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2013-10-23chan_sip: Fix an issue where an incompatible audio format may be added to SDP.Joshua Colp
If preferred codecs included any non-audio format the code would mistakenly add the audio format, even if it was not a joint capability with the remote side. (closes issue ASTERISK-21131) Reported by: nbougues Patches: patch_unsupported_codec_1.8.patch uploaded by nbougues (license 6470) ........ Merged revisions 401497 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 401498 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 401499 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401500 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-23chan_iax2: Fix Binding To Multiple Addresses AgainMichael L. Young
When reworking chan_iax2 for IPv6, the ability to bind to multiple addresses was removed by mistake. This patch restores this functionality and adds notes about IPv6 addresses in the sample config. (closes issue ASTERISK-22741) Reported by: Joshua Colp Tested by: Michael L. Young Patches: asterisk-22741-fix-binding-multiple-addr.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2945/ ........ Merged revisions 401488 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401489 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-22res_rtp_asterisk: Fix crash when RTCP is not available during SSRC changeMatthew Jordan
In r400089, a patch was put in to correct erroneous RTCP statistic resets. Unfortunately, ast_rtp_read can be called on an RTP instance that does not have RTCP information. This patch prevents that crash by only resetting the statistics if we do actually have an RTCP instance. (issue AST-1174) (closes issue ASTERISK-22667) Reported by: John Bigelow ........ Merged revisions 401445 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 401446 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 401447 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401450 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-22app_queue: Fix CLI "queue remove member" queue_log entry.Richard Mudgett
The queue_log entry resulting from CLI "queue remove member" when log_membername_as_agent is enabled is wrong. It always uses the interface name instead of the member name in the queue_log entry. * Get the queue member before removing it from the queue so the member name is available for the queue_log entry. (closes issue ASTERISK-21826) Reported by: Oscar Esteve Patches: fix_membername.diff (license #6505) patch uploaded by Oscar Esteve (modified to fix potential ref leak) ........ Merged revisions 401433 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 401434 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-22Bridging: Fix orphaned bridge if neither of the joining channels can join.Richard Mudgett
The original issue noted that the bridge is orphaned when res_parking.so is not loaded and a call uses the dial kK flags. A similar issue happens when only one of the park flags is used. In this case you have the bridge with one or the other channel left in it. The channel and bridge will stay around until the channel hangs up. * Fixed the initial bridge channel push failure to act as if the channel were kicked out of the bridge. The bridge then decides if it needs to be dissolved. (closes issue ASTERISK-22629) Reported by: Kevin Harwell Review: https://reviewboard.asterisk.org/r/2928/ ........ Merged revisions 401424 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401425 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-22res_parking: Give parking timeout comebacktoorigin channel DTMF features.Richard Mudgett
Parking timeouts did not set any DTMF features for the channel calling the parker back. * Added code to set the parkedcalltransfers, parkedcallreparking, parkedcallhangup, and parkedcallrecording options appropriately for the channels when a parking timeout occurs. The recall channel DTMF options are set using the BRIDGE_FEATURES channel variable to allow the other timeout options to have the DTMF features available. (closes issue ASTERISK-22630) Reported by: Kevin Harwell Review: https://reviewboard.asterisk.org/r/2942/ ........ Merged revisions 401422 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-22res_parking: Update XML documention for DTMF features after parking timeout.Richard Mudgett
* Updated the XML documentation to indicate that the parkedcalltransfers, parkedcallreparking, parkedcallhangup, and parkedcallrecording configuration options also apply to parking timeouts. (issue ASTERISK-22630) Reported by: Kevin Harwell Review: https://reviewboard.asterisk.org/r/2942/ ........ Merged revisions 401420 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401421 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-22Add an 'R' option to Dial which sends ringing until early media has been ↵Joshua Colp
received. (closes issue ASTERISK-10487) Reported by: Gaspar Zoltan Patches: 10487.patch uploaded by n8ideas (license 6075) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401411 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-21Remove a noisy debug message from bridging code.Mark Michelson
This particular debug message, during a stress test, was logged so often that it appeared that there may be a memory leak in the logger code. In actuality, there was no memory leak, but the logger thread was having a hard time keeping up with the demands of the rest of the system. Since this debug message has no value at all, the best way to fix the problem was to just remove the message. (closes issue AST-1225) reported by John Bigelow Patches: spammy_log.diff uploaded by Mark Michelson (License #5049) ........ Merged revisions 401364 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-21Segfault in LIBEDIT_INTERNAL after tgetstr(), when libncurses5-devKevin Harwell
isn't installed Include the appropriate declarations when not using termcap, but term+curses and [n]curses do not exist. (closes issue ASTERISK-22351) Reported by: A. Iglesias Patches: issueA22351_libedit_internal_without_ncurses_dev.patch uploaded by wdoekes (license 5674) ........ Merged revisions 401325 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 401326 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 401327 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401328 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-21Fixing r401281; the model name is Channel, with a capital CDavid M. Lee
........ Merged revisions 401315 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401317 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-21Fixed malformed Access-Control-Allow-Methods header. Was causing Safari to ↵David M. Lee
barf on POST and DELETE. ........ Merged revisions 401106 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401316 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-19Fix IAX2 incoming call address lookupsKinsey Moore
This fixes address lookup for incoming calls without a peer definition. The address family was unset instead of being set to AST_AF_UNSPEC which was causing lookup failures on "127.0.0.1". This is one of the causes of the current failure of the app_page integration test. Review: https://reviewboard.asterisk.org/r/2933/ ........ Merged revisions 401291 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401292 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-19Return a channel snapshot when originating using ARI, and subscribe the ↵Joshua Colp
Stasis application to it. This change allows a user of ARI to know what channel it has originated and also follow any progress. If a Stasis application is provided it will be automatically subscribed to the originated channel immediately. (closes issue ASTERISK-22485) Reported by: David Lee Review: https://reviewboard.asterisk.org/r/2910/ ........ Merged revisions 401281 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401282 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-18res_parking: Remove setting useless flag.Richard Mudgett
........ Merged revisions 401271 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401272 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-18This is just a quick script for dumping swagger-ui into static-http,David M. Lee
so that it can be served by the Asterisk web server. I had to change the Makefile in order to recursively install content from the static-http directory, hence the code review instead of just putting it in. Review: https://reviewboard.asterisk.org/r/2924/ ........ Merged revisions 401261 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401263 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-18Resolve some memory leaks due to incorrect for loop / ao2 ref usage.Mark Michelson
A common idiom in Asterisk is to due something like: for (ao2_obj = list_beginning; ao2_obj = next_item; ao2_ref(ao2_obj, -1)) { ...do stuff... } This is nice because it automatically takes care of the object references for you. However, there is a pitfall here. If a break statement is in the for loop, then the current reference is not cleaned up. In some cases, this is on purpose, but in others there is a leak. This commit fixes the leak cases. ........ Merged revisions 401248 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401249 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-18Add channel lock protection around translation path setup.Richard Mudgett
Most callers of ast_channel_make_compatible() happen before the channels enter a two party bridge. With the new bridging framework, two party bridging technologies may also call ast_channel_make_compatible() when there is more than one thread involved with the two channels. * Added channel lock protection in set_format() and ast_channel_make_compatible_helper() when dealing with the channel's native formats while setting up a translation path. * Fixed best_src_fmt and best_dst_fmt usage consistency in ast_channel_make_compatible_helper(). The call to ast_translator_best_choice() got them backwards. * Updated some callers of ast_channel_make_compatible() and the function documentation. There is actually a difference between the two channels passed in. * Fixed the deadlock potential in res_fax.c dealing with ast_channel_make_compatible(). The deadlock potential was already there anyway because res_fax called ast_channel_make_compatible() with chan locked. (closes issue ASTERISK-22542) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2915/ ........ Merged revisions 401239 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401240 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-18Tweak ast_bridge_depart() doxygen.Richard Mudgett
........ Merged revisions 401232 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401233 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-18Remove the bit about requiring ast_bridge_depart() to be called before ↵Mark Michelson
ast_bridge_destroy(). ........ Merged revisions 401223 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401224 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-18Clarify in ast_bridge_destroy() about how departable channels must be handled.Mark Michelson
........ Merged revisions 401212 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401216 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-18Remove Port Restriction When Checking For NATMichael L. Young
When trying to determine if a peer is behind NAT, we should not be using the ports when comparing addresses. This patch removes the port from being checked and just useds the addresses now. (closes issue ASTERISK-22729) Reported by: Michael L. Young Tested by: Michael L. Young Patches: asterisk-remove-using-port-for-nat-check.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2927/ ........ Merged revisions 401182 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 401183 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401184 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-18Properly copy/remove the device state cache flag over a masquerade.Walter Doekes
In r378303 the AST_FLAG_DISABLE_DEVSTATE_CACHE flag was added that tells the devstate system to not cache states for non-real devices. However, when optimizing away channels (ast_do_masquerade), that flag wasn't copied. In my case, using Local devices as queue members created a situation where the endpoint was considered in use, but the state change of the device being available again was ignored (not cached). The endpoint channel was optimized into the (previously) Local channel, but kept the do-not-cache flag. The end result being that the queue member apparently stayed in use forever. (closes issue ASTERISK-22718) Reported by: Walter Doekes Review: https://reviewboard.asterisk.org/r/2925/ ........ Merged revisions 401178 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 401179 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 401180 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401181 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-17Fix Setting A chan_sip Dialog's SIP_NAT_FORCE_RPORT FlagMichael L. Young
A condition was added in a commit to fix ASTERISK-21374, that, if the SIP_PAGE3_NAT_AUTO_RPORT flag was set, to then copy a peer's SIP_NAT_FORCE_RPORT flag to the dialog. This condition should not have been there since it assumed that if Asterisk is in an environment where NAT is involved, that the auto_* nat settings or force_rport setting would be on in the global settings. If the nat setting in the global setting is set to 'nat=no' and then turned on for peers (which is not quite the recommended way, although it is allowed) this flag is never copied to the dialog resulting in problems like, REGISTER replies going to the wrong port. This patch removes this conditional check and will now always use the peer's flag which by this point in the code the checks on whether the peer is behind NAT or not (if using auto_force_rport) have already been run. (closes issue ASTERISK-22236) Reported by: Filip Frank Tested by: Michael L. Young Patches: asterisk-2236-always-set-rport.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2919/ ........ Merged revisions 401167 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 401168 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401169 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-17res_parking: Fix bug where reloading immediately wipes new parkpos extensionsJonathan Rose
(closes issue ASTERISK-22631) Reported by: Kevin Harwell ........ Merged revisions 401158 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401159 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-17Reduce log level of a non-pubsub error messageKinsey Moore
Drop an error log message to debug level 1 since distributed device state functions correctly when receiving this message and it spams the logs. (closes issue ASTERISK-22410) Reported by: abelbeck Patches: asterisk-1.8-res_jabber-log-nonpubsub-error-to-debug.patch uploaded by abelbeck (License 5903) asterisk-11-res_xmpp-log-nonpubsub-error-to-debug.patch uploaded by abelbeck (License 5903) ........ Merged revisions 401119 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 401120 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 401121 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401122 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-16ARI: Fix crash when POST /playback/{id}/control does not have an operation ↵Richard Mudgett
parameter. (closes issue ASTERISK-22680) Reported by: John Bigelow ........ Merged revisions 401107 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401108 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-16Oops. Leftover /stasis referenceDavid M. Lee
........ Merged revisions 401096 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401097 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-16Clarify documentation for channel and bridge listKinsey Moore
This makes it clear that the ARI API calls for listing channels and bridges will list all channels or bridges in the system and not just those that are in or are controlled by a Stasis application. (closes issue ASTERISK-22635) Reported by: Kevin Harwell ........ Merged revisions 401087 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401088 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-16Don't check all realtime queues when doing "queue show some_queue".Walter Doekes
When using realtime queues, queues have to be fetched from the database every now and then to see if any info has been changed or to see if the queue has been removed. When fetching info for an individual queue, the pruning of other queues is unnecessarily costly. Review: https://reviewboard.asterisk.org/r/2907/ ........ Merged revisions 401049 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 401076 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 401077 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401079 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-16Use POST / DELETE to toggle ARI bridge mohPaul Belanger
Review: https://reviewboard.asterisk.org/r/2911/ ........ Merged revisions 401040 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401041 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-15translate.c: Some minor code tweaks.Richard Mudgett
* Consistently compare format2index() return value so matrix_get() cannot get passed negative values. * Optimize ast_translator_best_choice() to defer initializing things until needed. Also cached the matrix_get() return value rather than repeatedly calling it. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401039 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-15bridge_native_dahdi: Return channel join failure if could not make the ↵Richard Mudgett
channels compatible. ........ Merged revisions 401030 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401031 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-15chan_iax2: Fix channel left locked in off nominal code path.Richard Mudgett
........ Merged revisions 401016 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 401017 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401020 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-15Ensure bridge record error responses validateKinsey Moore
This adds the list of expected errors to the /bridges/{bridgeId}/record ARI documentation so that outbound 4xx errors validate properly. Previously, this would result in a response validation failure. (closes issue ASTERISK-22627) Reported by: Joshua Colp ........ Merged revisions 401018 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401019 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-15Use POST / DELETE to toggle hold / moh for ARI channelsPaul Belanger
This change updates how we handle toggle events, rather then create two different function names, we'll just use POST / DELETE from HTTP to handle it. Review: https://reviewboard.asterisk.org/r/2906/ ........ Merged revisions 400999 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401007 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-15Prevent chan_sip from sending duplicate BYEs.Mark Michelson
When a 200 OK for an initial INVITE is received, we were doing the right thing by ACKing and sending an immediate BYE. However, we also were doing the wrong thing and queuing an answer frame, thus causing the call to be answered. This would cause the call to be hung up by the channel thread, thus resulting in a second BYE being sent out. In this fix, I also have set the hangupcause to be correct since the initial BYE being sent by Asterisk had an unknown hangup cause. I have changed to using "Bearer capabilty not available" since the call was hung up due to an SDP offer/answer error. (closes issue ASTERISK-22621) reported by Kinsey Moore ........ Merged revisions 400970 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 400971 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 400984 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400998 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-15My doc correction in r400842 had a silly bug.David M. Lee
Because I added a wiki_description to models and not their properties, the rendered wiki page had the model description instead of the property descriptions, which looks very silly indeed. (closes issue ASTERISK-22705) ........ Merged revisions 400958 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400959 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-14chan_dahdi: Add config support for hwgain settings.Richard Mudgett
* Add hwtxgain and hwrxgain config options to chan_dahdi.conf with documentation in chan_dahdi.conf.sample. (closes issue ASTERISK-22429) Reported by: Jaco Kroon Patches: jira_asterisk_22429_hwgain_trunk.patch (license #5621) patch uploaded by rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400950 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-14chan_dahdi: Reflect the set software gain in the CLI "dahdi show channel" ↵Richard Mudgett
output. * Remember the swgain setting from CLI "dahdi set swgain" command so the CLI "dahdi show channel" output will reflect the current setting. * Updated CLI "dahdi set hwgain" and "dahdi set swgain" documentation. (issue ASTERISK-22429) Reported by: Jaco Kroon Patches: jira_asterisk_22429_v1.8_v2.patch (license #5621) patch uploaded by rmudgett ........ Merged revisions 400907 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 400909 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 400911 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400913 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-14chan_sip: Do not increment the SDP version between 183 and 200 responses.Mark Michelson
Bumping the SDP version number can cause interoperability problems since receivers of the responses will expect that a 200 SDP will be identical to a previous 183 SDP. (closes issue ASTERISK-21204) reported by NITESH BANSAL Patches: dont-increment-session-version-in-2xx-after-183.patch uploaded by NITESH BANSAL (License #6418) ........ Merged revisions 400906 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 400908 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 400910 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400912 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-14pjsip outbound registration: Log message says received a 408 when we didn'tKevin Harwell
If the server didn't exist that we are trying to register to the log message would say that a 408 was received from that server when in reality one wasn't. Added log messages stating no response was received if the response does not exist. (closes issue ASTERISK-22554) Reported by: Rusty Newton Review: https://reviewboard.asterisk.org/r/2893/ ........ Merged revisions 400890 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400891 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-14Remove duplicate module info blockMatthew Jordan
The module info block was repeated twice. Once is sufficient. ........ Merged revisions 400881 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400882 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-13Fix a race condition in res_pjsip_session with rapidly terminating the session.Joshua Colp
The INVITE session state callback wrongly assumes that a session will always exist, but when rapidly terminating the session this assumption goes out the window. As all handler code for the INVITE session state callback requires the session it will now just exit immediately if no session exists. (closes issue ASTERISK-22668) Reported by: John Bigelow ........ Merged revisions 400872 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-12Fix realm comparison for outbound authKinsey Moore
When generating the list of authentication credentials to pass to PJSIP, Asterisk was using the raw pointer of a pj_str_t which is not always NULL-terminated. This sometimes resulted in incorrect text for the realm and a failure to match the realm for authentication purposes which was causing the outbound nominal auth pjsip basic call test to bounce. This now uses the pj_str_t that contains the realm instead of generating a new one. Thanks to John Bigelow for helping to narrow this down. ........ Merged revisions 400863 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400864 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-11channel.h: whitespace changes.Richard Mudgett
........ Merged revisions 400854 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400855 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-11Multiple revisions 400508,400842-400843,400848David M. Lee
........ r400508 | dlee | 2013-10-03 23:54:51 -0500 (Thu, 03 Oct 2013) | 1 line Corrected response class for stopPlayback ........ r400842 | dlee | 2013-10-10 14:23:24 -0500 (Thu, 10 Oct 2013) | 1 line Correct some ARI wiki rendering errors ........ r400843 | dlee | 2013-10-10 14:26:19 -0500 (Thu, 10 Oct 2013) | 1 line Updated /play resource docs. The playback of http: resources isn't implemented... yet ........ r400848 | dlee | 2013-10-11 11:18:46 -0500 (Fri, 11 Oct 2013) | 5 lines Fix a stupid copy/paste error in ARI docs. Patches: ari-doc-patch.txt uploaded by jbigelow (license 5091) ........ Merged revisions 400508,400842-400843,400848 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400852 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-11Fixed merge tracking for r400360, which was somehow lostDavid M. Lee
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400851 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-11Softmix: Fix crash when switching from softmix to another bridge technology.Richard Mudgett
The crash is caused by a race condition when switching between native RTP and softmix bridging technologies. In this situation, the bridging technology is switched from native RTP to softmix, and then back to native RTP fast enough that the softmix private data gets destroyed before the softmix mixing thread gets started. Thanks to Kinsey Moore for the crash analysis. * Fix race condition when starting the softmix mixing thread and switching to another bridge technology. (closes issue ASTERISK-22678) Reported by: John Bigelow Patches: jira_asterisk_22678_v12.patch (license #5621) patch uploaded by rmudgett Tested by: John Bigelow ........ Merged revisions 400849 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400850 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-10Perform validation of permanent contacts on AORs in res_pjsip.Joshua Colp
........ Merged revisions 400833 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400834 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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