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2013-03-27AST-2013-002: Prevent denial of service in HTTP serverMatthew Jordan
AST-2012-014, fixed in January of this year, contained a fix for Asterisk's HTTP server for a remotely-triggered crash. While the fix put in place fixed the possibility for the crash to be triggered, a denial of service vector still exists with that solution if an attacker sends one or more HTTP POST requests with very large Content-Length values. This patch resolves this by capping the Content-Length at 1024 bytes. Any attempt to send an HTTP POST with Content-Length greater than this cap will not result in any memory allocation. The POST will be responded to with an HTTP 413 "Request Entity Too Large" response. This issue was reported by Christoph Hebeisen of TELUS Security Labs (closes issue ASTERISK-20967) Reported by: Christoph Hebeisen patches: AST-2013-002-1.8.diff uploaded by mmichelson (License 5049) AST-2013-002-10.diff uploaded by mmichelson (License 5049) AST-2013-002-11.diff uploaded by mmichelson (License 5049) ........ Merged revisions 383978 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383980 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-27AST-2013-001: Prevent buffer overflow through H.264 format negotiationMatthew Jordan
The format attribute resource for H.264 video performs an unsafe read against a media attribute when parsing the SDP. The value passed in with the format attribute is not checked for its length when parsed into a fixed length buffer. This patch resolves the vulnerability by only reading as many characters from the SDP value as will fit into the buffer. (closes issue ASTERISK-20901) Reported by: Ulf Harnhammar patches: h264_overflow_security_patch.diff uploaded by jrose (License 6182) ........ Merged revisions 383973 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383975 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-27Fix skinny encall button to not blind xfer.Damien Wedhorn
The softbutton endcall should not turn a transfer into a blind transfer but hangup the exten being called and leave the original call on hold. This does that. (closes issue ASTERISK-21321) Reported by: wedhorn Tested by: snuffy, myself Patches: skinny-xferendcall01.diff uploaded by wedhorn (license 5019) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383948 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-26Remove the noop handler from sorcery so it does not produce an empty value.Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383925 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-26Resolve deadlock between SIP registration and channel based functionsMatthew Jordan
In r373424, several reentrancy problems in chan_sip were addressed. As a result, the SIP channel driver is now properly locking the channel driver private information in certain operations that it wasn't previously. This exposed two latent problems either in register_verify or by functions called by register_verify. This includes: * Holding the private lock while calling sip_send_mwi_to_peer. This can create a new sip_pvt via sip_alloc, which will obtain the channel container lock. This is a locking inversion, as any channel related lock must be obtained prior to obtaining the SIP channel technology private lock. Note that this issue was already fixed in Asterisk 11. * Holding the private lock while calling sip_poke_peer. In the same vein as sip_send_mwi_to_peer, sip_poke_peer can create a new SIP private, causing the same locking inversion. Note that this locking inversion typically occured when CLI commands were run while a SIP REGISTER request was being processed, as many CLI commands (such as 'sip show channels', 'core show channels', etc.) have to obtain the channel container lock. (issue ASTERISK-21068) Reported by: Nicolas Bouliane (issue ASTERISK-20550) Reported by: David Brillert (issue ASTERISK-21314) Reported by: Badalian Vyacheslav (issue ASTERISK-21296) Reported by: Gabriel Birke ........ Merged revisions 383863 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 383878 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383879 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-26Resolve deadlock between pending CDR and batch CDR locksMatthew Jordan
r375757 attempted to resolve a race condition between multiple submissions of CDRs while in batch mode from attempting to destroy the scheduled batch submission by extending the batch CDR lock. Unfortunately, this causes a deadlock between the pending CDR lock and the batch CDR lock. This patch resolves the intent of r375757 by simply providing a new lock that protects the scheduling of the batches. The original batch CDR lock is kept to protect manipulation of the batch CDR settings, but has been placed such that it is not held when the pending lock is held. Thanks to Chase Venters for providing lock analysis on the issue. (issue ASTERISK-21162) Reported by: Chase Venters ........ Merged revisions 383839 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 383840 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383841 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-26Suppress compiler warning.Russell Bryant
This code caused a compiler warning when --enable-dev-mode was not used. The warning was that this variable was set but not used. That was indeed the case as the only place this is used is as an argument to SKINNY_DEBUG which is compiled out when not in dev mode. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383838 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-26Fix multi-station answer race condition.Russell Bryant
When an SLA trunk is ringing (inbound call on the trunk) Asterisk will make outbound calls to the stations that have that trunk. If more than one station answers the call at the same time, all channels other than the first one to answer are left in a bad state. The channel gets leaked, is not connected to anything, and there's no way to get rid of it. We now properly clean up these losing channels by hanging up on them. Since they lost the race, as we process their answer, there is no ringing trunk for them to answer. ........ Merged revisions 383835 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 383836 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383837 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-25Set the CALLERID(dnid-num-plan) for incoming ISDN calls.Richard Mudgett
The CALLEDTON channel variable is set for incoming ISDN calls to the lower 7 bits of the Q.931 type-of-number/numbering-plan octet. The CALLERID(dnid-num-plan) should have the same value. (closes issue ASTERISK-21248) Reported by: rmudgett ........ Merged revisions 383796 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 383798 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383799 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-25Fix typoKinsey Moore
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383754 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-25Fix missing ' ' around '='Kinsey Moore
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383753 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-25install_prereq: removed some out-of-date commentsDavid M. Lee
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383747 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-25install_prereq: Adding jansson-devel to RH packagesDavid M. Lee
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383728 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-25Move NewCallerid, HangupRequest and SoftHangupRequest to StasisDavid M. Lee
HangupRequest and SoftHangupRequest are now ast_channel_blob Stasis messages, with the cause code as an optional field in the blob. NewCallerid now simply watches for changes in the callerid information in channel snapshots, and creates the AMI event appropriately. Since the original NewCallerid event honored the channelvars setting in manager.conf, the channel variables configured there had to become a part of the channel snapshot. These are now a part of every snapshot based event, making the configuration description "every time a channel-oriented event is emitted" less of a lie. There a a few other changes wrapped up in here as well. * When ast_channel_topic() is given NULL for a channel, it returns the ast_channel_topic_all() topic instead of NULL. This can clean up a lot of NULL checking we're doing currently. * The fields Cause and Cause-txt were removed from the base channel information and put only on the Hangup events, since those fields are meaningless outside of a Hangup event. * Removed the pipe-delimiter processing of the channelvars field, since that's been deprecated forever. (closes issue ASTERISK-21096) Review: https://reviewboard.asterisk.org/r/2405/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383726 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-25Properly delimit post data in res_config_curl.Sean Bright
........ Merged revisions 383667 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 383668 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383669 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-22Fixed another issue from r383579.David M. Lee
Core modules don't honor <depend> flags in MODULEINFO, which broke jansson if specified --with-jansson to configure. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383633 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-22Fix StopMixMonitor Hanging Up When Unable To Stop MixMonitor On A ChannelMichael L. Young
A regression was accidentally introduced when allowing an optional ID to be used when calling StopMixMonitor. When we are unable to stop MixMonitor on a channel, -1 is being returned which triggers the hangup of the channel. This patch restores the prior behavior by returning 0 whether we were successful or not. It also allows the call from the manager to use the return code when the action fails. (closes issue ASTERISK-21294) Reported by: daroz Tested by: daroz Patches: asterisk-21294-stop_mixmonitor_hangingup.diff Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2404/ ........ Merged revisions 383631 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383632 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-22Corrected some module issues introduced by r383579.David M. Lee
When I moved res_json.c to json.c, I left the MODULE_INFO stuff in there, which was interesting if you ran module show. I also forgot to call what was in module_load() from asterisk main(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383611 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-22Move more channel events to Stasis; move res_json.c to main/json.c.David M. Lee
This patch started out simply as fixing the bouncing tests introduced in r382685, but required some other changes to give it a decent implementation. To fix the bouncing tests, the UserEvent and Newexten AMI events needed to be refactored to dispatch via Stasis. Dispatching directly to AMI resulted in those events sometimes getting ahead of the associated Newchannel events, which would understandably confuse anyone. I found that instead of creating a zillion different message types and structures associated with them, it would be preferable to define a message type that has a channel snapshot and a blob of structured data with a small bit of additional information. The JSON object model provides a very nice way of representing structured data, so I went with that. * Move JSON support from res_json.c to main/json.c * Made libjansson-dev a required dependency * Added an ast_channel_blob message type, which has a channel snapshot and JSON blob of data. * Changed UserEvent and Newexten events so that they are dispatched via ast_channel_blob messages on the channel's topic. * Got rid of the ast_channel_varset message; used ast_channel_blob instead. * Extracted the manager functions converting Stasis channel events to AMI events into manager_channel.c. (issue ASTERISK-21096) Review: https://reviewboard.asterisk.org/r/2381/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383579 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-22Fix skinny voicemail indication issues.Damien Wedhorn
Unsubscribe from MWI stasis event on channel reload. (closes issue ASTERISK-21216) Reported by: wedhorn Tested by: snuffy, myself Patches: skinny-mwiind02.diff uploaded by snuffy (license 5024) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383560 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-21Corrected doc error for Stasis. I guess the mutex isn't necessary.David M. Lee
Thanks, rmudgett! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-21Fix astobj2 doxygen comment.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383519 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-20Have func_curl log a warning when a curl request fails.Walter Doekes
Review: https://reviewboard.asterisk.org/r/2403/ ........ Merged revisions 383460 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 383461 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383462 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-20Minor cleanup in func_curl near hashcompat code.Walter Doekes
Review: https://reviewboard.asterisk.org/r/2402/ ........ Merged revisions 383457 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383458 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-20Resolve a race condition in StasisKinsey Moore
Because of the way that topics were handled when publishing, it was possible to dispatch a message to a subscription after that subscription had been unsubscribed such that the dispatched message arrived at the callback after the callback had received its final message. In callbacks that cleaned up user data, this would often cause a segfault. This has been resolved by locking the topic during the entirety of dispatch. To prevent long publishing and topic locking times, forwarding subscriptions have been made to be standard subscriptions instead of mailboxless subscriptions which were dispatched at publishing time. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383422 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-20Pass the sorcery instance to wizards for CUD operations as well as retrieve.Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383405 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-19Fix lock destruction/unlock inversionKinsey Moore
When using scoped locks, the unref of an AO2 object should happen after the unlock occurs which requires usage of scoped refs. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383377 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-19Multiple revisions 383341-383342David M. Lee
........ r383341 | dlee | 2013-03-19 10:57:29 -0500 (Tue, 19 Mar 2013) | 5 lines Removed codecs/g722/*.i on make clean ........ Merged revisions 383340 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r383342 | dlee | 2013-03-19 10:58:33 -0500 (Tue, 19 Mar 2013) | 1 line Remove codecs/speex/*.i on make clean ........ Merged revisions 383341-383342 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383343 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-16Make sure things compile...Kinsey Moore
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383287 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-16Transition MWI to Stasis-coreKinsey Moore
Remove MWI's dependency on the event system by moving it to Stasis-core. This also introduces forwarding topic pools in Stasis-core which aggregate many dynamically allocated topics into a single primary topic. Review: https://reviewboard.asterisk.org/r/2368/ (closes issue ASTERISK-21097) Patch-by: Kinsey Moore git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383284 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-16Add support for using XMPP buddy state via device state.Joshua Colp
This change allows you to use XMPP buddy state in places where device state can be used be used, such as dialplan hints. If at least one resource is available the buddy is considered available. Now your phone can reflect their IM status too! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383283 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-16Fix a bug where resources were not found due to hashing on the priority itself.Joshua Colp
........ Merged revisions 383266 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383267 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-15A simplistic router for stasis_message's.David M. Lee
Often times, when subscribing to a topic, one wants to handle different message types differently. While one could cascade if/else statements through the subscription handler, it is much cleaner to specify a different callback for each message type. The stasis_message_router is here to help! A stasis_message_router is constructed for a particular stasis_topic, which is subscribes to. Call stasis_message_router_unsubscribe() to cancel that subscription. Once constructed, routes can be added using stasis_message_router_add() (or stasis_message_router_set_default() for any messages not handled by other routes). There may be only one route per stasis_message_type. The route's callback is invoked just as if it were a callback for a subscription; but it only gets called for messages of the specified type. (issue ASTERISK-20887) Review: https://reviewboard.asterisk.org/r/2390/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383242 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-15Sample config file for stasis-core.David M. Lee
(issue ASTERISK-20887) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383225 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-15Take advantage of the fact that stasis_unsubscribe now returns NULLKinsey Moore
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383169 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-15Make stasis unsubscription functions return NULLKinsey Moore
Unsubscribing things in Asterisk seems to very commonly follow with NULLing out the variable that was unsubscribed. This change makes that a bit simpler. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383168 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-15tcptls: Prevent unsupported options from being setKinsey Moore
AMI, HTTP, and chan_sip all support TLS in some way, but none of them support all the options that Asterisk's TLS core is capable of interpreting. This prevents consumers of the TLS/SSL layer from setting TLS/SSL options that they do not support. This also gets tlsverifyclient closer to a working state by requesting the client certificate when tlsverifyclient is set. Currently, there is no consumer of main/tcptls.c in Asterisk that supports this feature and so it can not be properly tested. Review: https://reviewboard.asterisk.org/r/2370/ Reported-by: John Bigelow Patch-by: Kinsey Moore (closes issue AST-1093) ........ Merged revisions 383165 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 383166 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383167 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-15When a session timer expires during a T.38 call, re-invite with correct SDPMatthew Jordan
When a session timer expires during a dialog that has re-negotiated to T.38 and Asterisk is the refresher, Asterisk will send a re-INVITE with an SDP containing audio media only. This causes some hilarity with the poor fax session under weigh. This patch corrects that by sending T.38 parameters if we are in the middle of a T.38 session. (closes issue ASTERISK-21232) Reported by: Nitesh Bansal patches: dont-send-audio-reinvite-for-sess-timer-in-t38-call.patch uploaded by nbansal (License 6418) ........ Merged revisions 383124 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 383125 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383126 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-15Fix processing of call files when using KQueue on OS XMatthew Jordan
In certain situations, call files are not processed when using KQueue with pbx_spool. Asterisk was sending an invalid timeout value when the spool directory is empty, causing the call to kevent to error immediately. This can create a tight loop, increasing the CPU load on the system. (closes issue ASTERISK-21176) Reported by: Carlton O'Riley patches: kqueue_osx.patch uploaded by coriley (License 6473) ........ Merged revisions 383120 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 383121 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383122 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-14Fix whitespace in AST_EXT_LIB_CHECK macro.Jason Parker
........ Merged revisions 383061 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 383062 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383063 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-13Always set the RTP instance data in the RTP engineMatthew Jordan
Not informing the RTP engine of the instance data creates shrapnel. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383008 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-12Update DoxygenAndrew Latham
Push some cleanups upstream before testing another ticket. (issue ASTERISK-20259) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382989 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-12Fix Sorting Order For Parking Lots Stored In Static RealtimeMichael L. Young
When retrieving the parking lots from a MySQL database table, the current order is "filename, cat_metric desc, var_metric asc, category". If there are multiple parking lots with the same cat_metric but different categories, everything is being sorted on cat_metric first resulting in errors when loading the parking lots. This patch fixes the problem by sorting on the category field first, then the cat_metric field. (closes issue ASTERISK-21035) Reported by: Alex Epshteyn Patches: asterisk-21035-orderby.diff Michael L. Young (license 5026) ........ Merged revisions 382942 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 382943 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382954 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-12Update Contributed Realtime Schema Files - IPv6 AddressesMichael L. Young
This commit updates some fields in the contributed realtime schema files to handle IPv6 addresses. (closes issue ASTERISK-21173) Reported by: Torrey Searle Patches: realtime_sql.patch Torrey Searle (license 5334) asterisk-21173-update-ip-fields.diff Michael L. Young (license 5026) ........ Merged revisions 382939 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 382940 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382941 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-12Fix a crash when res_xmpp is configured using a username without a domain.Joshua Colp
(closes issue ASTERISK-21156) Reported by: amsoft2001 ........ Merged revisions 382923 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382924 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-12Switch to using external pjproject libraries.Jason Parker
ICE/STUN/TURN support in res_rtp_asterisk is also now optional. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382900 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-12Include the Username field in SIP Registry events when Status is registeredMatthew Jordan
In ASTERISK-17888, the AMI Registry event during SIP registrations was supposed to include the Username field. Somehow, one of the events was missed. This patch corrects that - the Username field should be included in all AMI Registry events involving SIP registrations. (issue ASTERISK-17888) (closes issue ASTERISK-21201) Reported by: Dmitriy Serov patches: chan_sip.c.diff uploaded by Dmitriy Serov (license 6479) ........ Merged revisions 382847 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 382848 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382852 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-12Fix core dump on CLI usageIgor Goncharovskiy
Fix issue with 'unistim show info' CLI command when device connected not configured ........ Merged revisions 382827 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382828 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-11Added an option to disallow music on holdKevin Harwell
Added an option "discard_remote_hold_retrieval" (default "no") that if set does not trigger the music on hold event. This essentially stops telling the peer to start music on hold. (issue ABE-2899) Reported by: Denis Alberto Martinez Review: https://reviewboard.asterisk.org/r/2336/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382787 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-09confbridge: Rename items for clarity and consistency.Richard Mudgett
struct conference_bridge_user -> struct confbridge_user struct conference_bridge -> struct confbridge_conference struct conference_state -> struct confbridge_state struct conference_bridge_user *conference_bridge_user -> struct confbridge_user *user struct conference_bridge_user *cbu -> struct confbridge_user *user struct conference_bridge *conference_bridge -> struct confbridge_conference *conference The names are now generally shorter, consistently used, and don't conflict with the struct names. This patch handles the renaming part of the issue. (issue ASTERISK-20776) Reported by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382764 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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