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2014-09-19chan_iax2: Fix a crash when using chan_iax2 jitterbuffer settingsJonathan Rose
Caused by format changes in Asterisk 13 ASTERISK-24265 #close Reported by: Dafi Ni Review: https://reviewboard.asterisk.org/r/3999/ ........ Merged revisions 423524 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423526 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-19PJSIP: Prevent T38 framehook being put on wrong channelKinsey Moore
This change gives framehooks a reverse-direction masquerade callback in addition to chan_fixup_cb similar to the callback added to datastores to handle the same situation. The new callback provides the same parameters as the fixup callback, but is called on the new channel's framehooks before moving framehooks from the old channel to the new channel. This gives the framehooks an oppurtunity to decide whether they should remain on the new channel or be removed. This new callback is used to prevent the PJSIP T.38 framehook from remaining on a masqueraded channel if the new channel is not also a PJSIP channel. This was causing a crash when a local channel was masqueraded into a PJSIP channel and the framehook was executed on the local channel since the channel's tech private data was not structured as expected. Review: https://reviewboard.asterisk.org/r/4001/ ........ Merged revisions 423503 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 423504 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423505 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-18res_pjsip: Don't require a password when doing userpass authentication.Sean Bright
An empty password is valid for username/password authentication so we should allow password to be empty/not supplied. Review: https://reviewboard.asterisk.org/r/3988 ........ Merged revisions 423481 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 423482 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423483 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-18utils: Create ast_strsep function that ignores separators inside quotesGeorge Joseph
This function acts like strsep with three exceptions... * The separator is a single character instead of a string. * Separators inside quotes are treated literally instead of like separators. * You can elect to have leading and trailing whitespace and quotes stripped from the result and have '\' sequences unescaped. Like strsep, ast_strsep maintains no internal state and you can call it recursively using different separators on the same storage. Also like strsep, for consistent results, consecutive separators are not collapsed so you may get an empty string as a valid result. Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3989/ ........ Merged revisions 423476 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 423478 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423480 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-18Add subscription state test events.Mark Michelson
These are needed for a set of batched notification RLS tests that are about to be committed to the testsuite. Review: https://reviewboard.asterisk.org/r/3967 ........ Merged revisions 423462 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423463 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-18res_pjsip_endpoint_identifier_ip: Fix parsing of match value with CIDRJonathan Rose
Also fixes comma separates match lists ASTERISK-24290 #close Reported by: Ray Crumrine Review: https://reviewboard.asterisk.org/r/3995/ ........ Merged revisions 423417 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 423425 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423442 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-18bridge_softmix.c: Made use ao2_replace() instead of the inline equivalent.Richard Mudgett
* Clarified some read/write format comments. * Fixed a doxygen tag typo. ........ Merged revisions 423423 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423424 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-18astobj2.c/refcounter.py: Fix to deal with invalid object refs.Richard Mudgett
* Make astob2 REF_DEBUG output an invalid object line when an invalid ao2 object ref/unref is attempted. This is similar to the constructor/destructor lines. * Fixed refcounter.py to handle skewed objects that have constructor/destructor states. * Made refcounter.py highlight the invalid ao2 object refs by putting them in their own section of the processed output file. * Made refcounter.py highlight unreffing an object by more than one that results in a negative ref count and the object being destroyed. The abnormally destroyed object is reported in the invalid and finalized object sections of the output. Review: https://reviewboard.asterisk.org/r/3971/ ........ Merged revisions 423349 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 423400 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 423416 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 423418 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423422 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-18Add API call to determine if format capability structure is "empty".Mark Michelson
Empty here means that there are no formats in the format_cap structure or the only format in it is the "none" format. I've added calls to check the emptiness of a format_cap in a few places in order to short-circuit operations that would otherwise be pointless as well as to prevent some assertions from being triggered in cases where channels with no formats are used. ........ Merged revisions 423414 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423415 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-18res_fax_spandsp: Properly handle cleanup before starting FAXes.Mark Michelson
If faxing fails at a very early stage, then it is possible for us to pass a NULL t30 state pointer to spandsp, which spandsp is none too pleased with. This patch ensures that we pass the correct pointer to spandsp in the situation where we have not yet set our local t30 state pointer. ASTERISK-24301 #close Reported by Matt Jordan Patches: ASTERISK-24301-fax.diff Uploaded by Mark Michelson (License #5049) ........ Merged revisions 423360 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 423365 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 423372 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423380 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-18res_pjsip_pubsub: Add some type safety when generating NOTIFY bodies.Mark Michelson
res_pjsip_pubsub has two separate checks that it makes when a SUBSCRIBE arrives. * It checks that there is a subscription handler for the Event * It checks that there are body generators for the types in the Accept header The problem is, there's nothing that ensures that these two things will actually mesh with each other. For instance, Asterisk will accept a subscription to MWI that accepts pidf+xml bodies. That doesn't make sense. With this commit, we add some type information to the mix. Subscription handlers state they generate data of type X, and body generators state that they consume data of type X. This way, Asterisk doesn't end up in some hilariously mismatched situation like the one in the previous paragraph. ASTERISK-24136 #close Reported by Mark Michelson Review: https://reviewboard.asterisk.org/r/3877 Review: https://reviewboard.asterisk.org/r/3878 ........ Merged revisions 423344 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 423348 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423350 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-18res_pjsip: ami: Fix error in AMI output when an endpoint has no transportGeorge Joseph
When no transport is associated to an endpoint, the AMI output for PJSIPShowEndpoint indicates an error instead of silently ignoring the missing transport. This patch causes the error to appear only if a transport was specified on the endpoint and the transport doesn't exist. It also fixes an issue with counting the objects that were actually found. ASTERISK-24161 #close ASTERISK-24331 #close Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3998/ ........ Merged revisions 423282 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 423284 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423285 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-18Only install dahdi_span_config_hook if DAHDI is enabledDavid M. Lee
This patch changes the install to only install the hook script if DAHDI is enabled. It also adds the script to the uninstall task, and moves the DAHDI_UDEV_HOOK_DIR variable so that it's not between the _MAKEOPTS variables and their comment. This allows installs which specify a --prefix to work normally, as long as they don't enable DAHDI. Review: https://reviewboard.asterisk.org/r/3972/ ........ Merged revisions 423281 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423283 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-18config: bug: Fix SEGV in ast_category_insert when matching category isn't foundGeorge Joseph
If you call ast_category_insert with a match category that doesn't exist, the list traverse runs out of 'next' categories and you get a SEGV. This patch adds check for the end-of-list condition and changes the signature to return an int for success/failure indication instead of a void. The only consumer of this function is manager and it was also changed to use the return value. Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3993/ ........ Merged revisions 423276 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 423277 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 423278 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 423279 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423280 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-17res_rtp_asterisk: Ensure that the thread terminating pj stuff is registered.Joshua Colp
........ Merged revisions 423253 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 423254 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 423255 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423256 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-16pbx/Makefile: Revert r423237Matthew Jordan
This patch was supposed to go into a team branch, but I was a bit fast on the gun before 'svn switch' had apparently moved the target branch over. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423238 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-16Add some pbx python stuffMatthew Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423237 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-16Multiple revisions 423209,423212Joshua Colp
........ r423209 | file | 2014-09-16 17:35:34 -0300 (Tue, 16 Sep 2014) | 8 lines res_rtp_asterisk: Fix building when pjproject is not used. ........ Merged revisions 423207 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 423208 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ r423212 | file | 2014-09-16 18:03:59 -0300 (Tue, 16 Sep 2014) | 10 lines res_rtp_asterisk: Fix 100% CPU usage due to timer heap thread spinning. Side note: I need a vacation. ........ Merged revisions 423210 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 423211 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 423209,423212 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423213 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-16Voicemail: get correct duration when copying file to vmScott Griepentrog
Changes made during format improvements resulted in the recording to voicemail option 'm' of the MixMonitor app writing a zero length duration in the msgXXXX.txt file. This change introduces a new function ast_ratestream(), which provides the sample rate of the format associated with the stream, and updates the app_voicemail function for ast_app_copy_recording_to_vm to calculate the right duration. Review: https://reviewboard.asterisk.org/r/3996/ ASTERISK-24328 #close ........ Merged revisions 423192 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423193 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-16res_pjsip_session: Fix usage of wrong memory pool when creating local SDP.Joshua Colp
........ Merged revisions 423172 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 423173 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423174 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-16res_rtp_asterisk: Fix a myriad of TURN client issues.Joshua Colp
1. The number of file descriptors an ioqueue instance can handle is fixed, so we now spawn the required number to handle the load. 2. Our transport identifiers were exceeding the range supported by pjnath. 3. The TURN client did not set up client binding causing needless bandwidth usage. 4. The code no longer updates address information on each packet. 5. STUN traffic was getting looped back to Asterisk instead of going through the TURN server. 6. Synchronization now ensures things are completely setup or destroyed. 7. Logging now reflects the target the TURN server is sending to/receiving from on our behalf. ASTERISK-23577 #close Reported by: Jay Jideliov ASTERISK-23634 #close Reported by: Roman Skvirsky Review: https://reviewboard.asterisk.org/r/3982/ ........ Merged revisions 423150 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 423151 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 423152 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423153 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-15contrib: Fix verifyi typo in alembic DB script ps_transport table.Walter Doekes
Reported by: Zogot (on IRC) Patches: tmp.diff uploaded by Zogot, cleaned up by me. ........ Merged revisions 423128 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 423129 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423130 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-14chan_sip: Clarify that sipdebug=yes cannot be undone by the CLI.Walter Doekes
Document it in sip.conf. ASTERISK-24249 #close Reported by: Avinash Mohod Review: https://reviewboard.asterisk.org/r/3926/ ........ Merged revisions 423066 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 423067 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 423068 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 423069 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423070 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-14musiconhold: Add sort=randstart, and deprecate old stuff.Walter Doekes
- adds sort=randstart (next to sort=, sort=random, sort=alpha) - combines duplicate moh option parsing code into a single function - adds deprecationwarnings for application=r to sort randomly - adds deprecationwarnings for random=yes to sort randomly - removes invisible code that was supposed to stay until 1.8 The sort=randstart works like sort=alpha, except we start at a random position. Review: https://reviewboard.asterisk.org/r/3991/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423065 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-12chan_rtp: Add unicast RTP support.Joshua Colp
This module supports sending both unicast and multicast RTP to a specified target. Multicast functionality is the same as chan_multicast_rtp was. In the case of unicast a specific IP address and port can be specified, along with optional RTP engine and format in the form of: UnicastRTP/<ip address>:<port>/<engine>/<format> This can be useful for sending a copy of a media stream to another application for processing. Review: https://reviewboard.asterisk.org/r/3981/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423004 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-12Realtime: Fix a bug that caused realtime destroy command to crashJonathan Rose
Also has could affect with anything that goes through ast_destroy_realtime. If a CLI user used the command 'realtime destroy <family>' with only a single column/value pair, Asterisk would crash when trying to create a variable list from a NULL value. ASTERISK-24231 #close Reported by: Niklas Larsson Review: https://reviewboard.asterisk.org/r/3985/ ........ Merged revisions 422984 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 422985 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422991 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-11Remove undocumented default behavior of ast_play_and_record_full acceptdtmf.Mark Michelson
ast_play_and_record_full() has a parameter called "acceptdtmf" that is a string of acceptable DTMF digits that may be pressed by a caller to end and accept the recording. ARI uses this function in order to perform recording, and it provides options for what is passed as acceptdtmf to ast_play_and_record_full(). By default, ARI passes an empty string, with the intention that no DTMF can be used to end the recording. The problem is that ast_play_and_record_full() attempts to be "helpful" by setting "#" as the acceptdtmf if an empty string or NULL pointer has been passed in. With ARI, this results in unexpected behavior occurring if you have attempted to intercept "#" yourself in order to perform some other manipulation of the live recording. This change removes the "helpful" behavior by no longer accepting "#" as a default acceptdtmf if none is specified by the caller of ast_play_and_record_full(). This makes the ARI scenario work as expected. The other callers of ast_play_and_record_full() are app_voicemail and app_minivm, and in both cases, they pass an explicit "#" to ast_play_and_record_full() as acceptdtmf, so they are unaffected by this change. ........ Merged revisions 422964 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 422965 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422967 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-10config: bug: fix truncation of included config files on permissions errorGeorge Joseph
ast_config_text_file_save() currently truncates include files as they are processed. If a subsequent include file or the main config file has a permissions error that prevents writing, earlier include files are left truncated resulting in a frantic search for backups. This patch causes ast_config_text_file_save to check for write access on all files before it truncates any of them. Will be applied 1.8 > trunk. Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3986/ ........ Merged revisions 422900 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 422903 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 422904 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 422905 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422906 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-10pjsip/config_auth.c: Add missing whitespace to log messages.Sean Bright
The errors generated when validating 'auth' settings are missing a space which makes the messages a little confusing. ........ Merged revisions 422899 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 422901 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422902 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-09Update CHANGES for CHANNEL(onhold).Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422885 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-09Sounds/BuildSystem: Modifications to include new releases and Japanese language.Rusty Newton
Modifying Makefile and sounds.xml to include new core 1.4.26 and extra 1.4.15 sound prompt releases, plus the new Japanese core sound prompts contributed by QLOOG. ASTERISK-23324 Reported by: Kevin McCoy Tested by: Rusty Newton ........ Merged revisions 422789 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 422790 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 422791 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 422883 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422884 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-09func_channel: Add CHANNEL(onhold) item to get the current hold status of the ↵Richard Mudgett
channel. It would be useful to get the current hold status of a channel. Added CHANNEL(onhold) item that returns 1 (onhold) and 0 (not-onhold) for the hold status of a channel. ASTERISK-24038 Reported by: Matt Jordan AFS-113 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/3983/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422870 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-08Add note about configuring list_items on a single line.Mark Michelson
........ Merged revisions 422855 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422856 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-08Add sample configuration for resource lists.Mark Michelson
On review /r/3977, it was recommended to note in the sample configuration about the size limitation for resource lists. However, since there was no section in the sample configuration at all for resource list subscriptions, I decided to make a separate commit where I have added the necessary sample configuration as well as the size limitation warning. ........ Merged revisions 422853 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422854 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-08Pre-allocate transmission data buffer for RLS NOTIFY requests.Mark Michelson
PJSIP, unless a constant is modified at compilation time, limits SIP requests to 4000 bytes. Full-state RLS notifications can easily exceed this limit with moderately small lists. This changeset allows for Asterisk to work around this size limit by performing its own allocation of the transmission data buffer. This way, Asterisk can allocate a buffer that exceeds the built-in maximum. We still impose our own limit of 64000 bytes, mainly because making allocations larger than that is a bit absurd. ASTERISK-24181 #close Reported by Mark Michelson Review: https://reviewboard.asterisk.org/r/3977 ........ Merged revisions 422851 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422852 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-08res_pjsip_pubsub: Check supported headers for eventlist when subscribing toJonathan Rose
resource list https://wiki.asterisk.org/wiki/display/AST/Resource+List+Subscription+Test+Plan According to the off-nominal plan, if evenlist support is not specified in a SUBSCRIBE's supported header(s), that subscription should be rejected with an error. ASTERISK-23871 Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/3960/diff/#index_header ........ Merged revisions 422836 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422837 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-06main/cdr: Copy over location information during a forkMatthew Jordan
When a CDR is forked, a new CDR is created and appended to the CDR chain for the Party A. The forked CDR starts life off as a clone of the last non-finalized for the particular Party A. In the past, merely copying over the snapshots for Party A/Party B would be sufficient. However, as the CDRs now contain cached information from Party A - specifically application/data, context, and extension - we need to copy that over during a fork as well. Huzzah for unit tests catching this when the context/extension were derived from a cached value on the CDR instead of on Party A. ........ Merged revisions 422769 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 422770 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422771 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-06main/rtp_engine: Format NTP timestamps as unsigned intsMatthew Jordan
On some systems, a timeval's tv_sec/tv_usec will be unsigned lont ints, as opposed to long ints. When the RTP engine formats these as strings, it was previously formatting them as signed integers, which can result in some odd negative timestamp values (particularly on 32-bit systems). This patch formats the values as unsigned long integers. ........ Merged revisions 422766 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 422767 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422768 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-06res_pjsip_sdp_rtp: Fix retrieval of "ice-pwd" attribute if in session and ↵Joshua Colp
not media stream. ........ Merged revisions 422746 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 422747 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422748 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-05main/cdrs: Preserve context/extension when executing a Macro or GoSubMatthew Jordan
The context/extension in a CDR is generally considered the destination of a call. When looking at a 2-party call CDR, users will typically be presented with the following: context exten channel dest_channel app data default 1000 SIP/8675309 SIP/1000 Dial SIP/1000,,20 However, if the Dial actually takes place in a Macro, the current behaviour in 12 will result in the following CDR: context exten channel dest_channel app data macro-dial s SIP/8675309 SIP/1000 Dial SIP/1000,,20 The same is true of a GoSub: context exten channel dest_channel app data subs dial_stuff SIP/8675309 SIP/1000 Dial SIP/1000,,20 This generally makes the context/exten fields less than useful. It isn't hard to preserve these values in the CDR state machine; however, we need to have something that informs us when a channel is executing a subroutine. Prior to this patch, there isn't anything that does this. This patch solves this problem by adding a new channel flag, AST_FLAG_SUBROUTINE_EXEC. This flag is set on a channel when it executes a Macro or a GoSub. The CDR engine looks for this value when updating a Party A snapshot; if the flag is present, we don't override the context/exten on the main CDR object. In a funny quirk, executing a hangup handler must *not* abide by this logic, as the endbeforehexten logic assumes that the user wants to see data that occurs in hangup logic, which includes those subroutines. Since those execute outside of a typical Dial operation (and will typically have their own dedicated CDR anyway), this is unlikely to cause any heartburn. Review: https://reviewboard.asterisk.org/r/3962/ ASTERISK-24254 #close Reported by: tm1000, Tony Lewis Tested by: Tony Lewis ........ Merged revisions 422718 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 422719 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422720 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-05main/cdr: Fix crash/memory consumption in CDRs in multi-party bridge scenariosMatthew Jordan
This patch fixes an issue where CDRs would get stuck generating an infinite number of CDRs, eventually crashing Asterisk (and consuming a lot of memory along the way). When a channel enters into a multi-party bridge, the CDR engine creates mappings of each participant to each other participant, picking the 'A' party as it goes. So, if we have four channels in a multi-party bridge (Alice, Bob, Charlie, Denise), we would have something like: Alice => Bob Alice => Charlie Alice => Denise Bob => Charlie Bob => Denise Charlie => Denise This works fine when participants enter the bridge a single time. When a participant leaves a bridge, the CDRs for that channel are transitioned to a finalized state. The bug occurs if Bob rejoins. When the CDR engine creates mappings between the channels, it walks through all the participants currently in the bridge, and realizes that no one in the bridge can create a CDR with the channel (Bob). As such it creates a new CDR for the candidate and appends it to that candidate's chain. Unfortunately, on this particular code path, it doesn't stop traversing the candidate's chain. Since we just added ourselves to the chain, this causes the loop to keep going, constantly adding new CDRs. This patch makes it so the engine bails when it creates a CDR match in this case. Review: https://reviewboard.asterisk.org/r/3964/ ASTERISK-24241 #close Reported by: Deepak Singh Rawat Tested by: Deepak Singh Rawat ASTERISK-24208 Reported by: Frankie Chin ........ Merged revisions 422715 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 422716 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422717 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-05func_channel.c: Add missing locking to some CHANNEL() requests.Richard Mudgett
* The CHANNEL() audionativeformat, videonativeformat, audioreadformat, and audiowriteformat now need locking since the media format rework when accessing the channel's format pointers. * Increased the buffer size for CHANNEL() audionativeformat and videonativeformat output strings since the allow=all can be a lengthy list. * Tweaked the CHANNEL() XML documentation for secure_bridge_signaling, secure_bridge_media, and state. * Ensured the output buffer is initialized for secure_bridge_signaling and secure_bridge_media. * Made use the locked_copy_string() macro instead of inlining it for trace and checkhangup. ........ Merged revisions 422700 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422701 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-05Dial API: Add a dial option to indicate the dialed channel will replace dialerJonathan Rose
Adds an option to the dial API that marks an outgoing dial as replacing the dialing channel for the purpose of propagating accountcode. When it is used, AST_CHANNEL_REQUESTOR_REPLACEMENT is used instead of AST_CHANNEL_REQUESTOR_BRIDGE_PEER when setting accountcodes on the involved channels with ast_channel_req_accountcodes. Review: https://reviewboard.asterisk.org/r/3968/ ........ Merged revisions 422684 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422697 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-05Call IDs: Fix appearance of call ID in core show channels when NULLJonathan Rose
NULL call IDs were meant to appear as '(none)' but instead were showing the contents of an uninitialized character buffer. ASTERISK-24223 Review: https://reviewboard.asterisk.org/r/3979/ ........ Merged revisions 422664 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 422665 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422683 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-05devicestate.c: Minor tweaksRichard Mudgett
* In ast_state_chan2dev() use ARRAY_LEN() instead of a sentinel value in chan2dev[]. * Fix some comments in chan_iax2.c. ........ Merged revisions 422661 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422663 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-05Menuselect: Fix incorrect enabling on failed depsKinsey Moore
This corrects a situation where menuselect can incorrectly enable a module by default that has defaultenabled set to "no" and has failed/non-selected dependencies. The bug is due to an inverted test when checking for whether the given module should be set to enabled by default on load. Review: https://reviewboard.asterisk.org/r/3975/ Reported by: John Bigelow ........ Merged revisions 422646 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422647 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-04Manager: Require read permission for SYSTEM in order to send FullyBootedJonathan Rose
Review: https://reviewboard.asterisk.org/r/3969/ ........ Merged revisions 422584 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 422625 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 422626 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 422631 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422632 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-03res_pjsip_transport_websocket: Fix crash when the Contact header is not a URI.Joshua Colp
The code for changing the Contact header wrongly assumed that the Contact would always contain a URI. This is incorrect. ASTERISK-24271 Reported by: Dafi Ni ........ Merged revisions 422557 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 422558 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422559 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-02Resolve race condition where channels enter dialplan application before ↵Mark Michelson
media has been negotiated. Testsuite tests will occasionally fail because on reception of a 200 OK SIP response, an AST_CONTROL_ANSWER frame is queued prior to when media has finished being negotiated. This is because session supplements are called into before PJSIP's inv_session code has told us that media has been updated. Sometimes the queued answer frame is handled by the PBX thread before the ensuing media negotiations occur, causing a test failure. As it turns out, there is another place that session supplements could be called into, which is after media has finished getting negotiated. What this commit introduces is a means for session supplements to indicate when they wish to be called into when handling an incoming SIP response. By default, all session supplements will be run at the same point that they were prior to this commit. However, session supplements may indicate that they wish to be handled earlier than normal on redirects, or they may indicate they wish to be handled after media has been negotiated. In this changeset, two session supplements have been updated to indicate a preference for when they should be run: res_pjsip_diversion executes before handling redirection in order to get information from the Diversion header, and chan_pjsip now handles responses to INVITEs after media negotiation to fix the race condition mentioned previously. ASTERISK-24212 #close Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/3930 ........ Merged revisions 422536 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 422542 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422543 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-01main/cli: Do not attempt to show CDR data for internal channelsMatthew Jordan
Internal channels don't have CDRs. Querying the CDR engine for their variables will make it cranky. ........ Merged revisions 422506 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 422507 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422524 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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