2392 Commits (565198b44bc77792affcf9ce19909b63bd90870c)

Author	SHA1	Message	Date
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	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
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
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
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
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
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
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
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	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
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
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
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
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
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
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
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
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
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
Joshua Colp	f89ab79862	res_pjsip_nat: Don't assume a Contact header will always contain a URI. ... If the 'rewrite_contact' option was enabled and a Contact header was received which contained a '*' a crash would occur. This change makes the res_pjsip_nat module ignore the Contact header if it contains only a '*'. (closes issue ASTERISK-23101) Reported by: Matt Jordan ........ Merged revisions 405019 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405020 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	61f215250e	External MWI AMI support. ... The external MWI AMI interface provides a thin wrapper around the core external MWI resource. The resource adds the following AMI actions: MWIGet, MWIDelete, and MWIUpdate. (closes issue AFS-46) Review: https://reviewboard.asterisk.org/r/3061/ ........ Merged revisions 404954 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404955 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	9fa171e547	External MWI core support. ... * The core external MWI resource provides for MWI message counts persistence using sorcery. With sorcery, the user is able to configure which sorcery wizzard backend to use if the default astdb is not desired. * The core external MWI resoruce provides some debugging CLI commands enabled by defining MWI_DEBUG_CLI. The debugging CLI commands are: "mwi delete all", "mwi delete like <regex>", "mwi delete mailbox <mailbox>", "mwi list all", "mwi list like <regex>", "mwi show mailbox <mailbox>", and "mwi update mailbox <mailbox> [<new> [<old>]]". (closes issue AFS-43) Review: https://reviewboard.asterisk.org/r/3061/ ........ Merged revisions 404952 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404953 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	986c9e897d	res_pjsip_outbound_registration: Don't assume that a registration client will always exist. ... ........ Merged revisions 404935 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404936 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	11f18e4724	res_pjsip_outbound_registration: Create registration client in pj thread. ... Depending on which threading was loading the outbound registration it was possible for the registration client to be allocated outside of a pj thread. This change moves the creation inside the synchronous task where it is guaranteed it will occur in a pj thread. Reported by: Rob Thomas ........ Merged revisions 404923 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404924 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	f8b55f16d2	res_pjsip_logger: Add the ASTERISK_FILE_VERSION macro ... Registering yourself with the Asterisk core is the nice thing to do, even when you're a logging module. ........ Merged revisions 404855 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404856 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	c6df713da7	res_pjsip_authenticator_digest: Fix md5 hash buffer ... An md5 hash is 32 bytes long. The char buffer must be at least 33 bytes to avoid clobbering of the stack. This patch also fixes a potential clobbering in test_utils.c. Thanks to Andrew Nagy for reporting and testing this out in #asterisk-dev Reported by: Andrew Nagy Tested by: Andrew Nagy ........ Merged revisions 404843 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404844 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	0ae89e7b7e	res_pjsip: Ensure more URI validation happens in pj threads. ... ........ Merged revisions 404737 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404738 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	f0f23655c9	res_pjsip_outbound_registration: Ensure URI validation happens in a pjlib thread. ... This change moves outbound registration URI validation into the task executed within a pjlib thread. Reported by: Andrew Nagy ........ Merged revisions 404725 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404726 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	821ab51381	res_pjsip: add 'set_var' support on endpoints ... Added a new 'set_var' option for ast_sip_endpoint(s). For each variable specified that variable gets set upon creation of a pjsip channel involving the endpoint. (closes issue ASTERISK-22868) Reported by: Joshua Colp Review: https://reviewboard.asterisk.org/r/3095/ ........ Merged revisions 404663 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404664 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	f720a9ac89	chan_pjsip: Handle hanging up before calling. ... Channel creation in Asterisk is broken up into two steps: requesting and calling. In some cases a channel may be requested but never called. This happens in the ChanIsAvail dialplan application for determining if something is reachable or not. The PJSIP channel driver did not take this situation into account and attempted to end a session that was never called out on. The code now checks the session state to determine if the session has been called out on and if not terminates it instead of ending it. (closes issue ASTERISK-23074) Reported by: Kilburn ........ Merged revisions 404652 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404653 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	e583dca9dd	res_pjsip_endpoint_identifier_ip: Accept hostnames in the 'match' field. ... Hostnames specified in the 'match' field will be resolved and all addresses returned. Each address will be added to the endpoint identifier for the matching process. Reported by: Rob Thomas ........ Merged revisions 404613 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404620 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	135b9d3562	res_pjsip_outbound_registration: Add validation for 'server_uri' and 'client_uri'. ... When applying configuration for outbound registrations the 'server_uri' and 'client_uri' fields were not validated. The code will now confirm that they exist and that they contain parseable SIP URIs. Reported by: Andrew Nagy ........ Merged revisions 404592 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404593 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	c3d5c41dae	res_pjsip_pubsub: Ensure dialog manipulation happens on proper thread. ... When destroying a subscription we remove the serializer from its dialog and decrease its reference count. Depending on which thread dropped the subscription reference count to 0 it was possible for this to occur in a thread where it is not possible. (closes issue ASTERISK-22952) Reported by: Matt Jordan ........ Merged revisions 404553 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404554 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	5a15803a1b	res_pjsip/pjsip_cli: fix compilation error caused by passing ast_free ... When wanting to pass *free as a function pointer, ast_free_ptr has to be used instead of ast_free. This allows it to be compiled with MALLOC_DEBUG enabled. ........ Merged revisions 404531 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404532 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	40a7f68e4b	ari: Remove support for specifying channel vars during origination. ... When we added support for specifying channel variables for an origination, we didn't consider how that would interact with another feature, namely specifying request parameters in a JSON request body. The method of specifying channel variables (as a flat JSON object passed in the JSON body) interferes with parsing parameters out of the request body. Unfortunately, fixing this would be a backward incompatible change. In the interest of keeping the API sane and keeping our release schedule, we're dropping the feature for specifying channel variables in the origination request. We will bring the feature back soon, as a backward compatible addition to the API. (closes issue ASTERISK-23051) Review: https://reviewboard.asterisk.org/r/3088 ........ Merged revisions 404509 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404512 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	b172d369c4	res_pjsip: Add PJSIP CLI commands ... Implements the following cli commands: pjsip list aors pjsip list auths pjsip list channels pjsip list contacts pjsip list endpoints pjsip show aor(s) pjsip show auth(s) pjsip show channels pjsip show endpoint(s) Also... Minor modifications made to the AMI command implementations to facilitate reuse. New function ast_variable_list_sort added to config.c and config.h to implement variable list sorting. (issue ASTERISK-22610) patches: pjsip_cli_v2.patch uploaded by george.joseph (License 6322) ........ Merged revisions 404480 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404507 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	344cdab3a7	Fix issue where PJSIP blind transferer dialog may not complete as planned. ... When transferring to a dialplan extension that will not place any outbound calls, the only control frames that the PJSIP REFER framehook will receive are inconsequential (such as unhold or srcchange). As such, we shouldn't allow for the reception of those types of frames prevent us from signaling to the transferring party that the transfer has completed successfully once voice frames are read. Thanks to Jonathan Rose for pointing this out. ........ Merged revisions 404439 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404452 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	7ddfe12aff	res_stasis_device_state: Set resource type for subscriptions to deviceState ... The documentation for ARI already specifies that the device state resource when used for subscribing for events is "deviceState", not "device_state". The code, however, used "device_state"; although this was inconsistent as well in doxygen comments in resource_applications. Because the actual resource being subscribed to is /deviceStates/{device}/, it makes sense for the resource type specifier to be deviceState. Note that the key value in the events is still "device_state". ........ Merged revisions 404437 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404438 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	72c282cc66	ao2_iterator: Mini-audit of the ao2_iterator loops in the new code files. ... * Fixed several places where ao2_iterator_destroy() was not called. * Fixed several iterator loop object variable reference problems. * Fixed res_parking AMI actions returning non-zero. Only the AMI logoff action can return non-zero. Review: https://reviewboard.asterisk.org/r/3087/ ........ Merged revisions 404434 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404436 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	433c6f010f	res_pjsip: Ignore 401/407 responses for transactions and dialogs we don't know about. ... Under normal conditions it is unlikely we will ever receive a response for a transaction or dialog we don't know about but if any are received ignore them. ........ Merged revisions 404371 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404372 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago