asterisk

Commit Graph

Author	SHA1	Message	Date
Joshua Colp	d76f29a547	Merge "AST-2016-004: Fix crash on REGISTER with long URI."	10 years ago
Mark Michelson	7b8b6e2e4f	AST-2016-004: Fix crash on REGISTER with long URI. Due to some ignored return values, Asterisk could crash if processing an incoming REGISTER whose contact URI was above a certain length. ASTERISK-25707 #close Reported by George Joseph Patches: 0001-res_pjsip-Validate-that-URIs-don-t-exceed-pjproject-.patch AST-2016-004 Change-Id: I3ea7cee16f29c8088794de3085ca7523c1c4833d	10 years ago
Richard Mudgett	ff3af764de	bridge_softmix.c: Fix crash if could not allocate the dsp. Fix off nominal crash where we could not setup the channel to process frames for the softmix bridge technology because of allocation failure. Change-Id: Ic307a8386e46bf551e48fcd1eb97276714d56372	10 years ago
Joshua Colp	fddec0c266	Merge "app_voicemail: Fix test_voicemail_notify_endl test."	10 years ago
Joshua Colp	c714d0006b	Merge "res_pjsip_dialog_info: Add missing "direction" attribute in NOTIFY event"	10 years ago
Joshua Colp	5a0534dc62	app_voicemail: Fix test_voicemail_notify_endl test. The test_voicemail_notify_endl test checks the end-of-line characters of an email message to confirm that they are consistent. The test wrongfully assumed that reading from the email message into a buffer will always result in more than 1 character being read. This is incorrect. If only 1 character was read the test would go outside of the buffer and access other memory causing a crash. The test now checks to ensure that 2 or more characters are read in ensuring the test stays within the buffer. ASTERISK-25874 #close Change-Id: Ic2c89cea6e90f2c0bc2d8138306ebbffd4f8b710	10 years ago
Alexei Gradinari	49813bc9e5	res_pjsip: Add headers to AMI Event ContactStatusDetail * Added Useragent and RegExpire headers to AMI Event ContactStatusDetail with associated documentation. ASTERISK-25903 #close Change-Id: If3d121e943e588d016ba51d4eb9c6a421a562239	10 years ago
zuul	74951bd591	Merge "res_pjsip_outbound_publish: Add transport for outbound PUBLISH"	10 years ago
Joshua Colp	11ec5801d9	Merge "alembic: Remove batch operations (and sqlite support)"	10 years ago
Joshua Colp	4a7ed92308	Merge "core_unreal: Fix hangupcauses not getting set on Local channels"	10 years ago
zuul	9480ec2bd8	Merge "res_pjsip contact: Lock expiration/addition of contacts"	10 years ago
Alexei Gradinari	4e00e31ef1	res_pjsip_outbound_publish: Add transport for outbound PUBLISH The first available transport of the appropriate type is used now. This patch adds new config option 'transport' for outbound-publish. If transport is set then outbound PUBLISH requests will use this transport. ASTERISK-25901 #close Change-Id: Ib389130489b70e36795b0003fa5fd386e2680151	10 years ago
Jaco Kroon	2cc56573de	core_unreal: Fix hangupcauses not getting set on Local channels ASTERISK-25912 #close Change-Id: I8e72e6894feaf36c9450f2788d205d07baec23aa	10 years ago
zuul	c3339816f1	Merge "app_voicemail/IMAP: IMAP access FATAL error: Out of memory"	10 years ago
George Joseph	a621dd5e96	res_pjsip contact: Lock expiration/addition of contacts Contact expiration can occur in several places: res_pjsip_registrar, res_pjsip_registrar_expire, and automatically when anyone calls ast_sip_location_retrieve_aor_contact. At the same time, res_pjsip_registrar may also be attempting to renew or add a contact. Since none of this was locked it was possible for one thread to be renewing a contact and another thread to expire it immediately because it was working off of stale data. This was the casue of intermittent registration/inbound/nominal/multiple_contacts test failures. Now, the new named lock functionality is used to lock the aor during contact expire and add operations and res_pjsip_registrar_expire now checks the expiration with the lock held before deleting the contact. ASTERISK-25885 #close Reported-by: Josh Colp Change-Id: I83d413c46a47796f3ab052ca3b349f21cca47059	10 years ago
Joshua Colp	44fba00ca4	Merge "lock: Add named lock capability"	10 years ago
George Joseph	8637f29d24	pjproject: Add patch to fix Via IPv6 parsing There's a bug in pjproject's sip_parser where the ":" wasn't correctly interpreted. This is causing IPv6 addresses in the "received" parameter of the Via header to cause a syntax check failure. This patch was submitted to Teluu on 4/10/2016. ASTERISK-25910 #close Reported-by: Anthony Messina Change-Id: Ic7e4c4aa14ded61860401ec349f5177568c4d922	10 years ago
Joshua Colp	8610f4344f	Merge "pbx.h: Make ast_state_cb_type take more const."	10 years ago
George Joseph	216abb0ae7	lock: Add named lock capability Locking some objects like sorcery objects can be tricky because the underlying ao2 object may not be the same for all callers. For instance, two threads that call ast_sorcery_retrieve_by_id on the same aor name might actually get 2 different ao2 objects if the underlying wizard had to rehydrate the aor from a database. Locking one ao2 object doesn't have any effect on the other even if those objects had locks in the first place. Named locks allow access control by keyspace and key strings. Now an "aor" named "1000" can be locked and any other thread attempting to lock "aor" "1000" will wait regardless of whether the underlying ao2 object is the same or not. Mutex and rwlocks are supported. This capability will initially be used to lock an aor when multiple threads may be attempting to prune expired contacts from it. Change-Id: If258c0b7f92b02d07243ce70e535821a1ea7fb45	10 years ago
Alexei Gradinari	f9dab80816	app_voicemail/IMAP: IMAP access FATAL error: Out of memory Sometimes uw-imap function 'mail_fetchbody' returns huge len which then pass to uw-imap function 'rfc822_base64'. uw-imap tries to allocate huge memory and abort() on fail. This patch check the len. If the len more than max size (128 Mbytes) log error. This patch also set variables len, newlen to avoid uninizialezed len. This patch also check pointer returned by rfc822_base64. ASTERISK-25899 #close Change-Id: I4a0e7d655f11abef6a5224e2169df6d5c1f1caca	10 years ago
Joshua Colp	b47dfd1c6e	Merge "pbx.c: Minor code rearangements."	10 years ago
Alexei Gradinari	b3be945415	res_pjsip_dialog_info: Add missing "direction" attribute in NOTIFY event BLF pickup isn't working on Cisco SPA and Snom phones if the direction="recipient" attribute is missing in 'dialog' tag. This patch adds direction="recipient" if extension state is Ringing. ASTERISK-24601 #close Change-Id: I5b2c097ca29fd59e92ba237ca5d397cb1b0bcd8c	10 years ago
Richard Mudgett	6138a75e8e	pbx.h: Make ast_state_cb_type take more const. This eliminates some casts that I made a note saying v10 and above would no longer need them. Better late than never :) Change-Id: I346cdb3032b6478ceb40eb6fe732978b54035572	10 years ago
Richard Mudgett	72c19f7dc5	pbx.c: Minor code rearangements. * Pull out a loop invariant. * Convert an else-if ladder to a switch statement. Change-Id: I0a95cfa9474a4600b9865f7b444534d275b37e95	10 years ago
Richard Mudgett	28cefc3e88	pbx: Update doxygen for extension state watchers. Change-Id: Id1403b12136de62a272c01bb355aef65fd2c2d1e	10 years ago
Joshua Colp	724c16c543	Merge "pbx: Add support for autohints."	10 years ago
George Joseph	751d7a5a49	alembic: Remove batch operations (and sqlite support) Because SQLite doesn't support full ALTER capabilities, alembic scripts require batch operations. However, that capability wasn't available until 0.7.0 which some distributions haven't reached yet. Therefore, the batch operations introduced in commit `86d6e44cc` (review 2319) have been reverted and SQLite is unsupported again, for now anyway. Tested the full upgrade and downgrade on MySQL/Mariadb and Postgresql. ASTERISK-25890 #close Reported-by: Harley Peters Change-Id: I82eba5456736320256f6775f5b0b40133f4d1c80	10 years ago
Joshua Colp	2eaeea690d	res_pjsip_registrar_expire: Fix race condition at shutdown. When shutting down, the PJSIP sorcery is destroyed. The registrar expiration module queries the PJSIP sorcery to determine what to expire. As there was no synchronization between termination of the expiration thread and the unloading of the module it was possible for the thread to try to access the PJSIP sorcery after it had been destroyed. This change ensures that the thread is shut down before allowing the module to be considered unloaded. Change-Id: I69fd239edbaaf160c2d37ae00d3ac06e5596fe8b	10 years ago
Joshua Colp	3e5672d843	res_pjsip: Fix configuration setting of "regcontext". Due to a merge problem two options were swapped causing the regcontext setting to not get set. Change-Id: Icb33edc668e7357bacbaec2861a6b5ac64edaff1	10 years ago
Jacek Konieczny	8ed5f61152	frame.c: Copy the whole subclass in ast_frdup(). The problem is ast_frdup() does not copy whole frame.subclass for voice, video and image frames, only the format is copied. For video frames, the subclass structure contains the .frame_ending flag used to put the RTP marker where it needs to be. ASTERISK-25894 #close Change-Id: I812ca90e84ed5d4f473b997d0dd0d3c5a915fe33	10 years ago
Joshua Colp	97db0ca884	Merge "res_pjsip: Handle deferred SDP hold/unhold properly."	10 years ago
Joshua Colp	72ef79dc2d	Merge "ARI: Add method to Dial a created channel."	10 years ago
Joshua Colp	3b71f09bb7	Merge "ARI: Add method to create a new channel."	10 years ago
Mark Michelson	abbb2edd4c	ARI: Add method to Dial a created channel. This adds a new ARI method that allows for you to dial a channel that you previously created in ARI. By combining this with the create method for channels, it allows for a workflow where a channel can be created, manipulated, and then dialed. The channel is under control of the ARI application during all stages of the Dial and can even be manipulated based on channel state changes observed within an ARI application. The overarching goal for this is to eventually be able to add a dialed channel to a Stasis bridge earlier than the "Up" state. However, at the moment more work is needed in the Dial and Bridge APIs in order to facilitate that. ASTERISK-25889 #close Change-Id: Ic6c399c791e66c4aa52454222fe4f8b02483a205	10 years ago
Mark Michelson	dd48d60c5b	ARI: Add method to create a new channel. This adds a new ARI method to the channels resource that allows for the creation of a new channel. The channel is created and then placed into the specified Stasis application. This is different from the existing originate method that creates a channel, dials it, and then places the answered channel into the dialplan or a Stasis application. This method does not attempt to call the channel at all. Dialing is left as a later step after channel creation. This allows for pre-dialing channel manipulation if desired. ASTERISK-25889 Change-Id: I3c96a0aba914b08e39f6256371a5bd4c92cbded8	10 years ago
Joshua Colp	4d06a4f366	Merge "Dial: Add function to append already-created channel."	10 years ago
Joshua Colp	c49ee63db1	Merge "config: Allow filters when appending to a category"	10 years ago
Joshua Colp	1dc5e28624	pbx: Add support for autohints. This change introduces the concept of autohints. These are hints which are created as a result of device state changes occurring within the core. When this happens a hint will be created (if it does not exist already) using the device name as the extension. For example if a device state change is received for "PJSIP/bob" and autohints are enabled on a context then a hint will exist in that context for "bob" with a device of "PJSIP/bob". For virtual or custom device states the name after the type will be used. For example if the device state of "Custom:bob" changes then a hint will exist in that context for "bob" with a device of "Custom:bob". This functionality can be enabled in extensions.conf by placing "autohints=yes" in a context. ASTERISK-25881 #close Change-Id: I7e444c7da41b7b7d33374420fec658beeb18584e	10 years ago
Mark Michelson	a098251e7e	res_pjsip: Handle deferred SDP hold/unhold properly. Some SIP devices indicate hold/unhold using deferred SDP reinvites. In other words, they provide no SDP in the reinvite. A typical transaction that starts hold might look something like this: * Device sends reinvite with no SDP * Asterisk sends 200 OK with SDP indicating sendrecv on streams. * Device sends ACK with SDP indicating sendonly on streams. At this point, PJMedia's SDP negotiator saves Asterisk's local state as being recvonly. Now, when the device attempts to unhold, it again uses a deferred SDP reinvite, so we end up doing the following: * Device sends reinvite with no SDP * Asterisk sends 200 OK with SDP indicating recvonly on streams * Device sends ACK with SDP indicating sendonly on streams The problem here is that Asterisk offered recvonly, and by RFC 3264's rules, if an offer is recvonly, the answer has to be sendonly. The result is that the device is not taken off hold. What is supposed to happen is that Asterisk should indicate sendrecv in the 200 OK that it sends. This way, the device has the freedom to indicate sendrecv if it wants the stream taken off hold, or it can continue to respond with sendonly if the purpose of the reinvite was something else (like a session timer refresher). The fix here is to alter the SDP negotiator's state when we receive a reinvite with no SDP. If the negotiator's state is currently in the recvonly or inactive state, then we alter our local state to be sendrecv. This way, we allow the device to indicate the stream state as desired. ASTERISK-25854 #close Reported by Robert McGilvray Change-Id: I7615737276165eef3a593038413d936247dcc6ed	10 years ago
Mark Michelson	ef4d3f1328	Dial: Add function to append already-created channel. The Dial API takes responsiblity for creating an outbound channel when calling ast_dial_append(). This commit adds a new function, ast_dial_append_channel(), which allows us to create the channel outside the Dial API and then to append the channel to the ast_dial structure. This is useful for situations where the channel's creation and dialing are distinct operations. Upcoming ARI early bridge work will illustrate its usage. ASTERISK-25889 Change-Id: Id8179f64f8f99132f80dead8d5db2030fd2c0509	10 years ago
Joshua Colp	1805d8a57b	Merge "res_http_websocket: Make core supported."	10 years ago
Joshua Colp	245c9ca5ce	Merge "stringfields: Refactor to allow fields to be added to the end of structures"	10 years ago
George Joseph	984d6fd95c	config: Allow filters when appending to a category In sorcery based config files where there are multiple categories with the same name, you can't use the (+) operator to reliably append to a category because config.c stops looking when it finds the first one with the same name. Example: [1000] type = endpoint [1000] type = aor [1000](+) authenticate_qualify = yes This config will fail because config.c appends authenticate_qualify to the first category it finds, the endpoint, and that's not valid for endpoint. Solution: The capability to find a category that contains a certain variable already exists so the only real change was to parse anything after the '+' that's not a comma, as a filter string. [1000] type = endpoint [1000] type = aor [1000](+type=aor) authenticate_qualify = yes This now works as expected. Although the following example doesn't make any sense for pjsip, you can even specify multiple filters: [1000](+type=aor&qualify_frequency=10) ASTERISK-25868 #close Reported-by: Nick Repin Change-Id: I10773da4c79db36fbf1993961992af63d3441580	10 years ago
Joshua Colp	784fb43f43	res_http_websocket: Make core supported. Websockets are a core part of ARI support and as such this module should also be core supported. Change-Id: I8f9283c6a167152761b92984779bb39e3db51a9c	10 years ago
Joshua Colp	051da5c3af	Merge "res_rtp_asterisk: Use separate SRTP session for RTCP with DTLS"	10 years ago
George Joseph	4d40b161c3	stringfields: Refactor to allow fields to be added to the end of structures String fields are great, except that you can't add new ones without breaking ABI compatibility because it shifts down everything else in the structure. The only alternative is to add your own char * field to the end of the structure and manage the memory yourself which isn't ideal, especially since you then can't use the OPT_STRINGFIELD_T type. Background: The reason string fields had to be declared inside the AST_DECLARE_STRING_FIELDS block was to facilitate iteration over all declared fields for initialization, compare and copy. Since AST_DECLARE_STRING_FIELDS declared the pool, then the fields, then the manager, you could use the offsets of the pool and manager and iterate over the sequential addresses in between to access the fields. The actual pool, field allocation and field set operations don't actually care where the field is. It's just iteration over the fields that was the problem. Solution: Extended String Fields An extended string field is one that is declared outside the AST_DECLARE_STRING_FIELDS block but still (anywhere) inside the parent structure. Other than using AST_STRING_FIELD_EXTENDED instead of AST_STRING_FIELD, it looks the same as other string fields. It's storage comes from the pool and it participates in string field compare and copy operations peformed on the parent structure. It's also a valid target for the OPT_STRINGFIELD_T aco option type. Implementation: To keep track of the extended fields and make sure that ABI isn't broken, the existing embedded_pool pointer in the manager structure was repurposed to be a pointer to a separate header structure that contains the embedded_pool pointer plus a vector of fields. The length of the manager structure didn't change and the embedded_pool pointer isn't used in the macros, only the stringfields C code. A side benefit of this is that changing the header structure in the future won't break ABI. ast_string_fields_init initializes the normal string fields and appends them to the vector, and subsequent calls to ast_string_field_init_extended initialize and append the extended fields. Cleanup, ast_string_fields_cmp, and ast_string_fields_copy can now work on the vector instead of sequentially traversing the addresses between the pool and manager. The total size of a structure using string fields didn't change, whether using extended fields or not, nor have the offsets of any structure members, either inside the original block or outside. Adding an extended field to the end of a structure is the same as adding a char . Details: The stringfield C code was pulled out from utils.c and into stringfields.c. It just made sense. Additional work was done in ast_string_field_init and ast_calloc_with_stringfields to handle the allocation of the new header structure and the vector, and the associated cleanup. In the process some additional NULL pointer checking was added. A lot of work was done in stringfields.h since the logic for compare and copy is there. Documentation was added as well as somne additional NULL checking. The ability to call ast_calloc_with_stringfields with a number of structures greater than 1 never really worked. Well, the calloc worked but there was no way to access the additional structures or clean them up. It was agreed that there was no use case for requesting more than 1 structure so an ast_assert was added to prevent it and the iteration code removed. Testing: The stringfield unit tests were updated to test both normal and extended fields. Tests for ast_string_field_ptr_set_by_fields and ast_calloc_with_stringfields were also added. As an ABI test, 13 was compiled from git and the res_pjsip_ modules, except res_pjsip itself, saved off. The patch was then added and a full compile and install was performed. Then the older res_pjsip_* moduled were copied over the installed versions so res_pjsip was new and the rest were old. No issues. contact->aor, which is a char * at the end of contact, was then changed to an extended string field and a recompile and reinstall was performed, again leaving stock versions of the the res_pjsip_* modules. Again, no issues with the res_pjsip_* modules using the old stringfield implementation and with contact->aor as a char *, and res_pjsip itself using the new stringfield implementation and contact->aor being an extended string field. Finally, several existing string fields were converted to extended string fields to test OPT_STRINGFIELD_T. Again, no issues. Change-Id: I235db338c5b178f5a13b7946afbaa5d4a0f91d61	10 years ago
George Joseph	c07e1190ec	res_pjsip_mwi: Fix segv caused by `16c7d8e74a` I forgot the new voicemail_extension wasn't a stringfield and didn't check for NULL where I should have. Change-Id: I029482d5c2ab72474838750461bd46b0809c90fb	10 years ago
Joshua Colp	86ccd1bd2b	Merge "install_prereq: Fix check_installed_debs remove subversion"	10 years ago
Joshua Colp	abf241b94f	Merge "res_pjsip_mwi: Allow subscribe to vm access extension as an alias"	10 years ago
Joshua Colp	96dcd81110	Merge "res_pjsip_mwi: Add voicemail extension and mwi_subscribe_replaces_unsolicited"	10 years ago

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