asterisk

Commit Graph

Author	SHA1	Message	Date
Kevin Harwell	e29c5e0c5c	alembic: script modifications due to errors A couple of the scripts had errors that would not allow a full migration to take place. The extensions table needed to make its 'id' column a primary key in order to work with mysql. The other script ...add_endpoints... was missing tables that it was trying to add columns to. Added the primary key on id for extensions and added the tables in for the missing pjsip configuration options. While it is not ideal to modify already released scripts this was a case where it had to be done due to errors in the script and lacking a better alternative. Review: https://reviewboard.asterisk.org/r/3167/ ........ Merged revisions 407019 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407029 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	6b114b654e	res_pjsip_mwi: Subscribe fails when missing aor name When subscribing to MWI (res_pjsip_mwi) and the sip uri did not contain a name (ex: sip:<ip address>) then the subscription would fail since it would be unable to locate an associated aor. This patch makes it so that when a subscribe comes with no aor name then it will subscribe to all aors on the located endpoint. (closes issue ASTERISK-23072) Reported by: Bob M Review: https://reviewboard.asterisk.org/r/3164/ ........ Merged revisions 407014 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407015 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	d5431ed358	PJSIP: Fix address for ACK in NAT situations In NAT scenarios where a call is placed to a Grandstream phone, res_pjsip will sometimes send the ACK to a 200 OK to the private address of the device behind the NAT instead of the address of the NAT device. This corrects that behavior by rewriting the address in the Contact header in the incoming 200 OK and the dialog's target address if necessary (since it has already been rewritten to the incorrect private address). (closes issue ASTERISK-23106) Review: https://reviewboard.asterisk.org/r/3168/ Reported by: Matt Jordan ........ Merged revisions 407000 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407001 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Damien Wedhorn	e8eb6a9e30	Skinny: fix up possible double unlock of chan. Return before chan is possibly unlocked a second time when hanging up a channel in SUBSTATE_OFFHOOK. ........ Merged revisions 406987 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406988 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Corey Farrell	c35d07950f	res_rtp_asterisk & udptl: fix port selection to work with SELinux restrictions ast_bind to a port reserved for another program by SELinux causes errno == EACCES. This caused random failures when binding rtp or udptl sockets. Treat EACCES as a non-fatal error, try next port. (closes issue ASTERISK-23134) Reported by: Corey Farrell ........ Merged revisions 406933 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 406934 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406935 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406936 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Sean Bright	98de7719dd	Make a NOTICE about an invalid channel name more useful. ........ Merged revisions 406918 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406919 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406920 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Russell Bryant	e7b20b1c91	queues.conf.sample Fix documented default for persistentmembers Closes issue ASTERISK-22662 ........ Merged revisions 406860 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 406861 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406862 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406863 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	565198b44b	res_pjsip_pubsub: potential crash on timeout What seems to be happening is if a subscription has been terminated and the subscription timeout/expires is less than the time it takes for all pending transactions (currently on the subscription) to end then the subscription timer will not have been canceled yet and sub will be null. Since the subscription has already been canceled nothing needs to be done so a null check in the asterisk code is sufficient in working around this problem. (closes issue ASTERISK-23129) Reported by: Dan Jenkins ........ Merged revisions 406847 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406848 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	ade5c8a2a4	cdr_radius, cel_radius: build agains libfreeradius-client Asterisk's RADIUS module currently build against libradiusclient-ng, but this project has been superseeded by libfreeradius-client. The API is 99% compatible except that the header name has changed, the library name has changed, and the configuration file location has changed. (closes issue ASTERISK-22980) Reported by: Jeremy Lainé Patches: freeradius-client.patch uploaded by sharky (license 6561) ........ Merged revisions 406801 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 406802 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406803 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406825 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	01a537d515	res_pjsip,compat: INFINITY and NAN undefined On some systems the values for INFINITY and NAN are not defined thus causing a build error on those systems. Added definitions for those if they had not previously been defined. (closes issue ASTERISK-23056) Reported by: capouch Patches: inf-nan-patch.txt uploaded by capouch (license 6564) ........ Merged revisions 406788 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406789 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	3eae87fdfe	ARI: Make double subscribe respond with success Currently, attempting to subscribe an application to a device state that it has already subscribed to will generate a 500 error response. This will now be treated as a subscription refresh even though ARI subscriptions don't currently support lifetimes and will respond with the normal response for a successful subscription (200 OK). (closes issue ASTERISK-23143) Reported by: Matt Jordan ........ Merged revisions 406775 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406778 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	601692a7e4	rtp_engine: improved handling of get_rtp_info failure In ast_rtp_instance_make_compatible(), after a failure of channel tech call get_rtp_info() to return peer_instance, the null pointer would be passed to ao2_ref, producing an error that looked like a refernce counting problem but is not. This patch corrects that and adds helpful LOG_ERROR messages to indicate which failure path occurred. (issue AST-1276) Review: https://reviewboard.asterisk.org/r/3156/ ........ Merged revisions 406721 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 406722 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406723 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406724 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	aff9e1f583	test_cdr.c, test_cel.c: Correctly destroy created bridges. * Fixed the test_cel_attended_transfer_bridges_link unit test to also account for the local channel link being destroyed now that the bridges are actually destroyed. * Made CDR unit test use its own version of do_sleep() from the CEL unit tests. ........ Merged revisions 406707 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406710 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	8f82eb0098	manager: ExtensionStatus event status human readable Added a note in the changes file about the new 'StatusText' field that was added to the 'ExtensionStatus' event. (issue ASTERISK-23154) Reported by: Jonathan Rose git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406696 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	f9479fbcbd	manager: ExtensionStatus event status human readable When an 'ExtensionStatus' event was raised it included the status as a numerical value, but did not include a text description of the status. Added a 'StatusText' field to the event which is a string representation of the extension status. Also added this to the 'Extension State' command response. (closes issue ASTERISK-23154) Reported by: Jonathan Rose git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406647 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Russell Bryant	8a762efb35	Allow nested #includes in extconfig.conf extconfig.conf was hard-coded to not allow nested includes for some reason. The code has been this way since a patch was merged for ASTERISK-3333 (revision 4889), which was a significant update to this code ("Merge config updates"). I can't figure out any good reason why this should be limited. This patch just removes the limit and uses the default nesting depth limit. Closes issue ASTERISK-17837 Review: https://reviewboard.asterisk.org/r/3159/ ........ Merged revisions 406643 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 406644 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406645 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406646 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Walter Doekes	cc42229f26	manager: The eventfilter= option now takes an extended regex. In pre-trunk versions (...12) it accepts a basic regex, which is confusing because all other regexes in asterisk are of the extended kind. Review: https://reviewboard.asterisk.org/r/3147/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406618 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Russell Bryant	33071d636c	Protect ast_filestream object when on a channel The ast_filestream object gets tacked on to a channel via chan->timingdata. It's a reference counted object, but the reference count isn't used when putting it on a channel. It's theoretically possible for another thread to interfere with the channel while it's unlocked and cause the filestream to get destroyed. Use the astobj2 reference count to make sure that as long as this code path is holding on the ast_filestream and passing it into the file.c playback code, that it knows it's valid. Bug reported by Leif Madsen. Review: https://reviewboard.asterisk.org/r/3135/ ........ Merged revisions 406566 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 406567 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406574 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406595 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	45261449ec	tcptls.c: Add missing cleanup on off nominal path. ........ Merged revisions 406514 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 406515 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406516 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406517 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Tzafrir Cohen	6b019afada	live_ast: run wrapped programs with exec live_ast can be used as a wrapper script to run asterisk, gdb or valgrind. In those cases it runs them and returns the result. It is more useful to use 'exec' to avoid having another odd process in the chain. Review: https://reviewboard.asterisk.org/r/3110/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406503 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	ff455ee2aa	res_pjsip_session: Be less strict with core requested outgoing capabilities. The core may (depending on circumstances) request a single codec on outgoing calls. Many channel drivers ignore or treat this as a suggestion while still including configured codecs. The res_pjsip_session logic treated this as an explicit request, leaving out other configured codecs. This change makes res_pjsip_session behave like other channel driver and simply adds the requested codec to the list. (closes issue ASTERISK-23082) Reported by: xrobau Review: https://reviewboard.asterisk.org/r/3140/ ........ Merged revisions 406489 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406490 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	42c15dfa6e	CEL: Protect data structures during reload and shutdown. The CEL data structures need to be protected during a configuration reload and shutdown. Asterisk crashed during a shutdown because CEL events were still in flight and the CEL data structures were already destroyed. * Protected the cel_backends, cel_dialstatus_store, and cel_linkedids ao2 containers with a global ao2 object wrapper. * Added NULL checks before use of the cel_backends, cel_dialstatus_store, and cel_linkedids ao2 containers in case the CEL module is already shutdown. * Fixed overloading of the cel_linkedids held objects reference count. During shutdown any held objects would be leaked. * Fixed memory leak of cel_linkedids held objects if the LINKEDID_END is not being tracked. The objects in the cel_linkedids container were not removed if the LINKEDID_END event is not used. * Added access protection to the cel_backends container during the CLI "cel show status" command. * Made cel_backends, cel_dialstatus_store, and cel_linkedids use the standard ao2 callback templates for the hash and cmp functions. * Eliminated unnecessary uses of RAII_VAR(). * Made ast_cel_engine_init() cleanup alocated resources on failure. (closes issue AST-1253) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/3128/ ........ Merged revisions 406417 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 406418 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406465 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406466 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	2a9d15b400	Thread Debugging: Add LWP to core show locks output This patch adds the LWP to core show locks output if it is available. Review: https://reviewboard.asterisk.org/r/3142/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406416 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	a9911f027e	manager: Register atexit shutdown routine only once. * Made register atexit shutdown routine only once in __init_manager(). * Fixed some initial load failure conditions in __init_manager(). * Made reset options to defaults on reload when the reload will actually happen. * Removed unnecessary container traversals of the white/black filters during manager_free_user(). * ast_free() does not need a NULL check before calling. ........ Merged revisions 406359 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 406400 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406401 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406407 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	b78d0c0187	res_config_pgsql: Fix a memory leak and use RAII_VAR for cleanup when practical Review: https://reviewboard.asterisk.org/r/3141/ ........ Merged revisions 406360 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 406361 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406389 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406399 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	82cce81737	manager: Protect data structures during shutdown. Occasionally, the manager module would get an "INTERNAL_OBJ: bad magic number" error on a "core restart gracefully" command if an AMI connection is established. * Added ao2_global_obj protection to the sessions global container. * Fixed the order of unreferencing a session object in session_destroy(). * Removed unnecessary container traversals of the white/black filters during session_destructor(). (closes issue AST-1242) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/3144/ ........ Merged revisions 406341 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406342 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406343 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	799fc2493e	Today is not my day for writing code that compiles. ........ Merged revisions 406327 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406328 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	4e86602921	res_config_mysql: Fix Setting The Column Name Incorrectly When support for a realtime sorcery module was added in revision 386731, the wrong property was accidentally used for setting the column name to be updated in the database table. This patch fixes the typo. (closes issue ASTERISK-23177) Reported by: Denis Tested by: Denis Patches: asterisk-23177-use-field-name.diff by Michael L. Young (license 5026) ........ Merged revisions 406311 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406312 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	9b8f2db47e	Multiple revisions 406294-406295 ........ r406294 \| mmichelson \| 2014-01-23 15:00:24 -0600 (Thu, 23 Jan 2014) \| 11 lines Fix presence body errors found during testing: * PIDF bodies were reporting an "open" state in many cases where it should have been reporting "closed" * XPIDF bodies had XML nodes placed incorrectly within the hierarchy. * SIP URIs in XPIDF bodies did not go through XML sanitization * XML sanitization had some errors: * Right angle bracket was being replaced with "&rt;" instead of ">" * Double quote, apostrophe, and ampersand were not being escaped. ........ r406295 \| mmichelson \| 2014-01-23 15:09:35 -0600 (Thu, 23 Jan 2014) \| 11 lines Fix presence body errors found during testing: * PIDF bodies were reporting an "open" state in many cases where it should have been reporting "closed" * XPIDF bodies had XML nodes placed incorrectly within the hierarchy. * SIP URIs in XPIDF bodies did not go through XML sanitization * XML sanitization had some errors: * Right angle bracket was being replaced with "&rt;" instead of ">" * Double quote, apostrophe, and ampersand were not being escaped. ........ Merged revisions 406294-406295 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406298 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	64e2e1d5d8	pbx.c: Pre-initialize timezone to avoid crash on destroy In ast_build_timing, initialize the timezone value to NULL in order to avoid deferencing an uninitialized value later when calling ast_destroy_timing. The timezone value could be uninitialized if ast_build_timing were to fail due to a zero length time string. (closes issue ASTERISK-22861) Reported by: Sebastian Murray-Roberts Review: https://reviewboard.asterisk.org/r/3134/ Patches: ast_build_timing-initialize-timezone.patch uploaded by coreyfarrell (license 5909) ........ Merged revisions 406241 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 406245 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406264 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406269 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	3e6c4a6f89	ConfBridge: Fix channel parameter documentation Confbridge AMI and CLI commands for mute, unmute, and setting the single video source can accept channel prefixes in lieu of a full channel name, but documentation states only that it is required and is a channel name. This corrects the documentation. (closes issue PQ-1397) Reported by: Steve Pitts ........ Merged revisions 406217 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406223 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406224 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	0fbffdb3b2	chan_sip: Decline image streams on unsupported transports This change allows chan_sip to decline individual image streams over unsupported transports in the SDP of the 200 response. Previously, an image stream offer with RTP/AVP as the transport would cause chan_sip to respond with a 488. (closes issue ASTERISK-22988) Reported by: adomjan Original patch by: adomjan ........ Merged revisions 406170 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 406171 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406172 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406173 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	761d7271d4	res_stasis_playback: Correct error argument order Several of the playback error messages for invalid media input in res_stasis_playback.c had the media name and channel name reversed. They now correctly identify the channel name and media name. Reported by: skrusty ........ Merged revisions 406152 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406153 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Rusty Newton	a1d6e8ebab	res_pjsip: Documentation improvement for Endpoint and AOR mailbox options. Making the help text for both more explicit regarding the format of mailbox identifiers. i.e. clarifying the format for app_voicemail mailboxes vs mailboxes from external MWI sources through modules such as res_external_mwi. ........ Merged revisions 406133 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406134 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Walter Doekes	9a88cc33f8	manager: Clarify eventfilter documentation. Textual changes only. Review: https://reviewboard.asterisk.org/r/3133/ ........ Merged revisions 406079 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 406080 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406081 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406082 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	b2a682bae2	chan_mgcp: Enforce locking for oseq This restricts direct usage of global oseq so that all accesses are locked and threads are not racing to get oseq values that they did not claim. This also fixes a build error in res_pktccops under dev mode. (closes issue ASTERISK-23100) Reported by: adomjan Patch by: adomjan ........ Merged revisions 406037 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 406038 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406049 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406078 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	e0da867dbe	PJSIP: Handle headers in a list appropriately The PJSIP header parsing function (pjsip_parse_hdr) can generate more than one header instance from a single header field. These header instances exist as a list attached to the returned header and must be handled appropriately when they are added to a message or else only the first header instance will be used. This changes the linked list functions used in outbound proxy code to merge the lists properly. ........ Merged revisions 406020 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406022 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	1590d32ab0	ARI: Support channel variables in originate This adds back in support for specifying channel variables during an originate without compromising the ability to specify query parameters in the JSON body. This was accomplished by generating the body-parsing code in a separate function instead of being integrated with the URI query parameter parsing code such that it could be called by paths with body parameters. This is transparent to the user of the API and prevents manual duplication of code or data structures. (closes issue ASTERISK-23051) Review: https://reviewboard.asterisk.org/r/3122/ Reported by: Matt Jordan ........ Merged revisions 406003 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406006 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Damien Wedhorn	4bc84b1b9f	Skinny: fix up handling of fragmented packets. Bad offset in reading second or more fragment of skinny packets. Fixed to offset by char (single byte) rather than size of req. ........ Merged revisions 405982 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405985 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	eddbe10f91	chan_dahdi/PRI: Suppress CONNECTED_LINE updates when nothing in the udpate is valid. * Also simplified some subddress handling code. (closes issue ASTERISK-23008) Reported by: Michael Cargile ........ Merged revisions 405926 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 405927 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405928 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405947 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Damien Wedhorn	1f401eed45	Skinny: fix up session logging. Logging from the skinny session loop was providing some incorrect reasons for exiting the loop. Cleaned up messages and handling so correct reason displayed. ........ Merged revisions 405924 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405925 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	85fbbed45d	chan_pjsip: Provide a means for tracking device state when holding/unholding Previously PJSIP did not track hold/unhold and it would always simply be 'inuse'. This patch fixes that. review: https://reviewboard.asterisk.org/r/3129/ ........ Merged revisions 405908 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405910 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Damien Wedhorn	fcc492645b	Skinny: fix reversed device reset from CLI. Existing code would do a full device restart when "skinny reset device" was entered at the CLI and do a reset when "skinny reset device restart" entered. ........ Merged revisions 405893 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405894 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Sean Bright	778d74cacf	Make sure the maxptime attribute is added to the correct offers. ........ Merged revisions 405877 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405878 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	2b14601bdc	pjsip: fix support for allow=all This change adds improvements to support for allow=all in pjsip.conf so that it functions as intended. Previously, the allow/disallow socery configuration would set & clear codecs from the media.codecs and media.prefs list, but if all was specified the prefs list was not updated. Then a call would fail when create_outgoing_sdp_stream() created an SDP with no audio codecs. A new function ast_codec_pref_append_all() is provided to add all codecs to the prefs list - only those not already on the list. This enables the configuration to specify a codec preference, but still add all codecs, and even then remove some codecs, as shown in this example: allow = ulaw, alaw, all, !g729, !g723 Also, the display order of allow in cli output is updated to match the configuration by using prefs instead of caps when generating a human readable string. Finally, a change to create_outgoing_sdp_stream() skips a codec when it does not have a payload code instead of the call failing. (closes issue ASTERISK-23018) Reported by: xrobau Review: https://reviewboard.asterisk.org/r/3131/ ........ Merged revisions 405875 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405876 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	2704b49c1b	http: supported chunked Transfer-Encoding This change implements support for HTTP Transfer-Encoding chunked in both JSON and Form (post vars) body content. A new function ast_http_get_contents() handles both regular and chunked mode body, returning after the entire body is received. (closes issue ASTERISK-23068) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3125/ ........ Merged revisions 405861 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405862 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Rusty Newton	926081461b	Fixing some XML syntax issues with my previous commit at r405777 for ASTERISK-23071 ........ Merged revisions 405843 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405844 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Rusty Newton	f6647d2362	Documentation: doc fixes across various parts of the code for ASTERISK issues 23061,23028,23046,23027 Fixes typos of "transfered" instead of "transferred" in various code. Fixes incorrect gosub param help text for app_queue. Fixes Asterisk man pages containing unquoted minus signs. Adds note about the "textsupport" option in sip.conf.sample. (issue ASTERISK-23061) (issue ASTERISK-23028) (issue ASTERISK-23046) (issue ASTERISK-23027) (closes issue ASTERISK-23061) (closes issue ASTERISK-23028) (closes issue ASTERISK-23046) (closes issue ASTERISK-23027) Reported by: Eugene, Jeremy Laine, Denis Pantsyrev Patches: transferred.patch uploaded by Jeremy Laine (license 6561) hyphen.patch uploaded by Jeremy Laine (license 6561) sip.conf.sample.patch uploaded by Eugene (license 6360) ........ Merged revisions 405791 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 405792 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405829 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405830 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Rusty Newton	3fb2906955	res_pjsip: enhance documentation for mailboxes options, for both endpoints and aors Made documentation more explicit as to the use of the both options. (issue ASTERISK-23071) (closes issue ASTERISK-23071) Reported by: Matt Jordan ........ Merged revisions 405777 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405778 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Walter Doekes	72cb7a254f	Enable wide band audio in musiconhold streams. Review: https://reviewboard.asterisk.org/r/3112/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405766 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago

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