asterisk

Commit Graph

Author	SHA1	Message	Date
George Joseph	2b3261cd36	res_pjsip_mwi: Allow subscribe to vm access extension as an alias Background: If your extension is 1000 and the voicemail access extension is 1571 and you dial 1571, usually a dialplan rule calls voicemailmain with your extension and you are placed directly in your mailbox. Therefore most admins program the voicemail (or other speed dial) button on their phones to the access extension. Some phones (Snom at least) use whatever is programmed there to also subscribe for MWI and so can't dial one number and subscribe to another. This works fine in chan_sip because chan_sip completely ignores the user portion of the SUBSCRIBE message request URI. If it can match the peer, is subscribes to the peer's mailbox. The user could be set to anything or nothing and you'd still get subscribed to your mailbox. Issue: chan_pjsip actually uses the user portion of the URI to find an aor and its mailboxes. Therefore a subscribe to 1571 results in a 404. Sure, you can create an aor for 1571 but you certainly can't add your entire voicemail system's mailboxes to it and everyone would get notified of every MWI. Solution: When an MWI subscribe comes in and an aor can't be found that matches the resource directly, check the resource against the endpoint's aors. If an aor is found that has a voicemail_extension that matches the resource, use it. ASTERISK-25865 Reported-by: Ross Beer Change-Id: I770ea185f751f1ada888fafb4b452115f1c06e9e	10 years ago
George Joseph	e2524fcee3	res_pjsip_mwi: Add voicemail extension and mwi_subscribe_replaces_unsolicited res_pjsip_mwi was missing the chan_sip "vmexten" functionality which adds the Message-Account header to the MWI NOTIFY. Also, specifying mailboxes on endpoints for unsolicited mwi and on aors for subscriptions required that the admin know in advance which the client wanted. If you specified mailboxes on the endpoint, subscriptions were rejected even if you also specified mailboxes on the aor. Voicemail extension: * Added a global default_voicemail_extension which defaults to "". * Added voicemail_extension to both endpoint and aor. * Added ast_sip_subscription_get_dialog for support. * Added ast_sip_subscription_get_sip_uri for support. When an unsolicited NOTIFY is constructed, the From header is parsed, the voicemail extension from the endpoint is substituted for the user, and the result placed in the Message-Account field in the body. When a subscribed NOTIFY is constructed, the subscription dialog local uri is parsed, the voicemail_extension from the aor (looked up from the subscription resource name) is substituted for the user, and the result placed in the Message-Account field in the body. If no voicemail extension was defined, the Message-Account field is not added to the NOTIFY body. mwi_subscribe_replaces_unsolicited: * Added mwi_subscribe_replaces_unsolicited to endpoint. The previous behavior was to reject a subscribe if a previous internal subscription for unsolicited MWI was found for the mailbox. That remains the default. However, if there are mailboxes also set on the aor and the client subscribes and mwi_subscribe_replaces_unsolicited is set, the existing internal subscription is removed and replaced with the external subscription. This allows an admin to configure mailboxes on both the endpoint and aor and allows the client to select which to use. ASTERISK-25865 #close Reported-by: Ross Beer Change-Id: Ic15a9415091760539c7134a5ba3dc4a6a1217cea	10 years ago
zuul	23d2a561d5	Merge "res_pjsip/pjsip_options: Fix From generation on outgoing OPTIONS"	10 years ago
George Joseph	724b9ab28f	res_rtp_asterisk: Fix placement of txcount increment Commit `1bce690ccb` was incrementing txcount for rtcp packets as well as rtp packets and that was causing sender reports to be generated instead of receiver reports in cases where no rtp was actually being sent. Moved the txcount increment from __rtp_sento, which handles both rtp and rtcp, to rtp_sento which only handles rtp packets. Discovered by the hep/rtcp-receiver test. Change-Id: Ie442e4bb947a68847a676497021ba10ffaf376d5	10 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	10 years ago
Joshua Colp	03845666da	Merge "res_rtp_asterisk: Fix packet stats on bridged connection"	10 years ago
zuul	2a7bc6b2aa	Merge "sorcery/res_pjsip: Refactor for realtime performance"	10 years ago
Jacek Konieczny	970803efcb	res_rtp_asterisk: Use separate SRTP session for RTCP with DTLS Asterisk uses separate UDP ports for RTP and RTCP traffic and RFC 5764 explicitly states: There MUST be a separate DTLS-SRTP session for each distinct pair of source and destination ports used by a media session This means RTP keying material cannot be used for DTLS RTCP, which was the reason why RTCP encryption would fail. ASTERISK-25642 Change-Id: I7e8779d8b63e371088081bb113131361b2847e3a	10 years ago
Jacek Konieczny	9785e8d090	app_echo: forward and generate VIDUPDATE frames When using app_echo via WebRTC with VP8 video the video would appear only after a few minutes, because there would be nothing to request a full reference frame. This fixes the problem in both ways: - echos any VIDUPDATE frames received on the channel - sends one such frame when first video frame is to be forwarded This makes the echo work with Firefox and Chrome WebRTC implementation. ASTERISK-25867 #close Change-Id: I73bda87bf7532ee8bfb28d917045a21034908c1e	10 years ago
Joshua Colp	6ce25bd62a	Merge "res_parking: Misc fixes."	10 years ago
George Joseph	44ffb5105a	res_rtp_asterisk: Fix packet stats on bridged connection rxcount, txcount, rxoctetcount and txoctetcount weren't being calculated for bridged streams because the calulations were being done after the bridged short-circuit. Actually, rxoctetcount wasn't ever being calculated. Moved the calculations so they occur for all valid received packets and all transmitted packets. Also added rxoctetcount and txoctetcount to ast_rtp_instance_stat. Change-Id: I08fb06011a82d38c3b4068867a615068fbe59cbb	10 years ago
George Joseph	c971a64366	res_pjsip/pjsip_options: Fix From generation on outgoing OPTIONS No one seemed to notice but every time an OPTIONS goes out, it goes out with a From of "asterisk" (or whatever the default from_user is set to), even if you specify an endpoint. The issue had several causes... qualify_contact is only called with an endpoint if called from the CLI. If the endpoint is NULL, qualify_contact only looks up the endpoint if authenticate_qualify=yes. Even then, it never passes it on to ast_sip_create_request where the From header is set. Therefore From is always "asterisk" (or whatever the default from_user is set to). Even if ast_sip_create_request were to get an endpoint, it only sets the From if endpoint->from_user is set. The fix is 4 parts... First, create_out_of_dialog_request was modified to use the endpoint id if endpoint was specified and from_user is not set. Second, qualify_contact was modified to always look up an endpoint if one wasn't specified regardless of authenticate_qualify. It then passes the endpoint on to create_out_of_dialog_request. Third (and most importantly), find_an_endpoint was modified to find an endpoint by using an "aors LIKE %contact->aor%" predicate with ast_sorcery_retrieve_by_fields. As such, this patch will only work if the sorcery realtime optimizations patch goes in. Otherwise we'd be pulling the entire endpoints database every time we send an OPTIONS. Since we already know the contact's aor, the on_endpoint callback was also modified to just check if the contact->aor is an exact match to one of the endpoint's. Finally, since we now have an endpoint for every OPTIONS request, res_pjsip/endpt_send_request (which handles out-of-dialog reqests) was updated to get the transport from the endpoint and set it on tdata. Now the correct transport is used. Change-Id: I2207e12bb435e373bd1e03ad091d82e5aba011af	10 years ago
George Joseph	c948ce9651	sorcery/res_pjsip: Refactor for realtime performance There were a number of places in the res_pjsip stack that were getting all endpoints or all aors, and then filtering them locally. A good example is pjsip_options which, on startup, retrieves all endpoints, then the aors for those endpoints, then tests the aors to see if the qualify_frequency is > 0. One issue was that it never did anything with the endpoints other than retrieve the aors so we probably could have skipped a step and just retrieved all aors. But nevermind. This worked reasonably well with local config files but with a realtime backend and thousands of objects, this was a nightmare. The issue really boiled down to the fact that while realtime supports predicates that are passed to the database engine, the non-realtime sorcery backends didn't. They do now. The realtime engines have a scheme for doing simple comparisons. They take in an ast_variable (or list) for matching, and the name of each variable can contain an operator. For instance, a name of "qualify_frequency >" and a value of "0" would create a SQL predicate that looks like "where qualify_frequency > '0'". If there's no operator after the name, the engines add an '=' so a simple name of "qualify_frequency" and a value of "10" would return exact matches. The non-realtime backends decide whether to include an object in a result set by calling ast_sorcery_changeset_create on every object in the internal container. However, ast_sorcery_changeset_create only does exact string matches though so a name of "qualify_frequency >" and a value of "0" returns nothing because the literal "qualify_frequency >" doesn't match any name in the objset set. So, the real task was to create a generic string matcher that can take a left value, operator and a right value and perform the match. To that end, strings.c has a new ast_strings_match(left, operator, right) function. Left and right are the strings to operate on and the operator can be a string containing any of the following: = (or NULL or ""), !=, >, >=, <, <=, like or regex. If the operator is like or regex, the right string should be a %-pattern or a regex expression. If both left and right can be converted to float, then a numeric comparison is performed, otherwise a string comparison is performed. To use this new function on ast_variables, 2 new functions were added to config.c. One that compares 2 ast_variables, and one that compares 2 ast_variable lists. The former is useful when you want to compare 2 ast_variables that happen to be in a list but don't want to traverse the list. The latter will traverse the right list and return true if all the variables in it match the left list. Now, the backends' fields_cmp functions call ast_variable_lists_match instead of ast_sorcery_changeset_create and they can now process the same syntax as the realtime engines. The realtime backend just passes the variable list unaltered to the engine. The only gotcha is that there's no common realtime engine support for regex so that's been noted in the api docs for ast_sorcery_retrieve_by_fields. Only one more change to sorcery was done... A new config flag "allow_unqualified_fetch" was added to reg_sorcery_realtime. "no": ignore fetches if no predicate fields were supplied. "error": same as no but emit an error. (good for testing) "yes": allow (the default); "warn": allow but emit a warning. (good for testing) Now on to res_pjsip... pjsip_options was modified to retrieve aors with qualify_frequency > 0 rather than all endpoints then all aors. Not only was this a big improvement in realtime retrieval but even for config files there's an improvement because we're not going through endpoints anymore. res_pjsip_mwi was modified to retieve only endpoints with something in the mailboxes field instead of all endpoints then testing mailboxes. res_pjsip_registrar_expire was completely refactored. It was retrieving all contacts then setting up scheduler entries to check for expiration. Now, it's a single thread (like keepalive) that periodically retrieves only contacts whose expiration time is < now and deletes them. A new contact_expiration_check_interval was added to global with a default of 30 seconds. Ross Beer reports that with this patch, his Asterisk startup time dropped from around an hour to under 30 seconds. There are still objects that can't be filtered at the database like identifies, transports, and registrations. These are not going to be anywhere near as numerous as endpoints, aors, auths, contacts however. Back to allow_unqualified_fetch. If this is set to yes and you have a very large number of objects in the database, the pjsip CLI commands will attempt to retrive ALL of them if not qualified with a LIKE. Worse, if you type "pjsip show endpoint <tab>" guess what's going to happen? :) Having a cache helps but all the objects will have to be retrieved at least once to fill the cache. Setting allow_unqualified_fetch=no prevents the mass retrieve and should be used on endpoints, auths, aors, and contacts. It should NOT be used for identifies, registrations and transports since these MUST be retrieved in bulk. Example sorcery.conf: [res_pjsip] endpoint=config,pjsip.conf,criteria=type=endpoint endpoint=realtime,ps_endpoints,allow_unqualified_fetch=error ASTERISK-25826 #close Reported-by: Ross Beer Tested-by: Ross Beer Change-Id: Id2691e447db90892890036e663aaf907b2dc1c67	10 years ago
Joshua Colp	77a9272431	Merge "res_parking: Fix blind transfer dynamic lots creation."	10 years ago
zuul	36347dbc3c	Merge "res_parking: Cleanup find_channel_parking_lot_name() usage."	10 years ago
Richard Mudgett	8e8cf80cea	res_parking: Fix blind transfer dynamic lots creation. Blind transfers to a recognized parking extension need to use the parker's channel variable values to create the dynamic parking lot. This is because there is always only one parker while the parkee may actually be a multi-party bridge. A multi-party bridge can never supply the needed channel variables to create the dynamic parking lot. In the multi-party bridge blind transfer scenario, the parker's CHANNEL(parkinglot) value and channel variables are inherited by the local channel used to park the bridge. * In park_common_setup(), make use the parker instead of the parkee to supply the dynamic parking lot channel variable values. In all but one case, the parkee is the same as the parker. However, in the recognized parking extension blind transfer scenario for a two party bridge they are different channels. For consistency, we need to use the parker channel. * In park_local_transfer(), pass the CHANNEL(parkinglot) value to the local channel when blind transferring a multi-party bridge to a recognized parking extension. * When a local channel starts a call, the Local;2 side needs to inherit the CHANNEL(parkinglot) value from Local;1. The DTMF one-touch parking case wasn't even trying to create dynamic parking lots before it aborted the attempt. * In parking_park_call(), add missing code to create a dynamic parking lot. A DTMF bridge hook is documented as returning -1 to remove the hook. Though the hook caller is really coded to accept non-zero. See the ast_bridge_hook_callback typedef. * In feature_park_call(), don't remove the DTMF one-touch parking hook because of an error. ASTERISK-24605 #close Reported by: Philip Correia Patches: call_park.patch (license #6672) patch uploaded by Philip Correia Change-Id: I221d3a8fcc181877a1158d17004474d35d8016c9	10 years ago
Richard Mudgett	3cf714031c	res_parking: Cleanup find_channel_parking_lot_name() usage. Change-Id: I8f7a8890aef27824301c642d4d15407ac83e6f02	10 years ago
Richard Mudgett	13e75ee04f	res_parking: Misc fixes. res/parking/parking_applications.c: * Add malloc fail checks in setup_park_common_datastore(). * Fix playing parking failed announcement to only happen on non-blind transfers in park_app_exec(). It could never go out before because a test was provedly always false. res/parking/parking_bridge.c: * Fix NULL tolerance in generate_parked_user() because bridge_parking_push() can theoretically pass a NULL parker channel if the parker channel went away for some reason. * Clarify some weird code dealing with blind_transfer in bridge_parking_push(). res/parking/parking_bridge_features.c: * Made park_local_transfer() set BLINDTRANSFER on the Local;1 channel which will be bulk copied to the Local;2 channel on the subsequent ast_call(). The additional advantage is if the parker channel has the BLINDTRANSFER and ATTENDEDTRANSFER variables set they are now guaranteed to be overridden. res/parking/parking_manager.c: * Fix AMI Park action input range checking of the Timeout header in manager_park(). * Reduced locking scope to where needed in manager_park(). res/res_parking.c: * Fix some off nominal missing unlocks by eliminating the returns. Change-Id: Ib64945bc285acb05a306dc12e6f16854898915ca	10 years ago
Philip Correia	e2853ae337	res_parking: Update parking documentation for dynamic parking lots. * Remove duplicate res_parking.conf courtesytone config option documentation. ASTERISK-24596 #close Reported by: Philip Correia ASTERISK-24605 Reported by: Philip Correia Patches: call_park_app_doc.patch (license #6672) patch uploaded by Philip Correia Change-Id: I90a92a891c6494dc08173e675856afcc4764c5b5	10 years ago
zuul	1555cf8951	Merge "pjproject-bundled: Cleanups for reported issues"	10 years ago
zuul	2600a42e80	Merge "progdocs: Exclude ./third-party from documentation generation"	10 years ago
zuul	c0b802c575	Merge "tests/test_http_media_cache: Fix file descriptor leak in test."	10 years ago
zuul	4cf7458c2e	Merge "core/logging: Fix broken syslog levels on older glibc."	10 years ago
Joshua Colp	ffe345cf6e	Merge "media_cache: Demote warning to debug as it may occur often."	10 years ago
Joshua Colp	72a897c534	media_cache: Demote warning to debug as it may occur often. The file playback system will now query the media cache and then the old file functionality. Under normal conditions this will result in the cache failing to retrieve a file causing a warning message to get output each time a file is played back. This change demotes this warning to a debug message. Change-Id: Ib72246ba300b5cce32774bfb3c26634bfb708624	10 years ago
Joshua Colp	9462a08aee	Merge "musiconhold: Only warn if music class is not found in memory and database."	10 years ago
Mark Michelson	89e94e886c	Restrict CLI/AMI commands on shutdown. During stress testing, we have frequently seen crashes occur because a CLI or AMI command attempts to access information that is in the process of being destroyed. When addressing how to fix this issue, we initially considered fixing individual crashes we observed. However, the changes required to fix those problems would introduce considerable overhead to the nominal case. This is not reasonable in order to prevent a crash from occurring while Asterisk is already shutting down. Instead, this change makes it so AMI and CLI commands cannot be executed if Asterisk is being shut down. For AMI, this is absolute. For CLI, though, certain commands can be registered so that they may be run during Asterisk shutdown. ASTERISK-25825 #close Change-Id: I8887e215ac352fadf7f4c1e082da9089b1421990	10 years ago
Alexander Traud	3f720155b7	chan_sip: Do not send all codecs on INVITE. Do not break on Session-Timers. Asterisk 13.7.0 included a fix for ASTERISK-24543, not to send all those codecs, which the caller did not request/support. That fix was not complete because on the second Session Timer all codecs were sent again. Some VoIP/SIP clients interpreted that complete codec-list as a change in the SIP session. Because of that, Asterisk did not send the RTP audio via NAT anymore which created a non-audio scenario after the second Session Timer fired. ASTERISK-24543 #close Change-Id: I1881827816ab7fd47eb4287a95961179b34a0b66	10 years ago
Gianluca Merlo	894071ea2c	config: fix flags in uint option handler The configuration unsigned integer option handler sets flags for the parser as if the option should be a signed integer (PARSE_INT32), leading to errors on "out of range" values. Fix flags (PARSE_UINT32). A fix to res_pjsip is also present which stops invalid flags from being passed when registering sorcery object fields for qualify status. ASTERISK-25612 #close Change-Id: I96b539336275e0e72a8e8033487d2c3344debd3e	10 years ago
Walter Doekes	13cdf3e8a1	musiconhold: Only warn if music class is not found in memory and database. The log message when a MusicOnHold music class was not found was changed from debug level to WARNING level in Asterisk 11.19 and 13.5. For those using realtime musiconhold, this message is wrong because it warns before checking the database. This changeset delays the warning until after the database has been checked. Reported-by: Conrad de Wet ASTERISK-25444 #close Change-Id: I6cfb2db2f9cfbd2bb3d30566ecae361c4abf6dbf	10 years ago
Walter Doekes	87c9ab97ea	core/logging: Fix broken syslog levels on older glibc. The fix to ASTERISK-25407 introduced the usage of LOG_MAKEPRI. However this macro is broken in older glibc (< 2.17); it would left-shift the facility a second time, causing the resultant priority to become invalid. The syslog manpage mentions nothing about LOG_MAKEPRI and suggests this: The priority argument is formed by ORing the facility and the level values [...]. ASTERISK-25510 #close Reported by: Michael Newton Change-Id: Ia89debe7fac5ad090c7ef595c0707f31bb1e3d03	10 years ago
Joshua Colp	a72f3b5bb4	tests/test_http_media_cache: Fix file descriptor leak in test. Change-Id: Ie8a9ae3d13bdeaacafc8d28271adc6707f633a5f	10 years ago
Joshua Colp	d7ee89b499	Merge "main/app: Only look to end of file if ':end' is specified, and not just ':'"	10 years ago
Joshua Colp	b1c688acab	Merge "main/file: Add the ability to play media in the media cache"	10 years ago
zuul	d82e3a1f28	Merge "tests/test_http_media_cache: Add unit tests for res_http_media_cache"	10 years ago
zuul	58c4d48e11	Merge "res/res_http_media_cache: Add an HTTP(S) backend for the core media cache"	10 years ago
zuul	88cc68d9ed	Merge "main/media_cache: Provide an extension on the local file associated with a URI"	10 years ago
zuul	c530840dbd	Merge "funcs/func_curl: Add the ability for CURL to download and store files"	10 years ago
Matt Jordan	13efea24f7	main/app: Only look to end of file if ':end' is specified, and not just ':' There is a little known feature in app_controlplayback that will cause the specified offset to be used relative to the end of a file if a ':end' is detected within the filename. This feature is pretty bad, but okay. However, a bug exists in this code where a ':' detected in the filename will cause the end pointer to be non-NULL, even if the full ':end' isn't specified. This causes us to treat an unspecified offset (0) as being "start playing from the end of the file", resulting in no file playback occurring. This patch fixes this bug by resetting the end pointer if ':end' is not found in the filename. Change-Id: Ib4c7b1b45283e4effd622a970055c51146892f35	10 years ago
Matt Jordan	ca14b99e6e	main/file: Add the ability to play media in the media cache This patch allows applications/APIs that access media through the core file APIs to play media in the media cache. Prior to determining if a 'filename' exists, the filename is passed to the media cache's retrieve API call. If that call succeeds, the local file specified passed back by the API is opened for streaming. When used in this fashion, the 'filename' is actually a URI that the media cache process and understand. ASTERISK-25654 #close Change-Id: I73b6e2e90c3e91b8500581c45cdf9c0dc785f5f0	10 years ago
Matt Jordan	01962a3932	tests/test_http_media_cache: Add unit tests for res_http_media_cache This patch adds unit tests for res_http_media cache, that covers nominal creation and retrieval - and through them as well, staleness and deletion checks. In addition, this patch adds tests that covers the interaction of various HTTP headers, including Expires, Etag, and Cache-Control. ASTERISK-25654 Change-Id: I2db101e307c863857fe416d6f5bf4cace9ac7cf5	10 years ago
Matthew Jordan	22e2340813	res/res_http_media_cache: Add an HTTP(S) backend for the core media cache This patch adds a bucket backend for the core media cache that interfaces to a remote HTTP server. When a media item is requested in the cache, the cache will query its bucket backends to see if they can provide the media item. If that media item has a scheme of HTTP or HTTPS, this backend will be invoked. The backend provides callbacks for the following: * create - this will always retrieve the URI specified by the provided bucket_file, and store it in the file specified by the object. * retrieve - this will pull the URI specified and store it in a temporary file. It is then up to the media cache to move/rename this file if desired. * delete - destroys the file associated with the bucket_file. * stale - if the bucket_file has expired, based on received HTTP headers from the remote server, or if the ETag on the server no longer matches the ETag stored on the bucket_file, the resource is determined to be stale. Note that the backend respects the ETag, Expires, and Cache-Control headers provided by the HTTP server it is querying. ASTERISK-25654 Change-Id: Ie201c2b34cafc0c90a7ee18d7c8359afaccc5250	10 years ago
Matt Jordan	791b4c9f81	main/media_cache: Provide an extension on the local file associated with a URI This patch does the following: First, it addresses file extension handling in the media cache. The media core in Asterisk is a bit interesting in that it wants: * A file to have an extension on it. That extension is used to associate the file with a defined format module. * The filename passed to the core to not have an extension on it. This allows the core to match the available file formats with the format a channel is capable of handling. Unfortunately, this makes the current implementation a bit lacking in the media cache. By default, we do not store the extension of a retrieved URI on the local file that is created. As a result, the media core does not know what format the file is, and the file is ignored. Modifying the file outside of the media core is bad, as we would not be able to update the internal ast_bucket_file's path. At the same time, we do not want to pass the extension out in the file_path parameter in ast_media_cache_retrieve. This parameter is intended to be fed into the media core; if we passed the extension, all callers would have to strip it off. Thus, this patch does the following: * If there is an extension specified in the URL, we append it to the local file name (if a preferred file name isn't specified), and we store that in the local file path. * The extension, however, is stripped off of the file_path parameter passed back out of ast_media_cache_retrieve. Second, this patch causes stale items to be completely removed from the system. Prior to this patch, sound files could be orphaned due to the bucket referencing the file being deleted, but the file itself not being removed. This is now addressed by explicitly calling ast_bucket_file_delete on the bucket_file when it is deemed to be stale. Note that this only happen when we know we will attempt to retrieve the resource again. Finally, this patch changes the AO2 container holding media items to just use a regular mutex. The usage for this container already assumed it was a plain mutex, and - given that retrieval of an item can cause it to be replaced in the container - a mutex makes more sense than a read/write lock. Change-Id: I51667fff86ae8d2e4a663555dfa85b11e935fe0f	10 years ago
Matthew Jordan	6bbcfb34bd	funcs/func_curl: Add the ability for CURL to download and store files This patch adds a write option to the CURL dialplan function, allowing it to CURL files and store them locally. The value 'written' to the CURL URL specifies the location on disk to store the file. As an example: same => n,Set(CURL(http://1.1.1.1/foo.wav)=/tmp/foo.wav) Would retrieve the file foo.wav from the remote server and store it in the /tmp directory. Due to the potentially dangerous nature of this function call, APIs are forbidden from using the write functionality unless live_dangerously is set to True in asterisk.conf. ASTERISK-25652 #close Change-Id: I44f4ad823d7d20f04ceaad3698c5c7f653c41b0d	10 years ago
George Joseph	392341ba37	pjproject-bundled: Cleanups for reported issues PortAudio should no longer be required PJSIP_MAX_PKT_LEN is now 6000 Older autoconf issue fixed. (CentOS 6) Change-Id: I463fa9586cbe7c6b3b603289f535bd8e361611dd	10 years ago
Francesco Castellano	ac66999971	chan_sip.c: Space after port causes unnecessary resolution attempt check_via() already skips leading blanks where the sent-by address (with the optional port) should be placed. Since RFC 3261 allows for blanks between the port ant the Via parameters: > https://tools.ietf.org/html/rfc3261#section-20.42 (actually it allows a lot of blanks more ;-)). I just switched from ast_skip_blanks() to ast_strip() on the local copy of the string. ASTERISK-21301 #close Change-Id: Ie5b8fe5a07067b7c0dc9bcdd1707e99b23b02b06	10 years ago
zuul	c21cee80cc	Merge "func_aes: fix misuse of strlen on binary data"	10 years ago
George Joseph	1d3191b118	progdocs: Exclude ./third-party from documentation generation We don't need pjproject's documentation embedded in Asterisk's. Change-Id: Iea6f5a621c0f4e3168dda3321eaab258d9f24a17	10 years ago
Gianluca Merlo	8f94f947f5	func_aes: fix misuse of strlen on binary data The encryption code for AES_ENCRYPT evaluates the length of the data to be encoded in base64 using strlen. The data is binary, thus the length of it can be underestimated at the first NULL character. Reuse the write pointer offset to evaluate it, instead. ASTERISK-25857 #close Change-Id: If686b5d570473eb926693c73461177b35b13b186	10 years ago
Kevin Harwell	a3c9a74a02	chan_pjsip: ref leak when checking direct_media_glare Fix the reference leak introduced in the following commit: `c534bd5807` ASTERISK-25849 Change-Id: I5cfefd5ee6c1c3a1715c050330aaa10e4d2a5e85	10 years ago

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