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${ noResults }
27781 Commits (f5aa9f8034fb66c4606b2f5c4de173c22c977e7c)
Author | SHA1 | Message | Date |
---|---|---|---|
|
56c8182913 |
Merge "app_voicemail/IMAP: function 'save_to_folder' creates wrong folder" into 13
|
9 years ago |
|
5f78801859 |
transport management: Register thread with PJProject.
The scheduler thread that kills idle TCP connections was not registering with PJProject properly and causing assertions if PJProject was built in debug mode. This change registers the thread with PJProject the first time that the scheduler callback executes. AST-2016-005 Change-Id: I5f7a37e2c80726a99afe9dc2a4a69bdedf661283 |
9 years ago |
|
9740277713 |
res_pjsip: Add serialized scheduler (res_pjsip/pjsip_scheduler.c)
There are several places that do scheduled tasks or periodic housecleaning, each with its own implementation: * res_pjsip_keepalive has a thread that sends keepalives. * pjsip_distributor has a thread that cleans up expired unidentified requests. * res_pjsip_registrar_expire has a thread that cleans up expired contacts. * res_pjsip_pubsub uses ast_sched directly and then calls ast_sip_push_task. * res_pjsip_sdp_rtp also uses ast_sched to send keepalives. There are also places where we should be doing scheduled work but aren't. A good example are the places we have sorcery observers to start registration or qualify. These don't work when changes are made to a backend database without a pjsip reload. We need to check periodically. As a first step to solving these issues, a new ast_sip_sched facility has been created. ast_sip_sched wraps ast_sched but only uses ast_sched as a scheduled queue. When a task is ready to run, ast_sip_task_pusk is called for it. This ensures that the task is executed in a PJLIB registered thread and doesn't hold up the ast_sched thread so it can immediately continue processing the queue. The serializer used by ast_sip_sched is one of your choosing or a random one from the res_pjsip pool if you don't choose one. Another feature is the ability to automatically clean up the task_data when the task expires (if ever). If it's an ao2 object, it will be dereferenced, if it's a malloc'd object it will be freed. This is selectable when the task is scheduled. Even if you choose to not auto dereference an ao2 task data object, the scheduler itself maintains a reference to it while the task is under it's control. This prevents the data from disappearing out from under the task. There are two scheduling models. AST_SIP_SCHED_TASK_PERIODIC specifies that the invocations of the task occur at the specific interval. That is, every "interval" milliseconds, regardless of how long the task takes. If the task takes longer than the interval, it will be scheduled at the next available multiple of interval. For exmaple: If the task has an interval of 60 secs and the task takes 70 secs (it better not), the next invocation will happen at 120 seconds. AST_SIP_SCHED_TASK_DELAY specifies that the next invocation of the task should start "interval" milliseconds after the current invocation has finished. Also, the same ast_sched facility for fixed or variable intervals exists. The task's return code in conjunction with the AST_SIP_SCHED_TASK_FIXED or AST_SIP_SCHED_TASK_VARIABLE flags controls the next invocation start time. One res_pjsip.h housekeeping change was made. The pjsip header files were added to the top. There have been a few cases lately where I've needed res_pjsip.h just for ast_sip calls and had compiles fail spectacularly because I didn't add the pjsip header files to my source even though I never referenced any pjsip calls. Finally, a few new convenience APIs were added to astobj2 to make things a little easier in the scheduler. ao2_ref_and_lock() calls ao2_ref() and ao2_lock() in one go. ao2_unlock_and_unref() does the reverse. A few macros were also copied from res_phoneprov because I got tired of having to duplicate the same hash, sort and compare functions over and over again. The AO2_STRING_FIELD_(HASH|SORT|CMP)_FN macros will insert functions suitable for aor_container_alloc into your source. This facility can be used immediately for the situations where we already have a thread that wakes up periodically or do some scheduled work. For the registration and qualify issues, additional sorcery and schema changes would need to be made so that we can easily detect changed objects on a periodic basis without having to pull the entire database back to check. I'm thinking of a last-updated timestamp on the rows but more on this later. Change-Id: I7af6ad2b2d896ea68e478aa1ae201d6dd016ba1c |
9 years ago |
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13cb5ea73f |
Merge "res_pjsip_transport_management: Kill idle TCP connections." into 13
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9 years ago |
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120493d5c0 |
Merge "Rename res_pjsip_keepalive res_pjsip_transport_management" into 13
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9 years ago |
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6c9c714bb6 |
Merge "AST-2016-004: Fix crash on REGISTER with long URI." into 13
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9 years ago |
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7fb3724a77 |
res_pjsip_transport_management: Kill idle TCP connections.
"Idle" here means that someone connects to us and does not send a SIP request. PJProject will not automatically time out such connections, so it's up to Asterisk to do it instead. When we receive an incoming TCP connection, we will start a timer (equivalent to transaction timer D) waiting to receive an incoming request. If we do not receive a request in that timeframe, then we will shut down the TCP connection. ASTERISK-25796 #close Reported by George Joseph AST-2016-005 Change-Id: I7b0d303e5d140d0ccaf2f7af562071e3d1130ac6 |
9 years ago |
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707fd4dcd0 |
Rename res_pjsip_keepalive res_pjsip_transport_management
ASTERISK-25796 Reported by George Joseph AST-2016-005 Change-Id: Id322a05f927392293570599730050bc677d99433 |
9 years ago |
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0b4bb19e0b |
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: I0ed3898fe7ab10121b76c8c79046692de3a1be55 |
9 years ago |
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f6e080c6a4 |
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 |
9 years ago |
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1f853df29c |
Merge "app_voicemail: Fix test_voicemail_notify_endl test." into 13
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9 years ago |
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cf15a2f2d3 |
pjproject: Add patch for removing strip of '[]' from header params
From the patch submitted to Teluu on 4/12/2016 <<<<<<<<< The wholesale stripping of '[]' from header parameters causes issues if something (like a port) occurs after the final ']'. '[2001🅰️:b]' will correctly parse to '2001🅰️:b' '[2001🅰️:b]:8080' will correctly parse to '2001🅰️:b' but the scanner is left with ':8080' and parsing stops with a syntax error. I can't even find a case where stripping the '[]' is a good thing anyway. Even if you continued to parse and resulted in a string that looks like this... '2001🅰️🅱️8080', it's not valid. This came up in Asterisk because Kamailio sends us a Contact with an alias URI parameter that has an IPv6 address in it like this: Contact: <sip:1171@127.0.0.1:5080;alias=[2001:1:2::3]~43691~6> which should be legal but causes a syntax error because of the characters after the final ']'. Even if it didn't, the '[]' should still not be stripped. I've run the Asterisk Test Suite for PJSIP (252 tests) many of which are IPv6 enabled. No issues were caused by removing the code that strips the '[]'. >>>>>>>>>>> ASTERISK-25123 #close Reported-by: Anthony Messina Change-Id: I5cb33f4ebf07ee1f2b26d07caae715e2ec65595a |
9 years ago |
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4ab4fc9141 |
Merge "res_pjsip_dialog_info: Add missing "direction" attribute in NOTIFY event" into 13
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9 years ago |
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daa086fae4 |
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 |
9 years ago |
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70c788ec5e |
Merge "res_pjsip: Add headers to AMI Event ContactStatusDetail" into 13
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9 years ago |
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f896136460 |
app_voicemail/IMAP: function 'save_to_folder' creates wrong folder
If try to move message to Cust1 (number 5) the function 'save_to_folder' tries to create Greeting folder instead of Cust1. This patch fixed it by setting GREETINGS_FOLDER = -1 ASTERISK-24927 #close Change-Id: I03d1a761894bcc2d130ec9b003bbcddc28e25c51 |
9 years ago |
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70b7673f09 |
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 |
9 years ago |
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cf1c0277b5 |
Merge "res_pjsip_outbound_publish: Add transport for outbound PUBLISH" into 13
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9 years ago |
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600148b5b0 |
Merge "alembic: Remove batch operations (and sqlite support)" into 13
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9 years ago |
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5eec2386cf |
Merge "core_unreal: Fix hangupcauses not getting set on Local channels" into 13
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9 years ago |
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df40173a00 |
Merge "res_pjsip contact: Lock expiration/addition of contacts" into 13
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9 years ago |
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64ecd41c8f |
Codecs: strip codec name while parsing allow/disallow options
Failed registration using PJSIP/Realtime if one of the codec name in allow/disallow option is wrong or contains space. This patch strip codec name. ASTERISK-25914 Change-Id: Ifdf02de94e5ddbce305640f6f0666084a3b9283d |
9 years ago |
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3f6c4667b8 |
core_unreal: Fix hangupcauses not getting set on Local channels
ASTERISK-25912 #close Change-Id: I8e72e6894feaf36c9450f2788d205d07baec23aa |
9 years ago |
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f2edcfe62e |
Merge "app_voicemail/IMAP: IMAP access FATAL error: Out of memory" into 13
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9 years ago |
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fe7e48db03 |
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 |
9 years ago |
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736a2c303d |
Merge "lock: Add named lock capability" into 13
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9 years ago |
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0c414eaf35 |
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 |
9 years ago |
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772ff3048f |
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 |
9 years ago |
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fd601f26f7 |
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 |
9 years ago |
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e7e16b7465 |
Merge "pbx.c: Minor code rearangements." into 13
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9 years ago |
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5f768d2a9c |
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 |
9 years ago |
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82638fb0c7 |
pbx.c: Minor code rearangements.
* Pull out a loop invariant. * Convert an else-if ladder to a switch statement. Change-Id: I0a95cfa9474a4600b9865f7b444534d275b37e95 |
9 years ago |
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bc320df173 |
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 |
9 years ago |
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2ef8a954b3 |
pbx: Update doxygen for extension state watchers.
Change-Id: Id1403b12136de62a272c01bb355aef65fd2c2d1e |
9 years ago |
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d312fdeb1b |
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
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9 years ago |
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901e8d78c4 |
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 |
9 years ago |
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8207372e66 |
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 |
9 years ago |
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0735a4d6d7 |
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 |
9 years ago |
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587a97a515 |
Merge "res_pjsip: Handle deferred SDP hold/unhold properly." into 13
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9 years ago |
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c61dca6419 |
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 |
9 years ago |
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edf2ce2eff |
Merge "config: Allow filters when appending to a category" into 13
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9 years ago |
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05a9b3eaf7 |
Merge "res_http_websocket: Make core supported." into 13
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9 years ago |
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50b0922a22 |
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 |
9 years ago |
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cb56ef8069 |
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 |
9 years ago |
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2b84290386 |
Merge "stringfields: Refactor to allow fields to be added to the end of structures" into 13
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9 years ago |
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bb7214180c |
Merge "res_rtp_asterisk: Use separate SRTP session for RTCP with DTLS" into 13
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9 years ago |
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f6f4cf459f |
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 |
9 years ago |
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cac6453e9a |
Merge "res_pjsip_mwi: Fix segv caused by 16c7d8e74a9af13f98c3c22aa9c43ce39965f6b7" into 13
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9 years ago |
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5de637df4d |
Merge "install_prereq: Fix check_installed_debs remove subversion" into 13
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9 years ago |
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fe448ac8a7 |
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 |
9 years ago |