asterisk

Commit Graph

Author	SHA1	Message	Date
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
Kevin Harwell	1f6c34a6c9	res_pjsip: AOR option qualify_frequency not respected on startup If an endpoint had previously dynamically registered a contact and the contact information was successfully stored in astdb then upon restart the qualify notifications would not be sent out if the qualify_frequency was set. This was due to the fact that only permanent contacts were being checked and scheduled for qualifies on startup. Modified the code to check and schedule all registered contacts at startup. (closes issue ASTERISK-23062) Reported by: Rusty Newton Review: https://reviewboard.asterisk.org/r/3124/ ........ Merged revisions 405748 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405749 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	7054e12ef2	manager: Originate doesn't abort on failed format_cap allocation action_originate responds to the remote system with an error when cap==NULL, but doesn't return (abort the originate). Patched to return. (closes issue ASTERISK-23034) Reported by: Corey Farrell Patches: ASTERISK-23034.patch uploaded by coreyfarrell (license 5909) ........ Merged revisions 405745 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405746 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405747 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	fc241d6f52	PJSIP: Fix outbound OPTIONS support When path support was added and contacts were made available during request creation and transmission, the code path used by outbound qualify support was not modified correctly and was causing request creation to fail. This ensures that outbound request creation with only a contact and no dialog, endpoint, or uri can succeed which restores qualify support. Reported by: gtjoseph Reported by: kharwell ........ Merged revisions 405743 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405744 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	a48798ce95	res_fax: check_modem_rate() returned incorrect rate for V.27 According to the new standard for V.27 and V.32 they are able to transmit at a bit rate of 4,800 or 9,600. The check_mode_rate function needed to be updated to reflect this. Also, because of this change the default 'minrate' value was updated to be 4800. (closes issue ASTERISK-22790) Reported by: Paolo Compagnini Patches: res_fax.txt uploaded by looserouting (license 6548) ........ Merged revisions 405656 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 405693 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405694 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405695 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	3373a5332b	chan_pjsip: initial device state on endpoints is INVALID When endpoints get loaded their device state gets set to 'INVALID' because the channel driver has not been loaded yet. Fixed by updating the device state for every endpoint upon load of the channel driver. (closes issue ASTERISK-23065) Reported by: Rusty Newton Review: https://reviewboard.asterisk.org/r/3123/ ........ Merged revisions 405643 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405644 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	3c90fc0bfd	Make 12 - 12.1 CHANGES log the same as in 12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405589 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	3e499ccd56	Blocked revisions 405587 ........ Remove subversion conflict tag accidentally left in CHANGES git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405588 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	8ba05ae67e	Include CHANGES info for r405553 ........ Merged revisions 405585 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405586 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	07481baafd	cel_manager: Don't crash if configuration file is invalid. The cel_manager module did not properly handle the case where the configuration file was invalid. The module will now output a warning message and disable itself if this occurs. Reported by: Bryan Walters ........ Merged revisions 405581 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 405582 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405583 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405584 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	7cbb6eab15	PJSIP: Add Path header support This adds Path support to chan_pjsip in res_pjsip_path.c with minimal additions in res_pjsip_registrar.c to store the path and additions in res_pjsip_outbound_registration.c to enable advertisement of path support to registrars and intervening proxies. Path information is stored on contacts and is enabled via Address of Record (AoRs) and Registration configuration sections. While adding path support, it became necessary to be able to add SIP supplements that handled messages outside of sessions, so a framework for handling these types of hooks was added in parallel to the already-existing session supplements and several senders of out-of-dialog requests were refactored as a result. (closes issue ASTERISK-21084) Review: https://reviewboard.asterisk.org/r/3050/ ........ Merged revisions 405565 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405566 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	aa9db707c5	ARI: Add mailboxes resource for controlling and polling external MWI Adds the following AMI commands: PUT mailboxes/mailboxName modifies mailbox state and implicitly creates new mailboxes GET mailboxes/mailboxName retrieves a JSON representation of a single mailbox if it exists GET mailboxes retrieves a JSON array of all mailboxes DELETE mailbox/mailboxName deletes a mailbox Note that res_mwi_external must be loaded for these functions to actually do anything. Review: https://reviewboard.asterisk.org/r/3117/ ........ Merged revisions 405553 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405554 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	ed0d083596	string container: Remove unnecessary RAII_VAR usage and string object lock. ........ Merged revisions 405541 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405542 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	5516cda6af	chan_sip: fix Local From tag on outbound register regression In ASTERISK-12117, an improvement to insure consistant local from tags on outbound registrations resulted in an undesirable behavior - caused by leftover unexpired sip_pvt dialogs (with the previous cseq number), resulting in many uncessary REGISTER requests. Instead of significant rework of transmit_register(), this change deletes the dialogs after a 200 OK response indiciating a successful registration, keeping the old dialogs from interfering with normal operation. (closes issue ASTERISK-22946) Reported by: Stephan Eisvogel Review: https://reviewboard.asterisk.org/r/3109/ ........ Merged revisions 405433 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 405434 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405435 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405437 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	828f339a9c	verbosity: Fix performance of console verbose messages. The per console verbose level feature as previously implemented caused a large performance penalty. The fix required some minor incompatibilities if the new rasterisk is used to connect to an earlier version. If the new rasterisk connects to an older Asterisk version then the root console verbose level is always affected by the "core set verbose" command of the remote console even though it may appear to only affect the current console. If an older version of rasterisk connects to the new version then the "core set verbose" command will have no effect. * Fixed the verbose performance by not generating a verbose message if nothing is going to use it and then filtered any generated verbose messages before actually sending them to the remote consoles. * Split the "core set debug" and "core set verbose" CLI commands to remove the per module verbose support that cannot work with the per console verbose level. * Added a silent option to the "core set verbose" command. * Fixed "core set debug off" tab completion. * Made "core show settings" list the current console verbosity in addition to the root console verbosity. * Changed the default verbose level of the 'verbose' setting in the logger.conf [logfiles] section. The default is now to once again follow the current root console level. As a result, using the AMI Command action with "core set verbose" could again set the root console verbose level and affect the verbose level logged. (closes issue AST-1252) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/3114/ ........ Merged revisions 405431 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405432 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405436 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	aced8bdd2e	Fix erroneous behavior when sending auth rejection to artificial endpoint. We were not including an authentication challenge when sending a 401 response to unmatched endpoints. This was due to the conversion to use a vector for authentication section names on an endpoint. The vector for artificial endpoints was empty, resulting in the challenge being sent back containing no challenges. This is worked around by placing a bogus value in the artificial endpoint's auth vector. This value is never looked up by anything, since they instead will directly call ast_sip_get_artificial_auth(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405420 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Damien Wedhorn	9ec4d15c8e	Skinny: do not add call to missed calls list if answered elsewhere. Patch updates skinny devices with a SKINNY_CONNECTED callstate if an inbound ringing or callwaiting call is answered elsewhere. ........ Merged revisions 405367 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405369 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	545593fe7b	Blocked revisions 405350 ........ PJSIP: Backport r405270 - Unhold on reinvite without SDP Adds behavior to unhold on a reinvite without an SDP section Review: https://reviewboard.asterisk.org/r/3106/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405351 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	f1497fe220	res_pjsip: Fix CLI tab completion issues This fixes several issues with the new res_pjsip CLI tab completion such as output of headers during tab completion and being able to tab-complete more items than the code actually handled (further items would simply be ignored). (closes issue ASTERISK-23081) Review: https://reviewboard.asterisk.org/r/3115/ Reported by: xrobau ........ Merged revisions 405338 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405339 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	8585340b87	res_ari: Fix various memory leaks. This change fixes a few memory leaks that were found based on a mailing list post. 1. Some JSON response messages were never freed. This was caused by the documentation stating that message references were stolen when in reality they were not. The code now follows the documentation and usage has been updated. 2. HTTP response headers were never freed. 3. The variable list for wildcards paths was never freed. (closes issue ASTERISK-23128) Reported by: Kenneth Watson (on list) Review: https://reviewboard.asterisk.org/r/3119/ ........ Merged revisions 405325 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405326 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	373965dbff	CDRs: Synchronize dialplan applications that manipulate CDRs with the engine In https://reviewboard.asterisk.org/r/3057/, applications and functions that manipulate CDRs were made to interact over Stasis. This was done to synchronize manipulations of CDRs from the dialplan with the updates the engine itself receives over the message bus. This change rested on a faulty premise: that messages published to the CDR topic or to a topic that forwards to the CDR topic are synchronized with the messages handled by the CDR topic subscription in the CDR engine. This is not the case. There is no ordering guaranteed for two messages published to the same topic; ordering is only guaranteed if a message is published to the same subscriber. Stasis was modified in r405311 to allow a publisher to synchronize on the subscriber. This patch uses that API to synchronize the CDR publishers with the CDR engine message router, which maintains the overall topic subscription. (closes issue ASTERISK-22884) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3099/ ........ Merged revisions 405312 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405314 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	f8aaf585a3	stasis: Add methods to allow for synchronous publishing to subscriber This patch adds an API call to Stasis that allows a publisher to publish a stasis message that will not return until a specific subscriber handles the message. Since a subscriber can have their own forwarding topic which orders messages from many topics, this allows a publisher who knows of that subscriber to synchronize to that subscriber regardless of the forwarding relationships between topics. This is of particular use for dialplan applications that need to synchronize on a particular subscriber's handling of a message. (issue ASTERISK-22884) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3099/ ........ Merged revisions 405311 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405313 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	60ed8159a1	Print "<unknown>" for artificial endpoint in PJSIP security events. Previously, this printed a UUID, which was not very clear when dealing with an artificial endpoint. Review: https://reviewboard.asterisk.org/r/3113 ........ Merged revisions 405298 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405299 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	4bde62eb40	Logging callid: Fix some sizeof() references per coding guidelines. ........ Merged revisions 405281 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405282 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405284 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	42b087c2df	PJSIP: Add unhold on reinvite without SDP behavior Review: https://reviewboard.asterisk.org/r/3106/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405270 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Damien Wedhorn	de6563c618	Fix chan_dahdi copile issue in dev-mode. Error "unused variable i in dahdi_create_channel_range" when compiling in dev-mode. Small restructure to dahdi_create_channel_range to move the for(x) loop and int i,x to a block within the IFDEF. ........ Merged revisions 405268 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405269 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	04c5c39d56	res_pjsip_messaging: potential for field values in from/to headers to be missing Added in ability to specify display name format ("name" <sip:name@ipaddr:port>) for a given URI and made sure it was fully propagated to the outgoing message. Also made it so outoing messages in res_pjsip always send as "sip:". (closes issue ASTERISK-22924) Reported by: Anthony Messina Review: https://reviewboard.asterisk.org/r/3094/ ........ Merged revisions 405266 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405267 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	51901aa2ed	astobj2: Correct ao2_iterator opacity violations This corrects the ao2_iterator opacity violations in res_pjsip_session.c by adding a global function to get the number of elements inside the container hidden behind the iterator. (closes issue ASTERISK-23053) Review: https://reviewboard.asterisk.org/r/3111/ Reported by: Richard Mudgett ........ Merged revisions 405253 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405254 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	bce38c0cc5	res_rtp_asterisk: Fails to resume WebRTC call from hold In ast_rtp_ice_start if the ice session create check list failed, start check was never initiated and ice_started was never set to true. Upon re-entering the function (for instance, [un]hold) it would try to create the check list again with duplicate remote candidates. Fixed so that if the create check list fails the necessary data structures are properly re-initialized for any subsequent retries. Note, it was decided to not stop ice support (by calling ast_rtp_ice_stop) on a check list failure because it possible things might still work. However, a debug message was added to help with any future troubleshooting. (closes issue ASTERISK-22911) Reported by: Vytis Valentinavičius Patches: works_on_my_machine.patch uploaded by xytis (license 6558) ........ Merged revisions 405234 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405235 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405236 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	50b2d6eec1	app_confbridge: Fix crash caused when waitmarked/marked users leave together When waitmarked users join a ConfBridge, the conference state is transitioned from EMPTY -> INACTIVE. In this state, the users are maintined in a waiting users list. When a marked user joins, the ConfBridge conference transitions from INACTIVE -> MULTI_MARKED, and all users are put onto the active list of users. This process works correctly. When the marked user leaves, if they are the last marked user, the MULTI_MARKED state does the following: (1) It plays back a message to the bridge stating that the leader has left the conference. This requires an unlocking of the bridge. (2) It moves waitmarked users back to the waiting list (3) It transitions to the appropriate state: in this case, INACTIVE However, because it plays the prompt back to the bridge before moving the users and before finishing the state transition, this creates a race condition: with the bridge unlocked, waitmarked users who leave the conference (or are kicked from it) can cause a state transition of the bridge to another state before the conference is transitioned to the INACTIVE state. This causes the state machine to get a bit wonky, often leading to a crash when the MULTI_MARKED state attempts to conclude its processing. This patch fixes this problem: (1) It prevents kicked users from being kicked again. That's just a nicety. (2) More importantly, it fixes the race condition by only playing the prompt once the state has transitioned correctly to INACTIVE. If waitmarked users sneak out during the prompt being played, no harm no foul. Review: https://reviewboard.asterisk.org/r/3108/ Note that the patch committed here is essentially the same as uploaded by Simon Moxon on ASTERISK-22740, with the addition of the double kick prevention. (closes issue AST-1258) Reported by: Steve Pitts (closes issue ASTERISK-22740) Reported by: Simon Moxon patches: ASTERISK-22740.diff uploaded by Simon Moxon (license 6546) ........ Merged revisions 405215 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405216 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405217 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Walter Doekes	c94e4ee1ac	"Minimun" typo. ........ Merged revisions 405160 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 405161 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405162 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405163 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	9674196009	Use proper case for checking if digest authentication is used. ........ Merged revisions 405131 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405144 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	537f4a9dac	pbx_lua: Add support for Lua 5.2 This adds support for Lua 5.2 in pbx_lua which is available on newer operating systems. (closes issue ASTERISK-23011) Review: https://reviewboard.asterisk.org/r/3075/ Reported by: George Joseph Patch by: George Joseph ........ Merged revisions 405090 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 405091 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405124 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405130 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	522593f901	Add the missing part of r400140 When the patch to add retry-on-forbidden-response was committed, part of the patch for chan_sip was not committed which caused the feature to be entirely nonfunctional. This corrects the code in question. (closes issue ASTERISK-17138) Review: https://reviewboard.asterisk.org/r/2874 ........ Merged revisions 405033 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 405081 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405083 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405129 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	34c595beb9	res_pjsip_acl: Fix another case of assuming a contact will always contain a URI. ........ Merged revisions 405034 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405035 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago

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