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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2013-03-25install_prereq: removed some out-of-date commentsDavid M. Lee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
2013-03-08chan_sip: Update the via header when relaying SMS MESSAGEJonathan Rose
Prior to this change, certain conditions for sending the message would result in an address of '(null)' being used in the via header of the SIP message because a NULl value of pvt->ourip was used when initially generating the via header. This is fixed by adding a call to build_via when the address is set before sending the message. (closes issue ASTERISK-21148) Reported by: Zhi Cheng Patches: 700-sip_msg_send_via_fix.patch uploaded by Zhi Cheng (license 6475) ........ Merged revisions 382739 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382746 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-08Stasis documentation updates.David M. Lee
(issue ASTERISK-20887) (issue ASTERISK-20959) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382724 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-08Ensure dummy channels get a stasis topic.David M. Lee
Fixes test failure introduced in r382685. (issue ASTERISK-20887) (issue ASTERISK-20959) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382721 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-08Add message dump capability to stasis cache layerKinsey Moore
The cache dump mechanism allows the developer to retreive multiple items of a given type (or of all types) from the cache residing in a stasis caching topic in addition to the existing single-item cache retreival mechanism. This also adds to the caching unit tests to ensure that the new cache dump mechanism is functioning properly. Review: https://reviewboard.asterisk.org/r/2367/ (issue ASTERISK-21097) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-08This patch adds a new message bus API to Asterisk.David M. Lee
For the initial use of this bus, I took some work kmoore did creating channel snapshots. So rather than create AMI events directly in the channel code, this patch generates Stasis events, which manager.c uses to then publish the AMI event. This message bus provides a generic publish/subscribe mechanism within Asterisk. This message bus is: - Loosely coupled; new message types can be added in seperate modules. - Easy to use; publishing and subscribing are straightforward operations. In addition to basic publish/subscribe, the patch also provides mechanisms for message forwarding, and for message caching. (issue ASTERISK-20887) (closes issue ASTERISK-20959) Review: https://reviewboard.asterisk.org/r/2339/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382685 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-08Remove unused functionMatthew Jordan
After r382670, get_ip_and_port_from_sdp was no longer used. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382671 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-08Don't reset the RTP address on a glare re-INVITEMatthew Jordan
Originally, way back in r201583, we added the alternate RTP address so that the RTP engine would expect to receive audio from a new source when a glare re-INVITE occurred. In r382589, we remove the alternate RTP source, as the 'secret' probation mode allows for switching to a new RTP source when a previous source stops sending RTP. At the time, it seemed appropriate to set the RTP source based on the information in the glared re-INVITE. Unfortunately, that doesn't work so well - in a glared re-INVITE that occurs with no SDP - such as in a connected line update that glances - we'll set the RTP source to an invalid address. In subsequent re-INVITE requests from this Asterisk instance, we'll then send an invalid media address, which will result in the remote side sending a 488. Whoops. There isn't any need to reset the RTP source - if we're using strictrtp, we'll simply synchronize to a new source when we stop getting packets from the old one. If we aren't using strictrtp, then again there shouldn't be a problem. Note that the Asterisk Test Suite's connectedline test caught this error. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382670 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-07Changing log level of "Not changing threadpool size" from notice to debug.David M. Lee
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382648 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-07Load sorcery modules earlier, so they can actually be used.Jason Parker
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382636 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-07Let vm_mailbox_snapshot combine "Urgent" when no folder is specifiedMatthew Jordan
r381835 fixed a bug in vm_mailbox_snapshot where combining INBOX and Old forgot that Urgent also "counts" as new messages. This fixed the problem when any of the three folders was specified and the combine option was used. It missed the case where the folder isn't specified and we build a snapshot of all folders. This patch corrects that. ........ Merged revisions 382617 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382621 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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