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2013-06-07Refactor chan_dahdi/sig_analog/sig_pri and chan_misdn to use the common ↵Richard Mudgett
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
2013-06-06Refactor the features configuration scheme.Mark Michelson
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
2013-06-06Add a BUGBUG note.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390639 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-28Add attended transfer support for chan_sip.cMark Michelson
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
2013-05-28Adds support for a core attended transfer function plus adds some hiding of ↵Mark Michelson
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
2013-05-24Split Hold event into Hold/Unhold, and move it into core.Jason Parker
(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
2013-05-24Migrate a large number of AMI events over to Stasis-CoreMatthew Jordan
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
2013-05-24Fix several problems caused by multiple line usage with i2004 phones.Igor Goncharovskiy
Reported by: Daniel Bohling, MihaiMircea (closes issue ASTERISK-21061) (closes issue ASTERISK-21120) ........ Merged revisions 389661 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389663 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-21Merge in the bridge_construction branch to make the system use the Bridging API.Richard Mudgett
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
2013-05-19Add transfer softkey to ringout state to enable blond transfers.Damien Wedhorn
(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
2013-05-18Add call forward no answer to skinny and cleanup general callfwd handling.Damien Wedhorn
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
2013-05-17Fix shutdown assertions in stasis-coreDavid M. Lee
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
2013-05-17Remove Character Limit On "inkeys" For IAX2Michael L. Young
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
2013-05-17Stasis: Update security events to use StasisJonathan Rose
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
2013-05-13Fix Crash Caused By One-way Audio With auto_* NAT Settings FixMichael L. Young
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
2013-05-13chan_gulp: Minor readability Improvements to chan_gulpJonathan Rose
(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
2013-05-10Allow mISDN to send PROGRESS messsage.Richard Mudgett
* 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
2013-05-10Fix copy/paste error in one-touch-recording implementation.Sean Bright
........ 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
2013-05-08Initial support for endpoints.David M. Lee
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
2013-05-08chan_sip: NOTIFYs for BLF start queuing up and fail to be sent out after ↵Alec L Davis
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
2013-05-06Make a log NOTICE more explicit that the event comes from DAHDI and not PRI.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387740 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-03Stasis: Convert network change events into network change stasis messagesJonathan Rose
(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
2013-05-03Use the configured formats for Gulp sessions if there are no joint formats ↵Joshua Colp
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
2013-05-02chan_sip: Session-Expires: Set timer to correctly expire at (~2/3) of the ↵Alec L Davis
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
2013-05-02chan_sip: Honor Session-Expires in 200OK response when it's a RE-INVITE when ↵Alec L Davis
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
2013-05-02chan_dahdi: fix lower bound check with -ve integer conversion from a float Alec L Davis
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
2013-05-01Simplify chan_local.c:manager_optimize_away() using ao2_find().Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387261 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-01Cleanup chan_local.c:local_new().Richard Mudgett
* 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
2013-05-01Trivial changes. Comments, parentheses, spelling, wording.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387212 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-01Make chan_local locals container an explicit list container.Richard Mudgett
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
2013-05-01Whitespace changes.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387210 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-01Remove some unnecessary calls to ast_bridged_channel() in chan_unistim.cRichard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387185 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-01Remove some unnecessary calls to ast_bridged_channel() in chan_mgcp.cRichard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387184 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-01Remove some unnecessary calls to ast_bridged_channel() in chan_skinny.cRichard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387183 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-01Remove some unnecessary calls to ast_bridged_channel() in chan_iax2.cRichard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387182 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-01Remove some unnecessary calls to ast_bridged_channel() in ↵Richard Mudgett
chan_dahdi.c/sig_analog.c git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387181 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-01Prevent crash in 'sip show peers' when the number of peers on a system is largeMatthew Jordan
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
2013-05-01Move some annoying chan_dahdi debug messages to level 5.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387108 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-04-30Stasis Core: Refactor ACL Change events to go out over the stasis core msg busJonathan Rose
(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
2013-04-30Fix a log message.Jason Parker
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386990 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-04-25Merge the pimp_my_sip branch into trunk.Mark Michelson
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
2013-04-25Fix Displaying Symmetric RTP Global SettingMichael L. Young
* 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
2013-04-25Change Case On Forcerport For ConsistencyMichael L. Young
* 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
2013-04-14Don't attempt to create a voice frame on a read errorMatthew Jordan
Prior to this patch, a read error in snd_pcm_readi would still be treated as a nominal result when constructing a voice frame from the expected data. Since the value returned is negative, as opposed to the number of samples read, this could result in a crash. With this patch, we now return a null frame when a read error is detected. Note that the patch on ASTERISK-21329 was modified slightly for this commit, in that we bail immediately on detecting the read error, rather than bypassing the construction of the voice frame. (closes issue ASTERISK-21329) Reported by: Keiichiro Kawasaki patches: chan_alsa.diff uploaded by kawasaki (License 6489) ........ Merged revisions 385633 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 385634 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385635 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-04-12Fix One-Way Audio With auto_* NAT Settings When SIP Calls Initiated By PBXMichael L. Young
When we reload Asterisk or chan_sip, the flags force_rport and comedia that are turned on and off when using the auto_force_rport and auto_comedia nat settings go back to the default setting off. These flags are turned on when needed or off when not needed at the time that a peer registers, re-registers or initiates a call. This would apply even when only the default global setting "nat=auto_force_rport" is being used, which in this case would only affect the force_rport flag. Everything is good except for the following: The nat setting is set to auto_force_rport and auto_comedia. We reload Asterisk and the peer's registration has not expired. We load in the settings for the peer which turns force_rport and comedia back to off. Since the peer has not re-registered or placed a call yet, those flags remain off. We then initiate a call to the peer from the PBX. The force_rport and comedia flags stay off. If NAT is involved, we end up with one-way audio since we never checked to see if the peer is behind NAT or not. This patch does the following: * Moves the checking of whether a peer is behind NAT into its own function * Create a function to set the peer's NAT flags if they are using the auto_* NAT settings * Adds calls in sip_request_call() to these new functions in order to setup the dialog according to the peer's settings (closes issue ASTERISK-21374) Reported by: Michael L. Young Tested by: Michael L. Young Patches: asterisk-21374-auto-nat-outgoing-fix_v2.diff Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2421/ ........ Merged revisions 385473 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385474 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-04-12IAX2 defer_full_frames fail to get sentAlec L Davis
Ensure iax2_process_thread is signalled when a deferred frame is queued to it. (closes issue ASTERISK-18827) Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2426/ ........ Merged revisions 385429 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 385430 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385431 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-04-12IAX2, prevent network thread starting before all helper threads are readyAlec L Davis
On startup, it's possible for a frame to arrive before the processing threads were ready. In iax2_process_thread() the first pass through falls into ast_cond_wait, should a frame arrive before we are at ast_cond_wait, the signal will be ignored. The result iax2_process_thread stays at ast_cond_wait forever, with deferred frames being queued. Fix: When creating initial idle iax2_process_threads, wait for init_cond to be signalled after each thread is started. (issue ASTERISK-18827) Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2427/ ........ Merged revisions 385402 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 385403 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385406 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-04-10Fix crash in chan_sip when a core initiated op occurs at the same time as a BYEMatthew Jordan
When a BYE request is processed in chan_sip, the current SIP dialog is detached from its associated Asterisk channel structure. The tech_pvt pointer in the channel object is set to NULL, and the dialog persists for an RFC mandated period of time to handle re-transmits. While this process occurs, the channel is locked (which is good). Unfortunately, operations that are initiated externally have no way of knowing that the channel they've just obtained (which is still valid) and that they are attempting to lock is about to have its tech_pvt pointer removed. By the time they obtain the channel lock and call the channel technology callback, the tech_pvt is NULL. This patch adds a few checks to some channel callbacks that make sure the tech_pvt isn't NULL before using it. Prime offenders were the DTMF digit callbacks, which would crash if AMI initiated a DTMF on the channel at the same time as a BYE was received from the UA. This patch also adds checks on sip_transfer (as AMI can also cause a callback into this function), as well as sip_indicate (as lots of things can queue an indication onto a channel). Review: https://reviewboard.asterisk.org/r/2434/ (closes issue ASTERISK-20225) Reported by: Jeff Hoppe ........ Merged revisions 385170 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 385173 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385174 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-04-05Fix For Not Overriding The Default Settings In chan_sipMichael L. Young
The initial report was that the "nat" setting in the [general] section was not having any effect in overriding the default setting. Upon confirming that this was happening and looking into what was causing this, it was discovered that other default settings would not be overriden as well. This patch works similar to what occurs in build_peer(). We create a temporary ast_flags structure and using a mask, we override the default settings with whatever is set in the [general] section. In the bug report, the reporter who helped to test this patch noted that the directmedia settings were being overriden properly as well as the nat settings. This issue is also present in Asterisk 1.8 and a separate patch will be applied to it. (issue ASTERISK-21225) Reported by: Alexandre Vezina Tested by: Alexandre Vezina, Michael L. Young Patches: asterisk-21225-handle-options-default-prob_v4.diff Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2385/ ........ Merged revisions 384827 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384828 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-04-03chan_dahdi: Change inband_on_proceeding option default to no/disabled.Richard Mudgett
(issue ASTERISK-21151) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384711 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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