7734 Commits (cc06020f23fd175061a65a36c8b6809feca0cc18)

Author	SHA1	Message	Date
Igor Goncharovskiy	2053fc3159	Fix issue with no sound in both way in case of previous call to chan_unistim phone was canceled. ...	13 years ago
Alec L Davis	b0143074fa	IAX2: Transfer Reject: Lock bridgecallno before touching it, refactor ... 1). When touching the bridgecallno, we need to lock it. 2). Remove magic number '0' and replace with TRANSFER_NONE. 3). Exit early if no bridgecallno. 4). Reduce indentation. Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2613/ ........ Merged revisions 391333 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 391334 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391335 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Damien Wedhorn	75e7c3a8fa	Change chan_skinny to use core transfer API. ... Changes for both attended and blind transfers in chan_skinny to use the new transfer API instead of masquerade. (closes issue ASTERISK-21526) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2557/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391297 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	0cec7dcdcd	chan_iax2: nativebridge refactor, missed unlock bridgecallno ... ........ Merged revisions 391143 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 391148 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391154 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	a6ab25a004	fix bad edit after conflict resolution ... ........ Merged revisions 391107 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 391111 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391112 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	d264518524	IAX2: refactor nativebridge transfer ... remove triple checking of iaxs[fr->callno]->transferring reduce indentation. Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2602/ ........ Merged revisions 391065 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 391084 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391102 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	1c8c91b63a	IAX2: fix race condition with nativebridge transfers. ... 1). When touching the bridgecallno, we need to lock it. 2). stop_stuff() which calls iax2_destroy_helper() Assumes the lock on the pvt is already held, when iax2_destroy_helper() is called. Thus we need to lock the bridgecallno pvt before we call stop_stuff(iaxs[fr->callno]->bridgecallno); 3). When evaluating the state of 'callno->transferring' of the current leg, we can't change it to READY unless the bridgecallno is locked. Why, if we are interrupted by the other call leg before 'transferring = TRANSFER_RELEASED', the interrupt will find that it is READY and that the bridgecallno is also READY so Releases the legs. (closes issue ASTERISK-21409) Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2594/ ........ Merged revisions 391062 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 391063 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391064 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	6114166237	Refactor chan_dahdi/sig_analog/sig_pri and chan_misdn to use the common transfer functions. ... (closes issue ASTERISK-21523) Reported by: Matt Jordan (closes issue ASTERISK-21524) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2600/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390804 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	2dc8a06006	Refactor the features configuration scheme. ... Features configuration is handled in its own API in features_config.h and features_config.c. This way, features configuration is accessible to anything that needs it. In addition, features configuration has been altered to be more channel-oriented. Most callers of features API code will be supplying a channel so that the individual channel's settings will be acquired rather than the global setting. Missing from this commit is XML documentation for the features configuration. That will be handled in a separate commit. Review: https://reviewboard.asterisk.org/r/2578/ (issue ASTERISK-21542) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390751 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	8e3f37adef	Add a BUGBUG note. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390639 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	cfe32ec1da	Add attended transfer support for chan_sip.c ... This now uses the core API for performing attended transfers. Review https://reviewboard.asterisk.org/r/2513 (Closes issue ASTERISK-21520) reported by Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389869 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	fac3839e68	Adds support for a core attended transfer function plus adds some hiding of masquerades. ... The attended transfer API call can complete the attended transfer in a number of ways depending on the current bridged states of the channels involved. The hiding of masquerades is done in some bridging-related functions, such as the manager Bridge action and the Bridge dialplan application. In addition, call pickup was edited to "move" a channel rather than masquerade it. Review: https://reviewboard.asterisk.org/r/2511 (closes issue ASTERISK-21334) Reported by Matt Jordan (closes issue Asterisk-21336) Reported by Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389848 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jason Parker	154fbf8cae	Split Hold event into Hold/Unhold, and move it into core. ... (closes issue ASTERISK-21487) Review: https://reviewboard.asterisk.org/r/2565/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389746 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	06be8463b6	Migrate a large number of AMI events over to Stasis-Core ... This patch moves a number of AMI events over to the Stasis-Core message bus. This includes: * ChanSpyStart/Stop * MonitorStart/Stop * MusicOnHoldStart/Stop * FullyBooted/Reload * All Voicemail/MWI related events In addition, it adds some Stasis-Core and AMI support for generic AMI messages, refactors the message router in AMI to use a single router with topic forwarding for the topics that AMI cares about, and refactors MWI message types and topics to be more name compliant. Review: https://reviewboard.asterisk.org/r/2532 (closes issue ASTERISK-21462) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389733 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Igor Goncharovskiy	1fb6f365ec	Fix several problems caused by multiple line usage with i2004 phones. ...	13 years ago
Richard Mudgett	3d63833bd6	Merge in the bridge_construction branch to make the system use the Bridging API. ... Breaks many things until they can be reworked. A partial list: chan_agent chan_dahdi, chan_misdn, chan_iax2 native bridging app_queue COLP updates DTMF attended transfers Protocol attended transfers git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389378 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Damien Wedhorn	8d6b0c9f3b	Add transfer softkey to ringout state to enable blond transfers. ... (closes issue ASTERISK-21327) Reported by: wedhorn Tested by: myself Patches: skinny-blindxfer01.diff uploaded by wedhorn (license 5019) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389164 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Damien Wedhorn	01d6e8dbc9	Add call forward no answer to skinny and cleanup general callfwd handling. ... CallforwardNoAnswer uses a sched to determine when to forward the call. Defaults to 20secs but configurable in skinny.conf. Adds dialType to each subchannel structure to be used to differentiate between normal dials that result in a call being placed (default) and other uses for the skinny_dialer (such as cfwd digit collection). Restructured all cfwd handling to use this new arrangement. (closes issue ASTERISK-21292) Reported by: wedhorn Tested by: myself Patches: skinny-callfwdnoans03.diff uploaded by wedhorn (license 5019) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389097 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	b97c71bb11	Fix shutdown assertions in stasis-core ... In r388005, macros were introduced to consistently define message types. This added an assert if a message type was used either before it was initialized or after it had been cleaned up. It turns out that this assertion fires during shutdown. This actually exposed a hidden shutdown ordering problem. Since unsubscribing is asynchronous, it's possible that the message types used by the subscription could be freed before the final message of the subscription was processed. This patch adds stasis_subscription_join(), which blocks until the last message has been processed by the subscription. Since joining was most commonly done right after an unsubscribe, a stasis_unsubscribe_and_join() convenience function was also added. Similar functions were also added to the stasis_caching_topic and stasis_message_router, since they wrap subscriptions and have similar problems. Other code in trunk was refactored to join() where appropriate, or at least verify that the subscription was complete before being destroyed. Review: https://reviewboard.asterisk.org/r/2540 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389011 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Michael L. Young	91bab76422	Remove Character Limit On "inkeys" For IAX2 ... Currently, the buffer for processing "inkeys" is limited to 256 characters. If the user has many keys and the names of those key files are long, the 256 character limit is not enough. * Change inkeys buffer to be dynamic (closes issue ASTERISK-21398) Reported by: Pavel Kopchyk Tested by: Pavel Kopchyk, Michael L. Young Patches: asterisk-21398-iax2-inkeys-dynamic-buffer_v3.diff by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2501/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389009 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	b90bba7a30	Stasis: Update security events to use Stasis ... Also moves ACL messages to the security topic and gets rid of the ACL topic (closes issue ASTERISK-21103) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2496/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388975 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Michael L. Young	4ff6e61808	Fix Crash Caused By One-way Audio With auto_* NAT Settings Fix ... The prior code committed, r385473, failed to take into consideration that not all outgoing calls will be to a peer. My fault. This patch does the following: * Check if there is a related peer involved. If there is, check and set NAT settings according to the peer's settings. * Fix a problem with realtime peers. If the global setting has auto_force_rport set and we issued a "sip reload" while a peer is still registered, the peer's flags for NAT are reset to off. When this happens, we were always setting the contact address of the peer to that of the full contact info that we had. (closes issue ASTERISK-21374) Reported by: jmls Tested by: Michael L. Young Patches: asterisk-21374-fix-crash-and-rt-peers.diff by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2524/ ........ Merged revisions 388601 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388602 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	f2ba7c04fb	chan_gulp: Minor readability Improvements to chan_gulp ... (closes issue ASTERISK-21670) Reported by: Snuffy Review: https://reviewboard.asterisk.org/r/2473/ Patches: gulp-coding-guide.diff uploaded by snuffy (license 5024) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388526 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	785be439f9	Allow mISDN to send PROGRESS messsage. ... * Made isdn_msg_parser.c build a progress message with the mandatory progress indicator IE. (The mISDNuser NT state machine rejected sending the incomplete message.) Note: The associated mISDN and mISDNuser patches respectively are viewable here: http://svnview.digium.com/svn/thirdparty?view=rev&rev=200 http://svnview.digium.com/svn/thirdparty?view=rev&rev=201 (closes issue AST-1153) Reported by: Guenther Kelleter Patches: progress-chan_misdn.diff (license #6372) patch uploaded by Guenther Kelleter progress-misdn.diff (license #6372) mISDN patch uploaded by Guenther Kelleter progress-misdnuser.diff (license #6372) mISDNuser patch uploaded by Guenther Kelleter ........ Merged revisions 388425 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 388426 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388427 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Sean Bright	2cfedc12ad	Fix copy/paste error in one-touch-recording implementation. ... ........ Merged revisions 388253 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388254 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	e06e519a90	Initial support for endpoints. ... An endpoint is an external device/system that may offer/accept channels to/from Asterisk. While this is a very useful concept for end users, it is surprisingly not a core concept within Asterisk itself. This patch defines ast_endpoint as a separate object, which channel drivers may use to expose their concept of an endpoint. As the channel driver creates channels, it can use ast_endpoint_add_channel() to associate channels to the endpoint. This updated the endpoint appropriately, and forwards all of the channel's events to the endpoint's topic. In order to avoid excessive locking on the endpoint object itself, the mutable state is not accessible via getters. Instead, you can create a snapshot using ast_endpoint_snapshot_create() to get a consistent snapshot of the internal state. This patch also includes a set of topics and messages associated with endpoints, and implementations of the endpoint-related RESTful API. chan_sip was updated to create endpoints with SIP peers, but the state of the endpoints is not updated with the state of the peer. Along for the ride in this patch is a Stasis test API. This is a stasis_message_sink object, which can be subscribed to a Stasis topic. It has functions for blocking while waiting for conditions in the message sink to be fulfilled. (closes issue ASTERISK-21421) Review: https://reviewboard.asterisk.org/r/2492/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387932 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	efd28c676a	chan_sip: NOTIFYs for BLF start queuing up and fail to be sent out after retries fail ... RFC6665 4.2.2: ... after a failed State NOTIFY transaction remove the subscription The problem is that the State Notify requests rely on the 200OK reponse for pacing control and to not confuse the notify susbsystem. The issue is, the pendinginvite isn't cleared if a response isn't received, thus further notify's are never sent. The solution, follow RFC 6665 4.2.2's 'SHOULD' and remove the subscription after failure. (closes issue ASTERISK-21677) Reported by: Dan Martens Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2475/ ........ Merged revisions 387875 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 387880 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387885 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	cc855cf9d8	Make a log NOTICE more explicit that the event comes from DAHDI and not PRI. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387740 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	1eac5a7988	Stasis: Convert network change events into network change stasis messages ... (issue ASTERISK-21103) Review: https://reviewboard.asterisk.org/r/2490/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387594 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	e236c2ea36	Use the configured formats for Gulp sessions if there are no joint formats between requested formats and configured formats. ... (closes issue ASTERISK-21756) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387545 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	f7f58b7bc2	chan_sip: Session-Expires: Set timer to correctly expire at (~2/3) of the interval when not the refresher ... RFC 4028 Section 10 if the side not performing refreshes does not receive a session refresh request before the session expiration, it SHOULD send a BYE to terminate the session, slightly before the session expiration. The minimum of 32 seconds and one third of the session interval is RECOMMENDED. Prior to this asterisk would refresh at 1/2 the Session-Expires interval, or if the remote device was the refresher, asterisk would timeout at interval end. Now, when not refresher, timeout as per RFC noted above. (closes issue ASTERISK-21742) Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2488/ ........ Merged revisions 387344 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 387345 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387369 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	7f0f53958b	chan_sip: Honor Session-Expires in 200OK response when it's a RE-INVITE when asterisk is the refresher. ... RFC 4028 Section 7.2 "UACs MUST be prepared to receive a Session-Expires header field in a response, even if none were present in the request." What changed After ASTERISK-20787, inbound calls to asterisk with no Session-Expires in the INVITE are now are offered a Session-Expires (1800 asterisk default) in the response, with asterisk as the refresher. Symptom: After 900 seconds (asterisk default refresher period 1800), asterisk RE-INVITEs the device, the device may respond with a much lower Session-Expires (180 in our case) value that it is now using. Asterisk ignores this response, as it's deemed both an INBOUND CALL, and a RE-INVITE. After 180 seconds the device times out and sends BYE (hangs up), asterisk is still working with the refresher period of 1800 as it ignored the 'Session Expires: 180' in the previous 200OK response. Fix: handle_response_invite() when 200OK, remove check for outbound and reinvite. (closes issue ASTERISK-21664) Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2463/ ........ Merged revisions 387312 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 387319 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387327 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	0b020e8c0b	chan_dahdi: fix lower bound check with -ve integer conversion from a float ... Lower bound of a 16bit signed int is -32768 not -32767 (closes issue ASTERISK-21744) Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) ........ Merged revisions 387297 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 387298 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387299 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	eea6df46d5	Simplify chan_local.c:manager_optimize_away() using ao2_find(). ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387261 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	e4db1903a2	Cleanup chan_local.c:local_new(). ... * Remove t and ama local variables. There is no way they could be anything other than default because p->owner can only be NULL at this point. * Rename tmp and tmp2 to owner and chan respectively. * Remove redundant initialization of channel context, exten, priority. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387260 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	a2a8afe07a	Trivial changes. Comments, parentheses, spelling, wording. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387212 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	e6f7de360d	Make chan_local locals container an explicit list container. ... Pretending that chan_local locals container can have more than one bucket is silly. The container has no key to help search. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387211 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	3992df6d41	Whitespace changes. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387210 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	5fec9b8d1f	Remove some unnecessary calls to ast_bridged_channel() in chan_unistim.c ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387185 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	213834ffd4	Remove some unnecessary calls to ast_bridged_channel() in chan_mgcp.c ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387184 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	30cf1a590c	Remove some unnecessary calls to ast_bridged_channel() in chan_skinny.c ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387183 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	b050778ce8	Remove some unnecessary calls to ast_bridged_channel() in chan_iax2.c ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387182 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	5ab5715646	Remove some unnecessary calls to ast_bridged_channel() in chan_dahdi.c/sig_analog.c ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387181 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	b693a72378	Prevent crash in 'sip show peers' when the number of peers on a system is large ... When you have lots of SIP peers (according to the issue reporter, around 3500), the 'sip show peers' CLI command or AMI action can crash due to a poorly placed string duplication that occurs on the stack. This patch refactors the command to not allocate the string on the stack, and handles the formatting of a single peer in a separate function call. (closes issue ASTERISK-21466) Reported by: Guillaume Knispel patches: fix_sip_show_peers_stack_overflow_asterisk_11.3.0-v2.patch uploaded by gknispel (License 6492) ........ Merged revisions 387134 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387135 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	3a5a0f3f26	Move some annoying chan_dahdi debug messages to level 5. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387108 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	8e257fe819	Stasis Core: Refactor ACL Change events to go out over the stasis core msg bus ... (issue ASTERISK-21103) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2481/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387037 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jason Parker	cc9b4d8da4	Fix a log message. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386990 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	74f2318051	Merge the pimp_my_sip branch into trunk. ... The pimp_my_sip branch is being merged at this point because it offers basic functionality, and from an API standpoint, things are complete. SIP work is *not* feature-complete; however, with the completion of the SUBSCRIBE/NOTIFY API, all APIs (except a PUBLISH API) have been created, and thus it is possible for developers to attempt to create new SIP work. API documentation can be found in the doxygen in the code, but usability documentation is still lacking. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386540 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Michael L. Young	b4c881c86e	Fix Displaying Symmetric RTP Global Setting ... * Use comedia_string() to display correctly the symmetric rtp setting when running "sip show settings" ........ Merged revisions 386486 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386487 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Michael L. Young	735026ccf6	Change Case On Forcerport For Consistency ... * Change "ForcerPort" to "Forcerport" to match everywhere else it is displayed ........ Merged revisions 386483 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 386484 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386485 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago