asterisk

Commit Graph

Author	SHA1	Message	Date
Jonathan Rose	862adf23cf	chan_sip: Send 408 on retransmit timeout instead of 603 (closes issue ASTERISK-20124) Reported by: Walter Doekes ........ Merged revisions 371824 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371825 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@371845 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	cb85c66baf	Fix misleading documentation in agents.conf.sample regarding ackcall usage. The documentation made it sound as if the DTMF acknowledgment was needed at the time the agent logs in, rather than when the agent is called. This is likely a relic from the days when there were multiple ways of logging in agents. (closes issue AST-962) reported by Steve Pitts ........ Merged revisions 371787 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371789 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@371790 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	29bda732a8	Fix incorrect documentation of the MailboxStatus manager command. The "Waiting" field was misdocumented as reporting the number of messages waiting. In reality, it simply indicated the presence or absence of waiting messages. ........ Merged revisions 371782 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371783 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@371784 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	cc2171a003	svn:ignore pjproject bin & output for all platforms. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@371753 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	adbadf6253	Fix incorrectly documented option in queues.conf sharedlastcall defaults to "no" not "yes" (closes issue AST-979) reported by Steve Pitts ........ Merged revisions 371747 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371748 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@371750 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	50d55266a1	Re-add merge and block properties. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@371749 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	8a2234472b	Fixes ast_rwlock_timed[rd\|wr]lock for BSD and variants. The original implementations simply wrap pthread functions, which take absolute time as an argument. The spinlock version for systems without those functions treated the argument as a delta. This patch fixes the spinlock version to be consistent with the pthread version. (closes issue ASTERISK-20240) Reported by: Egor Gorlin Patches: lock.c.patch uploaded by Egor Gorlin (license 6416) ........ Merged revisions 371718 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@371720 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	a97c6e85f8	Implement workaround for BETTER_BACKTRACES crash When compiling with BETTER_BACKTRACES enabled, Asterisk will sometimes crash when "core show locks" is run. This happens regularly in the testsuite since several tests run "core show locks" to help with debugging. This seems to be a fault with libraries on certain operating systems (notably CentOS 6.2/6.3) running on virtual machines and utilizing gcc 4.4.6. (closes issue ASTERISK-20090) ........ Merged revisions 371690 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371691 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@371692 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	2994f7ff41	mf_detect: incorrectly used DTMF_GSIZE instead of MF_GSIZE ........ Merged revisions 371662 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371663 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@371664 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	266d2cb75b	Add support for call-id logging to chan_motif. Review: https://reviewboard.asterisk.org/r/2077/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@371619 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	ff4674440d	Fix misuses of asprintf throughout the code. This fixes three main issues * Change asprintf() uses to ast_asprintf() so that it pairs properly with ast_free() and no longer causes MALLOC_DEBUG to freak out. * When ast_asprintf() fails, set the pointer NULL if it will be referenced later. * Fix some memory leaks that were spotted while taking care of the first two points. (Closes issue ASTERISK-20135) reported by Richard Mudgett Review: https://reviewboard.asterisk.org/r/2071 ........ Merged revisions 371590 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371591 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@371592 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	d131f3985a	Use thread-local storage to store pj_thread_descs. pj_thread_register() takes a parameter of type pj_thread_desc. It was assumed that pj_thread_register either used this item temporarily or made a copy of it. Unfortunately, all it does is keep a pointer to the structure in thread-local storage. This means that if our pj_thread_desc goes out of scope, then pjlib will be referencing bogus data quite often, most commonly on operations involving a pj_mutex_t. In our case, our pj_thread_desc was on the stack and went out of scope very shortly after registering our thread with pjlib. With this change, the pj_thread_desc is stored in thread-local storage so the pointer that pjlib keeps in thread-local storage will reference legitimate memory. (closes issue ASTERISK-20237) reported by Jeremy Pepper Patches: ASTERISK-20237.patch uploaded by Mark Michelson (license #5049) Tested by Jeremy Pepper git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@371571 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	3f789aa865	Ignore recovered zero-length secondary UDPTL packets In some cases, recovering lost packets using the secondary packet recovery mechanism with UDPTL/T.38 can result in the recovery of zero-length packets. These must be ignored or the frame generated from them can cause segfaults and allocation failures. (closes issue ASTERISK-19762) (closes issue ASTERISK-19373) Reported-by: Benjamin (bulkorok) Reported-by: Rob Gagnon (rgagnon) ........ Merged revisions 371544 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371545 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@371546 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	73755ee4c2	Recorded merge of revisions 371529 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Remove old debug code from http configuration loading (closes issue ASTERISK-20254) Reported by: Andrew Latham Patches: http.diff uploaded by Andrew Latham (license #5985) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@371530 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	80900ac680	Remove old debug code from http configuration loading (closes issue ASTERISK-20254) Reported by: Andrew Latham Patches: http.diff uploaded by Andrew Latham (license #5985) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@371520 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	85084f390a	Fix typo in JabberSend that looked for '2' instead of '@' in recipient argument The summary says about all there is to say. (closes issue ASTERISK-20239) Reported by: Gregory Porras git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@371518 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	b86c036cd8	Make the name of the "HangupCauseClear" application consistent The name of the "HangupCauseClear" application is "HangupCauseClear", not "HangupcauseClear". The incorrect case of 'cause' caused the XML documentation to not register properly. As an aside, this commit message felt very awkward, but I'm not sure how else to note that "X", which has to be "X", was referred to as "x". (closes issue ASTERISK-20253) Reported by: Andrew Latham Patches: hangupcause.diff uploaded by Andrew Latham (license #5985) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@371516 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	999d44a199	Update module support level on a variety of modules and compiler options Some core support modules and compiler options were no longer tagged with a module support level. This patch adds 'core' back to those options. Note that this patch modifies a few of the patches provided by Andrew Latham slightly. res_curl and res_fax are both 'core' supported modules. (closes issue ASTERISK-20215) Reported by: Andrew Latham Tested by: mjordan Patches: astcanary.diff (license #5985) uploaded by Andrew Latham cflagsxml.diff (license #5985) uploaded by Andrew Latham curl_fax.diff (license #5985) uploaded by Andrew Latham soundsxml.diff (license #5985) uploaded by Andrew Latham git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@371507 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	099107b0aa	Fix memory leak in XML documentation When formatting documentation fields, the XML documentation parser calls xmldoc_get_formatted. This function allocates a string buffer at the beginning of its routine. Unfortunately, on certain code paths, it also calls xmldoc_string_cleanup, which assumes that it will create the string buffer. The previously allocated string buffer is then leaked by the xmldoc_string_cleanup routine. Now: we don't do that. (closes issue AST-932) Reported by: Alexander Homig ........ Merged revisions 371469 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371491 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@371492 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	ef1f1b16a8	When a peer registers using WebSocket do not resolve the Contact provided. (closes issue ASTERISK-20238) Reported by: james.mortensen git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@371482 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	33c2ee2e90	Add instrumentation to subsystem reloads When Asterisk is built with TEST_FRAMEWORK defined, Asterisk will now generate TestEvent AMI events on subsystem reloads such as cdr, dnsmgr, extconfig, etc. (issue PQ-1126) ........ Merged revisions 371436 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371437 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@371438 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	bda007b5c2	Add some additional H.264 attributes, "max-smbps" and "max-fps", for passthrough. (closes issue ASTERISK-20206) Reported by: ddkprog Patches: res_format_attr_h264.c.diff uploaded by ddkprog (license 6008) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@371426 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Russell Bryant	c9cf719b36	rtp: Ensure defaults are set without rtp.conf. While building up a new install to test chan_motif, I ran into a failure due to icesupport being disabled. This was due to me not having an rtp.conf. It was intended in the code for it to be enabled by default, but it was only applied if rtp.conf existed. This patch updates res_rtp_asterisk to be consistent in how it handles defaults. A few options didn't have their default values set globally, including icesupport. They are now set and icesupport is enabled by default, even if you do not have an rtp.conf. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@371425 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	ed2b01f301	Handle integer over/under-flow in ast_parse_args The strtol family of functions will return _MIN/_MAX on overflow. To detect when an overflow has happened, errno must be set to 0 before calling the function, then checked afterward. (closes issue ASTERISK-20120) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2073/ ........ Merged revisions 371392 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371398 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@371399 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	94bbcafcd2	Add module reload instrumentation for TEST_FRAMEWORK This adds AMI events for module reloads when Asterisk is built with TEST_FRAMEWORK enabled and corrects generation of the module load AMI event. (issue PQ-1126) ........ Merged revisions 371393 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371394 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@371395 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	cf9265008d	chan_sip: Use pvt outgoing_call variable to set Remote-Party-ID Header Previously the pvt SIP_OUTGOING flag was used instead, which will frequently flip during reinvites. (closes issue AST-897) Reported by: Thomas Arimont ........ Merged revisions 371357 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371358 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@371382 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	80ee807c13	chan_sip: Trigger reinvite if the SDP answer is included in the SIP ACK Under certain conditions, a SIP transaction involving directmedia wouldn't trigger a re-invite because the SDP answer was included in an ACK instead of in a message that we would have triggered the invite with. This patch just queues a source change control frame if the dialog is using directmedia when we find sdp for an ACK. (closes issue AST-913) Reported by: Thomas Arimont ........ Merged revisions 371337 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371338 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@371355 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	760909af3c	Fix bug where final queue member would not be removed from memory. If a static queue had realtime members, then there could be a potential for those realtime members not to be properly deleted from memory. If the queue's members were loaded from realtime and then all the members were deleted from the backend, then the queue would still think these members existed. The reason was that there was a short- circuit in code such that if there were no members found in the backend, then the queue would not be updated to reflect this. Note that this only affected static queues with realtime members. Realtime queues with realtime members were unaffected by this issue. (closes issue ASTERISK-19793) reported by Marcus Haas ........ Merged revisions 371306 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371313 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@371324 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Michael L. Young	75f68294fc	Fix Segfault When Registering SIP Over WebSockets The helper function, get_address_family_filter, in chan_sip for dns resolution by address family was not recognizing the websockets transport and resulting in a null pointer being sent to functions in netsock2, in an attempt to determine if we are bound to ANY address ([::]) or not. This patch fixes this issue by handling the transport types SIP_TRANSPORT_WS and SIP_TRANSPORT_WSS which results in a sock address being set properly for use in determining the address family. (closes issue ASTERISK-20221) Reported by: Sven Beisiegel Tested by: Sven Beisiegel, James Mortensen Patches: asterisk-20221-ws-family-filter.diff uploaded by Michael L. Young (license 5026) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@371295 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	5add0570b5	Avoid unconditional NULLing of mwipvt on relatedpeer on SIP dialog destruction The other instance of this bug was fixed by jcolp/file in r121496. If we are destroying a dialog only set the MWI dialog pointer on the related peer to NULL if it is the dialog currently being destroyed. (closes issue ASTERISK-20119) Patch-by: Misha Vodsedalek ........ Merged revisions 371270 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371271 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@371272 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	d7fbceb55b	Add HANGUPCAUSE information to callee channels This adds HANGUPCAUSE information to called channels so that hangup handlers can, in conjunction with predial dialplan execution, access the hangupcause information when the dialed channel hangs up on a one-to-one basis instead of a many-to-one basis as with HANGUPCAUSE usage on the caller channel. Review: https://reviewboard.asterisk.org/r/2069/ (closes issue ASTERISK-20198) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@371258 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	121495f7a9	Add test instrumentation This adds test instrumentation for loading and unloading of modules and for certain actions in MeetMe to be used in the testsuite or any other consumer of AMI events. These will only be generated when Asterisk is built with TEST_FRAMEWORK enabled. (issue PQ-1131) (issue PQ-1133) ........ Merged revisions 371201 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371203 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@371227 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	85a6ab78ce	Fix problem where incorrect pointer was checked for nullity. ........ Merged revisions 371198 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371199 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@371200 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	90b098d7b9	Update CHANGES for private party ID. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@371146 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	bdcdb0dc87	Fix a couple of documentation problems in app_queue.c * The RemoveQueueMember app made mention of options that could be passed in, but no options are supported. I have removed the listing of options from the documentation. * The RQMSTATUS variable did not list "NOTDYNAMIC" as a possible value that could be set. (closes issue AST-949) reported by Steve Pitts (closes issue AST-954) reported by Steve Pitts ........ Merged revisions 371141 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371142 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@371143 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	81abe96456	_ _ _ _ _ _ / \ ___\| \|_ ___ _ __(_)___\| \| __ / \| / \| / _ \ / __\| __/ _ \ '__\| / __\| \|/ / \| \| \| \| / ___ \__ \\| \| __/ \| \| \__ \ < \| \| \| \| /_/ \_\___/\__\___\|_\| \|_\|___/_\|\_\ \|_\| \|_\| Because it's one greater than 10. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@371121 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	fb6238899b	Add private representation of caller, connected and redirecting party ids. This patch adds the feature "Private representation of caller, connected and redirecting party ids", as previously discussed with us (DATUS) and Digium. 1. Feature motivation Until now it is quite difficult to modify a party number or name which can only be seen by exactly one particular instantiated technology channel subscriber. One example where a modified party number or name on one channel is spread over several channels are supplementary services like call transfer or pickup. To implement these features Asterisk internally copies caller and connected ids from one channel to another. Another example are extension subscriptions. The monitoring entities (watchers) are notified of state changes and - if desired - of party numbers or names which represent the involving call parties. One major feature where a private representation of party names is essentially needed, i.e. where a party name shall be exclusively signaled to only one particular user, is a private user-specific name resolution for party numbers. A lookup in a private destination-dependent telephone book shall provide party names which cannot be seen by any other user at any time. 2. Feature Description This feature comes along with the implementation of additional private party id elements for caller id, connected id and redirecting ids inside Asterisk channels. The private party id elements can be read or set by the user using Asterisk dialplan functions. When a technology channel is initiating a call, receives an internal connected-line update event, or receives an internal redirecting update event, it merges the corresponding public id with the private id to create an effective party id. The effective party id is then used for protocol signaling. The channel technologies which initially support the private id representation with this patch are SIP (chan_sip), mISDN (chan_misdn) and PRI (chan_dahdi). Once a private name or number on a channel is set and (implicitly) made valid, it is generally used for any further protocol signaling until it is rewritten or invalidated. To simplify the invalidation of private ids all internally generated connected/redirecting update events and also all connected/redirecting update events which are generated by technology channels -- receiving regarding protocol information - automatically trigger the invalidation of private ids. If not using the private party id representation feature at all, i.e. if using only the 'regular' caller-id, connected and redirecting related functions, the current characteristic of Asterisk is not affected by the new extended functionality. 3. User interface Description To grant access to the private name and number representation from the Asterisk dialplan, the CALLERID, CONNECTEDLINE and REDIRECTING dialplan functions are extended by the following data types. The formats of these data types are equal to the corresponding regular 'non-private' already existing data types: CALLERID: priv-all priv-name priv-name-valid priv-name-charset priv-name-pres priv-num priv-num-valid priv-num-plan priv-num-pres priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag CONNECTEDLINE: priv-name priv-name-valid priv-name-pres priv-name-charset priv-num priv-num-valid priv-num-pres priv-num-plan priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag REDIRECTING: priv-orig-name priv-orig-name-valid priv-orig-name-pres priv-orig-name-charset priv-orig-num priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type priv-orig-subaddr-odd priv-orig-tag priv-from-name priv-from-name-valid priv-from-name-pres priv-from-name-charset priv-from-num priv-from-num-valid priv-from-num-pres priv-from-num-plan priv-from-subaddr priv-from-subaddr-valid priv-from-subaddr-type priv-from-subaddr-odd priv-from-tag priv-to-name priv-to-name-valid priv-to-name-pres priv-to-name-charset priv-to-num priv-to-num-valid priv-to-num-pres priv-to-num-plan priv-to-subaddr priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd priv-to-tag Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2030/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371120 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	5ff199d99a	Fix a comparison that was causing presence tests to fail. A recent change made it so that device state changes that were not actual "changes" would not get reported to subscribers. The problem was that this inadvertently blocked presence updates as well. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371113 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alexandr Anikin	cb525e5c38	remove ALREADYGONE flag on ooh323 call data by ooh323_indicate (CONGESTION/BUSY) due to call hasn't gone there really. This indication arrive from asterisk core not h.323 stack (closes issue ASTERISK-19308) Reported by: Dmitry Melekhov Patches: ASTERISK-19308.patch ........ Merged revisions 371089 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371090 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371091 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alexandr Anikin	bcc1a0142f	Send re-register packets by GRQ (gatekeeper request) interval (close issue ASTERISK-20094) Patches: ASTERISK-20094-2.patch ........ Merged revisions 371060 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371061 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371081 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alexandr Anikin	c7b2858322	restore calling cb functions by timer expire this was broken in rev 369602 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371059 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	ca481359b9	Fix pickup extension channel reference error. You cannot unref a pointer and then expect to ref it again later. * Fix potential NULL pointer deref if the call pickup search fails. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371052 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alexandr Anikin	7709885125	Introdue 'ooh323 show gk' cli command that show status of connection to H.323 Gatekeeper (GkClient state) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371043 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alexandr Anikin	6153acebe8	Fix to resend GRQ/RRQ if RRJ (registration reject) is received (close issue ASTERISK-20094) Patches: ASTERISK-20094.patch ........ Merged revisions 371011 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371022 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371036 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	18d5041981	Use better libss7 detection test and move libpri compile test. ........ Merged revisions 371012 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371013 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371030 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alexandr Anikin	0d82844cad	change opening h323 logfile with append mode instead of overwrite ........ Merged revisions 370988 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370989 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371010 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	609061a8c0	Correct documentation for the MeetMe x flag The documentation for the x flag for MeetMe incorrectly described its function as closing down the conference when the last marked user left. It actually causes the users with that flag to leave the conference when the last marked user exits. The functionality of this flag is not changing. ........ Merged revisions 370985 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370986 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370987 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	9ee8b3c0f6	Extend extension state callbacks to have more information. Quote from review board: This patch extends the extension state callbacks so that monitoring channels (as chan_sip) get more information of the devices which are responsible for an extension state change. The additional information is needed by chan_sip to present names/numbers of the caller and callee in an early-state SIP notification. Users of extenstion state callback not interested in the additional information are not affected by the changes. Motivation: to present the involved party's name/number in an early-state nofification (used by the notified device as a pickup offer) one after another so that a user can see which call he will pick up in an undirected pickup. Such a pickup offer to a user shall indicate the same call (number/name-A calls number/name-B) as the call which would be picked up when an undirected pickup is executed. Users interested in additional state info must use the new functions ast_extension_state_add_extended() resp. ast_extension_state_add_destroy_extended() to register an extended state callback. When the callback is registered this way, an extra member device_state_info of struct ast_state_cb_info is passed to the callback in addition to the aggregated extension state. This container holds an object for every device of the monitored extension hint consisting of the device name, the device state and a channel reference to the channel which (presumably) caused the device state. The information is used by chan_sip for early-state notifications. When the state of a device changes and the new state contains AST_EVENT_RINGING, an early-state notification is sent to the subscribed devices with the caller/callee names/numbers of the oldest ringing channel of the monitored extension. The notified user may then invoke a direct pickup, which will pickup exactly this channel. Users of the old non-extended callbacks will only be called when the aggregated state did change (same behavior as before). Users of the extended callback will also be called when the state is unchanged but does contain AST_EVENT_RINGING. That could be the case if two channels are ringing at one device and one of them hangs up, so the aggregated state does not change. This way the monitoring channel can create a new early-state notification with the now ringing party-ids. Review: https://reviewboard.asterisk.org/r/2048 This contribution comes from Guenther Kelleter git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370979 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	1067294065	DUNDi: Add CLI commands DUNDi show cache and DUNDi show hints (closes issue ASTERISK-18390) Reported by: Peter Racz Patches: dundi_cli_cache.patch.v2 uploaded by Peter Racz (license #6290) ASTERISK-18390_dundi_cli_cache_jrose_mods_v2.diff uploaded by Jonathan Rose (license #6182) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370978 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Michael L. Young	5cf9eb4645	Fix Not Unreferencing A Spied Channel When a channel hangs up while being spied upon and the option to exit the ChanSpy application when the spied on channel hangs up is set, ast_autochan_destroy is not being called and therefore a reference to the spied upon channel is not removed. The symptom being reported was that when using func_group in the dialplan and calling "group show channels" at the cli, the spied upon channel was still being shown while "core show channels" showed that the channel was not up. This patch calls ast_autochan_destroy when a spied upon channel hangs up and the option to exit the ChanSpy application is set, removing the reference to the channel allowing the count for the group that the spied channel was part of to be decremented. (closes issue ASTERISK-17515) Reported by: Arkadiusz Malka Tested by: Alexandr Gordeev, Michael L. Young Patches: asterisk-17515-destroy-autochan.diff uploaded by Michael L. Young (license 5026) ........ Merged revisions 370952 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370954 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370955 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago

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