asterisk

Commit Graph

Author	SHA1	Message	Date
Joshua Colp	1a95c9a906	When a peer registers using WebSocket do not resolve the Contact provided. (closes issue ASTERISK-20238) Reported by: james.mortensen ........ Merged revisions 371482 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371483 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	064c7bd456	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 ........ Merged revisions 371438 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371439 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Russell Bryant	b8b425971c	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. ........ Merged revisions 371425 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371428 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	1f64b85106	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) ........ Merged revisions 371426 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371427 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	69dc8e3adb	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 ........ Merged revisions 371399 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371400 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	34265d5265	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 ........ Merged revisions 371395 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371396 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	d4879edd8e	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 ........ Merged revisions 371382 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371383 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	70ca2e51a1	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 ........ Merged revisions 371355 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371356 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	f4a34ee89c	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 ........ Merged revisions 371324 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371325 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Michael L. Young	7aac43b4b1	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) ........ Merged revisions 371295 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371296 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	837e00a5cc	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 ........ Merged revisions 371272 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371277 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	76d642ff69	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) ........ Merged revisions 371258 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371259 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	45c6620d74	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 ........ Merged revisions 371227 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371228 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	5d02d8e016	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 ........ Merged revisions 371200 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371202 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	9d79ccd3db	Add UPGRADE-11.txt file; update UPGRADE.txt to reflect Asterisk 12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371170 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	e2cd045c7e	Update CHANGES for private party ID. ........ Merged revisions 371146 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371147 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	567b35e547	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 ........ Merged revisions 371143 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371144 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	938ba94264	Remove 10 properties, add 11 properties git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371134 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
Mark Michelson	b03e7cc4c7	Move a SIP change up to the other SIP changes in the CHANGES file. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370953 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	eb9e645a27	Allow support for early media on AMI originates and call files. This is based on the work done by Olle Johansson on review board. The idea is that the channel specified in an AMI originate or call file is typically not connected to the outgoing extension until the channel has been answered. With this change, an EarlyMedia header can be specified for AMI originates and an early_media option can be specified in call files. With this option set, once early media is received on a channel, it will be connected with the outgoing extension. (closes issue ASTERISK-18644) Reported by Olle Johansson Review: https://reviewboard.asterisk.org/r/1472 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370951 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	ee849b461f	Add AMI_CLIENT dialplan function Implementation of a dialplan function for checking manager accounts. Right now it only returns the number of logged in sessions for a manager account, but other attributes can be added later. Patch by: Olle Johansson Review: https://reviewboard.asterisk.org/r/421/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370943 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	4a389854a4	Create the payload type if it does not exist when setting information based on the 'm' line. An rtpmap attribute is not required for defined payload numbers. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370927 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	062becab80	Convert sig_analog to use a global callback table. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370926 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	e571897441	Do not define a cause that doesn't actually exist AST_CAUSE_NOTDEFINED is a placeholder for usage when there is no cause information. As such, it should not be defined and translatable as a cause. ........ Merged revisions 370923 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370924 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370925 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	f1dce57742	Fix the analog dial *0 flash-hook of bridged peer feature. The flash-hook the bridged peer feature now correctly determines if the bridged peer is another chan_dahdi channel, that it is an analog channel, and that it has the correct signaling for an FXO port. It now also flash-hooks the correct channel. ........ Merged revisions 370900 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370901 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370902 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	35bf5efeaf	Convert sig_pri to use a global callback table. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370893 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	f24be2740b	Convert sig_ss7 to use a global callback table. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370887 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Damien Wedhorn	f4d1b7ab12	Rewrite of skinny debugging. Debugging messages and associated controls only compiled in if configured with --enable-dev-mode. Debug messages provide more detail (including thread id) and are grouped so the user/dev can limit the type of messages displayed. Functionally no real change to chan_skinny. Review: https://reviewboard.asterisk.org/r/2040/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370881 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	8c5333f34e	Payload and RTP code are must remain separate since in non-Asterisk format cases they differ. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370860 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	ca68390f0b	Recorded merge of revisions 370858 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Add missing AST_CAUSE_* -> text translations ........ Merged revisions 370856 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370859 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	3d212da105	Add missing AST_CAUSE_* -> text translations A few of these were missing from the list and are necessary for the Who Hung Up? functionality. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370851 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	da808a0b66	Fix a bug uncovered by the test suite where the RTP payload number was not getting set. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370845 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	15e41c7542	Reduce memory consumption significantly for users of the RTP engine API by storing only the payloads present and in use instead of every possible one. Review: https://reviewboard.asterisk.org/r/2052/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370832 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	5c4578f4ad	Add named callgroups/pickupgroups This patch adds named calledgroups/pickupgroups to Asterisk. Named groups are implemented in parallel to the existing numbered callgroup/pickupgroup implementation. However, unlike the existing implementation, which is limited to a maximum of 64 defined groups, the number of defined groups allowed for named callgroups/pickupgroups is effectively unlimited. Named groups are configured with the keywords "namedcallgroup" and "namedpickupgroup". This corresponds to the numbered group definitions of "callgroup" and "pickupgroup". Note that as the implementation of named groups coexists with the existing numbered implementation, a defined named group of "4" does not equate to numbered group 4. Support for the named groups has been added to the SIP, DAHDI, and mISDN channel drivers. Review: https://reviewboard.asterisk.org/r/2043 Uploaded by: Guenther Kelleter(license #6372) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370831 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	096baa0897	Revert r370820 That change is wrong, wrong, wrong. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370821 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	4ec5c83f69	Update the MySQL voicemail_data contrib script to reflect Asterisk 11 changes All voicemails now have a 'msg_id' included in their metadata. The ODBC message storage backend now requires this column; as such, the MySQL contrib script that creates the voicemail_data table has been updated with the appropriate column information. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370820 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago

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