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2013-03-08This patch adds a new message bus API to Asterisk.David M. Lee
For the initial use of this bus, I took some work kmoore did creating channel snapshots. So rather than create AMI events directly in the channel code, this patch generates Stasis events, which manager.c uses to then publish the AMI event. This message bus provides a generic publish/subscribe mechanism within Asterisk. This message bus is: - Loosely coupled; new message types can be added in seperate modules. - Easy to use; publishing and subscribing are straightforward operations. In addition to basic publish/subscribe, the patch also provides mechanisms for message forwarding, and for message caching. (issue ASTERISK-20887) (closes issue ASTERISK-20959) Review: https://reviewboard.asterisk.org/r/2339/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382685 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-08Remove unused functionMatthew Jordan
After r382670, get_ip_and_port_from_sdp was no longer used. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382671 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-08Don't reset the RTP address on a glare re-INVITEMatthew Jordan
Originally, way back in r201583, we added the alternate RTP address so that the RTP engine would expect to receive audio from a new source when a glare re-INVITE occurred. In r382589, we remove the alternate RTP source, as the 'secret' probation mode allows for switching to a new RTP source when a previous source stops sending RTP. At the time, it seemed appropriate to set the RTP source based on the information in the glared re-INVITE. Unfortunately, that doesn't work so well - in a glared re-INVITE that occurs with no SDP - such as in a connected line update that glances - we'll set the RTP source to an invalid address. In subsequent re-INVITE requests from this Asterisk instance, we'll then send an invalid media address, which will result in the remote side sending a 488. Whoops. There isn't any need to reset the RTP source - if we're using strictrtp, we'll simply synchronize to a new source when we stop getting packets from the old one. If we aren't using strictrtp, then again there shouldn't be a problem. Note that the Asterisk Test Suite's connectedline test caught this error. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382670 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-07Changing log level of "Not changing threadpool size" from notice to debug.David M. Lee
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382648 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-07Load sorcery modules earlier, so they can actually be used.Jason Parker
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382636 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-07Let vm_mailbox_snapshot combine "Urgent" when no folder is specifiedMatthew Jordan
r381835 fixed a bug in vm_mailbox_snapshot where combining INBOX and Old forgot that Urgent also "counts" as new messages. This fixed the problem when any of the three folders was specified and the combine option was used. It missed the case where the folder isn't specified and we build a snapshot of all folders. This patch corrects that. ........ Merged revisions 382617 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382621 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-07Fix a memory leak in xmldocKinsey Moore
Another instance of attribute retrieval not being freed properly. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382604 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-07Resolve more memory leaks in xmldocKinsey Moore
Many places that allocated to pull out an attribute are now freed properly. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382600 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-07Add a 'secret' probation strictrtp mode to handle delayed changes in RTP sourceMatthew Jordan
Often, Asterisk may realize that a change in the source of an RTP stream is about to occur and ask that the RTP engine reset it's lock on the current RTP source. In certain scenarios, it may take awhile for the new remote system to send RTP packets, while the old remote system may continue providing RTP during that time period. This causes Asterisk to re-lock onto the old source, thereby rejecting the new source when the old source stops sending RTP and the new source begins. This patch prevents that by having a constant secondary, 'secret' probation mode enabled when an RTP source has been chosen. RTP packets from other sources are always considered, but never chosen unless the current RTP source stops sending RTP. Review: https://reviewboard.asterisk.org/r/2364 (closes issue AST-1124) Reported by: John Bigelow Tested by: John Bigelow (closes issue AST-1125) Reported by: John Bigelow Tested by: John Bigelow ........ Merged revisions 382573 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-07Fix minor memory leak in xmldocKinsey Moore
Strings retrieved via ast_xml_get_text() must be freed with ast_xml_free_text(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382587 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-07Ensure that logmsgs are freed properlyKinsey Moore
Messages sent while the logger thread is shutting down will now have their associated callid freed properly. ........ Merged revisions 382574 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382575 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-07Fix ref leak in threadpool.cKinsey Moore
If ast_threadpool_set_size with a size equal to the current size, a reference to a set_size_data structure would be leaked. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382555 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-06Resolve a ref leak in threadpool.cKinsey Moore
Ownership of the listener reference is not transferred because the listener is reffed when placed into the taskprocessor. Ensure that the listener is dereffed properly. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382489 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-05Add RFC 3327 Path header support to chan_sipMatthew Jordan
This patch adds support for RFC 3327 "Path" headers. This can be enabled in sip.conf using the 'supportpath' setting, either on a global basis or on a peer basis. This setting enables Asterisk to route outgoing out-of-dialog requests via a set of proxies by using a pre-loaded route-set defined by the Path headers in the REGISTER request. This patch also adds Realtime support for dynamically updating the Path information for a peer. A huge thank-you to Klaus Darillion and Olle E Johansson for their efforts in writing this patch. Review: https://reviewboard.asterisk.org/r/2235/ Review: https://reviewboard.asterisk.org/r/991/ (closes issue ASTERISK-16884) Reported by: klaus3000 Tested by: klaus3000, oej, mjordan patches: path-1.8.0-patch.txt uploaded by klaus3000 (License 5054) oolong-path-support-trunk in team branch by oej (License 5267) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382440 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-05Fix several unreleased mutex locks that cause problem with processing callsIgor Goncharovskiy
Reported by: Daniel Bohling Tested by: Daniel Bohling (Closes issue ASTERISK-21119) ........ Merged revisions 382409 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 382410 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382411 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-04Fixup some bridge and format capabilities comments and whitespace.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382392 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-04Fix comparison of presence state in event subsystem.Jason Parker
Several new IEs were not given types (or names), causing the comparison function to improperly succeed. This adds those. (closes issue AST-1128) ........ Merged revisions 382390 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382391 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-04Confbridge CLI new record file name check.Kevin Harwell
This fix checks to make sure that if a confbridge record start command is issued from the CLI it will always use the file name given on the CLI even if it changes between start/stop records for a conference. Previously it had been reusing the same file between start/stops even if a new filename was given. (issue AST-1088) Reported by: John Bigelow ........ Merged revisions 382385 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382386 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-01Add support for registering a sorcery handler which supports multiple fields ↵Joshua Colp
using a regex. Review: https://reviewboard.asterisk.org/r/2332/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382340 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-01Fix / Clean Up Some Items To Handle The New auto_* NAT OptionsMichael L. Young
The original report had to do with a realtime peer behind NAT being pruned and the peer's private address being used instead of its external address. Upon debugging, it was discovered that this was being caused by the addition of the auto_force_rport and auto_comedia settings. This patch does the following: * Adds a missing note to the CHANGES file indicating that the default global nat setting is auto_force_rport * Constify the 'req' parameter for check_via() * Add calls to check_via() in a couple of places in order for the auto_* settings to do their job in attempting to determine if NAT is involved * Set the flags SIP_NAT_FORCE_RPORT and SIP_PAGE2_SYMMETRICRTP if the auto_* settings are in use where it was needed * Moves the copying of peer flags up in build_peer() to before they are used; this fixes the realtime prune issue * Update the contrib/realtime schemas to allow the nat column to handle the different nat setting combinations we have This patch received a review and "Ship It!" on the issue itself. (closes issue ASTERISK-20904) Reported by: JoshE Tested by: JoshE, Michael L. Young Patches: asterisk-20904-nat-auto-and-rt-peersv2.diff Michael L. Young (license 5026) ........ Merged revisions 382322 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382323 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-28While the ICE negotiation is occurring leave strictrtp in an open state, ↵Joshua Colp
media can and will come from different places. ........ Merged revisions 382298 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382299 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-28Fix a bug with ICE and strictrtp where media could get dropped.Joshua Colp
If the end result of the ICE negotiation resulted in the path for media changing it was possible for the strictrtp code to discard the RTP packets. This change causes strictrtp to enter learning mode once again when the ICE negotiation has completed successfully. ........ Merged revisions 382296 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382297 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-28threadpool: Make ast_threadpool_push() return -1 if shutting_downRichard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382295 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-28threadpool: Whitespace and comment corrections.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382294 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-28Don't undefine bzero()/bcopy().Jason Parker
This was causing build failures against external libraries that happened to use them, unless silly hacks were added to the modules that used those headers. Review: https://reviewboard.asterisk.org/r/2359/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382292 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-28Prevent deadlock in chan_iax2 when attempting to set caller IDMatthew Jordan
A deadlock can occur in chan_iax2 when it attempts to set the caller ID, as it already holds the iax2 private lock and improperly fails to obtain the channel lock before calling ast_set_callerid. By not safely obtaining the channel lock, a locking inversion can take place, causing a deadlock. This patch solves this by calling the required deadlock avoidance functions that obtain the channel lock before setting the caller ID. Thanks to Pavel for fixing my syntax errors and testing this patch out. (closes issue ASTERISK-21128) Reported by: Pavel Troller Tested by: Pavel Troller patches: ASTERISK-21128-1.8.diff uploaded by mjordan (license 6283) ASTERISK-21128-modified-1.8.diff uploaded by Pavel Troller (license 6302) ........ Merged revisions 382233 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 382234 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382236 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-28Let channels joining a MeetMe conference opt out of the denoiserMatthew Jordan
For some channel drivers, specifically those that have a varying rate in the number of audio samples, the audio quality for a MeetMe conference can be exceedingly poor. This is due to a unilateral application of the DENOISE function in func_speex to channels joining the conference. The denoiser function in the speex library is initialized with the number of audio samples in each sample that will be provided to it. If the number of audio samples changes, the denoiser has to be thrown away and re-initialized. While this could be worked around by removing func_speex, that doesn't help if you actually use the denoiser with other channels on the system. This patches does the following: * Checks for the presence of func_speex as opposed to codec_speex when determining if the DENOISE function is present (which is where the function is actually implemented) * Adds an option to MeetMe 'n' that causes the denoiser to not be applied to a channel when it joins. This keeps the current behavior the default, but let's users disable the denoiser if it causes problems on their system. Review: https://reviewboard.asterisk.org/r/2358 (closes issue AST-1062) Reported by: Thomas Arimont ........ Merged revisions 382227 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 382230 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382232 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-27More places to eliminate the cast to argv but were not giving warnings.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382204 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-27Fix compiler warning by eliminating the need for a cast.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382203 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-27Relax dialog checking in get_sip_pvt_byid_locked so it works when the dialog ↵Joshua Colp
is forked. (closes issue ASTERISK-20638) Reported by: eelcob Patches: pedantic-call-pickup-from-tag.patch uploaded by eelcob (license 6442) ........ Merged revisions 382171 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 382174 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382182 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-26Consider linux-gnuspe as linux-gnuTzafrir Cohen
* The powerpcspe Linux port uses linux-gnuspe as the OS string. * Our build system shouldn't really care for that, so just call it linux-gnu. * Original report: Roland Stigge , http://bugs.debian.org/701505 Review: https://reviewboard.asterisk.org/r/2357/ ........ Merged revisions 382110 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 382111 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382113 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-26Correct RPID parsing for unquoted display-name.Walter Doekes
Parsing Remote-Party-ID will now succeed if display-name is of the *(token LWS) kind and not just the quoted-string kind. Review: https://reviewboard.asterisk.org/r/2341/ ........ Merged revisions 382107 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 382108 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382109 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-26Remove unneeded linux-gnueabi*Tzafrir Cohen
As of r380522 the configure scripts converts the value of linux-gnueabi* of OSARCH to "linux-gnu". So no point in testing for those values. ........ Merged revisions 382087 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 382096 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382106 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-26Clean up ConfBridge commands to account for wait_marked usersMatthew Jordan
When ConfBridge was refactored to better handle the concept of marked, wait_marked, and normal users co-existing in a conference (thereby implementing a state machine for the conference), the wait_marked users were put into their own list of conference participants, separate from the active users. This list is used for wait_marked users when they are waiting in a conference but no marked user has joined; normal users may have joined at this point however. There are several AMI/CLI commands that affect conference users that were not checking the wait_marked users list: * CLI/AMI commands that mute/unmute a participant. In this case, wait_marked users have to remain in their particular state and should not be affected - however, the commands would return "Channel not found" as opposed to the appropriate error condition. * CLI/AMI commands that kick a participant. An admin should always be able to kick a participant out of the conference. This patch fixes both sets of commands, and cleans up the CLI commands slightly by allowing them to complete a participant name (this was supposed to have been added, but the function call was commented out and wasn't implemented). Review: https://reviewboard.asterisk.org/r/2346/ (closes issue AST-1114) Reported by: John Bigelow Tested by: John Bigelow ........ Merged revisions 382068 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382070 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-26Ensure that the default bridge/user profiles are always availableMatthew Jordan
ConfBridge and Page require that there always be a default bridge and user profile available. While properties of the default profiles can be overriden in the configuration file, removing them can create situations where neither application can function properly. This patch ensures that if an administrator removes the profiles from the confbridge.conf configuration file, the profiles are added upon load. Documentation clarifying this has been added to the confbridge.conf.sample file. Review: https://reviewboard.asterisk.org/r/2356/ (closes issue AST-1115) Reported by: John Bigelow Tested by: John Bigelow ........ Merged revisions 382066 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382067 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-25Clean up use of va_end/va_args in res_config_mysqlMatthew Jordan
There were several problems using variadic argument macros in res_config_mysql. * Improper use of va_end. Multiple calls to va_end were possible resulting in an unbalanced matching of va_start/va_end. * Calls to va_arg after a possible encounter of a SENTINEL value. This patch corrects those errors. (closes issue ASTERISK-19451) Reported by: wdoekes patches: ASTERISK-19451-1.8--2.diff uploaded by wdoekes (License 5674) ........ Merged revisions 382021 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 382022 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382023 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-25More called details fixup for skinny.Damien Wedhorn
Basically sets the callerid and callername to the first device talked to for the purposes of putting the the calls made log on the device. Does not affect the device displaying who the device is currently talking to. Also some minor changes to use sub->exten in lieu of l->lastnumberdialed. (closes issue ASTERISK-21095) Reported by: wedhorn Tested by: snuffy, myself Patches: skinny-calllogsoutbound03.diff uploaded by wedhorn (license 5019) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382008 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-25Add prinotify messages to skinny.Damien Wedhorn
Adds both fixed and variable prinotify messages and clearprinotify messages to skinny. Also adds cli function for pushing messages to devices. i Initial code by snuffy, expanded by myself to include fixed messages. (closes issue ASTERISK-21091) Reported by: snuffy Tested by: snuffy, myself Patches: skinny-prinotify02.diff uploaded by wedhorn (license 5019) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382007 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-24Set the sin_family on the bind address socket during initializationMatthew Jordan
Somehow, chan_jingle has managed to operate for years without setting the sin_family on its bindaddr socket. This patch properly sets the field during initial module load to AF_INET. Note that the patch on the issue was modified slightly to change the initialization of the socket from allocation of a chan_jingle private to the module initialization, as the bindaddr object (which is static) only needs to have the address set once. (closes issue ASTERISK-19341) Reported by: andre valentin patches: 0105-chan_jingle.patch uploaded by avalentin (License 6064) ........ Merged revisions 381975 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 381976 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381977 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-24Don't display the AMI ALL class authorization for users if they don't have itMatthew Jordan
When converting AMI class authorizations to a string representation, the method always appends the ALL class authorization. This is especially important for events, as they should always communicate that class authorization - even if the event itself does not specify ALL as a class authorization for itself. (Events have always assumed that the ALL class authorization is implied when they are raised) Unfortunately, this did mean that specifying a user with restricted class authorizations would show up in the 'manager show user' CLI command as having the ALL class authorization. Rather then modifying the existing string manipulation function, this patch adds a function that will only return a string if the field being compared explicitly matches class authorization field it is being compared against. This prevents ALL from being returned unless it is actually specified for the user. (closes issue ASTERISK-20397) Reported by: Johan Wilfer ........ Merged revisions 381939 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 381943 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381949 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-24Make ParkAndAnnounce return to priority + 1 when return context is not definedMatthew Jordan
The ParkAndAnnounce application documentation for the optional return_context parameter states the following: return_context The goto-style label to jump the call back into after timeout. Default 'priority+1'. Unfortunately, the application was sending the channel back into the dialplan at 'priority', which is the ParkAndAnnounce application call. This causes an infinite loop of the channel constantly being parked, announced, timed out, parked, announced, timed out... while fun, especially for those callers you wish to drive to the end of madness, this was not the intent of the application. (closes issue ASTERISK-20113) Reported by: serginuez patches: app_parkandannounce.diff uploaded by serginuez (License 6405) ........ Merged revisions 381916 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 381917 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381918 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-22Fix FastAGI To Properly Check For A ConnectionMichael L. Young
When IPv6 support was added to FastAGI, the intent was to have the ability to check all addresses resolved for a host since we might receive an IPv4 address and an IPv6 address. The problem with the current code, is that, since we are doing O_NONBLOCK, we get EINPROGRESS when calling ast_connect() but are ignoring this instead of handling it. We break out of the loop and continue on. When we later call ast_poll(), it succeeds but we never check if we have a connection or not on the socket level. We then attempt to send data to the host address that we think is setup and it fails. We then check the errno and see that we have "connection refused" and then return with agi failed. This patch does the following: * Handles EINPROGRESS by creating the function handle_connection() - ast_poll() was moved into this function - This function checks the results of the connection on the socket level after calling ast_poll() * Continues to the next address if the above fails to create a connection * Once all addresses resolved are tried and we still are unable to establish a connection, then we return that the FastAGI call failed (closes issue ASTERISK-21065) Reported by: Jeremy Kister Tested by: Jeremy Kister, Michael L. Young Patches: asterisk-21065_poll_correctly_v4.diff Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2330/ ........ Merged revisions 381893 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381894 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-22app_dial: Honor the 'c' flag when the calling party hangs upJonathan Rose
Apparently this feature became broken in 11, probably as a result of the Hangup Cause project. (closes issue ASTERISK-21113) Reprted by: Heiko Wundram Patches: app_dial.patch uploaded by Heiko Wundram (license 5822) ........ Merged revisions 381880 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381881 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-22Properly detect launchdMatthew Jordan
Asterisk was a little too pro-active in claiming that it found launchd. On systems without launchd - such as FreeBSD - this resulted in certain items in Asterisk that conflict with launchd to not be selectable, such as res_timing_kqueue. (closes issue ASTERISK-20749) Reported by: Oleg Baranov ........ Merged revisions 381847 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 381848 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381869 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-19Write the correct callid to the data1 field in queue_log for transfer events.Kevin Harwell
The incorrect callid was being written to the "data1" field in queue_log table for transfer events. The callid of the queue was being written instead of the transfer target's callid. This now gets the correct "transfer to" number and places that in the "data1" field of the queue_log table when a transfer event is triggered. (closes issue ASTERISK-19960) Reported by: vladimir shmagin ........ Merged revisions 381770 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 381791 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381792 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-19Add The Status Of A Module To The Output Of "CLI> module show"Michael L. Young
When a module's configuration is not loadable, we still load the module but it is not in a running state. When trying to troubleshoot, let's say, why chan_motif is ignoring inbound XMPP traffic, there is no way to indicate that a loaded module is not currently running. (closes issue ASTERISK-21108) Reported by: Rusty Newton Tested by: Michael L. Young Patches: asterisk-21108_add_status-v2.diff Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2331/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381749 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-19Confbridge channels staying active when all participants leave.Kevin Harwell
If you started/stopped recording of a conference multiple times channels would remain active even when all participants left the conference. This was due to the fact that a reference to the confbridge was being added every time a start record command was issued, but when the recording was stopped there was no matching de-reference thus keeping the conference alive. Made sure only a single reference is added for the record thread no matter how many times recording is started/stopped. A de-reference is issued upon thread ending. Note, this issue is being fixed under AST-1088 since it relates to it and should have been corrected along with those modifications. (issue AST-1088) Reported by: John Bigelow ........ Merged revisions 381737 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381741 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-19Added Confbridge record_file_append option.Kevin Harwell
Currently, if one starts, stops, and then starts a recording again for a conference the recorded data is appended to the file originally created on the first record start. An option record_file_append has been added that defaults to "yes", but when set to "no" will force creation of a new file between every record start/stop. (issue AST-1088) Reported by: John Bigelow Review: http://reviewboard.digium.internal/r/374/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381729 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-19Add serviceURL stuff to skinny.Damien Wedhorn
Patch adds all the packet and structure stuff to skinny to enable setting service URLs in skinny, such as corporate directories. This stuff is only relevant during load/unload as when activated. Also some minor changes removing duplicated counting of addons and speedials in handle_skinny_show_devices. Review: https://reviewboard.asterisk.org/r/2321/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381718 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-19Fixup skinny CLI completion.Damien Wedhorn
Auto complete for skinny debug allows multiple options and negation, also add debug all option. Usage example: 'skinny debug all -packets' (each can be autocompleted including -packet). Change show device to use device name. Remove the duplicate ast_strdup's from place calling device complete return immediately from complete devicename and complete linename so that multiple options are displayed on the CLI if more than one option available. Review: https://reviewboard.asterisk.org/r/2333/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381717 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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