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2014-11-09app_voicemail: Fix enhancement that allowed multiple recipients in To: headerMatthew Jordan
An issue existed in r420577, which added multiple recipients to voicemail emails. The patch, when looking at the intended recipients, looked ahead for the '|' character inside a while loop which already had pulled out the appropriate field parsing on the '|' character. This would cause it to skip the recipients. This patch fixes it such that it relies completely on the while loop to parse through the e-mail fields. Note that the original author of the patch looked at this fix and approved it. ASTERISK-24250 #close Reported by: abelbeck patches: voicemail-420577-to-comma-fix.diff uploaded by abelbeck (License 5903) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427585 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-11-09bridge_native_rtp: Fix T.38 issues with remote bridgesMatthew Jordan
After r425242 the fax/sip/directmedia_reinvite_t38 test started failing due to the surviving channel not being re-INVITEd back from T.38 to audio. This patch fixes that bug - a deeper explanation of what happened follows. When two RTP channels are in a native bridge, the bridging layer will investigate each via the get_rtp_info glue callback. This callback returns the native bridge preference of the channel *at that moment in time* (that part is key). At different points during the bridging, the native bridging layer will inform the RTP capable channels of the status of the bridge via the update_peer glue callback. In a T.38 scenario with audio direct media, the sequence of events will often look like the following: * SIP/A and SIP/B both have audio and enter a native bridge. * Asterisk re-INVITEs audio between SIP/A and SIP/B directly (via an update_peer callback). * SIP/A sends a re-INVITE to T.38, which causes Asterisk to send a re-INVITE to T.38 to SIP/B. Assuming everyone 200 OKs the process, the UDPTL stack receives UDPTL packets in Asterisk from both endpoints. From the perspective of the channels, we are now in a local bridge for T.38, even though we are technically still in a remote bridge in bridge_native_rtp. (YAY!) * When one side hangs up, bridge_native_rtp is told to stop bridging. It then re-evaluates the channels and asks them how they are bridged - and since T.38 is enabled, they reply with a Local bridge (which is correct), but is wrong because the audio portion is still technically in a remote bridge. * Asterisk releases the surviving channel, whose audio is *not* re-INVITED back to Asterisk as bridge_native_rtp incorrectly assumes that it was in a local bridge. Ironically, prior to r425242, this used to work mostly due to a fluke in the bridging layer. The purpose of the get_rtp_info callback shouldn't be modified: it should tell the bridging layer what kind of bridge the channel prefers at that moment in time. If you have T.38 enabled, that *must* be a local bridge, as the UDPTPL stack must be in the media path. As such, this patch does not modify that part of the code. However, we have to tell the channels to re-evaluate themselves when they come out of a native bridge, since we can no longer trust the get_rtp_info callbacks when the native bridge is being stopped. Something else may have changed in the channels, and they may now be lying to us. As such, this patch makes it so that we unilaterally tell the channels that they are no longer bridged via the update_peer callback. This is actually what the channels expect anyway: code in both chan_sip and chan_pjsip's callbacks look at the T.38 state and - if they were in T.38 - send a re-INVITE to get the audio back to Asterisk. Review: https://reviewboard.asterisk.org/r/4157/ ........ Merged revisions 427582 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427583 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-11-08chan_console: Fix reference leaks to pvt.Corey Farrell
Fix a bunch of calls to get_active_pvt where the reference is never released. ASTERISK-24504 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4152/ ........ Merged revisions 427554 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427555 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427557 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-11-06app_agent_pool: Made agent alert interruptable by DTMF.Richard Mudgett
Made agent able to interrupt the alerting beep playback with DTMF. Any digit can interrupt if the call does not need to be acknowledged. Only the first digit of the acknowledgement can interrupt if the call needs to be acknowledged. The agent interrupting the alerting playback builds on the ASTERISK-24447 patch because it knows what digit interrupted the playback and needs to be able to pass that digit to the DTMF hook digit collection code. ASTERISK-24257 #close Reported by: Steve Pitts Review: https://reviewboard.asterisk.org/r/4123/ ........ Merged revisions 427508 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427512 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-11-06Bridge DTMF hooks: Made audio pass from the bridge while waiting for more ↵Richard Mudgett
matching digits. * Made collecting DTMF digits for the DTMF feature hooks pass frames from the bridge. * Made collecting DTMF digits possible by other bridge hooks if there is a need. ASTERISK-24447 #close Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/4123/ ........ Merged revisions 427493 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427494 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-11-06res_pjsip: Ensure in-dialog responses have an endpoint associated.Joshua Colp
When handling incoming messages we determine if it is associated with a dialog. If so we use that to determine what serializer and endpoint to use for the message. Previously this would pass the endpoint to the endpoint lookup module to actually place the endpoint completely on the message. For in-dialog responses, however, this did not occur as dialog processing took over and the endpoint lookup did not occur. This change just places the endpoint in the expected spot immediately instead of relying on the endpoint lookup module. In-dialog responses thus have the expected endpoint. AST-1459 #close Review: https://reviewboard.asterisk.org/r/4146/ ........ Merged revisions 427490 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427491 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-11-06main/file.c: fix possible extra ast_module_unref to format modules.Corey Farrell
fn_wrapper only adds a reference to the format's module if the file was able to be opened. If not this causes an unmatched ast_module_unref in filestream_destructor. Move ast_module_ref to get_stream. ASTERISK-24492 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4149/ ........ Merged revisions 427464 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427465 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427466 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-11-06res_hep: fix major leak that occurs when config is missing or enabled=no.Corey Farrell
Add missing unreference in hepv3_send_packet. ASTERISK-24491 #close Reported by: Zane Conkle Tested by: Zane Conkle Review: https://reviewboard.asterisk.org/r/4150/ ........ Merged revisions 427400 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427405 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-11-06Fix unintential memory retention in stringfields.Corey Farrell
* Fix missing / unreachable calls to __ast_string_field_release_active. * Reset pool->used to zero when the current pool->active reaches zero. ASTERISK-24307 #close Reported by: Etienne Lessard Tested by: ibercom, Etienne Lessard Review: https://reviewboard.asterisk.org/r/4114/ ........ Merged revisions 427380 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 427381 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427382 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-11-06test_strings: Remove string tests that exercise asserts.George Joseph
Since unit tests are run with DO_CRASH, those tests were causing the test to fail. Tested-by: George Joseph ........ Merged revisions 427354 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427355 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427356 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-11-05Make the disable_tcp_switch PJSIP system object enabled by default.Mark Michelson
Testing has shown repeatedly that PJSIP's default behavior of switching automatically to TCP for large messages can cause issues. The most common issues are that devices that we are communicating with do not handle the switch to TCP gracefully, thus causing situations such as broken calls or broken subscriptions. Now, in order to have this behavior happen, you must opt into it. The sample file has been updated to warn that enabling the TCP switch behavior may cause issues for you, so use at your own risk. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427334 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-11-05res_pjsip_multihomed: Add logging during startup to aid debugging if local ↵Joshua Colp
DNS is misbehaving. This change adds a bit of logging so if the local DNS is misbehaving it is easier to track down what is going on and where Asterisk may be hanging. ASTERISK-24438 #close Reported by: Melissa Shepherd Review: https://reviewboard.asterisk.org/r/4148/ ........ Merged revisions 427300 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427303 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-11-05config: Make text_file_save and 'dialplan save' escape semicolons in values.George Joseph
When a config file is read, an unescaped semicolon signals comments which are stripped from the value before it's stored. Escaped semicolons are then unescaped and become part of the value. Both of these behaviors are normal and expected. When the config is serialized either by 'dialplan save' or AMI/UpdateConfig however, the now unescaped semicolons are written as-is. If you actually reload the file just saved, the unescaped semicolons are now treated as start of comments. Since true comments are stripped on read, any semicolons in ast_variable.value must have been escaped originally. This patch re-escapes semicolons in ast_variable.values before they're written to file either by 'dialplan save' or config/ast_config_text_file_save which is called by AMI/UpdateConfig. I also fixed a few pre-existing formatting issues nearby in pbx_config.c Tested-by: George Joseph ASTERISK-20127 #close Review: https://reviewboard.asterisk.org/r/4132/ ........ Merged revisions 427275 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427276 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-11-04config: BUG: Restore ability for non-templ to be used as base objs in config.George Joseph
My recent refactor of config.c accidentally removed the capability for an object to inherit from a non-template object. This patch restores the capability to inherit from both template and non-template objects. Tested-by: George Joseph Reported-by: Scott Griepentrog ASTERISK-24487 #close Review: https://reviewboard.asterisk.org/r/4147/ ........ Merged revisions 427227 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427228 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-11-04func_talkdetect: Fix stasis message leak in audiohook callback.Corey Farrell
ASTERISK-24482 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4142/ ........ Merged revisions 427203 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427204 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-11-04res_http_websockets: Fix extra unref of moduleCorey Farrell
In websocket_add_protocol_internal is used to add the "echo" protocol, but ast_websocket_remove_protocol is used to remove it. This causes an extra call to ast_module_unref. ASTERISK-24480 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4140/ ........ Merged revisions 427200 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427201 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-11-04Fix crash caused by merge error on review 4138Corey Farrell
When merging from 12 to 13 there were conflicts, I mistakenly had the loop run ast_closestream(others[0]) when it should be ast_closestream(others[x]). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427181 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-11-03res_pjsip: Add disable_tcp_switch option.Richard Mudgett
When a packet exceeds the MTU, pjproject will switch from UDP to TCP. In some circumstances (on some networks), this can cause some issues with messages not getting sent to the correct destination - and can also cause connections to get dropped due to quirks in pjproject deciding to terminate TCP connections with no messages. While fixing the routing/messaging issues is important, having a configuration option in Asterisk that tells pjproject to not switch over to TCP would be useful. That way, if some glitch is discovered on some other network/site, we can at least disable the behavior until a fix is put into place. AFS-197 #close Review: https://reviewboard.asterisk.org/r/4137/ ........ Merged revisions 427129 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427130 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-11-03Fix compile error caused by review 4138Corey Farrell
There is no procedure called ast_closeframe, fix code to use ast_closestream. Reported By: Matt Jordan ........ Merged revisions 427087 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427088 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427089 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-11-02Fix ast_writestream leaksCorey Farrell
Fix cleanup in __ast_play_and_record where others[x] may be leaked. This was caught where prepend != NULL && outmsg != NULL, once realfile[x] == NULL any further others[x] would be leaked. A cleanup block was also added for prepend != NULL && outmsg == NULL. 11+: Fix leak of ast_writestream recording_fs in app_voicemail:leave_voicemail. ASTERISK-24476 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4138/ ........ Merged revisions 427023 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 427024 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427025 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427026 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-11-02func_jitterbuffer: fix frame leaks.Corey Farrell
Fix code paths where it is possible for frames to leak. Fix uninitialized variable in jb_get_fixed and jb_get_adaptive. ASTERISK-22409 #related Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4128/ ........ Merged revisions 427019 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427020 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427021 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-11-02res/res_stasis: Fix crash on module unload while performing operationMatthew Jordan
When the res_stasis module is unloaded, it will dispose of the apps_registry container. This is a problem if an ARI operation is in flight that attempts to use the registry, as the shutdown occurs in a separate thread. This patch adds some sanity checks to the various routines that access the registry which cause the operations to fail if the apps_registry does not exist. Crash caught by the Asterisk Test Suite. ........ Merged revisions 426995 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426996 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-31install init.d files on GNU/kFreeBSDTzafrir Cohen
Review: https://reviewboard.asterisk.org/r/4118/ ........ Merged revisions 426926 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 426927 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 426933 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426934 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-31pjsip: clarify tls cert and key file usageScott Griepentrog
A question arose as to whether a .pem file could be provided in place of the .crt and .key files in a PJSIP TLS configuration. I tested this and discovered that although a cert will be read from the pem file, a key will not, and thus the priv_key_file entry is still required. This update to the fine documentation clarifies the option usage. AST-1448 #close Review: https://reviewboard.asterisk.org/r/4129/ Reported by: John Bigelow ........ Merged revisions 426928 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426930 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-31pjsip: Handle outbound unregister correctlyScott Griepentrog
This updates the status of the outbound registration to reflect when it has been unregistered. Since the registration is unregistered but is not stopped, the registration schedule remains active as before. The patch also updates the documentation of both the AMI and CLI commands. ASTERISK-24411 #close Review: https://reviewboard.asterisk.org/r/4119/ Reported by: John Bigelow patches: unregister-patch1.txt uploaded by John Bigelow (License 5091) ........ Merged revisions 426923 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426924 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-31channels/sip/reqresp_parser: Fix unit tests for r426594Matthew Jordan
When r426594 was made, it did not take into account a unit test that verified that the function properly populated the unsupported buffer. The function would previously memset the buffer if it detected it had any contents; since this function can now be called iteratively on successive headers, the unit tests would now fail. This patch updates the unit tests to reset the buffer themselves between successive calls, and updates the documentation of the function to note that this is now required. ........ Merged revisions 426858 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 426860 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 426863 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426865 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-31REF_DEBUG: Install refcounter.py to $(ASTDATADIR)/scriptsCorey Farrell
This change ensures refcounter.py is installed to a place where it can be found by the Asterisk testsuite if REF_DEBUG is enabled. ASTERISK-24432 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4094/ ........ Merged revisions 426830 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 426831 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 426832 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426833 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-30app_queue: fix a couple leaks to struct call_queue in set_member_valueCorey Farrell
set_member_value has a couple leaks to references in the variable q found through testsuite tests/queues/set_penalty. Also remove the REF_DEBUG_ONLY_QUEUES compiler declaration, this is no longer possible with the updated REF_DEBUG code. ASTERISK-24466 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4125/ ........ Merged revisions 426805 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 426806 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426807 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-30audiohooks: Clean references to formatsCorey Farrell
Cleanup references to in_translate[x].format and out_translate[x].format in ast_audiohook_detach_list. ASTERISK-24465 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4124/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426803 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-30res_pjsip_exten_state: PJSIPShowSubscriptionsInbound causes crashKevin Harwell
Currently, it is possible for some subscriptions to get into a NULL state. When this occurs and the PJSIPShowSubscriptionsInbound ami action is issued and a device is subscribed for extension state then the associated subscription state object can't be located. The code then attempts to dereference a NULL object. Added a NULL check to avoid the problem. Reported by: John Bigelow ........ Merged revisions 426779 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426780 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-30res_pjsip: incorrect qualify statistics after disabling for contactKevin Harwell
When removing the qualify_frequency from an AoR or a contact the statistics shown when issuing "pjsip show aors" from the CLI are incorrect. This patch deletes the contact's status object from sorcery, disassociating it from the contact, if the qualify_freqency is removed from configuration. ASTERISK-24462 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4116/ ........ Merged revisions 426755 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426757 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-30app_voicemail: Fix unchecked bounds of myArray in IMAP_STORAGE.Walter Doekes
In update_messages_by_imapuser(), messages were appended to a finite array which resulted in a crash when an IMAP mailbox contained more than 256 entries. This memory is now dynamically increased as needed. Observe that this patch adds a bunch of XXX's to questionable code. See the review (url below) for more information. ASTERISK-24190 #close Reported by: Nick Adams Tested by: Nick Adams Review: https://reviewboard.asterisk.org/r/4126/ ........ Merged revisions 426691 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 426692 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 426696 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426702 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-30Add additional checks for NULL pointers to fix several crashes reported.Igor Goncharovskiy
ASTERISK-24304 #close Reported by: dhanapathy sathya ........ Merged revisions 426666 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 426667 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426668 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-30channels/chan_sip: Add improved support for 4xx error codesMatthew Jordan
This patch adds support for 414, 493, 479, and a stray 400 response in REGISTER response handling. This helps interoperability in a number of scenarios. Review: https://reviewboard.asterisk.org/r/3437 patches: rb3437.patch uploaded by oej (License 5267) ........ Merged revisions 426599 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 426600 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 426601 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426602 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-30channels/chan_sip: Support mutltiple Supported and Required headersMatthew Jordan
A SIP request may contain multiple Supported: and Required: headers. Currently, chan_sip only parses the first Supported/Required header it finds. This patch adds support for multiple Supported/Required headers for INVITE requests. Review: https://reviewboard.asterisk.org/r/2478 ASTERISK-21721 #close Reported by: Olle Johansson patches: rb2478.patch uploaded by oej (License 5267) ........ Merged revisions 426594 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 426595 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 426596 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426597 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-29Fix building chan_phone on big endian systemsTzafrir Cohen
A left over from the formats conversion (Corey Farrell). ASTERISK-24458 #close Review: https://reviewboard.asterisk.org/r/4117/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426570 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-28bridge_builtin_features: Add missing channel locks around ↵Richard Mudgett
ast_get_chan_features_general_config(). The feature_automonitor() and feature_automixmonitor() functions were not locking the channel around ast_get_chan_features_general_config(). Accessing the channel datastore list without the channel locked is a good way to corrupt the list or follow the pointer chain into oblivion. ........ Merged revisions 426531 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426552 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-28res_fax: Resolve T38 gateway frame leak.Corey Farrell
When frames are translated by a fax gateway they need to be freed. The existing call to ast_frfree was unreachable. This change reorganizes fax_gateway_framehook to ensure that ast_frfree is called when needed. ASTERISK-24457 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4115/ ........ Merged revisions 426527 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 426528 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426529 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-28manager: Unsubscribe from acl_change_sub at shutdown.Corey Farrell
ASTERISK-24453 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4110/ ........ Merged revisions 426524 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426525 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-28ASTERISK-23512, correct inaccurate comment in manager.conf.sampleMalcolm Davenport
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426459 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-28main/bridge: Destroy features struct on off nominal path during bridge impartMatthew Jordan
When a channel is imparted to a bridge, the invocation of the function may provide an ast_bridge_features struct. Upon passing this to ast_bridge_impart, the caller must assume that ownership has passed to the function, as in all paths the function destroys the struct prior to returning (as its purpose is to configure the behavior of the channel while in the bridge). On one off nominal path - where the channel already has a PBX thread - the struct was not being destroyed. This patch fixes that glitch. ASTERISK-24437 #close Reported by: Scott Griepentrog ........ Merged revisions 426431 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426432 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-28main/manager: Fix typo in AMI event documentation of "OriginateResponse"Matthew Jordan
The parameter name is "Response", not "Resonse". ASTERISK-24430 #close Reported by: Dafi Ni ........ Merged revisions 426366 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 426367 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426368 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-28ASTERISK-24323, fix bug in documentation of AGI STREAM FILE CONTROLMalcolm Davenport
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426362 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-28ASTERISK-24419, fix incorrect syntax for setting language in ↵Malcolm Davenport
extensions.conf.sample git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426294 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-28app_queue: Cleanup ao2_iteratorCorey Farrell
Clean ao2_iterator, resolving reference leak to queue members. ASTERISK-24454 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4111/ ........ Merged revisions 426255 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 426260 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426266 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-28func_cdr: Fix CDR_PROP payload leakCorey Farrell
Remove duplicate allocation of payload, preventing leak. ASTERISK-24455 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4113/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426252 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-27configure: Add autoconf check for libopus.Sean Bright
Because opus transcoding support cannot be included in the standard Asterisk distribution, a few codec_opus implementations have popped up. To make it easier for people to drop in opus support in their own installations, this patch adds configure checks for libopus. Review: https://reviewboard.asterisk.org/r/4106/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426234 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-27res/res_http_websocket: Fix minor nits found by wdoekes on r409681Matthew Jordan
When Moises committed the fixes for WSS (which was a great patch), wdoekes had a few style nits that were on the review that got missed. This patch resolves what I *think* were all of the ones that were still on the review. Thanks to both moy for the patch, and wdoekes for the reviews. Review: https://reviewboard.asterisk.org/r/3248/ ........ Merged revisions 426209 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 426210 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426211 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-27res/res_phoneprov: Fix crash on shutdown caused by container cleanupMatthew Jordan
In res_phoneprov, unloading the module first destroys the http_routes container, followed by the users. However, users may have a route in the http_routes container; the validity of this container is not checked in the users destructor. Hence, we hit an assert as the container has already been set to NULL. This patch does two things: (1) It adds a sanity check in the user destructor (because why not) (2) It switches the order of destruction, so that users are disposed of prior to the HTTP routes they may hold a reference to. Note that this crash was caught by the Test Suite (go go testing!) ........ Merged revisions 426174 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426176 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-27res/res_srtp: Fix include issue for libsrtp 1.5.0Matthew Jordan
In libsrtp 1.5.0, crypto_get_random is no longer resolved simply by including srtp.h. Now, one must include crypto_kernel.h as well. As it turns out, this header file has been provided by the library since 2006, so this is a relatively benign change. ASTERISK-24436 #close Reported by: Patrick Laimbock ........ Merged revisions 426140 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 426141 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 426142 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426143 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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