afd8dd2eb538806a39b887af0abd046266e14c7	8 years ago
Richard Mudgett	11ec2945c7	chan_pjsip: Fix PJSIP_MEDIA_OFFER dialplan function read. ... The construction of the returned string assumed incorrectly that the supplied buffer would always be initialized as an empty string. If it is not an empty string we could overrun the supplied buffer by the length of the non-empty buffer string plus one. It is also theoreticaly possible for the supplied buffer to be overrun by a string terminator during a read operation even if the supplied buffer is an empty string. * Fix the assumption that the supplied buffer would already be an empty string. The buffer is not guaranteed to contain an empty string by all possible callers. * Fix string terminator buffer overrun potential. Change-Id: If6a0806806527678c8554b1dcb34fd7808aa95c9	8 years ago
Joshua Colp	861984eac0	res_pjsip: Add support for returning only reachable contacts and use it. ... This introduces the ability for PJSIP code to specify filtering flags when retrieving PJSIP contacts. The first flag for use causes the query code to only retrieve contacts that are not unreachable. This change has been leveraged by both the Dial() process and the PJSIP_DIAL_CONTACTS dialplan function so they will now only attempt calls to contacts which are not unreachable. ASTERISK-26281 Change-Id: I8233b4faa21ba3db114f5a42e946e4b191446f6c	8 years ago
Joshua Colp	12460b05c1	chan_pjsip: Don't assume a session will have a channel. ... When querying for PJSIP specific information using the dialplan function CHANNEL() it is possible that the underlying session will no longer have a channel associated with it. This is most likely to occur when the RTCP HEP module attempts to get the channel name. If this happens then a crash will occur. This change just adds a check that the channel exists on the session before querying it. ASTERISK-26857 Change-Id: I113479cffff6ae64cf8ed089e9e1565223426f01	8 years ago
Daniel Journo	36fed72614	pjsip/cli_commands: pjsip show channelstats shows wrong codec ... * cli_commands.c Fixed CLI output ASTERISK-26822 #close Change-Id: I3889ef6a8f6738fc312fab42db5efacd6e452b01	8 years ago
Joshua Colp	ae57652983	chan_pjsip: Use session for retrieving CHANNEL() information. ... The CHANNEL() dialplan function implementation for PJSIP allows querying of PJSIP specific information. This used the channel passed in to get the PJSIP session and associated information. It is possible for this channel to be masqueraded and end up as a different channel type by the time the information request is actually acted upon. This change retrieves the PJSIP session safely and accesses data from it (including channel). This provides a guarantee that the session and channel will not be altered when the request is being acted upon. ASTERISK-26673 Change-Id: I335e12b89e1820cafdd92b3e7526b8ba649eb7e6	8 years ago
Corey Farrell	a6e5bae3ef	Remove ASTERISK_REGISTER_FILE. ... ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes all traces of it. Previously exported symbols removed: * __ast_register_file * __ast_unregister_file * ast_complete_source_filename This also removes the mtx_prof static variable that was declared when MTX_PROFILE was enabled. This variable was only used in lock.c so it is now initialized in that file only. ASTERISK-26480 #close Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966	9 years ago
Corey Farrell	824a4e84d1	Refactor usage pattern of xmldoc info tag. ... This updates func_channel.c and main/message.c to use a generic xpointer include instead of including info from each channel driver. Now the name attribute of info is CHANNEL or CHANNEL_EXAMPLES to be included in documentation for func_channel. Setting the name attribute of info to MessageToInfo or MessageFromInfo causes it to be included in the MessageSend application and AMI action. Change-Id: I89fd8276a3250824241a618009714267d3a8d1ea	9 years ago
Matt Jordan	ddab42e296	func_channel: Reorganize documentation ... * Following the example of the PJSIP channel driver, the channel technology specific documentation has been moved to the respective channel drivers that provide that functionality. This has the benefit of locating the documentation of items with those modules that provide it. * Examples of using the CHANNEL function for both standard items as well as for PJSIP have been added. * The 'max_forwards' standard item has been documented. Change-Id: Ifaa79a232c8ac99cf8da6ef6cc7815d398b1b79b	9 years ago
Matt Jordan	c315460abb	channels/chan_pjsip: Add PJSIP_SEND_SESSION_REFRESH ... This patch adds a new PJSIP specific dialplan function, PJSIP_SEND_SESSION_REFRESH. When invoked on a PJSIP channel, the media session will be refreshed via either an UPDATE or re-INVITE request. When used in conjunction with the PJSIP_MEDIA_OFFER dialplan function, the formats in use on a PJSIP channel can be re-negotiated and changed dynamically after call setup. ASTERISK-26277 #close Change-Id: Ib98fe09ba889aafe26d58d32f0fd1323f8fd9b1b (cherry picked from commit eec60dd773)	9 years ago
George Joseph	c4064727d2	chan_pjsip: Add 'pjsip show channelstats' ... Added the ability to show channel statistics to chan_pjsip (cli_functions.c) Moved the existing 'pjsip show channel(s)' functionality from pjsip_configuration to cli_functions.c. The stats needed chan_pjsip's private header so it made sense to move the existing channel commands as well. Now using stasis_cache_dump to get the channel snapshots rather than retrieving all endpoints, then getting each one's channel snapshots. Much more efficient. Change-Id: I03b114522126d27434030b285bf6d531ddd79869	9 years ago
George Joseph	d2eb65f71e	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
Matt Jordan	86b02228f5	channels/pjsip/dialplan_functions: Add an option for extracting the SIP call-id ... This patch adds a new option to the CHANNEL function that allows for the extraction of the SIP call-id. It is used in conjunction with the 'pjsip' option, and will return the Call-ID of the INVITE request that established the PJSIP channel. ASTERISK-25352 Change-Id: I278d1f8bcfe3a53c5aa1dadebc14e92b0abd476a	10 years ago
Matt Jordan	4a58261694	git migration: Refactor the ASTERISK_FILE_VERSION macro ... Git does not support the ability to replace a token with a version string during check-in. While it does have support for replacing a token on clone, this is somewhat sub-optimal: the token is replaced with the object hash, which is not particularly easy for human consumption. What's more, in practice, the source file version was often not terribly useful. Generally, when triaging bugs, the overall version of Asterisk is far more useful than an individual SVN version of a file. As a result, this patch removes Asterisk's support for showing source file versions. Specifically, it does the following: * Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and remove passing the version in with the macro. Other facilities than 'core show file version' make use of the file names, such as setting a debug level only on a specific file. As such, the act of registering source files with the Asterisk core still has use. The macro rename now reflects the new macro purpose. * main/asterisk: - Refactor the file_version structure to reflect that it no longer tracks a version field. - Remove the "core show file version" CLI command. Without the file version, it is no longer useful. - Remove the ast_file_version_find function. The file version is no longer tracked. - Rename ast_register_file_version/ast_unregister_file_version to ast_register_file/ast_unregister_file, respectively. * main/manager: Remove value from the Version key of the ModuleCheck Action. The actual key itself has not been removed, as doing so would absolutely constitute a backwards incompatible change. However, since the file version is no longer tracked, there is no need to attempt to include it in the Version key. * UPGRADE: Add notes for: - Modification to the ModuleCheck AMI Action - Removal of the "core show file version" CLI command Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e	10 years ago
Matthew Jordan	ea0098724e	clang compiler warnings: Fix autological comparisons ... This fixes autological comparison warnings in the following: * chan_skinny: letohl may return a signed or unsigned value, depending on the macro chosen * func_curl: Provide a specific cast to CURLoption to prevent mismatch * cel: Fix enum comparisons where the enum can never be negative * enum: Fix comparison of return result of dn_expand, which returns a signed int value * event: Fix enum comparisons where the enum can never be negative * indications: tone_data.freq1 and freq2 are unsigned, and hence can never be negative * presencestate: Use the actual enum value for INVALID state * security_events: Fix enum comparisons where the enum can never be negative * udptl: Don't bother to check if the return value from encode_length is less than 0, as it returns an unsigned int * translate: Since the parameters are unsigned int, don't bother checking to see if they are negative. The cast to unsigned int would already blow past the matrix bounds. * res_pjsip_exten_state: Use a temporary value to cache the return of ast_hint_presence_state * res_stasis_playback: Fix enum comparisons where the enum can never be negative * res_stasis_recording: Add an enum value for the case where the recording operation is in error; fix enum comparisons * resource_bridges: Use enum value as opposed to -1 * resource_channels: Use enum value as opposed to -1 Review: https://reviewboard.asterisk.org/r/4533 ASTERISK-24917 Reported by: dkdegroot patches: rb4533.patch submitted by dkdegroot (License 6600) ........ Merged revisions 434469 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 434470 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434471 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Joshua Colp	cc96e4a7ef	Multiple revisions 431751-431752 ... ........ r431751 | file | 2015-02-14 14:19:07 -0400 (Sat, 14 Feb 2015) | 5 lines chan_pjsip: Fix crash when CHANNEL dialplan function is invoked with pjsip argument and no type. ASTERISK-24771 #close Reported by: Niklas Larsson ........ r431752 | file | 2015-02-14 14:20:27 -0400 (Sat, 14 Feb 2015) | 2 lines 'information' ends with an 'n'. ........ Merged revisions 431751-431752 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431753 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Joshua Colp	0e631a541d	chan_pjsip: Add configure check for 'pjsip_get_dest_info' function. ... The 'pjsip_get_dest_info' function is used to determine if the signaling transport of the dialog is secure or not. This function was added in PJSIP 2.3 and does not exist in earlier versions. This configure check allows Asterisk to build and run with older versions at the loss of the 'secure' argument for the PJSIP CHANNEL dialplan function. Usage of this argument will require upgrading to PJSIP 2.3. ASTERISK-24665 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4329/ ........ Merged revisions 430546 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430547 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Joshua Colp	f7cf988a82	pjsip: Add 'PJSIP_AOR' and 'PJSIP_CONTACT' dialplan functions. ... The PJSIP_AOR dialplan function allows inspection of configured AORs including what contacts are currently bound to them. The PJSIP_CONTACT dialplan function allows inspection of contacts in existence. These can include both externally added (by way of registration) or permanent ones. ASTERISK-24341 Reported by: xrobau Review: https://reviewboard.asterisk.org/r/4308/ ........ Merged revisions 430179 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430180 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago