2012-08-10 Asterisk Development Team * Asterisk 11.0.0-beta1 Released. 2012-08-10 20:08 +0000 [r371121] Matthew Jordan * Created Asterisk 11 branch. Because its one greater then 10. 2012-08-10 19:54 +0000 [r371120] Richard Mudgett * main/channel.c, channels/chan_misdn.c, channels/chan_sip.c, main/channel_internal_api.c, main/features.c, include/asterisk/channel.h, channels/sig_pri.c, funcs/func_callerid.c, main/cli.c: Add private representation of caller, connected and redirecting party ids. This patch adds the feature "Private representation of caller, connected and redirecting party ids", as previously discussed with us (DATUS) and Digium. 1. Feature motivation Until now it is quite difficult to modify a party number or name which can only be seen by exactly one particular instantiated technology channel subscriber. One example where a modified party number or name on one channel is spread over several channels are supplementary services like call transfer or pickup. To implement these features Asterisk internally copies caller and connected ids from one channel to another. Another example are extension subscriptions. The monitoring entities (watchers) are notified of state changes and - if desired - of party numbers or names which represent the involving call parties. One major feature where a private representation of party names is essentially needed, i.e. where a party name shall be exclusively signaled to only one particular user, is a private user-specific name resolution for party numbers. A lookup in a private destination-dependent telephone book shall provide party names which cannot be seen by any other user at any time. 2. Feature Description This feature comes along with the implementation of additional private party id elements for caller id, connected id and redirecting ids inside Asterisk channels. The private party id elements can be read or set by the user using Asterisk dialplan functions. When a technology channel is initiating a call, receives an internal connected-line update event, or receives an internal redirecting update event, it merges the corresponding public id with the private id to create an effective party id. The effective party id is then used for protocol signaling. The channel technologies which initially support the private id representation with this patch are SIP (chan_sip), mISDN (chan_misdn) and PRI (chan_dahdi). Once a private name or number on a channel is set and (implicitly) made valid, it is generally used for any further protocol signaling until it is rewritten or invalidated. To simplify the invalidation of private ids all internally generated connected/redirecting update events and also all connected/redirecting update events which are generated by technology channels -- receiving regarding protocol information - automatically trigger the invalidation of private ids. If not using the private party id representation feature at all, i.e. if using only the 'regular' caller-id, connected and redirecting related functions, the current characteristic of Asterisk is not affected by the new extended functionality. 3. User interface Description To grant access to the private name and number representation from the Asterisk dialplan, the CALLERID, CONNECTEDLINE and REDIRECTING dialplan functions are extended by the following data types. The formats of these data types are equal to the corresponding regular 'non-private' already existing data types: CALLERID: priv-all priv-name priv-name-valid priv-name-charset priv-name-pres priv-num priv-num-valid priv-num-plan priv-num-pres priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag CONNECTEDLINE: priv-name priv-name-valid priv-name-pres priv-name-charset priv-num priv-num-valid priv-num-pres priv-num-plan priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag REDIRECTING: priv-orig-name priv-orig-name-valid priv-orig-name-pres priv-orig-name-charset priv-orig-num priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type priv-orig-subaddr-odd priv-orig-tag priv-from-name priv-from-name-valid priv-from-name-pres priv-from-name-charset priv-from-num priv-from-num-valid priv-from-num-pres priv-from-num-plan priv-from-subaddr priv-from-subaddr-valid priv-from-subaddr-type priv-from-subaddr-odd priv-from-tag priv-to-name priv-to-name-valid priv-to-name-pres priv-to-name-charset priv-to-num priv-to-num-valid priv-to-num-pres priv-to-num-plan priv-to-subaddr priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd priv-to-tag Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2030/ 2012-08-10 17:56 +0000 [r371113] Mark Michelson * channels/chan_sip.c: Fix a comparison that was causing presence tests to fail. A recent change made it so that device state changes that were not actual "changes" would not get reported to subscribers. The problem was that this inadvertently blocked presence updates as well. 2012-08-10 16:49 +0000 [r371059-371091] Alexandr Anikin * addons/chan_ooh323.c, /: remove ALREADYGONE flag on ooh323 call data by ooh323_indicate (CONGESTION/BUSY) due to call hasn't gone there really. This indication arrive from asterisk core not h.323 stack (closes issue ASTERISK-19308) Reported by: Dmitry Melekhov Patches: ASTERISK-19308.patch ........ Merged revisions 371089 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371090 from http://svn.asterisk.org/svn/asterisk/branches/10 * addons/ooh323c/src/ooGkClient.c, /: Send re-register packets by GRQ (gatekeeper request) interval (close issue ASTERISK-20094) Patches: ASTERISK-20094-2.patch ........ Merged revisions 371060 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371061 from http://svn.asterisk.org/svn/asterisk/branches/10 * addons/ooh323c/src/ooTimer.c: restore calling cb functions by timer expire this was broken in rev 369602 2012-08-10 02:07 +0000 [r371052] Richard Mudgett * main/features.c: Fix pickup extension channel reference error. You cannot unref a pointer and then expect to ref it again later. * Fix potential NULL pointer deref if the call pickup search fails. 2012-08-09 21:35 +0000 [r371036-371043] Alexandr Anikin * addons/chan_ooh323.c: Introdue 'ooh323 show gk' cli command that show status of connection to H.323 Gatekeeper (GkClient state) * addons/ooh323c/src/ooGkClient.c, /: Fix to resend GRQ/RRQ if RRJ (registration reject) is received (close issue ASTERISK-20094) Patches: ASTERISK-20094.patch ........ Merged revisions 371011 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371022 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-08-09 19:22 +0000 [r371030] Richard Mudgett * channels/chan_dahdi.c, /, configure, include/asterisk/autoconfig.h.in, configure.ac, channels/sig_pri.c, channels/sig_ss7.c: Use better libss7 detection test and move libpri compile test. ........ Merged revisions 371012 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371013 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-08-09 18:28 +0000 [r371010] Alexandr Anikin * /, addons/ooh323c/src/ooh323ep.c: change opening h323 logfile with append mode instead of overwrite ........ Merged revisions 370988 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370989 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-08-09 17:40 +0000 [r370987] Kinsey Moore * /, apps/app_meetme.c: Correct documentation for the MeetMe x flag The documentation for the x flag for MeetMe incorrectly described its function as closing down the conference when the last marked user left. It actually causes the users with that flag to leave the conference when the last marked user exits. The functionality of this flag is not changing. ........ Merged revisions 370985 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370986 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-08-09 14:52 +0000 [r370979] Mark Michelson * main/pbx.c, channels/chan_sip.c, include/asterisk/pbx.h, channels/sip/include/sip.h: Extend extension state callbacks to have more information. Quote from review board: This patch extends the extension state callbacks so that monitoring channels (as chan_sip) get more information of the devices which are responsible for an extension state change. The additional information is needed by chan_sip to present names/numbers of the caller and callee in an early-state SIP notification. Users of extenstion state callback not interested in the additional information are not affected by the changes. Motivation: to present the involved party's name/number in an early-state nofification (used by the notified device as a pickup offer) one after another so that a user can see which call he will pick up in an undirected pickup. Such a pickup offer to a user shall indicate the same call (number/name-A calls number/name-B) as the call which would be picked up when an undirected pickup is executed. Users interested in additional state info must use the new functions ast_extension_state_add_extended() resp. ast_extension_state_add_destroy_extended() to register an extended state callback. When the callback is registered this way, an extra member device_state_info of struct ast_state_cb_info is passed to the callback in addition to the aggregated extension state. This container holds an object for every device of the monitored extension hint consisting of the device name, the device state and a channel reference to the channel which (presumably) caused the device state. The information is used by chan_sip for early-state notifications. When the state of a device changes and the new state contains AST_EVENT_RINGING, an early-state notification is sent to the subscribed devices with the caller/callee names/numbers of the oldest ringing channel of the monitored extension. The notified user may then invoke a direct pickup, which will pickup exactly this channel. Users of the old non-extended callbacks will only be called when the aggregated state did change (same behavior as before). Users of the extended callback will also be called when the state is unchanged but does contain AST_EVENT_RINGING. That could be the case if two channels are ringing at one device and one of them hangs up, so the aggregated state does not change. This way the monitoring channel can create a new early-state notification with the now ringing party-ids. Review: https://reviewboard.asterisk.org/r/2048 This contribution comes from Guenther Kelleter 2012-08-09 14:36 +0000 [r370978] Jonathan Rose * pbx/pbx_dundi.c, CHANGES: DUNDi: Add CLI commands DUNDi show cache and DUNDi show hints (closes issue ASTERISK-18390) Reported by: Peter Racz Patches: dundi_cli_cache.patch.v2 uploaded by Peter Racz (license #6290) ASTERISK-18390_dundi_cli_cache_jrose_mods_v2.diff uploaded by Jonathan Rose (license #6182) 2012-08-08 22:45 +0000 [r370955] Michael L. Young * /, apps/app_chanspy.c: Fix Not Unreferencing A Spied Channel When a channel hangs up while being spied upon and the option to exit the ChanSpy application when the spied on channel hangs up is set, ast_autochan_destroy is not being called and therefore a reference to the spied upon channel is not removed. The symptom being reported was that when using func_group in the dialplan and calling "group show channels" at the cli, the spied upon channel was still being shown while "core show channels" showed that the channel was not up. This patch calls ast_autochan_destroy when a spied upon channel hangs up and the option to exit the ChanSpy application is set, removing the reference to the channel allowing the count for the group that the spied channel was part of to be decremented. (closes issue ASTERISK-17515) Reported by: Arkadiusz Malka Tested by: Alexandr Gordeev, Michael L. Young Patches: asterisk-17515-destroy-autochan.diff uploaded by Michael L. Young (license 5026) ........ Merged revisions 370952 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370954 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-08-08 22:41 +0000 [r370951-370953] Mark Michelson * CHANGES: Move a SIP change up to the other SIP changes in the CHANGES file. * main/channel.c, main/pbx.c, main/manager.c, pbx/pbx_spool.c, apps/app_originate.c, include/asterisk/channel.h, include/asterisk/pbx.h, CHANGES, res/res_clioriginate.c: Allow support for early media on AMI originates and call files. This is based on the work done by Olle Johansson on review board. The idea is that the channel specified in an AMI originate or call file is typically not connected to the outgoing extension until the channel has been answered. With this change, an EarlyMedia header can be specified for AMI originates and an early_media option can be specified in call files. With this option set, once early media is received on a channel, it will be connected with the outgoing extension. (closes issue ASTERISK-18644) Reported by Olle Johansson Review: https://reviewboard.asterisk.org/r/1472 2012-08-08 21:22 +0000 [r370943] Terry Wilson * main/manager.c, CHANGES: Add AMI_CLIENT dialplan function Implementation of a dialplan function for checking manager accounts. Right now it only returns the number of logged in sessions for a manager account, but other attributes can be added later. Patch by: Olle Johansson Review: https://reviewboard.asterisk.org/r/421/ 2012-08-08 20:47 +0000 [r370927] Joshua Colp * main/rtp_engine.c: Create the payload type if it does not exist when setting information based on the 'm' line. An rtpmap attribute is not required for defined payload numbers. 2012-08-08 20:32 +0000 [r370926] Richard Mudgett * channels/chan_dahdi.c, channels/sig_analog.c, channels/sig_analog.h: Convert sig_analog to use a global callback table. 2012-08-08 20:30 +0000 [r370925] Kinsey Moore * main/channel.c, /: Do not define a cause that doesn't actually exist AST_CAUSE_NOTDEFINED is a placeholder for usage when there is no cause information. As such, it should not be defined and translatable as a cause. ........ Merged revisions 370923 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370924 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-08-08 20:17 +0000 [r370887-370902] Richard Mudgett * channels/chan_dahdi.c, channels/sig_analog.c, /, channels/sig_analog.h: Fix the analog dial *0 flash-hook of bridged peer feature. The flash-hook the bridged peer feature now correctly determines if the bridged peer is another chan_dahdi channel, that it is an analog channel, and that it has the correct signaling for an FXO port. It now also flash-hooks the correct channel. ........ Merged revisions 370900 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370901 from http://svn.asterisk.org/svn/asterisk/branches/10 * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c: Convert sig_pri to use a global callback table. * channels/chan_dahdi.c, channels/sig_ss7.h, channels/sig_ss7.c: Convert sig_ss7 to use a global callback table. 2012-08-07 21:58 +0000 [r370881] Damien Wedhorn * build_tools/cflags-devmode.xml, channels/chan_skinny.c: Rewrite of skinny debugging. Debugging messages and associated controls only compiled in if configured with --enable-dev-mode. Debug messages provide more detail (including thread id) and are grouped so the user/dev can limit the type of messages displayed. Functionally no real change to chan_skinny. Review: https://reviewboard.asterisk.org/r/2040/ 2012-08-07 19:59 +0000 [r370860] Joshua Colp * main/rtp_engine.c, include/asterisk/rtp_engine.h: Payload and RTP code are must remain separate since in non-Asterisk format cases they differ. 2012-08-07 19:26 +0000 [r370851-370859] Kinsey Moore * /: Recorded merge of revisions 370858 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Add missing AST_CAUSE_* -> text translations ........ Merged revisions 370856 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/channel.c: Add missing AST_CAUSE_* -> text translations A few of these were missing from the list and are necessary for the Who Hung Up? functionality. 2012-08-07 17:47 +0000 [r370832-370845] Joshua Colp * main/rtp_engine.c: Fix a bug uncovered by the test suite where the RTP payload number was not getting set. * res/res_rtp_asterisk.c, main/rtp_engine.c, channels/chan_sip.c, channels/chan_motif.c, include/asterisk/rtp_engine.h: Reduce memory consumption significantly for users of the RTP engine API by storing only the payloads present and in use instead of every possible one. Review: https://reviewboard.asterisk.org/r/2052/ 2012-08-07 12:46 +0000 [r370820-370831] Matthew Jordan * main/channel.c, channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, channels/chan_misdn.c, channels/chan_sip.c, main/channel_internal_api.c, channels/misdn/chan_misdn_config.h, main/features.c, configs/misdn.conf.sample, include/asterisk/channel.h, configs/sip.conf.sample, CHANGES, channels/sip/include/sip.h, channels/misdn_config.c: Add named callgroups/pickupgroups This patch adds named calledgroups/pickupgroups to Asterisk. Named groups are implemented in parallel to the existing numbered callgroup/pickupgroup implementation. However, unlike the existing implementation, which is limited to a maximum of 64 defined groups, the number of defined groups allowed for named callgroups/pickupgroups is effectively unlimited. Named groups are configured with the keywords "namedcallgroup" and "namedpickupgroup". This corresponds to the numbered group definitions of "callgroup" and "pickupgroup". Note that as the implementation of named groups coexists with the existing numbered implementation, a defined named group of "4" does not equate to numbered group 4. Support for the named groups has been added to the SIP, DAHDI, and mISDN channel drivers. Review: https://reviewboard.asterisk.org/r/2043 Uploaded by: Guenther Kelleter(license #6372) * contrib/realtime/mysql/voicemail_data.sql: Revert r370820 That change is wrong, wrong, wrong. * contrib/realtime/mysql/voicemail_data.sql: Update the MySQL voicemail_data contrib script to reflect Asterisk 11 changes All voicemails now have a 'msg_id' included in their metadata. The ODBC message storage backend now requires this column; as such, the MySQL contrib script that creates the voicemail_data table has been updated with the appropriate column information. 2012-08-06 15:18 +0000 [r370801] Mark Michelson * /, channels/chan_sip.c: Improve debug message for temporary outbound proxies. Thanks to Paul Belanger for pointing this out. ........ Merged revisions 370797 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370798 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-08-03 21:52 +0000 [r370773] Mark Michelson * /, channels/chan_sip.c, channels/sip/config_parser.c, channels/sip/include/sip.h: Multiple revisions 370769-370771 ........ r370769 | mmichelson | 2012-08-03 16:35:00 -0500 (Fri, 03 Aug 2012) | 24 lines Fix error in the "IPorHost" section of a SIP dialstring. This is based on the review request posted by Walter Doekes (referenced lower in the commit message) The main fix here is to treat the IPorHost portion of the dial string as a temporary outbound proxy. This ensures requests get sent to the proper location. Due to the age of the request, some parts were no longer relevant. For instance, the request moved outbound proxy parsing code into a single method. This is done in a previous commit, so it was not necessary to do again. Also, the review request fixed some errors with regards to request routing for CANCEL and ACK requests. This has also been fixed in more recent commits. (closes issue ASTERISK-19677) reported by Walter Doekes Review https://reviewboard.asterisk.org/r/1859 ........ r370770 | mmichelson | 2012-08-03 16:39:35 -0500 (Fri, 03 Aug 2012) | 3 lines Remove unused variable. ........ r370771 | mmichelson | 2012-08-03 16:43:52 -0500 (Fri, 03 Aug 2012) | 5 lines Seriously? Another compilation error fixed. Somebody beat me. ........ Merged revisions 370769-370771 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370772 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-08-02 15:51 +0000 [r370740] Kinsey Moore * channels/chan_sip.c: Fix regression from r370636 When the chan_sip cleanup went in, a typo was included that caused some subscriptions of non-Polycom phones to be limited to the same capabilities as Polycom phones. This resolves the failures in the test suite resulting from this regression. 2012-08-01 19:37 +0000 [r370726] Mark Michelson * main/manager.c: Fix a possible crash due to passing NULL to ast_variables_dup() 2012-08-01 18:52 +0000 [r370720] Richard Mudgett * include/asterisk/astobj2.h, main/astobj2.c: Make astobj2.h not include linkedlists.h. Using astobj2 does not require linkedlists.h be included even though astob2 uses linked lists internally. 2012-08-01 02:26 +0000 [r370699] Kinsey Moore * /, utils/extconf.c: Revert alloca changes for utils These changes were a tad overzealous in the utils directory. Unfortunately, these don't compile with a "make". ........ Merged revisions 370697 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370698 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-07-31 22:28 +0000 [r370681-370691] Mark Michelson * channels/chan_sip.c, configs/sip.conf.sample, CHANGES, channels/sip/include/sip.h: Add headers from SIPAddHeader to outbound REFER requests. This is a patch from kkm from review board. This is useful for adding headers to REFER requests that emanate from a Transfer() dialplan application call. This also fixes some uses of the Referred-by header, removing an extra set of angle brackets. I've modified the reporter's original patch to not require any additions to the sip_refer header and to just remove the referred_by_name from sip_refer since it is no longer needed or used. (closes Issue ASTERISK-17639) reported by Kirill Katsnelson Patches: 019059-sip-refer-addheaders-trunk-353549.diff uploaded by Kirill Katsnelson (license #5845) Review: https://reviewboard.asterisk.org/r/1159 * main/manager.c, configs/manager.conf.sample, CHANGES: Add "setvar" option to manager.conf. With this option set, channel variables can be set on every manager originate. The Variable header can still be used to set additional channel variables for individual calls if desired. This work was completed by Olle Johansson on review board. I have applied the review feedback and am committing it in order to get this into trunk before Asterisk 11 is branched. Review: https://reviewboard.asterisk.org/r/1412 2012-07-31 21:20 +0000 [r370677] Matthew Jordan * /, channels/chan_sip.c: Schedule pokes of registered SIP peers within a given timespan after SIP reload With a large number of SIP peers registered, performing a SIP reload causes a flood of SIP OPTIONS request packets. These are immediately sent out, and, as responses come back, can cause peers to be flagged as 'lagged' due to handling of the many response messages. This fix prevents this "packet storm" and schedules the pokes for a random time. That time varies between 1 ms and the peer's qualify time, or, if the qualify time is unknown, the global qualifyfreq setting. The committed patch has some very small modifications to the patch schmidts wrote for the review. (closes issue ASTERISK-19154) Reported by: Nicolo Mazzon patches: issue19154.patch license #6034 uploaded by schmidts Review: https://reviewboard.asterisk.org/r/1652 ........ Merged revisions 370666 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370672 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-07-31 20:33 +0000 [r370664] Russell Bryant * main/event.c: Move event cache updates into event processing thread. Prior to this patch, updating the device state cache was done by the thread that originated the event. It would update the cache and then queue the event up for another thread to dispatch. This thread moves the cache updating part to be in the same thread as event dispatching. I was working with someone on a heavily loaded Asterisk system and while reviewing backtraces of the system while it was having problems, I noticed that there were a lot of threads contending for the lock on the event cache. By simply moving this into a single thread, this helped performance *a lot* and alleviated some deadlock-like symptoms. Review: https://reviewboard.asterisk.org/r/2066/ 2012-07-31 20:21 +0000 [r370655] Kinsey Moore * /, main/say.c, main/threadstorage.c, funcs/func_strings.c, channels/chan_iax2.c, main/config.c, channels/chan_dahdi.c, pbx/pbx_spool.c, channels/sig_analog.c, main/strcompat.c, main/features.c, pbx/pbx_ael.c, main/http.c, pbx/pbx_realtime.c, channels/chan_alsa.c, channels/sig_ss7.c, main/db.c, include/asterisk/utils.h, main/pbx.c, funcs/func_cut.c, tests/test_linkedlists.c, funcs/func_channel.c, apps/app_macro.c, apps/app_mixmonitor.c, main/asterisk.c, apps/app_voicemail.c, addons/app_mysql.c, apps/app_meetme.c, apps/app_dictate.c, main/utils.c, funcs/func_logic.c, cdr/cdr_pgsql.c, channels/chan_gtalk.c, res/res_jabber.c, res/res_http_websocket.c, res/ael/pval.c, main/channel.c, main/manager.c, apps/app_osplookup.c, res/res_agi.c, apps/app_minivm.c, main/logger.c, main/app.c, addons/chan_mobile.c, apps/app_while.c, res/res_config_pgsql.c, channels/chan_sip.c, apps/app_festival.c, pbx/pbx_lua.c, channels/sig_pri.c, apps/app_getcpeid.c, funcs/func_global.c, channels/chan_jingle.c, main/tcptls.c, apps/app_directed_pickup.c, main/file.c, main/callerid.c, apps/app_sms.c, main/astmm.c, main/event.c, pbx/pbx_dundi.c, include/asterisk/strings.h, utils/extconf.c, main/dsp.c, addons/res_config_mysql.c: Clean up and ensure proper usage of alloca() This replaces all calls to alloca() with ast_alloca() which calls gcc's __builtin_alloca() to avoid BSD semantics and removes all NULL checks on memory allocated via ast_alloca() and ast_strdupa(). (closes issue ASTERISK-20125) Review: https://reviewboard.asterisk.org/r/2032/ ........ Merged revisions 370642 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370643 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-07-31 19:57 +0000 [r370644] Mark Michelson * CHANGES, pbx/pbx_config.c: Add "dialplan remove context" and modify "dialplan add include" From corruptor's review board posting: "I've noticed that we can remove particular extension from context with dialplan remove extension command but in order to remove all extensions in the context we should delete them on by one. I've created dialplan remove context command which uses ast_context_destroy to destroy the whole context with all extensions. I've created to functions for in pbx_config.c: handle_cli_dialplan_remove_context which actually removes context and complete_dialplan_remove_context which completes input. They are based on other similar functions and pretty trivial but I can be mistaken somewhere. "I've also modified dialplan add include into . I've made it similar dialplan add extension ... command. It creates if it doesn't exist and I've also modified complete_dialplan_add_include and removed check for existance of because we can include non-existent context into another one. (I usually include empty (non-existent) contexts in advance). Should we raise warning in this case as it's raised while reading extensions.conf? "I use those functions with AMI. I think manager commands should be created in addition to those CLI commands." I've addressed the latest comments on review board and have made some other coding guidelines-related cleanup. I also have modified the CHANGES file to mention these new commands. (closes issue ASTERISK-19292) reported by Andrey Solovyev Patches: dialplan_add_include.patch uploaded by Andrey Solovyev (license #5214) dialplan_remove_context.patch uploaded by Andrey Solovyev (license #5214) Review: https://reviewboard.asterisk.org/r/2042 2012-07-31 19:10 +0000 [r370636] Kinsey Moore * channels/chan_sip.c, channels/sip/security_events.c, channels/sip/include/sip.h: Clean up chan_sip This clean up was broken out from https://reviewboard.asterisk.org/r/1976/ and addresses the following: - struct sip_refer converted to use the stringfields API. - sip_{refer|notify}_allocate -> sip_{notify|refer}_alloc to match other *alloc functions. - Replace get_msg_text, get_msg_text2 and get_pidf_body -> No, not get_pidf_msg_text_body3 but get_content, to match add_content. - get_body doesn't get the request body, renamed to get_content_line. - get_body_by_line doesn't get the body line, and is just a simple if test. Moved code inline and removed function. - Remove camelCase in struct sip_peer peer state variables, onHold -> onhold, inUse -> inuse, inRinging -> ringing. - Remove camelCase in struct sip_request rlPart1 -> rlpart1, rlPart2 -> rlpart2. - Rename instances of pvt->randdata to pvt->nonce because that is what it is, no need to update struct sip_pvt because _it already has a nonce field_. - Removed struct sip_pvt randdata stringfield. - Remove useless (and inconsistent) 'header' suffix on variables in handle_request_subscribe. - Use ast_strdupa on Event header in handle_request_subscribe to avoid overly complicated strncmp calls to find the event package. - Move get_destination check in handle_request_subscribe to avoid duplicate checking for packages that don't need it. - Move extension state callback management in handle_request_subscribe to avoid duplicate checking for packages that don't need it. - Remove duplicate append_date prototype. - Rename append_date -> add_date to match other add_xxx functions. - Added add_expires helper function, removed code that manually added expires header. - Remove _header suffix on add_diversion_header (no other header adding functions have this). - Don't pass req->debug to request handle_request_XXXXX handlers if req is also being passed. - Don't pass req->ignore to check_auth as req is already being passed. - Don't create a subscription in handle_request_subscribe if p->expiry == 0. - Don't walk of the back of referred_by_name when splitting string in get_refer_info - Remove duplicate check for no dialog in handle_incoming when sipmethod == SIP_REFER, handle_request_refer checks for that. Review: https://reviewboard.asterisk.org/r/1993/ Patch-by: gareth 2012-07-30 23:26 +0000 [r370565-370598] Richard Mudgett * main/test.c: Tweak unit test warning message. * funcs/func_presencestate.c, main/test.c: Fix some presence-state unit test typos. * apps/app_confbridge.c: DECLINE to load confbridge if the config fails to load. * channels/chan_misdn.c, /: Release B channel allocation on error path in chan_misdn. ........ Merged revisions 370563 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370564 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-07-30 14:52 +0000 [r370548] Jonathan Rose * /, apps/app_meetme.c: app_meetme: Change app_meetme support level to extended from deprecated (closes issue ASTERISK-20134) Reported by: Leif Madsen ........ Merged revisions 370547 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-07-30 13:45 +0000 [r370534-370541] Russell Bryant * tests/test_event.c: Fix ast_event_new unit test. One of my recent commits broke this test. The error was: [test_event.c:event_new_test:214]: Events expected to be identical have different size: 69 != 59 The difference in size occurred because the first event had the EID IE added to the event twice. ast_event_new() now always adds it automatically. Previously it only added it if there were no IEs specified, which was kind of weird. * include/asterisk/event_defs.h, res/res_corosync.c, main/event.c: Add a "corosync ping" CLI command. This patch adds a new CLI command to the res_corosync module. It is primarily used as a debugging tool. It lets you fire off an event which will cause res_corosync on other nodes in the cluster to place messages into the logger if everything is working ok. It verifies that the corosync communication is working as expected. I didn't put anything in the CHANGES file for this, because this module is new in Asterisk 11. There is already a generic "res_corosync new module" entry in there so I figure that covers it just fine. * addons/app_mysql.c, CHANGES: Allow specifying a port number for the MySQL server. This patch allows you to specify a port number for the MySQL server. It's useful if a MySQL server is running on a non-standard port. Even though this module is deprecated in favor of func_odbc, someone asked for this feature and it seems pretty harmless to add. It has been tested using a number of combinations of with/without a port number specified in the dialplan and changing the port number for mysqld. 2012-07-26 15:31 +0000 [r370510-370518] Jonathan Rose * channels/chan_sip.c, CHANGES: chan_sip: Add SIPpeerstatus command to AMI This patch was submitted by mnicholson a while back. It adds a new AMI action which allows users to request SIP peer status on demand similar to existing PeerStatus events and to the output you would see from CLI with sip show peer Review: https://reviewboard.asterisk.org/r/1098/ * /, res/res_agi.c: res_agi: Add message indicating need for \n character in verbose message The while loop responsible for reading AGI messages from a fastAGI service can end up looping indefinitely when an AGI script fails to indicate the end of a message with a \n character. This patch adds an indication that we are expecting a \n character to end the message to make it more clear to users that this is necessary if they are receiving this warning over and over. (issue ASTERISK-20061) Reported by: Eike Kuiper ........ Merged revisions 370494 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370495 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-07-25 14:27 +0000 [r370481-370488] Kevin P. Fleming * main/Makefile: Repair editline builds using in-tree editline sources. The previous change to the build system for using a system-provided editline library was missing a crucial include directory for building against the copy of the library in the Asterisk source tree. * main/Makefile: Use an absolute path when referring to the embedded editline directory. This patch changes the build system to refer to the embedded editline directory using an absolute path, which will resolve a problem seen on the CentOS automated build agents. * build_tools/menuselect-deps.in, configure, include/asterisk/autoconfig.h.in, main/Makefile, main/editline/configure, configure.ac, main/editline/readline (removed), main/editline/readline.c, main/editline/configure.in, CHANGES, makeopts.in, main/editline/readline.h (added), main/asterisk.c, contrib/scripts/install_prereq, main/cli.c: Enable usage of system-provided NetBSD editline library if available. This patch changes the Asterisk configure script and build system to detect the presence of the NetBSD editline library (libedit) on the system. If it is found, it will be used in preference to the version included in the Asterisk source tree. (closes issue ASTERISK-18725) Reported by: Jeffrey C. Ollie Review: https://reviewboard.asterisk.org/r/1528/ Patches: 0001-Allow-linking-building-against-an-external-editline.patch uploaded by jcollie (license #5373) (heavily modified by kpfleming) 2012-07-25 03:51 +0000 [r370474] Terry Wilson * main/pbx.c, /: Revert a change that broke compilation 1) There is no such function as ast_ref() 2) The patch was originally credited as the one uploaded by Guenther Kelleter (license 6372) via issue AST-921, but the patch committed was not the patch referenced on the issue. 3) Guenther Kelleter's patch was actually correct. It moved the ast_free above the presencechange_cleanup label. I am not committing his change as it is not technically necesary--calling ast_free(NULL) is perfectly safe and I worry that moving the ast_free outside of the label could lead to future bugs if someone ever adds another failure conditional and expects 'goto presencechange_cleanup;' to clean up after everything. 2012-07-24 21:30 +0000 [r370466] Jonathan Rose * main/pbx.c, /: Don't attempt free of NULL ptr in pbx.c handle_presencechange (closes issue AST-921) Reported by: Guenther Kelleter Patches: nullptr.patch uploaded by Guenther Kelleter (license 6372) 2012-07-24 19:12 +0000 [r370453] Kevin P. Fleming * tests/test_acl.c: Silence a warning message from older versions of GCC. Revision 370426 introduced the use of a nested function in tests/test_acl.c, but the lack of the 'auto' scope specifier on the function and a forward declaration resulted in compilation errors on the automated test systems. 2012-07-24 17:16 +0000 [r370433] Tzafrir Cohen * /, channels/chan_oss.c: chan_oss: fix "sample rate" error message Merged revisions 370428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 Merged revisions 370432 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-07-24 16:54 +0000 [r370426-370431] Kevin P. Fleming * main/frame.c, /: Rewrite a comment that didn't adequately explain the code it was documenting. ........ Merged revisions 370429 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370430 from http://svn.asterisk.org/svn/asterisk/branches/10 * CHANGES: Update CHANGES for list/negation ACL feature. * tests/test_acl.c, main/acl.c: Allow permit/deny ACL lines to contain multiple items and negated entries. Rules in ACLs (specified using 'permit' and 'deny') can now contain multiple items (separated by commas), and items in the rule can be negated by prefixing them with '!'. This simplifies Asterisk Realtime configurations, since it is no longer necessray to control the order that the 'permit' and 'deny' columns are returned from queries. Review: https://reviewboard.asterisk.org/r/1592/ Initial patch contributed by Tilghman Lesher Unit tests written by Kevin P. Fleming 2012-07-24 16:15 +0000 [r370419-370420] Joshua Colp * res/res_rtp_asterisk.c: Build is underway so logging can go away. * res/res_rtp_asterisk.c: Temporarily enable pj logging to console for debugging pjnath issue exposed by build slave. 2012-07-24 08:53 +0000 [r370413] Igor Goncharovskiy * channels/chan_unistim.c: Remove code, that operate with cdr in attempt_transfer(). That was removed somewhere between 1.2 and 1.4 and acidentaly put back in chan_unistim. (closes issue ASTERISK-19628) Reported by: Igor Olhovskiy 2012-07-23 21:27 +0000 [r370407] Kevin P. Fleming * codecs/Makefile, build_tools/menuselect-deps.in, configure, include/asterisk/autoconfig.h.in, configure.ac, codecs/codec_ilbc.c, CHANGES, makeopts.in: Enable usage of system-provided iLBC library. The WebRTC version of the iLBC codec is now package as a library and is available on some platforms. This patch allows codec_ilbc to be built against that library if it is present. Review: https://reviewboard.asterisk.org/r/1964/ 2012-07-23 21:15 +0000 [r370387] Matthew Jordan * tests/test_abstract_jb.c (added), main/abstract_jb.c, funcs/func_jitterbuffer.c, include/asterisk/abstract_jb.h: Unit tests for the Jitter Buffer API; remove unnecessary resync This patch includes the following: * Unit tests for the abstract Jitter Buffer API. This includes both fixed and adaptive flavors, testing nominal creation, frame input, frame retrieval, resyncing; off nominal frame input overflow, out of order, and others. * Tweaks to the abstract_jb API to remove the unnecessary resync_threshold parameter from the create function (resync_threshold is already in the struct passed into the create function) * Ensure the fixed jitter buffer is empty before destroying it, to avoid an ASSERT * Don't "resync" the adaptive jitter buffer. The mechanism that was being used actually causes the jitter buffer to think its being overflowed by going around the jitterbuf API and attempting to 'resynch' it improperly. If a resync is needed, the jitter buffer will do it properly by itself. Note that this is only an optimization needed for trunk, as the worst that happens is the loss of three voice packets before the adaptive jitter buffer will resync anyway. Review: https://reviewboard.asterisk.org/r/2035 2012-07-23 21:10 +0000 [r370386] Mark Michelson * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add separate configuration options for subscription and registration minexpiry and maxexpiry. This offers more fine-grained control over how long subscriptions last without negatively affecting the expiration range for registrations. Uploaded by: Guenther Kelleter(license #6372) Review: https://reviewboard.asterisk.org/r/2051 2012-07-23 21:10 +0000 [r370385] Kevin P. Fleming * /, funcs/func_shell.c: Improve documentation for the SHELL() dialplan function. ........ Merged revisions 370383 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370384 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-07-23 21:02 +0000 [r370382] Mark Michelson * UPGRADE.txt: Add notes to UPGRADE.txt about addition of msg_id to VoiceMails. 2012-07-23 00:15 +0000 [r370354] Joshua Colp * UPGRADE.txt: Update UPGRADE.txt with notes about ICE support and res_xmpp. 2012-07-22 23:37 +0000 [r370353] Matthew Jordan * CHANGES: Update CHANGES for Asterisk 11 This updates the CHANGES file with things that were committed for Asterisk 11, but were not noted in that file. 2012-07-22 17:03 +0000 [r370347] Joshua Colp * res/res_rtp_asterisk.c, channels/chan_sip.c, configs/sip.conf.sample, channels/sip/include/sip.h: Prevent multiple local candidates from being added with the same information and add support for disabling ICE on a per-peer basis. (closes issue ASTERISK-20088) Reported by: wimpy Review: https://reviewboard.asterisk.org/r/2044/ 2012-07-21 13:25 +0000 [r370341] Terry Wilson * main/config_options.c, apps/app_confbridge.c, apps/confbridge/conf_config_parser.c: Fix segfault introduced by conversion to ACO API The value "none" is specified in the config file as a valid value for the "video_mode" option. The code prior to the ACO conversion did not check for "none", but just ignored it and relied on the default zero value. The parsing with ACO is more strict, so without handling "none" specifically, parsing would fail. When parsing failed, but the module loaded anyway, the config info would never be stored, and one place in the code did not check for this case and would segfault. It was also possible that the aco_info struct's internals would be destroyed and used as well. This patch keeps the module from loading after parse failures, adds the "none" option to "video_mode", registers CLI functions only after parsing has completed, checks the config data for NULL before accessing it, and returns -1 on some allocation failures when initializing. (closes issue ASTERISK-20159) Reported by: Birger "WIMPy" Harzenetter Tested by: Birger "WIMPy" Harzenetter Patches: confbridge_fix3.txt uploaded by Terry Wilson 2012-07-20 19:36 +0000 [r370335] Jonathan Rose * channels/chan_iax2.c: chan_iax2: Fix a segfault introduced by call ID logging Didn't previously check that a non NULL IAX channel was stored in the array at the requested position before attempting iax_pvt_callid_get (closes issue ASTERISK-20145) Reported by: Birger "WIMPy" Harzenetter 2012-07-20 19:08 +0000 [r370329] Matthew Jordan * apps/app_dial.c: Clean up ManagerEvent Dial documentation The paragraph describing the SubEvent belongs with the SubEvent parameter itself, and not with its enum values. The order of parsing was placing the description after the last enum, which isn't correct. 2012-07-20 18:37 +0000 [r370328] Kinsey Moore * channels/chan_misdn.c: Fix build error in chan_misdn from commit 370316 chan_misdn was not updated properly to account for a change in parameters for HANGUPCAUSE functionality. It now builds properly. 2012-07-20 16:25 +0000 [r370322] Joshua Colp * res/res_http_websocket.exports.in: Export the ast_websocket_set_nonblock function for use by other modules. 2012-07-20 15:48 +0000 [r370316] Kinsey Moore * funcs/func_hangupcause.c (added), main/channel.c, channels/chan_dahdi.c, channels/sig_analog.c, main/rtp_engine.c, channels/chan_sip.c, main/channel_internal_api.c, UPGRADE.txt, include/asterisk/channel.h, channels/chan_iax2.c, channels/sig_pri.c, include/asterisk/frame.h, channels/sig_ss7.c: Add hangupcause translation support The HANGUPCAUSE hash (trunk only) meant to replace SIP_CAUSE has now been replaced with the HANGUPCAUSE and HANGUPCAUSE_KEYS dialplan functions to better facilitate access to the AST_CAUSE translations for technology-specific cause codes. The HangupCauseClear application has also been added to remove this data from the channel. (closes issue SWP-4738) Review: https://reviewboard.asterisk.org/r/2025/ 2012-07-20 15:40 +0000 [r370309-370315] Richard Mudgett * CHANGES: Update CHANGES about adding the AccountCode header to the AMI Hangup event. (issue ASTERISK-19963) * main/channel.c: Add the AccountCode header to the AMI Hangup event. It's harder to correlate the Newchannel and Hangup AMI events without specifying "AccountCode" in both. (closes issue ASTERISK-19963) Reported by: Oleg A. Arkhangelsky Patches: hangup_acctcode.diff (license #6397) patch uploaded by Oleg A. Arkhangelsky 2012-07-19 23:21 +0000 [r370303] Terry Wilson * include/asterisk/config_options.h, apps/confbridge/include/confbridge.h, main/config_options.c, apps/confbridge/conf_config_parser.c: Convert app_confbridge to use the config options framework Review: https://reviewboard.asterisk.org/r/2024/ 2012-07-19 22:25 +0000 [r370298] Richard Mudgett * /, main/cel.c: Fix compiler warnings. gcc (GCC) 4.2.4 has problems casting away constness. ........ Merged revisions 370275 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370277 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-07-19 22:17 +0000 [r370272-370278] Matthew Jordan * channels/chan_sip.c, res/res_xmpp.c, doc/appdocsxml.dtd, main/message.c, main/xmldoc.c: Add the ability to specify technology specific documentation A number of applications/AMI commands in Asterisk have specific behavioral differences depending on the resource or channel technology those applications are executed on. For example, the MessageSend application/ command is technology agnostic, but how the channel drivers that support that functionality behave is dependant on the protocols and channel driver implementation. Prior to this patch, those details were either documented in the application/command documentation itself, or were left undocumented. This patch adds a new element to the documentation schema, . An info node is essentially a piece of technology specific reference information that can be included by any top level XML documentation node. For example, the MessageSend application can now include XMPP/SIP specific information, where that technology specific information can be defined in chan_motif/res_xmpp/ chan_sip. Likewise, that information can also be included in the MessageSend AMI command. Review: https://reviewboard.asterisk.org/r/2049 * /, main/cel.c: Fix compilation error when MALLOC_DEBUG is enabled To fix a memory leak in CEL, a channel datastore was introduced whose destruction function pointer was pointed to the ast_free macro. Without MALLOC_DEBUG enabled this compiles as fine, as ast_free is defined as free. With MALLOC_DEBUG enabled, however, ast_free takes on a definition from a different place then utils.h, and became undefined. This patch resolves this by using a reference to ast_free_ptr. When MALLOC_DEBUG is enabled, this calls ast_free; when MALLOC_DEBUG is not enabled, this is defined to be ast_free, which is defined to be free. (issue AST-916) Reported by: Thomas Arimont ........ Merged revisions 370273 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370274 from http://svn.asterisk.org/svn/asterisk/branches/10 * res/res_rtp_asterisk.c, /: Handle extremely out of order RFC 2833 DTMF The current implementation of RFC 2833 DTMF handling in res_rtp_asterisk will, if a packet arrives out of order, drop the packet. This is to prevent duplicate ton generation in the Asterisk core. Since the RTP layer does not buffer data itself, this is the only option the RTP layer currently has for handling packets that arrive out of order. For the most part, this doesn't matter. For a particular digit, so long as a BEGIN packet arrives before the first END packet, the digit will be produced. If subsequent BEGIN packets arrive interleaved with the ENDs, they will be dropped; likewise, if the BEGIN or END packets themselves are out of order, those packets are dropped but sufficient information is conveyed to the Asterisk core to produce the appropriate digit. For certain sequences of DTMF packets - most notably when, for a particular digit, an END packet arrives before any BEGIN packet for that digit - this is a real problem. When an END arrives before any BEGINs, the END packet is dropped - but at the same time, it causes subsequent BEGIN packets for that digit to be ignored. When the next in order END packet arrives, it too is dropped - Asterisk believes that there was no initial BEGIN. The solution this patch provides is to trust the END packet to convey the information needed for the Asterisk core to produce the DTMF digit. If we receive an END packet, and it: * Has a timestamp greater then the last timestamp received from an END packet * Does not have the same sequence number as the last received sequence number (and is thus not an END packet retransmission) Then we send the END frame up to the Asterisk core. It contains enough DTMF information for Asterisk to produce the digit. On the other hand, if we receive a BEGIN or continuation packet that occurs with a timestamp equal to or less then the last END timestamp, then we've received something out of order - but we already have received enough information to produce the digit. These packets are dropped. Much thanks goes to Olle Johansson (oej) for providing the idea for this solution. Review: https://reviewboard.asterisk.org/r/2033/ (closes issue ASTERISK-18404) Reported by: Stephane Chazelas Tested by: Matt Jordan ........ Merged revisions 370252 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370271 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-07-19 20:37 +0000 [r370246-370265] Jonathan Rose * main/named_acl.c, configs/acl.conf.sample: named_acl: Remove systemname option from acl.conf, use asterisk.conf value Review: https://reviewboard.asterisk.org/r/2057/ * main/channel_internal_api.c: CallID Logging: Remove new line/carriage return from callID change test event 2012-07-19 12:14 +0000 [r370234-370240] Joshua Colp * res/Makefile, res/pjproject/build/os-auto.mak.in: Use the bruteforce method to get debugging enabled for pjproject. * res/Makefile: Turn on debugging for pjproject so we can get a better idea of what is causing the generic CCSS test crash. 2012-07-18 19:48 +0000 [r370225] Jonathan Rose * main/channel_internal_api.c: callid logging: Issue test events when the callid is changed for a channel Review: https://reviewboard.asterisk.org/r/2054/ 2012-07-18 19:18 +0000 [r370187-370211] Kevin P. Fleming * /, main/cel.c: Resolve severe memory leak in CEL logging modules. A customer reported a significant memory leak using Asterisk 1.8. They have tracked it down to ast_cel_fabricate_channel_from_event() in main/cel.c, which is called by both in-tree CEL logging modules (cel_custom.c and cel_sqlite3_custom.c) for each and every CEL event that they log. The cause was an incorrect assumption about how data attached to an ast_channel would be handled when the channel is destroyed; the data is now stored in a datastore attached to the channel, which is destroyed along with the channel at the proper time. (closes issue AST-916) Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2053/ ........ Merged revisions 370205 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370206 from http://svn.asterisk.org/svn/asterisk/branches/10 * main/channel.c, addons/app_mysql.c, main/pbx.c, funcs/func_curl.c, /, main/ccss.c, funcs/func_odbc.c, funcs/func_lock.c, apps/app_macro.c, channels/chan_iax2.c, apps/app_mixmonitor.c, apps/app_stack.c, funcs/func_global.c, res/res_odbc.c: Ensure that all ast_datastore_info structures are 'const'. While addressing a bug, I came across a instance of 'struct ast_datastore_info' that was not declared 'const'. Since the API already expects them to be 'const', this patch changes the declarations of all existing instances that were not already declared that way. ........ Merged revisions 370183 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370184 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-07-18 15:15 +0000 [r370171-370177] Joshua Colp * res/res_rtp_asterisk.c: Fix a crash in pjnath when starting an ICE connectivity check and immediately destroying the ICE session. The initial ICE connectivity check is scheduled as a timer item that is to be executed immediately. It is possible for this timer item to start executing while the ICE session it is working on is destroyed. To reduce the chance of this any timer items that need to be immediately executed will be executed within the thread that has started the initial ICE connectivity check. * channels/chan_sip.c, include/asterisk/rtp_engine.h: Fix a crash occurring as a result of excess stack usage. This fix involves moving the allocation of some temporary codec structures to the heap and also reduces the number of maximum payloads to something more sane for both regular and low memory builds. (closes issue ASTERISK-20140) Reported by: jonnt 2012-07-18 07:17 +0000 [r370165] Igor Goncharovskiy * channels/chan_unistim.c, configs/unistim.conf.sample, CHANGES: Added option 'interdigit_timer' to unistim.conf to make able controll hardcoded dial timeout constant. 2012-07-17 19:05 +0000 [r370152-370157] Joshua Colp * res/res_xmpp.c: Add pubsub unsubscription support so subscriptions do not linger for MWI and device state progatation. The pubsub code did not attempt to remove subscriptions at all. This has now changed so that if a client is being disconnected it will unsubscribe. It will also unsubscribe at connection time so if it unexpectedly disconnected duplicate subscriptions will not occur. (closes issue ASTERISK-19882) Reported by: mattvryan * include/asterisk/xmpp.h, res/res_xmpp.c: Fix a crash as a result of propagating MWI or device state over XMPP when the client is disconnected. The MWI and device state propagation code wrongly assumes that an XMPP client connection will remain established at all times. This fix corrects that by making the lifetime of the subscription the same as the lifetime of the connection itself. As the connection is established and disconnected the subscription itself is created and destroyed. (closes issue ASTERISK-18078) Reported by: elguero 2012-07-16 19:58 +0000 [r370133] Walter Doekes * /, channels/chan_sip.c: Code cleanup and bugfix in chan_sip outboundproxy parsing. The bug was clearing the global outboundproxy when a peer-specific outboundproxy was bad. The cleanup reduces duplicate code. Review: https://reviewboard.asterisk.org/r/2034/ Reviewed by: Mark Michelson ........ Merged revisions 370131 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370132 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-07-16 19:14 +0000 [r370111-370126] Joshua Colp * res/res_xmpp.c: Fix an issue where a service discovery request could crash Asterisk. A server sending a service discovery request to us may or may not put a from attribute in the message. If the from attribute is present use it in the to attribute for the result. If the from attribute is not present do not add a to attribute. (issue ASTERISK-16203) Reported by: wubbla * res/res_xmpp.c: Fix a bug where some XMPP servers would reject authentication. We need to use the user portion of the JID and not the full configured username. * res/res_xmpp.c: Add missing namespace for old non-SASL based authentication. * channels/chan_sip.c: Fix a bug exposed by the testsuite where text streams would no longer be parsed correctly. 2012-07-16 14:02 +0000 [r370083] Kinsey Moore * /, UPGRADE-10.txt, CHANGES, UPGRADE-1.8.txt: Add comments about the BUILD_NATIVE change This is a significant change and mention of it should have gone into UPGRADE.txt and CHANGES. ........ Merged revisions 370081 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370082 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-07-16 12:58 +0000 [r370072-370073] Joshua Colp * res/res_xmpp.c: Fix an issue where specifying the resource in the username would cause authentication to fail. * channels/sip/sdp_crypto.c, channels/chan_sip.c, channels/sip/security_events.c, include/asterisk/http_websocket.h, configs/sip.conf.sample, CHANGES, res/res_http_websocket.c, channels/sip/include/sip.h: Add support for SIP over WebSocket. This allows SIP traffic to be exchanged over a WebSocket connection which is useful for rtcweb. Review: https://reviewboard.asterisk.org/r/2008 2012-07-16 07:38 +0000 [r370066-370067] Igor Goncharovskiy * channels/chan_unistim.c: Deactivate timer for dialing entered number on hook switch hang up. (closes issue ASTERISK-19554) Reported by: Stefano Villani * channels/chan_unistim.c, contrib/unistimLang/fr.po (added), CHANGES: Add French translation for chan_unistim phones on-screen menus. 2012-07-13 18:41 +0000 [r370055-370060] Joshua Colp * include/asterisk/format.h, res/res_format_attr_h263.c (added), res/res_format_attr_h264.c (added): Reduce memory consumption and add the H.264 and H.263 modules I shamefully neglected to add. * main/format.c, channels/chan_sip.c, main/translate.c, include/asterisk/format.h, res/res_format_attr_silk.c, res/res_format_attr_celt.c: Add support for parsing SDP attributes, generating SDP attributes, and passing it through. This support includes codecs such as H.263, H.264, SILK, and CELT. You are able to set up a call and have attribute information pass. This should help considerably with video calls. Review: https://reviewboard.asterisk.org/r/2005/ 2012-07-13 00:05 +0000 [r370048] Tzafrir Cohen * contrib/scripts/live_ast: live_ast: don't set working directory contrib/scripts/live_ast currently assumes that it is being run from the top-level directory of the source tree. It creates a script that will also set the working directory. This fix avoids the need to set the working directory if the caller sets LIVE_AST_BASE_DIR instead. It relies on realpath for that. If realpath is not available, it will fall back to the original behaviour. Review: https://reviewboard.asterisk.org/r/2027/ 2012-07-12 21:43 +0000 [r370043] Terry Wilson * include/asterisk/config_options.h, configs/config_test.conf.sample, main/config_options.c, tests/test_config.c: Handle deprecated (aliased) option names with the config options api Add a simple way to register "deprecated" option names that alias to a different "current" name. Review: https://reviewboard.asterisk.org/r/2026/ 2012-07-12 20:28 +0000 [r370037] Richard Mudgett * channels/chan_dahdi.c, channels/sig_analog.c, /: Add missing ast_hangup() calls on some analog exception paths. Make starting analog_ss_thread() or __analog_ss_thread() failure paths hangup the channel. ........ Merged revisions 370017 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370025 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-07-12 20:06 +0000 [r369995-370016] Kinsey Moore * /, channels/chan_sip.c: Include Expires header for SIP PUBLISH requests RFC3903 requres SIP PUBLISH requests to have Expires headers, so add them. Review: https://reviewboard.asterisk.org/r/2003/ Patch-by: gareth ........ Merged revisions 370014 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370015 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, channels/chan_sip.c: Prevent double uri_escaping in chan_sip when pedantic is enabled If pedantic mode is enabled, outbound invites will have double-escaped contacts. This avoids setting an already-escaped string into a field where it is expected to be unescaped. (closes issue ASTERISK-20023) Reported by: Walter Doekes ........ Merged revisions 369993 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369994 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-07-12 14:38 +0000 [r369972-369974] Michael L. Young * /, funcs/func_math.c: Correct Documentation For DEC Function The documentation for DEC in func_math.c was incorrect. Looks like a copy and paste error. (Closes issue ASTERISK-20095) Reported by: Billy Chia Tested by: Michael L. Young Patches: func_math.patch uploaded by Billy Chia (license 6381) ........ Merged revisions 369970 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369971 from http://svn.asterisk.org/svn/asterisk/branches/10 * funcs/func_math.c: Reverting last merge since it wasn't completed properly. * funcs/func_math.c: Correct Documentation For DEC Function The documentation for DEC in func_math.c was incorrect. Looks like a copy and paste error. (Closes issue ASTERISK-20095) Reported by: Billy Chia Tested by: Michael L. Young Patches: func_math.patch uploaded by Billy Chia (license 6381) ........ Merged revisions 369970 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-07-11 18:33 +0000 [r369959] Jonathan Rose * include/asterisk/acl.h, channels/chan_sip.c, include/asterisk/config.h, main/acl.c, include/asterisk/channel.h, configs/manager.conf.sample, channels/chan_iax2.c, CHANGES, main/named_acl.c (added), main/config.c, main/loader.c, configs/iax.conf.sample, main/manager.c, include/asterisk/event_defs.h, configs/extconfig.conf.sample, configs/sip.conf.sample, channels/sip/include/sip.h, main/asterisk.c, configs/acl.conf.sample (added): Named ACLs: Introduces a system for creating and sharing ACLs This patch adds Named ACL functionality to Asterisk. This allows system administrators to define an ACL and refer to it by a unique name. Configurable items can then refer to that name when specifying access control lists. It also includes updates to all core supported consumers of ACLs. That includes manager, chan_sip, and chan_iax2. This feature is based on the deluxepine-trunk by Olle E. Johansson and provides a subset of the Named ACL functionality implemented in that branch. For more information on this feature, see acl.conf and/or the Asterisk wiki. Review: https://reviewboard.asterisk.org/r/1978/ 2012-07-11 17:16 +0000 [r369940] Tilghman Lesher * /, main/ast_expr2.h, main/ast_expr2f.c, res/ael/ael_lex.c, funcs/func_realtime.c, main/ast_expr2.c: Allow the REALTIME() function to report errors back to the caller. Also, do more error checking on the arguments specified to the REALTIME() function and clarify the documentation. While I was editing the file, a few coding guidelines fixups, as well. Review: https://reviewboard.asterisk.org/r/2031/ ........ Merged revisions 369937 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369938 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-07-11 17:14 +0000 [r369939] Matthew Jordan * main/features.c: Don't perform an XInclude to a document node that may not always be present Because some of the manager events are defined in the top of the source, due to the macro calls not containing all necessary information to have the documentation colocated with the call itself, several include statements were failing when built with 'make'. While this did not cause any problems in compilation or validation, it did result in a number of warnings being dumped to stderr. This patch changes those references such that they always resolve, regardless of the documentation build options. 2012-07-11 16:42 +0000 [r369936] Joshua Colp * channels/chan_motif.c: Do not consider failure to read the configuration file in chan_motif to be a show stopper for loading Asterisk by returning decline instead of failure. (closes issue ASTERISK-20103) Reported by: Terry Wilson 2012-07-11 02:06 +0000 [r369905-369910] Matthew Jordan * main/cdr.c, main/channel.c, channels/sig_analog.c, main/logger.c, channels/sig_pri.c, main/asterisk.c, main/loader.c: Fix validation errors when producing documentation using default build script The awk script parses out the first instance of the DOCUMENTATION tag that it finds within a file. If a file did not previously have a DOCUMENTATION tag but received one due to it having an AMI event, then the XML fragment associated with the AMI event was erroneously placed in the resulting XML file. Without the python scripts, these XML fragments will not validate. This patch adds DOCUMENTATION tags at the top of those files that did not previously have them to prevent the awk script from pulling AMI event documentation. * main/cdr.c, main/channel.c, channels/chan_dahdi.c, main/pbx.c, channels/chan_local.c, channels/sig_analog.c, main/manager.c, channels/chan_agent.c, main/features.c, main/logger.c, channels/sig_pri.c, doc/appdocsxml.dtd, main/asterisk.c, main/loader.c: Add some additional documentation for core AMI events This patch adds some basic documentation for a number of modules. This includes core source files in Asterisk (those in main), as well as chan_agent, chan_dahdi, chan_local, sig_analog, and sig_pri. The DTD has also been updated to allow referencing of AMI commands. 2012-07-10 15:36 +0000 [r369900] Kinsey Moore * channels/chan_sip.c: Fix failing SDP_offer_answer test Asterisk now generates image stream declinations with the same transport case that it used to before the stream declination improvements. (udptl vs UDPTL) (closes issue SWP-4736) 2012-07-10 15:25 +0000 [r369873-369898] Joshua Colp * channels/chan_motif.c: Add additional description stanza names from the old Google Talk protocol which is used with Google Voice. (closes issue ASTERISK-20114) Reported by: Malcolm Davenport * channels/chan_motif.c: Respect codec preference order when adding codecs to a media description. This change allows an endpoint in motif.conf to be configured with a preference of G.722 and fallback of ulaw. With Google this allows communication with Google Talk clients to use G.722 while when using Google Voice ulaw will be used. (closes issue ASTERISK-20114) Reported by: Malcolm Davenport 2012-07-10 13:40 +0000 [r369872] Kinsey Moore * main/pbx.c, /, apps/app_stack.c: Improve Goto and GotoIf related documentation Correct documentation on labeliftrue and labeliffalse parameters of GotoIf() and update several other locations that use the same syntax. (closes issue ASTERISK-20007) Patch-by: Leif Madsen Reported-by: WIMPy ........ Merged revisions 369869 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369871 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-07-10 13:34 +0000 [r369870] Matthew Jordan * main/libasteriskssl.c: Fix initial loading problem with res_curl When the OpenSSL duplicate initialization issues were resolved in r351447, res_curl could fail to load if it checked SSL_library_init after SSL initialization completed. This is due to the SSL_library_init stub returning a value of 0 for success, as opposed to a value of 1. OpenSSL uses a value of 1 to indicate success - in fact, SSL_library_init is documented to always return 1. Interestingly, the CURL libraries actually checked the return value - the fact that nothing else that depends on OpenSSL was having problems loading probably means they don't check the return value. (closes issue AST-924) Reported by: Guenther Kelleter patches: (AST-924.patch license #6372 uploaded by Guenther Kelleter) 2012-07-10 11:49 +0000 [r369837-369864] Joshua Colp * res/res_rtp_asterisk.c, channels/chan_motif.c: Add required items for Google video support. This adds legacy STUN support for RTCP sockets, adds RTCP candidates to the Google transport information, and adds required codec parameters. (closes issue ASTERISK-20106) Reported by: Malcolm Davenport * main/stun.c: When receiving a STUN binding request send one out as the Google Talk client uses this as a method to determine if the remote party is still reachable or not. Failure to do this results in the Google Talk client ignoring RTP packets after a specific period of time. This is also done as a result of receiving a STUN binding request so that the username information can be used from the inbound request, thus not requiring it to be stored on a per candidate basis. (closes issue ASTERISK-20107) Reported by: Malcolm Davenport * channels/chan_sip.c: Add support for exposing the received contact URI and also for setting the request URI in messages. (closes issue AST-911) * channels/chan_motif.c: Force the clock rate of G.722 to be 16000 when using the Google transports as it is 8000 elsewhere. (closes issue ASTERISK-20105) Reported by: Malcolm Davenport * configs/motif.conf.sample: Document that multiple endpoints using the same connection is not supported. (closes issue ASTERISK-20104) Reported by: Malcolm Davenport 2012-07-09 17:07 +0000 [r369820] Jason Parker * configs/sip_notify.conf.sample, /: Add Digium phones context to sip_notify sample config. This makes it so that they can be reconfigured remotely. (closes issue ASTERISK-19910) ........ Merged revisions 369818 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369819 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-07-09 16:44 +0000 [r369811-369817] Joshua Colp * res/res_rtp_asterisk.c: Fix an issue where media would not flow for situations where the legacy STUN code is in use. The STUN packets should *not* be blocked by strict RTP. (closes issue ASTERISK-20102) Reported by: Malcolm Davenport * res/res_xmpp.c: Add additional namespaces for Google Talk which are used for the gmail client. (closes issue ASTERISK-20101) Reported by: Malcolm Davenport * channels/chan_motif.c: Fix dependency to be on res_xmpp. Long ago in a galaxy far far away it used to use res_jabber. 2012-07-09 14:54 +0000 [r369794] Jonathan Rose * /, channels/chan_sip.c: chan_sip: Fix small behavioral change accidentally introduced in r369750 When removing the warning for AST_CONTROL_FLASH from sip_indicate, I also inadvertently changed the return value, which would likely make the indication not be sent in audio. This fixes that while still removing the warning message. ........ Merged revisions 369792 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369793 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-07-07 17:06 +0000 [r369769] Joshua Colp * res/res_xmpp.exports.in (added), include/asterisk/xmpp.h, channels/chan_motif.c (added), UPGRADE.txt, channels/chan_gtalk.c, res/res_xmpp.c, CHANGES, res/res_jabber.c, configs/motif.conf.sample (added): Add a new unified Jingle, Google Jingle, and Google Talk channel driver written from scratch called chan_motif. This channel driver is a replacement for both chan_gtalk and chan_jingle but adds additional features not found in either. These features include full configuration reload, video, full codec support, bidirectional cause code mapping, hold, unhold, and ringing indication. It is also compliant with the current published Jingle and Google Jingle specifications. The original Google Talk protocol is also supported for Google Voice interoperability. You may ask yourself though where the name motif comes from... and I would say to you... music! motif: a perceivable or salient recurring fragment or succession of notes Sorta like a jingle! Review: https://reviewboard.asterisk.org/r/1917/ 2012-07-06 22:03 +0000 [r369765] Kinsey Moore * channels/chan_dahdi.c, channels/sig_analog.c, channels/chan_iax2.c, channels/sig_pri.c, channels/sig_ss7.c: Remove unnecessary generation of informational cause frames It is not necessary to generate information cause code frames on every protocol event that occurs. This removes all the instances where the frame was not conveying a cause code and was instead just conveying a protocol-specific message. This also corrects the generation of the message associated with disconnects for MFC/R2 to use the MFC/R2 specific text for the disconnect cause. 2012-07-06 21:28 +0000 [r369764] Jonathan Rose * /, channels/chan_sip.c: chan_sip: Add case for FLASH control frames so that we don't display a warning. chan_sip channels can receive flash control frames when connected to analog phones and possibly for other reasons. There really isn't a reason to warn when these frames are received, we can safely ignore them. Patches: dahdi_sip_flash.diff uploaded by Jonathan Rose (license 6182) ........ Merged revisions 369750 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369751 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-07-06 18:49 +0000 [r369710-369733] Mark Michelson * main/tcptls.c, /: Remove a superfluous and dangerous freeing of an SSL_CTX. The problem here is that multiple server sessions share a SSL_CTX. When one session ended, the SSL_CTX would be freed and set NULL, leaving the other sessions unable to function. The code being removed is superfluous because the SSL_CTX structures for servers will be properly freed when ast_ssl_teardown is called. (closes issue ASTERISK-20074) Reported by Trevor Helmsley Patches: ASTERISK-20074.diff uploaded by Mark Michelson (license #5049) Testers: Trevor Helmsley ........ Merged revisions 369731 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369732 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, main/bridging.c: Fix bridging thread leak. The bridge thread was exiting but was never being reaped using pthread_join(). This has been fixed now by calling pthread_join() in ast_bridge_destroy(). (closes issue ASTERISK-19834) Reported by Marcus Hunger Review: https://reviewboard.asterisk.org/r/2012 ........ Merged revisions 369708 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369709 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-07-06 14:32 +0000 [r369703] Joshua Colp * res/pjproject/pjnath/include/pjnath/ice_session.h, res/pjproject/pjnath/src/pjnath/ice_session.c: Import revision 4196 from pjproject trunk. Fix a crash issue when starting ICE connectivity checks and immediately destroying the ICE session. This was exposed by the SIP CCSS test. Full fix for this issue will be worked on as a medium to long term roadmap item. pjroject issue viewable at https://trac.pjsip.org/repos/ticket/1548 2012-07-05 21:36 +0000 [r369681] Matthew Jordan * res/res_stun_monitor.c, CHANGES: Add 'stun show status' command This patch adds a new CLI command, 'stun show status'. This command will show a table describing all known STUN servers and statuses. (closes issue ASTERISK-18046) Reported by: Jeremy Kister Tested by: Jeremy Kister patches: (stun-show-status-v4-trunk.patch license #6232 uploaded by Jeremy Kister) Review: https://reviewboard.asterisk.org/r/2001 2012-07-05 19:36 +0000 [r369677] Richard Mudgett * res/pjproject/pjmedia/include/pjmedia, res/pjproject/pjsip/include/pjsip, res/pjproject/pjlib/include/pj/compat, res/pjproject/pjmedia/include/pjmedia-codec: Make res/pjproject ignore more files. 2012-07-05 19:36 +0000 [r369676] Kinsey Moore * /, apps/app_voicemail.c: AST-2012-011: Resolve heap corruption issue with voicemail The heard and deleted arrays in the voicemail state structure were not handled properly following the memory leak fix in r354890 and a fix for an invalid free in r356797. This could result in accessing and writing into freed memory. The allocation for these arrays has been reworked to avoid the possibility of invalid frees, access of freed memory, and crashes that were occurring as a result of this. Locking around accesses and modifications of the voicemail state structure members dh_arraysize, heard, and deleted has been added to prevent simultaneous modification and access when IMAP storage is in use. If IMAP storage is not in use, this locking is not compiled in. Review: https://reviewboard.asterisk.org/r/1994/ (closes issue ASTERISK-19923) Reported by: Dan Delaney Tested by: Dan Delaney, Julian Yap Patches: vm_alloc_fix.diff uploaded by kmoore (license 6273) ........ Merged revisions 369652 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369653 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-07-05 19:32 +0000 [r369666-369673] Richard Mudgett * res/pjproject/pjsip/src/pjsip-ua, res/pjproject/pjsip-apps/src/ipjsystest/ipjsystest.xcodeproj, res/pjproject/pjnath/src/pjnath-test, res/pjproject/third_party/build/speex, res/pjproject/third_party/build/gsm/output, res/pjproject/pjmedia/include/pjmedia-codec, res/pjproject/third_party/build/baseclasses, res/pjproject/third_party/build/srtp, res/pjproject/pjsip-apps/src/samples, res/pjproject/pjlib-util/lib, res/pjproject/pjmedia/bin, res/pjproject/pjlib/include/pj++, res/pjproject/tests/pjsua/scripts-call, res/pjproject/third_party/srtp/doc, res/pjproject/pjsip-apps/src/pocketpj/output, res/pjproject/pjnath/bin, res/pjproject/third_party/srtp/crypto/replay, res/pjproject/pjsip/include/pjsip, res/pjproject/third_party/build/speex/speex, res/pjproject/build.symbian, res/pjproject/third_party/bin, res/pjproject/pjsip/src/pjsua-lib, res/pjproject/third_party/srtp/include, res/pjproject/third_party/portaudio/doc, res/pjproject/lib, res/pjproject/pjmedia/include/pjmedia-videodev, res/pjproject/pjlib/bin, res/pjproject/third_party/srtp/crypto/cipher, res/pjproject/third_party/build/speex/output, res/pjproject/pjlib-util/src/pjlib-util, res/pjproject/third_party/portaudio/test, res/pjproject/third_party/build/gsm, res/pjproject/third_party/portaudio/include, res/pjproject/pjsip-apps/src/pjsua_wince, res/pjproject/pjsip/include/pjsip-simple, res/pjproject/pjmedia/src/pjmedia-codec, res/pjproject/tests/pjsua, res/pjproject/pjsip-apps/src/pocketpj/res, res/pjproject/pjsip-apps/src/3rdparty_media_sample, res/pjproject/third_party/gsm/inc, res/pjproject/pjsip-apps/build/wince-evc4, res/pjproject/pjsip-apps/src/ipjsua/Resources-iPad, res/pjproject/third_party/portaudio/src/hostapi, res/pjproject/third_party/portaudio/build, res/pjproject/build, res/pjproject/third_party/build/resample, res/pjproject/third_party/speex/include, res/pjproject/pjsip/src/pjsip, res/pjproject/pjlib/build/wince-evc4, res/pjproject/pjsip-apps/src/symbian_ua_gui/group, res/pjproject/pjsip-apps/src/symbian_ua, res/pjproject/tests/pjsua/wavs, res/pjproject/third_party/portaudio/src/os/win, res/pjproject/pjsip-apps/src/ipjsua/Classes, res/pjproject/pjmedia/include/pjmedia, res/pjproject/tests/pjsua/scripts-sendto, res/pjproject/third_party/gsm/src, res/pjproject/third_party/portaudio/build/msvc, res/pjproject/pjsip-apps/src/confbot, res/pjproject/pjnath/src/pjturn-client, res/pjproject/pjlib-util/build/output, res/pjproject/third_party/BaseClasses, res/pjproject/third_party/portaudio/src/hostapi/wasapi, res/pjproject/third_party/portaudio/src/hostapi/wdmks, res/pjproject/pjlib/src/pj/compat, res/pjproject/third_party/srtp/crypto/include, res/pjproject/third_party/speex/include/speex, res/pjproject/third_party/gsm/add-test, res/pjproject/pjsip/build, 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res/pjproject/third_party/portaudio/testcvs, res/pjproject/pjsip-apps/src/ipjsystest/Classes, res/pjproject/pjlib/build, res/pjproject/third_party/portaudio, res/pjproject/third_party/portaudio/src/hostapi/wmme, res/pjproject/pjlib-util/docs, res/pjproject/pjmedia/include/pjmedia-audiodev, res/pjproject/pjsip-apps/src/vidgui, res/pjproject/pjlib/src/pjlib-test, res/pjproject/pjsip-apps/src/py_pjsua, res/pjproject/third_party/portaudio/src/os, res/pjproject/pjsip/include, res/pjproject/pjmedia/build/wince-evc4, res/pjproject/pjmedia/src/pjmedia-videodev, res/pjproject/pjsip-apps/src, res/pjproject/third_party/speex, res/pjproject/third_party/gsm/tls, res/pjproject/third_party/g7221/common, res/pjproject/tests/pjsua/tools, res/pjproject/third_party/resample/include, res/pjproject/third_party/build/samplerate/output, res/pjproject/third_party/build/samplerate, res/pjproject/third_party/gsm/bin, res/pjproject/pjsip/src/pjsip-simple, res/pjproject/third_party/g7221/encode, res/pjproject/pjlib/src/pjlib-samples, res/pjproject/pjsip-apps/lib, res/pjproject/pjsip-apps/src/ipjsystest, res/pjproject/pjlib-util/include, res/pjproject/third_party/build/resample/output, res/pjproject/third_party/build/ilbc/output, res/pjproject/third_party/srtp/crypto, res/pjproject/pjsip-apps/src/python/samples, res/pjproject/tests, res/pjproject/pjsip-apps/src/symbian_ua_gui/sis, res/pjproject/pjnath/include, res/pjproject/pjsip-apps/src/symbian_ua_gui, res/pjproject/pjmedia/build, res/pjproject/pjmedia, res/pjproject/third_party/build/milenage/output, res/pjproject/pjlib-util/build, res/pjproject/pjsip/src, res/pjproject/pjmedia/build/wince-evc4/output, res/pjproject/third_party/portaudio/src/hostapi/alsa, res/pjproject/pjsip-apps/docs, res/pjproject/pjsip-apps/src/symbian_ua_gui/inc, res/pjproject/pjsip-apps/src/symbian_ua_gui/data, res/pjproject/tests/pjsua/scripts-pesq, res/pjproject/third_party/srtp/pjlib, res/pjproject/pjlib/include, res/pjproject/pjnath/build/output, res/pjproject/third_party/srtp/crypto/hash, res/pjproject/build/vs, res/pjproject/pjlib/docs, res/pjproject/third_party/build, res/pjproject/third_party/resample/src, res/pjproject/third_party, res/pjproject/pjlib/src/pjlib++-test, res/pjproject/third_party/build/g7221/output, res/pjproject/third_party/srtp/crypto/math, res/pjproject/pjsip/lib, res/pjproject/pjsip-apps/src/pocketpj, res/pjproject/tests/pjsua/scripts-recvfrom, res/pjproject/third_party/portaudio/build/dev-cpp, res/pjproject/pjsip/include/pjsua-lib, res/pjproject/pjsip-apps/src/symbian_ua_gui/src, res/pjproject, res/pjproject/third_party/portaudio/src/hostapi/oss, res/pjproject/pjlib-util/src, res/pjproject/third_party/ilbc: Make res/pjproject ignore some generated files. * include/asterisk/utils.h: Tweak some comments and whitespace in utils.h 2012-07-05 18:11 +0000 [r369644] Jonathan Rose * apps/app_mixmonitor.c: app_mixmonitor: Fix a reference leak in manager_mixmonitor function Manager_mixmonitor included an early return on failed executions of mixmonitor that would result in a leaked channel reference. (closes issue ASTERISK-19943) Reported by: Mark Murawski Patches: mixmonitor-trunk-368394.patch uploaded by Mark Murawski (license 5791) 2012-07-05 17:03 +0000 [r369628] Matthew Jordan * /, channels/chan_sip.c: Do not send a BYE when a provisional response arrives during a re-INVITE Commits r369557 and r369579 were done to improve handling of re-INVITEs when the UA that was supposed to receive the re-INVITE fails to respond. A limitation of those patches occurred when a UA sent a provisional response to the re-INVITE. This triggered a sending of a BYE in check_pending. This patch tweaks the handling of the re-INVITE such that a BYE is not sent in response to those messages. (issue ASTERISK-19992) Reported by: Steve Davies Tested by: Steve Davies patches: (reinvite_tweak.diff license #5012 by Steve Davies) ........ Merged revisions 369626 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369627 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-07-05 11:42 +0000 [r369602-369620] Alexandr Anikin * addons/ooh323c/src/ooCmdChannel.c, addons/ooh323c/src/ooStackCmds.c, addons/ooh323c/src/ooq931.c: Fix dev mode ooh323 warnings * addons/chan_ooh323.c, addons/ooh323c/src/ooq931.h, addons/ooh323c/src/ooCalls.h, configs/chan_ooh323.conf.sample (removed), addons/ooh323c/src/ooh323ep.c, CHANGES, configs/ooh323.conf.sample (added), addons/ooh323c/src/ooLogChan.c, addons/ooh323c/src/ooStackCmds.c, addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooLogChan.h, addons/ooh323c/src/ooStackCmds.h, addons/ooh323c/src/ooh245.c, addons/ooh323cDriver.c, addons/ooh323c/src/ooh245.h, addons/ooh323c/src/ooCmdChannel.c, addons/ooh323c/src/ooq931.c: Added direct media support to ooh323 channel driver options are documented in config sample sample config rename to proper name - ooh323.conf To change media address ooh323 send empty TCS if there was completed TCS exchange or send facility forwardedelements with new fast start proposal if not. Then close transmit logical channels and renew TCS exchange. If new fast start proposal is received then ooh323 stack call back channel driver routine to change rtp address in the rtp instance. If empty TCS is received then close transmit logical channels and renew TCS exchange Review: https://reviewboard.asterisk.org/r/1607/ * addons/ooh323cDriver.c: fix small mistake in the previous * addons/ooh323c/src/ooTimer.c, addons/ooh323c/src/ooCapability.c, addons/ooh323c/src/decode.c, addons/ooh323c/src/perutil.c, addons/ooh323cDriver.c, addons/ooh323c/src/ooSocket.c, addons/ooh323c/src/ooq931.c: Fix modern gcc warning Review: https://reviewboard.asterisk.org/r/1767 2012-07-03 17:07 +0000 [r369559-369581] Terry Wilson * /, channels/chan_sip.c: More improvements to re-INVITEs timing out after a provisional response There is no need to call check_pendings() on a final response to an INVITE when destroying the scheduler entry as it will be done later during normal processing. (issue ASTERISK-19992) ........ Merged revisions 369579 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369580 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, channels/chan_sip.c, channels/sip/include/sip.h: Better handle re-INVITEs with provisional but no final repsonses A previous attempt at fixing this issue had negative side effects related to attended transfers which this patch should resolve. Many thanks to Steve Davies for all of the good suggestions and testing. (closes issue ASTERISK-19992) Reported by: Steve Davies Tested by: Steve Davies, Terry Wilson Review: https://reviewboard.asterisk.org/r/2009/ ........ Merged revisions 369557 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369558 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-07-02 14:06 +0000 [r369517-369527] Joshua Colp * configs/xmpp.conf.sample (added), include/asterisk/xmpp.h (added), configs/cli_aliases.conf.sample, res/res_xmpp.c (added): Add a cleaned up drop-in replacement for res_jabber called res_xmpp. This provides the same externally facing functionality but is implemented differently internally. This is currently not built by default but this will be changed once chan_jingle2 (insert actual name in your head when reading this after it has been merged) is in the tree. Review: https://reviewboard.asterisk.org/r/1983/ * res/res_rtp_asterisk.c: Ensure the timer heap is protected by a lock. * res/pjproject/pjlib/include/pj/config_site.h: Enable IPv6 support in pjproject. * res/res_rtp_asterisk.c: Don't try to send connectivity checks on RTCP if RTCP is no longer present and don't do multiple ICE connectivity checks at once. * res/pjproject/pjlib/src/pj/sock_qos_common.c (added), res/pjproject/pjlib-util/src/pjlib-util/crc32.c (added), res/pjproject/pjsip/src/pjsip-simple/xpidf.c (added), res/pjproject/third_party/gsm/src/gsm_implode.c (added), res/pjproject/tests/pjsua/scripts-sipp/uas-cancel-no-final.xml (added), res/pjproject/build.symbian/pjmedia.mmp (added), res/pjproject/third_party/build/portaudio/src/pa_hostapi.h (added), res/pjproject/pjlib/src/pjlib-test/fifobuf.c (added), res/pjproject/pjlib/src/pj/file_access_unistd.c (added), res/pjproject/third_party/gsm/src/toast_ulaw.c (added), res/pjproject/pjsip/include/pjsip/sip_transport_tls.h (added), res/pjproject/pjsip/include/pjsip/sip_multipart.h (added), res/pjproject/pjmedia/src/pjmedia/errno.c (added), res/pjproject/pjsip-apps/src/pjsua_wince/pjsua_wince.vcp (added), res/pjproject/third_party/speex/COPYING (added), res/pjproject/pjlib/src/pj/os_core_darwin.m (added), res/pjproject/third_party/ilbc/packing.c (added), res/pjproject/third_party/build/portaudio/src/pa_mac_core_internal.h (added), res/pjproject/tests/pjsua/scripts-sendto/300_srtp_receive_crypto_tag_zero.py (added), res/pjproject/third_party/ilbc/packing.h (added), res/pjproject/pjlib/src/pj/pool_caching.c (added), res/pjproject/pjnath/include/pjnath/errno.h (added), res/pjproject/pjmedia/include/pjmedia-codec/h264_packetizer.h (added), res/pjproject/pjmedia/include/pjmedia/sdp_neg.h (added), res/pjproject/third_party/speex/libspeex/lsp_bfin.h (added), res/pjproject/third_party/portaudio/aclocal.m4 (added), res/pjproject/third_party/mp3/mp3_port.h (added), res/pjproject/third_party/BaseClasses/ctlutil.cpp (added), res/pjproject/pjsip-apps/src/pocketpj/PocketPJDlg.cpp (added), res/pjproject/tests/pjsua/scripts-recvfrom/240_publish_scenarios.py (added), res/pjproject/README-RTEMS (added), res/pjproject/third_party/build/portaudio/output (added), res/pjproject/pjsip-apps/build/Makefile (added), res/pjproject/tests/pjsua/scripts-sipp/prack_fork.xml (added), res/pjproject/pjlib-util/src/pjlib-util-test/stun.c (added), res/pjproject/pjlib-util/src/pjlib-util/dns_dump.c (added), res/pjproject/pjmedia/include/pjmedia/circbuf.h (added), res/pjproject/pjlib/build/os-darwinos.mak (added), res/pjproject/third_party/srtp/test/rtpw.c (added), res/pjproject/tests/pjsua/scripts-sipp/uas-answer-180-multiple-fmts.xml (added), res/pjproject/third_party/srtp/crypto/include/cryptoalg.h (added), res/pjproject/third_party/portaudio/bindings/cpp (added), res/pjproject/tests/pjsua/scripts-sipp/uas-answer-200-reinvite-without-sdp.xml (added), res/pjproject/third_party/portaudio/configure.in (added), res/pjproject/pjmedia/include/pjmedia-codec/g722.h (added), res/pjproject/pjsip-apps/src/vidgui/pj-pkgconfig.mak (added), res/pjproject/pjmedia/include/pjmedia-codec/speex.h (added), res/pjproject/config.guess (added), res/pjproject/tests/cdash/cfg_site_sample.py (added), res/pjproject/third_party/portaudio/src/common/pa_skeleton.c (added), res/pjproject/pjsip-apps/src/symbian_ua_gui/inc/symbian_ua_guiSettingItemList.hrh (added), res/pjproject/third_party/srtp/test/getopt_s.c (added), res/pjproject/pjmedia/src/pjmedia-codec/g722 (added), res/pjproject/tests/pjsua/scripts-pesq/201_codec_g722.py (added), res/pjproject/pjnath/src/pjturn-client/client_main.c (added), res/pjproject/third_party/gsm/src/short_term.c (added), res/pjproject/build.symbian/libg7221codec.mmp (added), res/pjproject/pjmedia/src/pjmedia/wsola.c (added), res/pjproject/pjlib-util/include/pjlib-util/hmac_sha1.h (added), res/pjproject/pjlib/include/pj++/list.hpp (added), res/pjproject/third_party/ilbc/anaFilter.c (added), res/pjproject/third_party/mp3 (added), res/pjproject/pjmedia/src/pjmedia/tonegen.c (added), res/pjproject/pjsip-apps/src/samples/stateful_proxy.c (added), res/pjproject/third_party/ilbc/anaFilter.h (added), res/pjproject/pjsip-apps/src/symsndtest/app_main.cpp (added), res/pjproject/pjsip-apps/src/pocketpj/SettingsDlg.cpp (added), res/pjproject/tests/pjsua/scripts-sipp/uas-invite.xml (added), res/pjproject/third_party/g7221/encode/sam2coef.c (added), res/pjproject/pjlib/src/pj/compat/string.c (added), res/pjproject/pjlib/include/pj/compat/cc_gcce.h (added), res/pjproject/pjlib/include/pj/config_site_sample.h (added), res/pjproject/third_party/build/srtp/output (added), res/pjproject/tests/pjsua/scripts-pesq/200_codec_speex_8000.py (added), res/pjproject/tests/pjsua/scripts-sipp/uac-options.xml (added), res/pjproject/third_party/ilbc/iCBConstruct.c (added), res/pjproject/tests/pjsua/scripts-sendto/153_err_sdp_unsupported_codec.py (added), res/pjproject/pjsip/build/wince-evc4 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res/pjproject/pjsip/src/pjsip/sip_auth_parser_wrap.cpp (added), res/pjproject/aconfigure.ac (added), res/pjproject/tests/pjsua/scripts-sendto/140_sdp_with_direction_attr_in_session_1.py (added), res/pjproject/pjsip-apps/src/pjsystest/pjsystest_wince.rc2 (added), res/pjproject/pjlib/include/pj/compat/os_win32.h (added), res/pjproject/pjmedia/include/pjmedia/doxygen.h (added), res/pjproject/pjsip/src/test/main_rtems.c (added), res/pjproject/pjlib-util/include/pjlib-util/scanner_cis_bitwise.h (added), res/pjproject/pjsip-apps/src/ipjsystest/main.m (added), res/pjproject/build.symbian/pjsip.mmp (added), res/pjproject/third_party/speex/include/speex/speex_jitter.h (added), res/pjproject/tests/pjsua/run.py (added), res/pjproject/third_party/speex/symbian (added), res/pjproject/tests/pjsua/scripts-pesq/200_codec_l16_8000_stereo.py (added), res/pjproject/pjsip-apps/src/samples/auddemo.c (added), res/pjproject/tests/pjsua/scripts-sendto/300_srtp_crypto_case_insensitive.py (added), 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res/pjproject/pjlib/build (added), res/pjproject/third_party/build/baseclasses/libbaseclasses.vcproj (added), res/pjproject/third_party/speex/include/speex/speex_preprocess.h (added), res/pjproject/pjlib/src/pjlib-test (added), res/pjproject/pjsip-apps/src/symbian_ua_gui/data/symbian_ua_gui.l01 (added), res/pjproject/pjlib/build/privkey.pem (added), res/pjproject/pjmedia/src/pjmedia/alaw_ulaw_table.c (added), res/pjproject/configure-legacy (added), res/pjproject/tests/pjsua/scripts-sendto/200_ice_success_1.py (added), res/pjproject/pjsip/include/pjsip/sip_transport.h (added), res/pjproject/pjnath/src/pjturn-srv/server.c (added), res/pjproject/pjmedia/build/os-linux.mak (added), res/pjproject/pjlib/include/pj/compat/os_win32_wince.h (added), res/pjproject/pjsip/src/pjsip-ua/sip_replaces.c (added), res/pjproject/third_party/portaudio/src/common/pa_util.h (added), res/pjproject/pjsip-apps/src/symbian_ua_gui/inc/symbian_ua_guiDocument.h (added), res/pjproject/pjlib/src/pj/fifobuf.c (added), res/pjproject/third_party/gsm/tls/sour1.dta (added), res/pjproject/pjsip/include/pjsip/sip_types.h (added), res/pjproject/pjlib/include/pj/compat/time.h (added), res/pjproject/pjsip/src/pjsip/sip_auth_msg.c (added), res/pjproject/tests/pjsua/scripts-sendto/001_torture_4475_3_1_1_1.py (added), res/pjproject/pjsip/include/pjsip_ua.h (added), res/pjproject/pjlib/build/Makefile (added), res/pjproject/third_party/srtp/README (added), res/pjproject/tests/pjsua/scripts-sendto/311_srtp1_recv_avp.py (added), res/pjproject/pjsip-apps/src/pjsua/main_rtems.c (added), res/pjproject/pjsip-apps/src/pocketpj/res/invisibl.bmp (added), res/pjproject/pjlib/src/pjlib-test/rtems_network_config.h (added), res/pjproject/third_party/srtp/crypto/math/stat.c (added), res/pjproject/third_party/srtp/test/replay_driver.c (added), res/pjproject/pjmedia/src/pjmedia-audiodev/audiotest.c (added), res/pjproject/pjlib/src/pjlib++-test (added), res/pjproject/pjsip-apps/src/samples/streamutil.c (added), res/pjproject/pjmedia/src/pjmedia/ffmpeg_util.c (added), res/pjproject/tests/pjsua/scripts-sendto/500_pres_subscribe_with_bad_event.py (added), res/pjproject/third_party/srtp/install-sh (added), res/pjproject/tests/pjsua/scripts-pesq/200_codec_speex_16000.py (added), res/pjproject/third_party/srtp/crypto/cipher/null_cipher.c (added), res/pjproject/pjmedia/src/pjmedia/ffmpeg_util.h (added), res/pjproject/pjlib-util/src (added), res/pjproject/pjsip/include/pjsip/sip_config.h (added), res/pjproject/pjlib/docs/doxygen.cfg (added): Add support for ICE/STUN/TURN in res_rtp_asterisk and chan_sip. Review: https://reviewboard.asterisk.org/r/1891/ 2012-06-29 20:32 +0000 [r369512] Mark Michelson * main/rtp_engine.c, /: Fix apparent copy and paste error where incorrect "glue" is used. ........ Merged revisions 369511 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-06-29 17:02 +0000 [r369493] Richard Mudgett * apps/app_dial.c, main/channel.c, main/autoservice.c, main/pbx.c, channels/chan_local.c, funcs/func_channel.c, main/channel_internal_api.c, main/features.c, configs/cdr.conf.sample, include/asterisk/channel.h, include/asterisk/pbx.h, CHANGES, apps/app_followme.c, apps/app_queue.c: Hangup handlers - Dialplan subroutines that run when the channel hangs up. Hangup handlers are an alternative to the h extension. They can be used in addition to the h extension. The idea is to attach a Gosub routine to a channel that will execute when the call hangs up. Whereas which h extension gets executed depends on the location of dialplan execution when the call hangs up, hangup handlers are attached to the call channel. You can attach multiple handlers that will execute in the order of most recently added first. (closes issue ASTERISK-19549) Reported by: Mark Murawski Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/2002/ 2012-06-29 16:56 +0000 [r369492] Joshua Colp * /, channels/chan_sip.c: With some configurations a transport is not actually specified so assume UDP in these cases. ........ Merged revisions 369490 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369491 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-06-29 16:42 +0000 [r369489] Richard Mudgett * main/channel_internal_api.c, .cleancount: Remove obsolete struct ast_channel note. The opaquing the ast_channel struct no longer requires .cleancount to be changed when the struct is changed. * Bump .cleancount value one last time because of struct ast_channel for old times sake. 2012-06-29 15:33 +0000 [r369473] Joshua Colp * /, channels/chan_sip.c: Make the address family filter specific to the transport. (closes issue ASTERISK-16618) Reported by: Leif Madsen Review: https://reviewboard.asterisk.org/r/1667/ ........ Merged revisions 369471 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369472 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-06-28 01:12 +0000 [r369449-369454] Terry Wilson * include/asterisk/config_options.h, configs/config_test.conf.sample, main/config_options.c, tests/test_config.c: Add the ability to set flags via the config options api Allows the setting of flags via the config options api. For example, code like this: #define OPT1 1 << 0 #define OPT2 1 << 1 #define OPT3 1 << 2 struct thing { unsigned int flags; }; and a config like this: [blah] opt1=yes opt2=no opt3=yes Review: https://reviewboard.asterisk.org/r/2004/ * /, channels/chan_sip.c, channels/sip/include/sip.h: AST-2012-010: Clean up after a reinvite that never gets a final response The basic problem is that if a re-INVITE is sent by Asterisk and it receives a provisional response, but no final response, then the dialog is never torn down. In addition to leaking memory, this also leaks file descriptors and will eventually lead to Asterisk no longer being able to process calls. This patch just keeps track of whether there is an outstanding re-INVITE, and if there is goes ahead and cleans up everything as though there was no outstanding reinvite. Review: https://reviewboard.asterisk.org/r/2009/ (closes issue ASTERISK-19992) Reported by: Steve Davies Tested by: Steve Davies, Terry Wilson ........ Merged revisions 369436 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369437 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-06-26 21:45 +0000 [r369414] Jonathan Rose * include/asterisk/logger.h, channels/chan_dahdi.c, main/autoservice.c, main/pbx.c, channels/chan_local.c, channels/sig_analog.c, main/channel_internal_api.c, channels/chan_agent.c, main/features.c, main/logger.c, channels/chan_iax2.c, channels/sig_pri.c, channels/sig_ss7.c, main/bridging.c, main/cli.c: Unique Call ID logging Phases III and IV Adds call ID logging changes to specific channel drivers that weren't handled handled in phase II of Call ID Logging. Also covers logging for threads for threads created by systems that may be involved with many different calls. Extra special thanks to Richard for rigorous review of chan_dahdi and its various signalling modules. review: https://reviewboard.asterisk.org/r/1927/ review: https://reviewboard.asterisk.org/r/1950/ 2012-06-26 13:23 +0000 [r369370-369392] Matthew Jordan * /, main/adsi.c: Fix crash in unloading of res_adsi module When res_adsi is unloaded, it removes the ADSI functions that it previously installed by passing a NULL adsi_funcs pointer to ast_adsi_install_funcs. This function was not checking whether or not the adsi_funcs pointer passed in was NULL before dereferencing it to check whether or not the version of the functions matches what the core was expecting it. This patch makes it so that the version is only checked if a potentially valid adsi_funcs pointer was passed in. Passing in NULL removes the installed functions, bypassing the version check. ........ Merged revisions 369390 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369391 from http://svn.asterisk.org/svn/asterisk/branches/10 * main/manager.c: Update "manager show event" to support tab completion Thank you rmudgett for pointing out that I was missing this in the initial check-in for AMI event documentation (r369346) * main/cdr.c, /: Fix incorrect duration reporting in CDRs created in batch mode Certain places in core/cdr.c would, if the duration value were 0, calculate the duration as being the delta between the current time and the time at which the CDR record was started. While this does not typically cause a problem in non-batch mode, this can cause an issue in batch mode where CDR records are gathered and written long after those calls have ended. In particular, this affects calls that were never answered, as those are expected to have a duration of 0. Often, this would result in CDR logs with a significant number of calls with lengthy durations, but dispositions of "BUSY". Note that this does not affect cdr_csv, as that backend does not use ast_cdr_getvar and instead directly reports the duration value. The affected core backends include cdr_apative_odbc and cdr_custom; other extended or deprecated CDR backends may potentially still directly manipulate the duration values. (issue ASTERISK-19860) Reported by: Thomas Arimont (issue AST-883) Reported by: Thomas Arimont Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1996/ ........ Merged revisions 369351 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369369 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-06-25 19:26 +0000 [r369367] Mark Michelson * /, channels/chan_sip.c, channels/sip/include/sip.h: Re-fix how local tag is generated when sending a 481 to an INVITE. Match our local tag to whatever to-tag was sent in the initial INVITE. Because the size of the to-tag may not fit in the buffer in the sip_pvt, it has been changed to a string field. (closes issue ASTERISK-19892) reported by Walter Doekes Review: https://reviewboard.asterisk.org/r/1977 ........ Merged revisions 369352 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369353 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-06-25 17:59 +0000 [r369346] Matthew Jordan * apps/app_dial.c, apps/app_meetme.c, configure.ac, apps/app_userevent.c, CHANGES, apps/app_queue.c, Makefile, build_tools/get_documentation.py (added), main/manager.c, configure, build_tools/post_process_documentation.py (added), include/asterisk/xmldoc.h, apps/app_confbridge.c, makeopts.in, apps/app_stack.c, apps/app_chanspy.c, doc/appdocsxml.dtd, main/xmldoc.c, apps/app_voicemail.c: Add AMI event documentation This patch adds the core changes necessary to support AMI event documentation in the source files of Asterisk, and adds documentation to those AMI events defined in the core application modules. Event documentation is built from the source by two new python scripts, located in build_tools: get_documentation.py and post_process_documentation.py. The get_documentation.py script mirrors the actions of the existing AWK get_documentation scripts, except that it will scan the entirety of a source file for Asterisk documentation. Upon encountering it, if the documentation happens to be an AMI event, it will attempt to extract information about the event directly from the manager event macro calls that raise the event. The post_process_documentation.py script combines manager event instances that are the same event but documented in multiple source files. It generates the final core-[lang].xml file. As this process can take longer to complete than a typical 'make all', it is only performed if a new make target, 'full', is chosen. Review: https://reviewboard.asterisk.org/r/1967/ 2012-06-25 16:07 +0000 [r369329] Richard Mudgett * /, main/features.c: Fix Bridge application occasionally returning to the wrong location. * Fix do_bridge_masquerade() getting the resume location from the zombie channel. The code must not touch a clone channel after it has masqueraded it. The clone channel has become a zombie and is starting to hangup. (closes issue ASTERISK-19985) Reported by: jamicque Patches: jira_asterisk_19985_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: jamicque ........ Merged revisions 369327 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369328 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-06-25 15:55 +0000 [r369304-369326] Mark Michelson * include/asterisk/adsi.h, /, main/Makefile, res/res_adsi.c, main/adsi.c (added), res/res_adsi.exports.in (removed): Multiple revisions 369323-369324 ........ r369323 | mmichelson | 2012-06-25 10:35:43 -0500 (Mon, 25 Jun 2012) | 9 lines Eliminate embedding of res_adsi.so module. The way this is done is to stop using the optional API. Instead, res_adsi.so, when loaded fills in a table of function pointers. Review: https://reviewboard.asterisk.org/r/1991 ........ r369324 | mmichelson | 2012-06-25 10:50:17 -0500 (Mon, 25 Jun 2012) | 2 lines Forgot to svn add this file in my last commit. ........ Merged revisions 369323-369324 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369325 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, channels/chan_sip.c: Be more consistent with the return code for requests received from invalid domain. When Asterisk receives an INVITE from an external domain when allowexternaldomains=no send a 403 instead of a 404. This is consistent with Asterisk's behavior when receiving a REGISTER in this situation. (Closes issue ASTERISK-19601) Reported by Matthew Jordan Patches: ASTERISK-19601-no401.patch uploaded by Mark Michelson (License #5049) ........ Merged revisions 369302 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369303 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-06-23 00:33 +0000 [r369237-369296] Richard Mudgett * main/features.c: Fix F and F(x) action logic in Bridge application. * /, main/features.c: Fix Bridge application and AMI Bridge action error handling. * Fix AMI Bridge action disconnecting the AMI link on error. * Fix AMI Bridge action and Bridge application not checking if their masquerades were successful. * Fix Bridge application running the h-exten when it should not. * Made do_bridge_masquerade() return if the masquerade was successful so the Bridge application and AMI Bridge action could deal with it correctly. * Made bridge_call_thread_launch() hangup the passed in channels if the bridge_call_thread fails to start. Those channels would have been orphaned. * Made builtin_atxfer() check the success of the transfer masquerade setup. ........ Merged revisions 369282 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369283 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, apps/app_queue.c: Explicitly check caller hangup in app Queue rather than a polluted res2 value. ........ Merged revisions 369262 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369263 from http://svn.asterisk.org/svn/asterisk/branches/10 * apps/app_queue.c: Fix F and F(x) action logic in Queue application. * apps/app_dial.c, /: Check if PBX was started and fix F and F(x) action logic in Dial application. ........ Merged revisions 369258 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369259 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, main/ccss.c: Check if PBX was started for generic CCSS recall. ........ Merged revisions 369238 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369239 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, channels/chan_sip.c: Change incorrect chan_sip zombie hangup debug message. They are all zombies now. ........ Merged revisions 369235 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369236 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-06-22 20:05 +0000 [r369217] Terry Wilson * /, channels/chan_sip.c: Don't crash on a guest directmedia call A sip_pvt may not have relatedpeer set if a call doesn't match up with a peer. If there is no relatedpeer, there is no direct media ACL to apply, so just return that it is allowed. (closes issue ASTERISK-20040) Reported by: Terry Wilson ........ Merged revisions 369214 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369215 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-06-22 19:54 +0000 [r369184-369216] Kinsey Moore * channels/chan_dahdi.c: Fix wrong variable name in the R2 disconnect callback * /, channels/chan_sip.c: Don't parse media stream state for SIP video streams The sendonly/recvonly/sendrecv/inactive media stream attributes were parsed for video, but nothing was ever done with them. With this code removed, an UNSUPPORTED message is produced when these attributes are used in conjunction with a video stream which is the better behavior since they were never really supported in the first place. ........ Merged revisions 369195 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369206 from http://svn.asterisk.org/svn/asterisk/branches/10 * channels/chan_dahdi.c: Add HANGUPCAUSE hash implementation for DAHDI MFC/R2 subtech This adds a minimal implementation of the "Who Hung Up?" Asterisk 11 work to chan_dahdi.c for the MFC/R2 DAHDI subtech. Given the way that OpenR2 interfaces with chan_dahdi, it is much harder to expose the type of protocol information that is available in PRI, SS7, or other channel technologies. * channels/sig_analog.c, channels/sig_pri.c: Add HANGUPCAUSE hash support for analog and PRI DAHDI subtechs This is part of the DAHDI support for the Asterisk 11 "Who Hung Up?" project and covers the implementation for the technologies implemented in sig_analog.c and sig_pri.c. Tested on a local machine to verify protocol and cause information is available. Review: https://reviewboard.asterisk.org/r/1953/ (issue SWP-4222) * channels/sig_ss7.c: Add "Who Hung Up?" implementation for DAHDI SS7 subtechnology Testing was done on a local machine to verify that protocol and cause information was being sent properly. Review: https://reviewboard.asterisk.org/r/1955/ (issue SWP-4222) 2012-06-20 21:33 +0000 [r369166-369167] Richard Mudgett * main/logger.c: Don't waste time initializing the whole call_identifer_str[]. The array is either setup with a callid string or only the first element needs to be initialized. * channels/chan_misdn.c: Fix chan_misdn compile error. 2012-06-20 17:48 +0000 [r369148] Alexandr Anikin * /, addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooCalls.c: fix locking issue on empty callList (issue ASTERISK-19298) Reported by: Dmitry Melekhov Patches: ASTERISK-18322-2.patch ........ Merged revisions 369146 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369147 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-06-20 11:47 +0000 [r369142] Sean Bright * apps/app_externalivr.c: Remove declaration of eivr_connect_socket because it no longer exists. 2012-06-20 11:20 +0000 [r369141] Alexandr Anikin * addons/chan_ooh323.c: use right definition for channel name 2012-06-20 03:18 +0000 [r369110-369126] Michael L. Young * main/manager.c, CHANGES: Add IPv6 Support To Manager This patch adds IPv6 support to AMI. (Closes issue ASTERISK-19965) Reported by: Michael L. Young Tested by: Michael L. Young Patches: ami_ipv6_v3.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/1968/ * main/netsock2.c, /, include/asterisk/netsock2.h: Fix NULL pointer segfault in ast_sockaddr_parse() While working with ast_parse_arg() to perform a validity check, a segfault occurred. The segfault occurred due to passing a NULL pointer to ast_sockaddr_parse() from ast_parse_arg(). According to the documentation in config.h, "result pointer to the result. NULL is valid here, and can be used to perform only the validity checks." This patch fixes the segfault by checking for a NULL pointer. This patch also adds documentation to netsock2.h about why it is necessary to check for a NULL pointer. (Closes issue ASTERISK-20006) Reported by: Michael L. Young Tested by: Michael L. Young Patches: asterisk-20006-netsock-null-ptr.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/1990/ ........ Merged revisions 369108 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369109 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-06-19 23:36 +0000 [r369092] Alexandr Anikin * addons/chan_ooh323.c, /: check rtptimeouts in ooh323 channels as per config file (rtp voice, video, udptl except rtcp) (closes issue ASTERISK-19179) Reported by: TSAREGORODTSEV Yury Patches: 19179-ooh323-ast10.patch ........ Merged revisions 369091 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-06-19 21:13 +0000 [r369086] Kinsey Moore * main/channel.c, channels/chan_dahdi.c, channels/chan_misdn.c, main/rtp_engine.c, include/asterisk/channel.h, channels/chan_iax2.c: Ensure that pvt cause information does not break native bridging Channel drivers that allow native bridging need to handle AST_CONTROL_PVT_CAUSE_CODE frames and previously did not handle them properly, usually breaking out of the native bridge. This change corrects that behavior and exposes the available cause code information to the dialplan while native bridges are in place. This required exposing the HANGUPCAUSE hash setter outside of channel.c, so additional documentation has been added. 2012-06-19 15:44 +0000 [r369068] Mark Michelson * /, channels/chan_sip.c: Fix request routing issue when outboundproxy is used. Asterisk was incorrectly setting the destination of CANCELs and ACKs for error responses to the URI of the initial INVITE. This resulted in further requests, such as INVITEs with authentication credentials, to be routed incorrectly. Instead, when these CANCEL or ACKs are to be sent, we should simply keep the destination the same as what it previously was. There is no need to alter it any. (closes issue ASTERISK-20008) Reported by Marcus Hunger Patches: ASTERISK-20008.patch uploaded by Mark Michelson (license #5049) ........ Merged revisions 369066 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369067 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-06-18 22:56 +0000 [r369061] Kinsey Moore * main/features.c: Fix AST_CONTROL_PVT_CAUSE_CODE handling When the IAX2 Who Hung Up? changes were added, they uncovered a bug in the way AST_CONTROL_PVT_CAUSE_CODE was handled in feature_request_and_dial(). This particular frame subtype was being treated like more terminal control frames causing the function to be exited prematurely. 2012-06-18 18:25 +0000 [r369057] Richard Mudgett * /, main/features.c: Fix monitoring calls put in a parking lot. * Fix a regression that was introduced by -r366167 which effectively disabled monitoring parked calls. (closes issue ASTERISK-20012) Reported by: sdolloff Tested by: rmudgett ........ Merged revisions 369043 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369044 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-06-15 21:18 +0000 [r369034] Damien Wedhorn * channels/chan_skinny.c: Various small chan_skinny fixes and cleanup Added test to skinny_register to only allow device to register against a device that is not already registered. Addback l->device test for skinny_show_lines. Fixes segfault if a line is configured but not configured to a device. Reverses part of r368680. Removed redundant l->device tests in subsubstate and dumpsub. l->device will always be valid if these routines are called. Reverses 368948 - discussed with mjordan on irc. Some indentation cleanup. 2012-06-15 17:13 +0000 [r369028] Kinsey Moore * channels/chan_sip.c, channels/sip/include/sip.h: Allow chan_sip to decline unwanted media streams This change replaces the static array of four representable media streams with an AST_LIST so that chan_sip can keep track of offered media streams. This allows chan_sip to deal with offers containing multiple same-type streams and many other situations without rejecting the SDP offer in its entirety, yet still generating a valid response. This also covers cases where Asterisk can not comprehend the offer if it is in the correct format. Previously, chan_sip would reject SDP offers or entirely ignore individual stream offers in an effort to be more compatible which would often result in invalid SDP responses. Review: https://reviewboard.asterisk.org/r/1988/ 2012-06-15 16:30 +0000 [r369027] Jason Parker * /, apps/app_voicemail.c: Fix voicemail API tests by using the correct argument order for create/destroy. ........ Merged revisions 369024 from http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11 ........ Merged revisions 369026 from http://svn.asterisk.org/svn/asterisk/branches/10-digiumphones 2012-06-15 16:20 +0000 [r369013] Kevin P. Fleming * main/format.c, main/udptl.c, main/netsock2.c, main/autoservice.c, main/rtp_engine.c, main/frame.c, main/security_events.c, /, main/say.c, main/threadstorage.c, channels/console_video.c, main/devicestate.c, main/astfd.c, main/taskprocessor.c, main/format_pref.c, main/astobj2.c, main/indications.c, main/config.c, main/loader.c, main/term.c, apps/confbridge/conf_config_parser.c, main/cli.c, channels/sig_analog.c, main/framehook.c, main/strcompat.c, main/plc.c, main/fskmodem_int.c, main/syslog.c, main/stdtime/localtime.c, main/bridging.c, main/db.c, channels/sig_ss7.c, main/datastore.c, main/sched.c, channels/sip/sdp_crypto.c, main/strings.c, main/pbx.c, channels/vcodecs.c, channels/sip/security_events.c, main/libasteriskssl.c, channels/iax2-provision.c, pbx/dundi-parser.c, main/aoc.c, main/cel.c, utils/astdb2bdb.c, channels/iax2-parser.c, main/chanvars.c, main/netsock.c, build_tools/find_missing_support_level (added), main/data.c, main/srv.c, channels/chan_misdn.c, main/privacy.c, main/fixedjitterbuf.c, channels/sip/dialplan_functions.c, main/test.c, main/audiohook.c, codecs/codec_dahdi.c, main/alaw.c, main/asterisk.c, main/timing.c, main/global_datastores.c, main/fskmodem_float.c, main/ccss.c, channels/sip/reqresp_parser.c, main/xml.c, channels/misdn/isdn_msg_parser.c, main/utils.c, main/autochan.c, channels/misdn/isdn_lib.c, main/enum.c, main/presencestate.c, main/fskmodem.c, channels/misdn_config.c, main/io.c, main/channel.c, main/cdr.c, res/ael/pval.c, main/ulaw.c, main/dial.c, main/format_cap.c, main/tdd.c, channels/console_gui.c, main/heap.c, channels/misdn/ie.c, main/logger.c, main/app.c, channels/console_board.c, main/image.c, main/message.c, main/dns.c, main/lock.c, main/stun.c, channels/sip/srtp.c, main/dnsmgr.c, main/slinfactory.c, main/channel_internal_api.c, main/translate.c, main/jitterbuf.c, main/acl.c, utils/astdb2sqlite3.c, channels/sip/utils.c, channels/sig_pri.c, apps/app_system.c, funcs/func_realtime.c, main/tcptls.c, main/hashtab.c, funcs/func_presencestate.c, apps/app_celgenuserevent.c, main/abstract_jb.c, main/callerid.c, main/file.c, main/config_options.c, res/snmp/agent.c, main/astmm.c, main/event.c, channels/misdn/portinfo.c, channels/sip/config_parser.c, channels/vgrabbers.c, main/dsp.c, main/xmldoc.c: Multiple revisions 369001-369002 ........ r369001 | kpfleming | 2012-06-15 10:56:08 -0500 (Fri, 15 Jun 2012) | 11 lines Add support-level indications to many more source files. Since we now have tools that scan through the source tree looking for files with specific support levels, we need to ensure that every file that is a component of a 'core' or 'extended' module (or the main Asterisk binary) is explicitly marked with its support level. This patch adds support-level indications to many more source files in tree, but avoids adding them to third-party libraries that are included in the tree and to source files that don't end up involved in Asterisk itself. ........ r369002 | kpfleming | 2012-06-15 10:57:14 -0500 (Fri, 15 Jun 2012) | 3 lines Add a script to enable finding source files without support-levels defined. ........ Merged revisions 369001-369002 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369005 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-06-15 16:17 +0000 [r369007] Kinsey Moore * main/frame.c, channels/chan_iax2.c, include/asterisk/frame.h: Add HANGUPCAUSE hash support to IAX2 Continuing with the Who Hung Up? project for Asterisk 11, this adds support to IAX2 for the HANGUPCAUSE hash. Additionally, this breaks out some functionality in frame.c for getting information about frame types and subclasses. Review: https://reviewboard.asterisk.org/r/1941/ (issue SWP-4222) 2012-06-15 15:33 +0000 [r369000] Jason Parker * /, apps/app_voicemail.exports.in: Remove some symbol exports that got missed in the removal of global symbols. (issue AST-807) (issue AST-901) (issue AST-908) ........ Merged revisions 368998 from http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11 ........ Merged revisions 368999 from http://svn.asterisk.org/svn/asterisk/branches/10-digiumphones 2012-06-15 00:55 +0000 [r368972-368991] Richard Mudgett * /: Remove remaining properties mmichelson left laying around from phones branch merge. * apps/app_dial.c, main/channel.c, include/asterisk/app.h, main/ccss.c, main/app.c, apps/app_followme.c, apps/app_queue.c, apps/app_stack.c: Allow non-normal execution routines to be able to run on hungup channels. * Make non-normal dialplan execution routines be able to run on a hung up channel. This is preparation work for hangup handler routines. * Fixed ability to support relative non-normal dialplan execution routines. (i.e., The context and exten are optional for the specified dialplan location.) Predial routines are the only non-normal routines that it makes sense to optionally omit the context and exten. Setting a hangup handler also needs this ability. * Fix Return application being able to restore a dialplan location exactly. Channels without a PBX may not have context or exten set. * Fixes non-normal execution routines like connected line interception and predial leaving the dialplan execution stack unbalanced. Errors like missing Return statements, popping too many stack frames using StackPop, or an application returning non-zero could leave the dialplan stack unbalanced. * Fixed the AGI gosub application so it cleans up the dialplan execution stack and handles the autoloop priority increments correctly. * Eliminated the need for the gosub_virtual_context return location. Review: https://reviewboard.asterisk.org/r/1984/ * main/pbx.c: Make the Hangup application set a softhangup flag. The Hangup application used to just return -1 to cause normal dialplan execution to hangup a channel. For the non-normal execution routines like predial and connected-line interception routines, the hangup request would exit the routine early but otherwise be ignored. * Made the Hangup application not allow setting a cause code of zero. A zero cause code is not defined. * include/asterisk/app.h: Move vm defines to group them better. 2012-06-14 19:40 +0000 [r368966] Jason Parker * include/asterisk/app.h, /, tests/test_voicemail_api.c, main/app.c, include/asterisk/app_voicemail.h (removed), apps/app_voicemail.c: Multiple revisions 368963,368965 ........ r368963 | qwell | 2012-06-14 13:47:03 -0500 (Thu, 14 Jun 2012) | 14 lines Remove global symbol requirement from app_voicemail. This uses the existing "function installation" stuff that already existed for other functions, like getting message counts. (closes issue AST-807) (issue AST-901) (issue AST-908) Review: https://reviewboard.asterisk.org/r/1965/ ........ Merged revisions 368962 from http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11 ........ r368965 | qwell | 2012-06-14 14:04:57 -0500 (Thu, 14 Jun 2012) | 11 lines These functions that were moved need to be static. Also wrap test functions in a #ifdef. (issue AST-807) (issue AST-901) (issue AST-908) ........ Merged revisions 368964 from http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11 ........ Merged revisions 368963,368965 from http://svn.asterisk.org/svn/asterisk/branches/10-digiumphones 2012-06-14 17:34 +0000 [r368948] Matthew Jordan * /, channels/chan_skinny.c: AST-2012-009: Fix crash in chan_skinny due to Key Pad Button Message handling AST-2012-008 (r367844) fixed a denial of service attack exploitable in the Skinny channel driver that occurred when certain messages are sent after a previously registered station sends an Off Hook message. Unresolved in that patch is an issue in the Asterisk 10 releases, wherein, if a Station Key Pad Button Message is processed after an Off Hook message, the channel driver will inappropriately dereference a NULL pointer. This patch fixes those places where the message handling or the channel callback functions would attempt to dereference the line's pointer to the device. (issue ASTERISK-19905) Reported by: Christoph Hebeisen Tested by: mjordan, Christoph Hebeisen Patches: AST-2012-009-10.diff uploaded by mjordan (license 6283) ........ Merged revisions 368947 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-06-14 15:28 +0000 [r368929] Mark Michelson * /, main/Makefile: Revert Makefile change to remove embedding res_adsi.so The change has resulted in a linking error for certain versions of GCC. This is much worse than the original issue, so for now, temporarily revert the change. A more thorough change will be sought out. ........ Merged revisions 368927 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368928 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-06-14 13:41 +0000 [r368920-368921] Terry Wilson * include/asterisk/config_options.h, main/config_options.c: Add a post_apply callback to the Config Options API This adds a callback that only fires when changes have been successfully applied via the Config Options API. Review: https://reviewboard.asterisk.org/r/1980/ * include/asterisk/config_options.h, main/config_options.c: Add filename alias support to the Config Options API This adds the ability to handle a single filename alias for a config file. This is useful if a config filename has changed, but the old filename should be supported for backwards compatibility. Review: https://reviewboard.asterisk.org/r/1981/ 2012-06-13 21:17 +0000 [r368900] Mark Michelson * /, funcs/func_volume.c: Fix a deadlock that occurs when func_volume is used on a local channel. This was discovered by trying to perform a call forward to an extension that makes use of func_volume. When the local channel is optimized away, the datastore on the local;2 channel would have its audiohook destroyed rather than detaching the audiohook from the channel and then destroying it. With this patch, func_volume's datastore destructor takes the proper route of detaching the audiohook and then destroying it. (closes issue ASTERISK-19611) reported by Volker Sauer Patches: ASTERISK-19611.patch uploaded by Mark Michelson (license #5049) ........ Merged revisions 368898 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368899 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-06-13 20:28 +0000 [r368896] Matthew Jordan * res/res_smdi.c, /, res/res_adsi.c: Mark res_smdi/res_adsi as 'core' supported modules Recently, various issues surrounding weak symbols have caused problems with modules that rely on that feature to be enabled in menuselect. This includes app_voicemail and chan_dahdi, as they both rely upon res_smdi and res_adsi, which, in certain circumstances, may not be enabled by default in menuselect. Because res_smdi/res_adsi are dependencies for chan_dahdi/app_voicemail, this patch marks both as 'core' supported modules. This will allow both app_voicemail and chan_dahdi to be enabled as well, regardless of whether or not that system supports weak symbols. (issue AST-900) Reported by: Thomas Arimont (issue AST-885) Reported by: Denis Alberto Martinez ........ Merged revisions 368894 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368895 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-06-13 19:51 +0000 [r368886] Mark Michelson * /, main/Makefile: Remove forced linking of res_adsi.o In GCC 4.5+ the result is that Asterisk has a phantom module loaded at startup, claiming to be res_adsi. (closes issue ASTERISK-19920) reported by Leif Madsen ........ Merged revisions 368873 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368885 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-06-13 14:55 +0000 [r368832-368855] Matthew Jordan * Makefile: Replace MODULES_DIR with ASTMODDIR in Makefile's INSTALLDIRS Post Asterisk 10, the MODULES_DIR variable no longer exists, and was replaced with ASTMODDIR. * Makefile, /: Do not install empty directories; add ASTLIBDIR r368830 modified the installation script to only create a directory if that directory does not exist. If some directory variable was empty, it would attempt to create the empty location. It also failed to create the ASTLIBDIR directory. This patch fixes it such that the correct directories are made and only created if a value specifying them actually exists. ........ Merged revisions 368852 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368853 from http://svn.asterisk.org/svn/asterisk/branches/10 * Makefile, /: Do not perform install on existing directories If a directory already exists, performing a 'make install' will remove the permissions associated with the current directory and replace them with the permissions of the user executing the install. This patch changes this behavior to only perform an install on the directory if the directory does not exist. Thus, if a user later changes the permissions on that directory, those permissions will be preserved in subsequent installs. Review: https://reviewboard.asterisk.org/r/1986 Review: https://reviewboard.asterisk.org/r/1864 (closes issue ASTERISK-19492) Reported by: Karl Fife Tested by: Paul Belanger, Tilghman Lesher patches: ASTERISK-19492 by pabelanger (uploaded by mjordan) ........ Merged revisions 368830 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368831 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-06-12 15:46 +0000 [r368809] Mark Michelson * /, channels/chan_sip.c: Set the Caller ID "tag" on peers even if remote party information is present. On incoming calls, we were setting the cid_tag on the dialog only if there was no remote party information (Remote-Party-ID or P-Asserted-Identity) present. The Caller ID tag is an invented parameter, though, and should be set no matter the circumstance. (closes issue ASTERISK-19859) Reported by Thomas Arimont (closes issue AST-884) Reported by Trey Blancher ........ Merged revisions 368807 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368808 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-06-12 14:09 +0000 [r368793-368794] Matthew Jordan * /: Update merge property information * channels/chan_sip.c: Fix deadlock in SIP transfers that involve a REFER request In r367163, "send to voicemail" functionality was added to the SIP channel driver. This required updating the party redirecting information for the channel based on the headers provided in the REFER request. When the redirecting party information is updated on the channel, a call to ast_indicate_data occurs. Because handle_request_refer still had the sip_pvt locked, a deadlock could occur between the pbx_thread and the do_monitor thread servicing the REFER request. This patch preserves the proper locking order between the channel and the sip_pvt by ensuring that the sip_pvt is unlocked prior to updating the party redirecting information on the channel. (closes issue AST-903) Reported by: Matt Jordan patches: jira_ast_903_trunk.patch by rmudgett (license 5621) 2012-06-12 04:03 +0000 [r368784] Kinsey Moore * channels/chan_sip.c, UPGRADE.txt: Parse ANI2 information from SIP From header parameters ANI2 information is now parsed out of SIP From headers when present in the oli, isup-oli, and ss7-oli parameters and is available via the CALLERID(ani2) dialplan function. (closes issue ASTERISK-19912) Patch-by: Rob Gagnon Review: https://reviewboard.asterisk.org/r/1947/ 2012-06-11 17:34 +0000 [r368772] Richard Mudgett * main/channel.c, channels/chan_dahdi.c, channels/sig_analog.c, /, channels/chan_sip.c, include/asterisk/channel.h, channels/chan_iax2.c: Fix deadlock potential with ast_set_hangupsource() calls. Calling ast_set_hangupsource() with the channel lock held can result in a deadlock because the function also locks the bridged channel. (issue ASTERISK-19537) (closes issue AST-891) Reported by: Guenther Kelleter Tested by: Guenther Kelleter (closes issue ASTERISK-19801) Reported by: Alec Davis ........ Merged revisions 368759 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368760 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-06-11 15:23 +0000 [r368722-368751] Kinsey Moore * channels/sip/sdp_crypto.c, /, channels/chan_sip.c, main/say.c, res/res_fax.c, channels/sip/reqresp_parser.c, apps/app_queue.c, main/loader.c, channels/chan_dahdi.c, res/res_config_odbc.c, channels/sip/dialplan_functions.c, apps/app_directory.c, pbx/pbx_config.c, res/res_odbc.c, res/res_speech.c, apps/app_voicemail.c: Fix coverity UNUSED_VALUE findings in core support level files Most of these were just saving returned values without using them and in some cases the variable being saved to could be removed as well. (issue ASTERISK-19672) ........ Merged revisions 368738 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368739 from http://svn.asterisk.org/svn/asterisk/branches/10 * /: Recorded merge of revisions 368721 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Fix compilation in dev-mode Backport a compilation fix in md5.c from trunk that only showed up in dev-mode under certain compiler versions. ........ Merged revisions 368719 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-06-08 21:08 +0000 [r368712-368714] Richard Mudgett * main/manager.c, main/utils.c, include/asterisk/strings.h: Fix error paths in action_hangup() for AMI Hangup action. * Check allocation function return values for failure. Crashing is bad. * Tweak ast_regex_string_to_regex_pattern() parameters for proper ast_str usage. * main/channel.c, include/asterisk/channel.h: Tweak ast_channel_softhangup_withcause_locked() to take a typed parameter. 2012-06-08 08:32 +0000 [r368688] Igor Goncharovskiy * channels/chan_unistim.c: Fix MWI update so LED display correct voicemail state after phone usage. Also fixes few warnings. (closes issue #19675) Reported by: dbohling Patches: fixmwi.patch uploaded by dbohling (license 6378) 2012-06-07 21:44 +0000 [r368680-368681] Damien Wedhorn * channels/chan_skinny.c: Skinny cleanup (mwi_event_cb). Original was testing for d->session, setting and testing again (all nested). Removed duplicate testing and restructured function to test/return and then the main code. * channels/chan_skinny.c: Skinny cleanup. Removed d->registered which was mirroring d->session. Changed relevant references to use d->session instead. Moved setting and unsetting of l->device from session register to device configuration. As such, l->device will always be valid unless it is has not been configured to a device. Revised various test where checking if a device is registered to use l->device->session. 2012-06-07 20:39 +0000 [r368674-368675] Richard Mudgett * apps/app_queue.c: Fix app_queue debug message use of args.options after the string has been parsed. * apps/app_queue.c: Fix inverted test in app_queue for ringinuse. Regression from -r367080 ringinuse commit. (issue ASTERISK-19536) 2012-06-07 20:32 +0000 [r368673] Terry Wilson * main/udptl.c, include/asterisk/config_options.h, apps/app_skel.c, main/config_options.c, tests/test_config.c: Fix reloading an unchanged file with the Config Options API Adding multiple file support broke reloading an unchanged file. This adds an enum for return values for the aco_process_* functions and ensures that the config is not applied if res is not ACO_PROCESS_OK. Review: https://reviewboard.asterisk.org/r/1979/ 2012-06-07 20:00 +0000 [r368668] Tzafrir Cohen * formats/format_ogg_vorbis.c: Fix a typo in format_ogg_vorbis.c: suport Review: https://reviewboard.asterisk.org/r/1970/ 2012-06-07 15:43 +0000 [r368663] Terry Wilson * include/asterisk/config_options.h, main/config_options.c, tests/test_config.c: Add default handler documentation and standardize acl handler Added documentation describing what flags and arguments to pass to aco_option_register for default option types. Also changed the ACL handler to use the flags parameter to differentiate between "permit" and "deny" instead of adding an additional vararg parameter. Review: https://reviewboard.asterisk.org/r/1969/ 2012-06-06 21:34 +0000 [r368646] Richard Mudgett * channels/chan_dahdi.c, channels/sig_analog.c, /: Fix POTS flash hook to orignate a second call deadlock. A deadlock can occur when a POTS phone tries to flash hook to originate a second call for 3-way or transfer. If another process is scanning the channels container when the POTS line flash hooks then a deadlock will occur. * Release the channel and private locks when creating a new channel as a result of a flash hook. (closes issue ASTERISK-19842) Reported by: rmudgett Tested by: rmudgett ........ Merged revisions 368644 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368645 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-06-06 19:25 +0000 [r368637] Mark Michelson * /, channels/chan_sip.c: Fix a specific scenario where ACKs are not matched. If a dialog-starting INVITE contains a to-tag, then Asterisk will respond with a 481. In this case, the resulting incoming ACK would not be matched, so Asterisk would continue retransmitting the 481 until the transaction times out. There were two issues. Asterisk, upon creating a sip_pvt would generate a local tag. However, when the time came to transmit the 481, since there was a to-tag in the INVITE, Asterisk would place this original to-tag in the 481 response. When the ACK came in, Asterisk would attempt to match the to-tag in the ACK to the generated local tag. Unfortunately, Asterisk never actually transmitted a response with the generated local tag, so the to-tag in the ACK would not match. The other problem was that when the 481 was sent, nothing was set on the sip_pvt to indicate what CSeq is expected in the ACK. To fix the first problem, we zero out the to-tag seen in the incoming INVITE. This way, Asterisk, when time to send a response, will send its generated local tag instead. To fix the second problem, we set the sip_pvt's pendinginvite to the CSeq of the INVITE when we send a 481. (closes issue ASTERISK-19892) Reported by Mark Michelson ........ Merged revisions 368625 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368629 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-06-06 17:22 +0000 [r368606] Matthew Jordan * /, build_tools/make_version: Add feature modifier to versions produced from branches Certain branches, such as Certified Asterisk, may have a modifier added to them that specifies the features available in that branch. For branches, this modifier is expected to be reflected in the location of the branch in subversion. For example, a subversion of URL of /certified/branches/1.8.11 would have a feature modifier of 'certified'. This is slightly different then how features are determined for tags, where the feature is part of the actual tag name, e.g., "10.5.0-digiumphones". In keeping with the nomenclature used for tags, the feature specifier for branches is translated and placed after the revision numbers. For the example given previously, this would result in a branch version of "Asterisk SVN-branch-1.8.11-cert-rXXXXXX". ........ Merged revisions 368604 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368605 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-06-06 16:11 +0000 [r368588] Kinsey Moore * /, channels/chan_sip.c: Ensure overlapping hold flags do not conflict When changing between different modes of hold, the flags were not being cleared out properly causing a failure to change hold states. (closes issue ASTERISK-19919) Patch-by: Morten Tryfoss Reported-by: Morten Tryfoss ........ Merged revisions 368586 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368587 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-06-06 01:11 +0000 [r368566-368569] Richard Mudgett * /, main/features.c: Fix parked call performing a DTMF blind transfer after being retrieved. When a parked call was retrieved from the parking lot, it could not do a blind transfer because it caused the involved calls to be hung up unconditionally. * Made the ParkedCall application return the ast_bridge_call() return value. (closes issue ABE-2862) Reported by: Vlad Povorozniuc ........ Merged revisions 368567 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368568 from http://svn.asterisk.org/svn/asterisk/branches/10 * main/features.c: Make builtin_blindtransfer() fully use ast_async_goto() abilities. 2012-06-05 16:25 +0000 [r368550] Jonathan Rose * CHANGES: Merge 'core' and 'core changes' sections in CHANGES file. 2012-06-05 15:28 +0000 [r368519-368537] Kinsey Moore * /: Recorded merge of revisions 368536 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Resolve some build warnings My newly upgraded compiler caught these usages of uninitialized values. They weren't actually used. ........ Merged revisions 368533 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, apps/app_voicemail.c: Ensure that pages and emails are sent using RFC822-compliant date format When localization was added to app_voicemail, these headers were altered when they should have remained in en_US format for RFC compliance. This reverts the changes to those two lines. (closes issue ASTERISK-19876) ........ Merged revisions 368520 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368524 from http://svn.asterisk.org/svn/asterisk/branches/10 * apps/app_dial.c, channels/chan_unistim.c, channels/chan_local.c, channels/chan_sip.c, main/channel_internal_api.c, main/features.c, include/asterisk/channel.h, apps/app_queue.c: Convert AST_FLAG_ANSWERED_ELSEWHERE usage to AST_CAUSE_ANSWERED_ELSEWHERE This was essentially duplicated functionality where normal channels used AST_CAUSE_ANSWERED_ELSEWHERE while local channels and queues used AST_FLAG_ANSWERED_ELSEWHERE. This removes the flag and converts that usage into AST_CAUSE_ANSWERED_ELSEWHER usage. Review: https://reviewboard.asterisk.org/r/1944 (closes issue ASTERISK-19865) Patch-by: Birger Harzenetter 2012-06-04 22:12 +0000 [r368500] Mark Michelson * /, channels/chan_sip.c: Relay proper SIP responses on calling side. Revision 351130 broke corect HANGUPCAUSE setting for the 404 case in chan_sip. Other cases were also potentially broken. This patch fixes the relaying of causes to be what they used to be. (closes issue ASTERISK-19914) Reported by Pavel Troller Tested by Walter Doekes (via a reviewboard test to be committed later) Patches: chan_sip.diff uploaded by Pavel Troller (license #6302) ........ Merged revisions 368498 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368499 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-06-04 21:18 +0000 [r368472] Richard Mudgett * /, UPGRADE.txt: Document BLINDTRANSFER behavior change. (issue ASTERISK-19322) (closes issue ASTERISK-19875) Reported by: call ........ Merged revisions 368469 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368470 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-06-04 20:53 +0000 [r368435-368467] Mark Michelson * contrib/editors/asterisk.vim: Also have vim syntax-highlight type=network. * contrib/editors/asterisk.vim: Add vim syntax highlighting for type=line, type=phone, and type=application. (closes issue ASTERISK-19800) Reported by: Billy Chia Patches: asterisk.vim.patch uploaded by Billy Chia (license #6381) * main/channel.c, apps/app_mixmonitor.c: Remove some extra debugging I forgot to remove in the merge of Digium phone support. * /: Remove automerge properties. * /, contrib/realtime/mysql/voicemail_messages.sql, main/presencestate.c (added), main/config.c, main/channel.c, include/asterisk/callerid.h, include/asterisk/file.h, main/manager.c, channels/chan_skinny.c, include/asterisk/event_defs.h, include/asterisk/sip_api.h (added), tests/test_voicemail_api.c (added), main/features.c, apps/app_voicemail.exports.in, main/app.c, main/message.c, channels/sip/include/sip.h, main/pbx.c, channels/chan_sip.c, include/asterisk/presencestate.h (added), include/asterisk/config.h, include/asterisk/app_voicemail.h (added), configs/manager.conf.sample, apps/app_queue.c, include/asterisk/manager.h, include/asterisk/app.h, funcs/func_presencestate.c (added), include/asterisk/message.h, main/file.c, main/callerid.c, main/event.c, include/asterisk/pbx.h, tests/test_config.c, channels/chan_sip.exports.in (added), apps/app_mixmonitor.c, main/asterisk.c, apps/app_voicemail.c: Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. 2012-06-04 19:46 +0000 [r368421] Richard Mudgett * main/channel.c, /: Fix potential deadlock between masquerade and chan_local. * Restructure ast_do_masquerade() to not hold channel locks while it calls ast_indicate(). * Simplify many calls to ast_do_masquerade() since it will never return a failure now. If it does fail internally because a channel driver callback operation failed, the only thing ast_do_masquerade() can do is generate a warning message about strange things may happen and press on. * Fixed the call to ast_bridged_channel() in ast_do_masquerade(). This change fixes half of the deadlock reported in ASTERISK-19801 between masquerades and chan_iax. (closes issue ASTERISK-19537) Reported by: rmudgett Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1915/ ........ Merged revisions 368405 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368407 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-06-02 21:13 +0000 [r368359] Joshua Colp * include/asterisk/utils.h, res/res_http_websocket.exports.in (added), include/asterisk/http_websocket.h (added), main/utils.c, res/res_http_websocket.c (added): Add res_http_websocket module which implements the WebSocket protocol according to RFC 6455. Review: https://reviewboard.asterisk.org/r/1952/ 2012-06-01 23:53 +0000 [r368311] Richard Mudgett * /, apps/app_stack.c: Fix deadlock when Gosub used with alternate dialplan switches. Attempting to remove a channel from autoservice with the channel lock held will result in deadlock. * Restructured gosub_exec() to not call ast_parseable_goto() and ast_exists_extension() with the channel lock held. (closes issue ASTERISK-19764) Reported by: rmudgett Tested by: rmudgett ........ Merged revisions 368308 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368310 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-06-01 20:42 +0000 [r368268-368269] Kevin P. Fleming * channels/chan_sip.c: Improve SDP offer/answer RFC compliance Asterisk should not accept SDP offers that contain unknown RTP profiles (for audio/video streams) or unknown top-level media types. When it does, it answers with an SDP that does not match the offer properly, and this will nearly always result in a broken call. This patch causes such offers to be rejected. Review: https://reviewboard.asterisk.org/r/1811/ * /, channels/chan_sip.c: Improve SDP parsing warning messages * 'Unsupported media type' is only reported when that is in fact the case, not when a supported media type is included in an 'm' line that has an invalid format. * All warning messages related to parsing 'm' lines now include the 'm' line contents. * (minor bugfix) newline added to port-number-zero warning messages. * Warning messages improved to use RFC-specified terminology for various items. * Warnings for offers that include more than one port for a single media type now include the media type. Review: https://reviewboard.asterisk.org/r/1811/ ........ Merged revisions 368218 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368267 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-06-01 18:20 +0000 [r368181-368221] Terry Wilson * configs/config_test.conf.sample (added): Add missing config for config API test * main/udptl.c, include/asterisk/utils.h, include/asterisk/astobj2.h, configure.ac, include/asterisk/config.h, main/astobj2.c, main/config.c, Makefile, include/asterisk/config_options.h (added), configure, main/asterisk.exports.in, apps/app_skel.c, main/config_options.c (added), tests/test_config.c, makeopts.in, configs/app_skel.conf.sample (added), include/asterisk/stringfields.h: Add new config-parsing framework This framework adds a way to register the various options in a config file with Asterisk and to handle loading and reloading of that config in a consistent and atomic manner. Review: https://reviewboard.asterisk.org/r/1873/ 2012-06-01 13:04 +0000 [r368143] Mark Michelson * channels/chan_sip.c, configs/sip.conf.sample, channels/sip/include/sip.h: Help mitigate potential reinvite glare scenarios. When Asterisk servers are set up back-to-back, and direct media is to be used betweeen endpoints, it is fairly common for the two Asterisk servers to send direct media reinvites to each other simultaneously. This results in 491s and ACKs being exchanged between the servers. While the media eventually gets set up properly, the problem is that there can be a noticeable delay for the streams to stabilize. This patch adds a new directmedia option called "outgoing". With this set, an immediate direct media reinvite will only be sent if the call direction is outgoing. For incoming dialogs, an immediate direct media reinvite will not be sent, but further "reactionary" direct media reinvites may be sent. Review: https://reviewboard.asterisk.org/r/1954 2012-06-01 03:30 +0000 [r368094] Michael L. Young * /, funcs/func_channel.c: Add documentation to function CHANNEL for options echocan_mode and buffers The ability to set "echocan_mode" and "buffers" through the dialplan was added to chan_dahdi some time ago. This patch adds some documentation to func_channel. (Closes issue ASTERISK-19911) Reported by: Dale Noll Tested by: Michael L. Young Patches: asterisk-19911-branch18.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/1949/ ........ Merged revisions 368092 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368093 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-05-31 18:39 +0000 [r368052] Richard Mudgett * res/ael/pval.c, main/tcptls.c, main/manager.c, res/res_config_odbc.c, /, channels/chan_sip.c, channels/chan_agent.c, funcs/func_math.c, main/features.c, apps/app_queue.c, channels/chan_iax2.c, pbx/pbx_config.c: Coverity Report: Fix issues for error type REVERSE_INULL (core modules) * Fixes findings: 0-2,5,7-15,24-26,28-31 (issue ASTERISK-19648) Reported by: Matt Jordan ........ Merged revisions 368039 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368042 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-05-30 18:08 +0000 [r367908-367982] Richard Mudgett * /: Use the DEADLOCK_AVOIDANCE() macro instead. (issue ASTERISK-19854) ........ Merged revisions 367980 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 367981 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, channels/sig_pri.c, channels/sig_ss7.c: Fix deadlock when executing CLI "pri show channels" and "ss7 show channels" commands. * Fix sig_pri_lock_owner() to avoid deadlock properly. * Code pri_grab() better. * Fix sig_ss7_lock_owner() to avoid deadlock properly. * Code ss7_grab() better. (closes issue ASTERISK-19854) Reported by: Jaxon Patches: jira_asterisk_19854_v1.8.6.patch (license #5621) patch uploaded by rmudgett (Modified to do the same thing to sig_ss7) Tested by: Jaxon ........ Merged revisions 367976 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 367978 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, apps/app_meetme.c: Coverity Report: Fix issues for error type REVERSE_INULL (deprecated modules) * Fix only issue pointed out by deprecated_REVERSE_INULL.txt for app_meetme.c in find_user(). * Change use of %i to %d in sscanf() in find_user(). The use of %i gives unexpected parsing because it can accept hex, octal, and decimal integer formats. * Changed other uses of %i in app_meetme() to use %d for consistency. (issue ASTERISK-19648) Reported by: Matt Jordan ........ Merged revisions 367906 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 367907 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-05-29 18:40 +0000 [r367845] Matthew Jordan * /, channels/chan_skinny.c: AST-2012-008: Fix remote crash vulnerability in chan_skinny When a skinny session is unregistered, the corresponding device pointer is set to NULL in the channel private data. If the client was not in the on-hook state at the time the connection was closed, the device pointer can later be dereferened if a message or channel event attempts to use a line's pointer to said device. The patches prevent this from occurring by checking the line's pointer in message handlers and channel callbacks that can fire after an unregistration attempt. (closes issue ASTERISK-19905) Reported by: Christoph Hebeisen Tested by: mjordan, Damien Wedhorn Patches: AST-2012-008-1.8.diff uploaded by mjordan (license 6283) AST-2012-008-10.diff uploaded by mjordan (licesen 6283) ........ Merged revisions 367844 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-05-25 16:33 +0000 [r367783] Richard Mudgett * /, channels/chan_iax2.c: AST-2012-007: Fix IAX receiving HOLD without suggested MOH class crash. * Made schedule_delivery() set the received frame f->data.ptr to NULL if the datalen is zero. * Fix queue_signalling() memcpy() size error. * Made queue_signalling() not use C++ keyword variable names. (closes issue ASTERISK-19597) Reported by: mgrobecker Patches: jira_asterisk_19597_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett, Michael L. Young ........ Merged revisions 367781 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 367782 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-05-25 02:31 +0000 [r367732] Michael L. Young * /, channels/chan_sip.c: Fix pvt_sip for inbound call to use peer's allowtransfer setting The pvt_sip allowtransfer was not being set to that of the peer's setting. Therefore, the global allowtransfer setting was being used instead which would lead to calls not being transfered if the global setting was set to 'no' despite the setting on the peer being 'yes' and vice versa, calls would be allowed to transfer even if the peer's setting was 'no' but the global setting was 'yes'. (Closes issue ASTERISK-19856) Reported by: Jacek Tested by: Michael L. Young, Jacek Patches: issue-asterisk-19856-branch10-v3.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/1923/ ........ Merged revisions 367730 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 367731 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-05-24 23:52 +0000 [r367693] Richard Mudgett * apps/app_dial.c, /, apps/app_queue.c: Fix Dial I option ignored if dial forked and one fork redirects. The Dial and Queue I option is intended to block connected line updates and redirecting updates. However, it is a feature that when a call is locally redirected, the I option is disabled if the redirected call runs as a local channel so the administrator can have an opportunity to setup new connected line information. Unfortunately, the Dial and Queue I option is disabled for *all* forked calls if one of those calls is redirected. * Make the Dial and Queue I option apply to each outgoing call leg independently. Now if one outgoing call leg is locally redirected, the other outgoing calls are not affected. * Made Dial not pass any redirecting updates when forking calls. Redirecting updates do not make sense for this scenario. * Made Queue not pass any redirecting updates when using the ringall strategy. Redirecting updates do not make sense for this scenario. * Fixed deadlock potential with chan_local when Dial and Queue send redirecting updates for a local redirect. * Converted the Queue stillgoing flag to a boolean bitfield. (closes issue ASTERISK-19511) Reported by: rmudgett Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1920/ ........ Merged revisions 367678 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 367679 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-05-24 18:56 +0000 [r367640] Jonathan Rose * main/rtp_engine.c, channels/chan_sip.c, include/asterisk/rtp_engine.h: chan_sip: fix problem directmediapermit/deny uses the wrong address When remotely bridging calls with directmedia, Asterisk would check the address of the peers/users holding directmedia ACLs (set via directmediapermit/directmediadeny) instead of the bridged peer. This is similar to r366547, but trunk specific and involves changes to the rtpengine instead of just chan_sip. (closes issue AST-876) review: https://reviewboard.asterisk.org/r/1924/ 2012-05-24 13:33 +0000 [r367563] Matthew Jordan * /, apps/app_confbridge.c: Fix crash in ConfBridge when user announcement is played for more than 2 users A patch introduced in r354938 made it so that ConfBridge would not attempt to play sound files if those files did not exist. Unfortunately, ConfBridge uses the same underlying function, play_sound_helper, to playback both sound files and numbers to callers. When a number is being played back, the name of the sound file is expected to be NULL. This NULL value was passed into a function that tested for the existance of a sound file and is not tolerant to NULL file names, causing a crash. This patch fixes the behavior, such that if a sound file does not exist we do not attempt to play it, but we only attempt that check if the a sound file was specified in the first place. If a sound file was not specified, we use the 'play number' logic in the helper function. (closes issue ASTERISK-19899) Reported by: Florian Gilcher Tested by: Florian Gilcher patches: asterisk-19899.diff uploaded by mjordan (license 6283) ........ Merged revisions 367562 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-05-24 00:36 +0000 [r367477-367520] Richard Mudgett * channels/iax2-parser.c: Made use IAX frame cache only for cacheable frame types. * main/pbx.c, /: Fix WaitExten(x,m(musicclass)) string termination. The AST_CONTROL_HOLD MOH class from the WaitExten application can now be queued onto a channel, passed over local channels with the /m option, and passed over IAX channels. ........ Merged revisions 367469 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 367470 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-05-23 20:39 +0000 [r367419] Jonathan Rose * main/pbx.c: logger: Fix a potential callid reference leak discovered in development Uncovered a nasty reference leak while I was writing some changes to chan_dahdi/sig_analog. Slapped myself around a bit after seeing that I performed the unchecked return causing this problem. 2012-05-23 20:30 +0000 [r367418] Mark Michelson * main/tcptls.c, /: Only call SSL_CTX_free if DO_SSL is defined. Thanks to Paul Belanger for pointing out this error. ........ Merged revisions 367416 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 367417 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-05-23 13:46 +0000 [r367376] Matthew Jordan * /, channels/chan_sip.c, channels/sip/include/sip.h: Re-add LastMsgsSent value for SIP peers Previously, MWI logic utilized a counter called 'lastmsgssent' to know whether or not MWI NOTIFY requests had been sent to a specific peer. When MWI notifications were changed to use the internal event framework, this value was no longer needed for its original purpose. Hence, it was no longer updated with the new/old message counts for a peer. The value was previously removed for Asterisk 10; however, since it was still present in Asterisk 1.8 and still useful for reporting purposes, it was decided to re-add the value. This patch re-adds the 'LastMsgsSent' field in the response to an AMI/CLI 'sip show peer [peer]' command, and makes it so that the value of lastmsgssent is updated appropriately. The value should now display the new/old message counts for a particular peer. (closes issue ASTERISK-17866) Reported by: Steve Davies patches by: ast-17866-rb1272.patch (License #5041 by irroot) Modified slightly for this commit Review: https://reviewboard.asterisk.org/r/1939 ........ Merged revisions 367362 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 367369 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-05-22 17:29 +0000 [r367274-367309] Terry Wilson * main/channel.c, /, include/asterisk/cel.h, main/channel_internal_api.c, include/asterisk/channel.h, main/cel.c, main/asterisk.c: Fix race condition for CEL LINKEDID_END event This patch fixes to situations that could cause the CEL LINKEDID_END event to be missed. 1) During a core stop gracefully, modules are unloaded when ast_active_channels == 0. The LINKDEDID_END event fires during the channel destructor. This means that occasionally, the cel_* module will be unloaded before the channel is destroyed. It seemed generally useful to wait until the refcount of all channels == 0 before unloading, so I added a channel counter and used it in the shutdown code. 2) During a masquerade, ast_channel_change_linkedid is called. It calls ast_cel_check_retire_linkedid which unrefs the linkedid in the linkedids container in cel.c. It didn't ref the new linkedid. Now it does. Review: https://reviewboard.asterisk.org/r/1900/ ........ Merged revisions 367292 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 367299 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, channels/chan_sip.c: Resolve crash in subscribing for MWI notifications ASTOBJ_UNREF sets the variable to NULL after unreffing it, so the variable should definitely not be used after that. To solve this in the two cases that affect subscribing for MWI notifications, we instead save the ref locally, and unref them in the error conditions. (closes issue ASTERISK-19827) Reported by: B. R Review: https://reviewboard.asterisk.org/r/1940/ ........ Merged revisions 367266 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 367267 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-05-21 22:45 +0000 [r367227] Richard Mudgett * main/channel.c: Made ast_queue_hangup() and ast_queue_hangup_with_cause() lock instead of trylock. It made no sense to trylock the channel and then unconditionally lock the channel right after. 2012-05-21 20:35 +0000 [r367189] Kinsey Moore * channels/chan_iax2.c: Make chan_iax2 reject cause code indications correctly If chan_iax2 does not reject the PVT_CAUSE_CODE frames, the cause will not be stored properly. 2012-05-21 20:31 +0000 [r367163-367183] Mark Michelson * include/asterisk/callerid.h, channels/chan_sip.c, main/callerid.c: Revert revision 367163. This should have been committed to my team trunk-digiumphones branch instead of trunk. * include/asterisk/callerid.h, channels/chan_sip.c, main/callerid.c: Add "send to voicemail" Digium phone functionality to Asterisk. This change accommodates two methods by which calls can be directed to a user's voicemail. * Incoming calls can be redirected to any user's voicemail. * Established calls can be blind transferred to any user's voicemail. Digium phones indicate the desire to direct a call to voicemail by using a Diversion header with a reason parameter of "send_to_vm". This patch adds the "send_to_vm" reason as a valid redirecting reason. In addition, chan_sip.c has been modified to update redirecting information on the transferred channel by reading a Diversion header on a REFER request. (closes issue AST-871) Reported by Malcolm Davenport Review: https://reviewboard.asterisk.org/r/1925 2012-05-21 17:39 +0000 [r367124] Terry Wilson * include/asterisk/astobj2.h: Minor documentation change 2012-05-18 19:39 +0000 [r367080] Jonathan Rose * configs/queues.conf.sample, CHANGES, apps/app_queue.c: app_queue: Per Member ringinuse option and deprecation of ignorebusy Adds a number of methods for controlling the setting of 'ringinuse' which is basically the same concept as the old ignorebusy setting, only now the per member setting always controls whether or not the member is actually ringed while in use. A CLI command and a manager action have been added to change a given queue member's ringinuse option while Asterisk is running and the an argument has been added for adding members with deliberately set ringinuse in queues.conf Some effort has been made to ensure compatability with dialplans and databases still referring to 'ignorebusy'. (issue ASTERISK-19536) reported by: Philippe Lindheimer Review: https://reviewboard.asterisk.org/r/1919/ 2012-05-18 17:54 +0000 [r367010-367029] Mark Michelson * channels/chan_dahdi.c, /, main/say.c: Address MISSING_BREAK static analysis reports some more. This addresses core findings 4 and 6. Moises Silva helped me by stating that a break could be safely added to the case where it is added in chan_dahdi.c In say.c, I have added a comment indicating that static analysis complains but that it is currently unknown if this is correct. This fixes all core findings of this type. (closes issue ASTERISK-19662) reported by Matthew Jordan ........ Merged revisions 367027 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 367028 from http://svn.asterisk.org/svn/asterisk/branches/10 * main/tcptls.c, /, channels/chan_sip.c, include/asterisk/tcptls.h: Fix memory leak of SSL_CTX structures in TLS core. SSL_CTX structures were allocated but never freed. This was a bigger issue for clients than servers since new SSL_CTX structures could be allocated for each connection. Servers, on the other hand, typically set up a single SSL_CTX for their lifetime. This is solved in two ways: 1. In __ssl_setup(), if a tcptls_cfg has an ssl_ctx on it, it is freed so that a new one can take its place. 2. A companion to ast_ssl_setup() called ast_ssl_teardown() has been added so that servers can properly free their SSL_CTXs. (issue ASTERISK-19278) ........ Merged revisions 367002 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 367003 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-05-18 15:51 +0000 [r366917-366955] Matthew Jordan * channels/chan_dahdi.c, /, channels/chan_sip.c, funcs/func_odbc.c, main/cli.c: Fix more memory leaks This patch adds to what was fixed in r366880. Specifically, it addresses the following: * chan_sip: dispose of an allocated frame in off nominal code paths in sip_rtp_read * func_odbc: when disposing of an allocated resultset, ensure that any rows that were appended to that resultset are also disposed of * cli: free the created return string buffer in another off nominal code path * chan_dahdi: free a frame that was allocated by the dsp layer if we choose not to process that frame (issue ASTERISK-19665) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1922/ ........ Merged revisions 366944 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 366948 from http://svn.asterisk.org/svn/asterisk/branches/10 * main/netsock2.c, res/res_rtp_asterisk.c, main/pbx.c, res/res_calendar_exchange.c, res/res_calendar_icalendar.c, apps/app_page.c, /, funcs/func_dialgroup.c, channels/chan_sip.c, apps/app_record.c, res/res_calendar_caldav.c, res/res_jabber.c, apps/app_queue.c, channels/chan_iax2.c, main/enum.c, main/editline/term.c, main/config.c, res/res_srtp.c, main/cli.c, main/editline/tokenizer.c, main/data.c, channels/chan_dahdi.c, funcs/func_odbc.c, main/features.c, apps/app_minivm.c, main/editline/readline.c, channels/sip/config_parser.c, main/xmldoc.c, res/res_calendar.c, apps/app_voicemail.c: Fix a variety of memory leaks This patch addresses a number of memory leaks in a variety of modules that were found by a static analysis tool. A brief summary of the changes: * app_minivm: free ast_str objects on off nominal paths * app_page: free the ast_dial object if the requested channel technology cannot be appended to the dialing structure * app_queue: if a penalty rule failed to match any existing rule list names, the created rule would not be inserted and its memory would be leaked * app_read: dispose of the created silence detector in the presence of off nominal circumstances * app_voicemail: dispose of an allocated unique ID field for MWI event un-subscribe requests in off nominal paths; dispose of configuration objects when using the secret.conf option * chan_dahdi: dispose of the allocated frame produced by ast_dsp_process * chan_iax2: properly unref peer in CLI command "iax2 unregister" * chan_sip: dispose of the allocated frame produced by sip_rtp_read's call of ast_dsp_process; free memory in parse unit tests * func_dialgroup: properly deref ao2 object grhead in nominal path of dialgroup_read * func_odbc: free resultset in off nominal paths of odbc_read * cli: free match_list in off nominal paths of CLI match completion * config: free comment_buffer/list_buffer when configuration file load is unchanged; free the same buffers any time they were created and config files were processed * data: free XML nodes in various places * enum: free context buffer in off nominal paths * features: free ast_call_feature in off nominal paths of applicationmap config processing * netsock2: users of ast_sockaddr_resolve pass in an ast_sockaddr struct that is allocated by the method. Failures in ast_sockaddr_resolve could result in the users of the method not knowing whether or not the buffer was allocated. The method will now not allocate the ast_sockaddr struct if it will return failure. * pbx: cleanup hash table traversals in off nominal paths; free ignore pattern buffer if it already exists for the specified context * xmldoc: cleanup various nodes when we no longer need them * main/editline: various cleanup of pointers not being freed before being assigned to other memory, cleanup along off nominal paths * menuselect/mxml: cleanup of value buffer for an attribute when that attribute did not specify a value * res_calendar*: responses are allocated via the various *_request method returns and should not be allocated in the various write_event methods; ensure attendee buffer is freed if no data exists in the parsed node; ensure that calendar objects are de-ref'd appropriately * res_jabber: free buffer in off nominal path * res_musiconhold: close the DIR* object in off nominal paths * res_rtp_asterisk: if we run out of ports, close the rtp socket object and free the rtp object * res_srtp: if we fail to create the session in libsrtp, destroy the temporary ast_srtp object (issue ASTERISK-19665) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1922 ........ Merged revisions 366880 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 366881 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-05-18 14:27 +0000 [r366896] Jonathan Rose * channels/sip/dialplan_functions.c: chan_sip: Fix a small TEST_FRAMEWORK related error that prevents compiling Introduced with r366842, a function call made only with TEST_FRAMEWORK enabled was missing an argument since the function arguments were changed. 2012-05-18 14:21 +0000 [r366843-366888] Kinsey Moore * /, channels/sip/config_parser.c: Reorder and renumber tests appropriately It appears that a patch did not apply properly when adding tests 12 and 13 and test 11 was duplicated. These tests have been reordered and renumbered such that they make sense. ........ Merged revisions 366882 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 366884 from http://svn.asterisk.org/svn/asterisk/branches/10 * main/channel.c: Make the new SIP_CAUSE backend behave more like the original SIP_CAUSE There was a slight discrepancy in the behaviors of the old SIP_CAUSE and the new SIP_CAUSE/HANGUPCAUSE when a channel had been originated and had not yet been answered. This caused the noload_res_srtp_attempt_srtp test to fail since the SIP_CAUSE variable was never actually set. This behavior has been restored. 2012-05-17 16:28 +0000 [r366842] Jonathan Rose * include/asterisk/logger.h, main/channel.c, channels/sip/include/dialog.h, main/pbx.c, channels/chan_sip.c, main/channel_internal_api.c, main/logger.c, include/asterisk/channel.h, CHANGES, channels/sip/include/sip.h, main/cli.c: logger: Adds additional support for call id logging and chan_sip specific stuff This patch improves the handling of call id logging significantly with regard to transfers and adding APIs to better handle specific aspects of logging. Also, changes have been made to chan_sip in order to better handle the creation of callids and to enable the monitor thread to bind itself to a particular call id when a dialog is determined to be related to a callid. It then unbinds itself before returning to normal monitoring. review: https://reviewboard.asterisk.org/r/1886/ 2012-05-17 13:21 +0000 [r366746] Matthew Jordan * channels/chan_dahdi.c, /, res/res_calendar_ews.c: Fix checking bounds of array index after using it; improper sizeof This patch fixes two problems pointed out by a static analysis tool. * In chan_dahdi, when an event is handled the index of the sub channel is first obtained. In very off nominal cases, the method that determines the index can return a negative value. In the event handling code, whether or not the index returned is valid was being checked after that value was used to index into an array. This patch makes it so the value is checked before any indexing is done. * In res_calendar_ews, sizeof was being passed a pointer instead of the struct to determine the amount of memory to allocate. (issue ASTERISK-19651) Reported by: Matt Jordan (closes issue ASTERISK-19671) Reported by: Matt Jordan ........ Merged revisions 366740 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 366741 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-05-16 18:00 +0000 [r366663-366700] Richard Mudgett * include/asterisk/astobj2.h: Remove missed idx parameter to some ao2 global holder macros. * include/asterisk/astobj2.h, tests/test_astobj2.c, main/astobj2.c: Change ao2 global array to ao2 global object holder. Review: https://reviewboard.asterisk.org/r/1921/ 2012-05-15 23:41 +0000 [r366599] Mark Michelson * /, channels/chan_sip.c: Correct misuse of ast_strip_quoted() when getting a Diversion header's reason parameter. The use here was assuming that the pointer would be updated, but the updated string is actually returned by ast_strip_quoted() instead. ........ Merged revisions 366597 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 366598 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-05-15 19:36 +0000 [r366462-366546] Richard Mudgett * channels/chan_local.c: The predial routine must be run on the local;1 channel. When ast_call() operates on a local channel, it copies a lot of things from the local;1 channel to the local;2 channel. This includes among other things, channel variables and party id information. Other reasons it was a bad idea to run predial on the local;2 channel: 1) The channel has not been completely setup. The ast_call() completes the setup. 2) The local;2 caller and connected line party information is opposite to any other channels predial runs on. (And it hasn't been setup yet.) * Partially back out -r366183 by removing the chan_local implementation of the struct ast_channel_tech.pre_call callback. * CHANGES, apps/app_followme.c: Add predial support to FollowMe. Like the new predial feature for Dial. This adds the same b/B options to FollowMe. Review: https://reviewboard.asterisk.org/r/1910/ * channels/chan_local.c: Make chan_local use the API call instead of inlining its own version. 2012-05-14 20:15 +0000 [r366413] Mark Michelson * /, pbx/dundi-parser.c: Fix two more coverity constant expression result findings. These correspond to findings 0 and 1 in the core findings of ASTERISK-19649. After contacting Mark Spencer, he was unsure of what the intent behind these lines of code were, so they are being axed. For Asterisk 1.8 and 10, the output of debugging DUNDi frames will not be changed, but for trunk the "Retry" portion will be omitted since it does not properly distinguish retransmissions from initial frames. (closes issue ASTERISK-19649) Reported by Matthew Jordan ........ Merged revisions 366409 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 366412 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-05-14 19:44 +0000 [r366408] Kinsey Moore * channels/chan_unistim.c, apps/app_dial.c, main/rtp_engine.c, channels/chan_vpb.cc, channels/chan_sip.c, UPGRADE.txt, channels/chan_gtalk.c, channels/chan_console.c, channels/chan_iax2.c, apps/app_queue.c, apps/app_followme.c, channels/chan_oss.c, channels/chan_jingle.c, main/channel.c, channels/chan_phone.c, main/dial.c, channels/chan_misdn.c, channels/chan_skinny.c, funcs/func_frame_trace.c, main/features.c, channels/chan_h323.c, main/file.c, channels/chan_alsa.c, configs/sip.conf.sample, include/asterisk/frame.h, channels/chan_mgcp.c: Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE) This is the starting point for the Asterisk 11: Who Hung Up work and provides a framework which will allow channel drivers to report the types of hangup cause information available in SIP_CAUSE without incurring the overhead of the MASTER_CHANNEL dialplan function. The initial implementation only includes cause generation for chan_sip and does not include cause code translation utilities. This change deprecates SIP_CAUSE and replaces its method of reporting cause codes with the new framework. This change also deprecates the 'storesipcause' option in sip.conf. Review: https://reviewboard.asterisk.org/r/1822/ (Closes issue SWP-4221) 2012-05-14 19:27 +0000 [r366401] Mark Michelson * /, channels/chan_sip.c: Fix broken reinvite glare scenario. To make a long story short, reinvite glares were broken because Asterisk would invert the To and From headers when ACKing a 491 response. The reason was because the initreq of the dialog was being changed to the incoming glared reinvite instead of being set to the outgoing glared reinvite. This change has three parts * In handle_incoming, we never will reject an ACK because it has a to-tag present, even if we think the request may be out of dialog. * In handle_request_invite, we do not change the initreq when receiving a reinvite to which we will respond with a 491. * In handle_request_invite, several superflous settings up pendinginvite have been removed since this is dones automatically by transmit_response_reliable Review: https://reviewboard.asterisk.org/r/1911 ........ Merged revisions 366389 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 366390 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-05-14 13:42 +0000 [r366351] Tzafrir Cohen * configure, configure.ac, autoconf/ast_pkgconfig.m4 (added): Macro AST_PKG_CONFIG_CHECK to use chkconfig AST_PKG_CONFIG_CHECK: Similar to AST_EXT_LIB_CHECK, but simply uses pkg-config data. This simple version only uses pkg-config(1)'s tests. This commit also uses the macro to test for GTK2 and GMIME (instead of the current direct usage of pkg-config). Review: https://reviewboard.asterisk.org/r/1906/ 2012-05-12 00:03 +0000 [r366298] Russell Bryant * /, addons/format_mp3.c: format_mp3: Fix a possible crash in mp3_read(). This patch fixes a potential crash in mp3_read() by not assuming that dbuf has enough data to finish filling up the output buffer. The patch also makes sure that the dbuf state gets reset after we know we read everything out of it already. In passing, this patch includes some other cleanups of this module, including stripping trailing whitespace, formatting fixes based on coding guidelines, and removing a number of unused members from the private state struct. (closes issue ASTERISK-19761) Reported by: Chris Maciejewsk Tested by: Chris Maciejewsk ........ Merged revisions 366296 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 366297 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-05-10 23:49 +0000 [r366183-366242] Richard Mudgett * main/channel.c, /: * Made ast_change_name() hold the channels container lock while changing the channel name. * Eliminate redundant list not empty check in clone_variables(). ........ Merged revisions 366240 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 366241 from http://svn.asterisk.org/svn/asterisk/branches/10 * apps/app_dial.c: Tweak app_dial predial documentation. * apps/app_dial.c, main/channel.c, channels/chan_local.c, include/asterisk/channel.h: Run predial routine on local;2 channel where you would expect. Before this patch, the predial routine executes on the ;1 channel of a local channel pair. Executing predial on the ;1 channel of a local channel pair is of limited utility. Any channel variables set by the predial routine executing on the ;1 channel will not be available when the local channel executes dialplan on the ;2 channel. * Create ast_pre_call() and an associated pre_call() technology callback to handle running the predial routine. If a channel technology does not provide the callback, the predial routine is simply run on the channel. Review: https://reviewboard.asterisk.org/r/1903/ 2012-05-10 20:56 +0000 [r366169] Kinsey Moore * funcs/func_speex.c, main/pbx.c, res/res_calendar_icalendar.c, /, channels/chan_sip.c, funcs/func_lock.c, channels/chan_agent.c, channels/sip/reqresp_parser.c, main/devicestate.c, pbx/dundi-parser.c, channels/chan_iax2.c, channels/iax2-parser.c, main/config.c, res/res_monitor.c, main/channel.c, main/cdr.c, res/ael/pval.c, main/data.c, channels/chan_dahdi.c, main/tcptls.c, main/manager.c, main/features.c, main/app.c, main/event.c, pbx/pbx_dundi.c, res/res_odbc.c, main/xmldoc.c, apps/app_voicemail.c: Resolve FORWARD_NULL static analysis warnings This resolves core findings from ASTERISK-19650 numbers 0-2, 6, 7, 9-11, 14-20, 22-24, 28, 30-32, 34-36, 42-56, 82-84, 87, 89-90, 93-102, 104, 105, 109-111, and 115. Finding numbers 26, 33, and 29 were already resolved. Those skipped were either extended/deprecated or in areas of code that shouldn't be disturbed. (Closes issue ASTERISK-19650) ........ Merged revisions 366167 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 366168 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-05-10 18:35 +0000 [r366126] Jonathan Rose * main/pbx.c, channels/sig_analog.c, /, channels/chan_sip.c, funcs/func_lock.c, main/features.c, main/acl.c, channels/iax2-provision.c, apps/app_queue.c, channels/chan_iax2.c, res/ael/ael.flex, funcs/func_devstate.c, main/asterisk.c, main/xmldoc.c, apps/app_voicemail.c: Coverity Report: Fix issues for error type CHECKED_RETURN for core (issue ASTERISK-19658) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1905/ ........ Merged revisions 366094 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 366106 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-05-10 16:22 +0000 [r366062] Mark Michelson * /, channels/chan_sip.c: Close the proper tcptls_session when session creation fails. (issue AST-998) Reported by: Thomas Arimont Tested by: Thomas Arimont ........ Merged revisions 366052 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 366053 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-05-10 15:57 +0000 [r366007-366051] Jonathan Rose * /, funcs/func_cdr.c, main/features.c, apps/app_disa.c, apps/app_chanspy.c: Coverity Report: Fix issues for error type UNINIT in Core supported modules (issue ASTERISK-19652) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1909/ ........ Merged revisions 366048 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 366049 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, codecs/codec_dahdi.c: Block on frameout if the hardware has enough samples to complete a frame. Fixes some problems with skipping audio in elaborate scenarios involving multiple codecs by making codec_dahdi operate in a more synchronous fashion similar to codec_g729. This change also fixes the use of file conversion tools from Asterisk's CLI. This change may cause the thread responsible for transcoding audio to block briefly (Shaun Ruffell describes this as 'several milliseconds') while waiting for the hardware transcoder. (closes issue ASTERISK-19643) reported by: Shaun Ruffell Patches: 0001-codec_dahdi-Block-on-frameout-the-hardware-has-enoug.patch uploaded by Shaun Ruffell (license 5417) ........ Merged revisions 365989 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 365990 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-05-09 19:26 +0000 [r366002] Tzafrir Cohen * Makefile: pass BUILD_CFLGAS and BUILD_LDFLAGS to menuselect Allow menuselect to get its set of CFLAGS and LDFLAGS through the environment of Make: make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever" Review: https://reviewboard.asterisk.org/r/1907/ 2012-05-09 17:58 +0000 [r365951] Richard Mudgett * configs/followme.conf.sample, apps/app_followme.c: Improve FollowMe accept/decline DTMF string matching. If you hit the wrong DTMF digit trying to accept/decline a FollowMe call, you had to wait for the prompt to repeat to try again. * Make FollowMe compare the last DTMF digits received to the accept/decline matching strings. 2012-05-09 16:36 +0000 [r365913] Mark Michelson * /, channels/chan_sip.c: Prevent sip_pvt refleak when an ast_channel outlasts its corresponding sip_pvt. chan_sip was coded under the assumption that a SIP dialog with an owner channel will always be destroyed after the owner channel has been hung up. However, there are situations where the SIP dialog can time out and auto destruct before the corresponding channel has hung up. A typical example of this would be if the 'h' extension in the dialplan takes a long time to complete. In such cases, __sip_autodestruct() would complain about the dialog being auto destroyed with an owner channel still in place. The problem is that even once the owner channel was hung up, the sip_pvt would still be linked in its ao2_container because nothing would ever unlink it. The fix for this is that if __sip_autodestruct() is called for a sip_pvt that still has an owner channel in place, the destruction is rescheduled for 10 seconds in the future. This will continue until the owner channel is finally hung up. (closes issue ASTERISK-19425) reported by David Cunningham Patches: ASTERISK-19425.patch uploaded by Mark Michelson (License #5049) (closes issue ASTERISK-19455) reported by Dean Vesvuio Tested by Dean Vesvuio ........ Merged revisions 365896 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 365898 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-05-09 02:35 +0000 [r365766-365856] Richard Mudgett * configs/followme.conf.sample, UPGRADE.txt, apps/app_followme.c: Keep answered FollowMe calls until call accepted or last step times out. * apps/app_followme.c: Put winning FollowMe outgoing call on hold if the caller put it on hold. The FollowMe caller call leg is usually answered and listening to MOH. The caller could put the call on hold while FollowMe is looking for a winner. The winning outgoing call is now immediately placed on hold if the caller has put the call on hold before the winning call was selected. * apps/app_followme.c: Restructure how the FollowMe outgoing channel list is handled. * apps/app_followme.c: Addendum to -r365766. Since it is no longer allocated. * apps/app_followme.c: Make FollowMe findmeexec() put the list head on the stack instead of mallocing it. Why this tiny struct was malloced instead of the 28k struct in the last change is beyond me. Just doing my part to help stamp out sillyness. 2012-05-08 21:46 +0000 [r365751] Sean Bright * apps/app_externalivr.c: Add interrupt ('I') command to ExternalIVR. Sending the 'I' command from an external process will cause the current playlist to be cleared, including stopping any audio file that is currently playing. This is useful when you want to interrupt audio playback only when specific DTMF is entered by the caller. 2012-05-08 21:41 +0000 [r365633-365749] Richard Mudgett * apps/app_followme.c: Make FollowMe app_exec() not declare a 28k struct on the stack. Helping to stamp out stack abuse. * apps/app_followme.c: Simplify findmeexec() parameter passing. * /, apps/app_followme.c: * Fix FollowMe memory leak on error paths in app_exec(). * Fix FollowMe leaving recorded caller name file on error paths in app_exec(). * Use correct buffer dimension define in struct fm_args.namerecloc[]. This fixes unexpected namerecloc filename length restriction. ........ Merged revisions 365692 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 365701 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, apps/app_followme.c: * Fix accept/decline DTMF buffer overwrite in FollowMe. * Made use MAX_YN_STRING define to make all accept/decline DTMF buffers the same size. Just using 20 isn't good enough when someone didn't get the memo. * Fix stupid use of a global variable in FollowMe. (ynlongest) * Fix bit field declarations in FollowMe. ........ Merged revisions 365631 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 365632 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-05-08 15:57 +0000 [r365576] Mark Michelson * /, channels/chan_sip.c: Send more accurate identification information in dialog-info SIP NOTIFYs. This uses the calling channel's caller ID and connected line information to populate the remote and local identities in the dialog-info NOTIFY when an extension is ringing. There is a bit of an oddity here, and that is that we seed the remote target with the To header of the outbound call rather than the from header. This is because it was reported that seeding with the from header caused hints to be broken with certain SNOM devices. A comment has been added to the code to explain this. (closes issue ASTERISK-16735) reported by Maciej Krajewski patches: local_remote_hint2.diff uploaded by Mark Michelson (license #5049) 16735_tweak1.diff uploaded by Mark Michelson (license #5049) Tested by Niccolo Belli ........ Merged revisions 365574 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 365575 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-05-07 20:08 +0000 [r365532] Richard Mudgett * main/features.c: Change comment to use local channel name designators in features.c 2012-05-07 18:58 +0000 [r365480] Matthew Jordan * main/pbx.c, apps/app_voicemail.c: Fix channel opaquification slip-up in r365477 Those channels are opaque now... 2012-05-07 18:51 +0000 [r365479] Richard Mudgett * /, tests/test_config.c: Fix type punned compiler warning in test_config.c ........ Merged revisions 365476 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 365478 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-05-07 18:42 +0000 [r365477] Matthew Jordan * main/pbx.c, /, apps/app_voicemail.c: Support VoiceMail d() option when extension does not exist in channel's context The VoiceMail d([c]) option is documented to accept digits for a new extension in context , if played during the greeting. This option works fine if the extension being redirected to has an extension with the same initial digit in the channel's current context. If that digit did not happen to exist in some extension, a dialplan match would fail and the user would not be redirected. This patch fixes it such that if the option is used, the extensions are matched in that context as opposed to the caller's original context. (closes issue ASTERISK-18243) Reported by: mjordan Tested by: mjordan Review: https://reviewboard.asterisk.org/r/1892 ........ Merged revisions 365474 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 365475 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-05-04 22:17 +0000 [r365400] Kinsey Moore * /, channels/chan_sip.c, funcs/func_aes.c, main/features.c, apps/app_followme.c, channels/chan_iax2.c, channels/sip/config_parser.c, pbx/pbx_config.c, apps/app_chanspy.c, apps/app_stack.c, main/config.c, apps/app_voicemail.c: Fix many issues from the NULL_RETURNS Coverity report Most of the changes here are trivial NULL checks. There are a couple optimizations to remove the need to check for NULL and outboundproxy parsing in chan_sip.c was rewritten to avoid use of strtok. Additionally, a bug was found and fixed with the parsing of outboundproxy when "outboundproxy=," was set. (Closes issue ASTERISK-19654) ........ Merged revisions 365398 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 365399 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-05-04 17:38 +0000 [r365356] Richard Mudgett * channels/chan_local.c, /: Fix local channel chains optimizing themselves out of a call. * Made chan_local.c:check_bridge() check the return value of ast_channel_masquerade(). In long chains of local channels, the masquerade occasionally fails to get setup because there is another masquerade already setup on an adjacent local channel in the chain. * Made the outgoing local channel (the ;2 channel) flush one voice or video frame per optimization attempt. * Made sure that the outgoing local channel also does not have any frames in its queue before the masquerade. * Made do the masquerade immediately to minimize the chance that the outgoing channel queue does not get any new frames added and thus unconditionally flushed. * Made block indication -1 (Stop tones) event when the local channel is going to optimize itself out. When the call is answered, a chain of local channels pass down a -1 indication for each bridge. This blizzard of -1 events really slows down the optimization process. (closes issue ASTERISK-16711) Reported by: Alec Davis Tested by: rmudgett, Alec Davis Review: https://reviewboard.asterisk.org/r/1894/ ........ Merged revisions 365313 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 365320 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-05-04 15:52 +0000 [r365300] Mark Michelson * res/res_rtp_asterisk.c, /: Fix core FINDING 2, FINDING 3, and FINDING 4 from Coverity's CONSTANT_EXPRESSION_RESULT report. These three all are in RTP code that attempts to print the number of sequence number cycles in an RTCP RR report. The code was masking out the upper 16 bits and then shifting the number right by 16 bits. This led to an all zero result in all cases. The fix is to do the shift without the bit masking. (issue ASTERISK-19649) ........ Merged revisions 365298 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 365299 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-05-03 19:36 +0000 [r365248] Michael L. Young * tests/test_security_events.c: Update security events unit tests The security events framework API was changed in Asterisk 10 but the unit tests were not updated at the same time. This patch does the following: * Adds two more security events that were added to the API * Add challenge, received_challenge and received_hash in the inval_password security event unit test (Closes issue ASTERISK-19760) Reported by: Michael L. Young Tested by: Michael L. Young Patches: issue-asterisk-19760-trunk.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/1897/ 2012-05-03 18:43 +0000 [r365213] Sean Bright * CHANGES: Update documentation references in CHANGES to reflect the correct pages on the wiki. The current CHANGES file refers to doc/ in many places and those files no longer exist. 2012-05-03 15:05 +0000 [r365161] Alexandr Anikin * addons/ooh323c/src/ooh323.c, /, addons/ooh323c/src/h323/H323-MESSAGES.h, addons/ooh323c/src/h323/H323-MESSAGESEnc.c: Fix warning of Coverity Static analysis, change H225ProtocolIdentifier from value to pointer per functions that use this. (close issue ASTERISK-19670) Reported by: Matt Jordan Patches: ASTERISK-19670.patch (License #5415) ........ Merged revisions 365159 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 365160 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-05-03 14:47 +0000 [r365158] Sean Bright * apps/app_externalivr.c, CHANGES: Add IPv6 support to ExternalIVR. Review: https://reviewboard.asterisk.org/r/1896/ 2012-05-03 14:35 +0000 [r365157] Alexandr Anikin * /, addons/ooh323c/src/ooq931.c: Fix coverity static analysis warning, allocate full ie structure instead of without data buffer (close issue ASTERISK-19674) Reported by: Matt Jordan Patches: ASTERISK-19674.patch (License #5415) ........ Merged revisions 365143 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 365155 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-05-02 17:43 +0000 [r365084] Terry Wilson * channels/chan_local.c, /, main/cel.c: Multiple revisions 365006,365068 ........ r365006 | twilson | 2012-05-02 10:49:03 -0500 (Wed, 02 May 2012) | 12 lines Fix a CEL LINKEDID_END race and local channel linkedids This patch has the ;2 channel inherit the linkedid of the ;1 channel and fixes the race condition by no longer scanning the channel list for "other" channels with the same linkedid. Instead, cel.c has an ao2 container of linkedid strings and uses the refcount of the string as a counter of how many channels with the linkedid exist. Not only does this eliminate the race condition, but it also allows us to look up the linkedid by the hashed key instead of traversing the entire channel list. Review: https://reviewboard.asterisk.org/r/1895/ ........ r365068 | twilson | 2012-05-02 12:02:39 -0500 (Wed, 02 May 2012) | 11 lines Don't leak a ref if out of memory and can't link the linkedid If the ao2_link fails, we are most likely out of memory and bad things are going to happen. Before those bad things happen, make sure to clean up the linkedid references. This patch also adds a comment explaining why linkedid can't be passed to both local channel allocations and combines two ao2_ref calls into 1. Review: https://reviewboard.asterisk.org/r/1895/ ........ Merged revisions 365006,365068 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 365083 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-05-02 15:59 +0000 [r365011] Jason Parker * channels/chan_sip.c: Save the address on which a MESSAGE was received, so it can be used in MESSAGE() This is useful in cases where chan_sip may be listening on multiple addresses. 2012-05-02 02:51 +0000 [r364966] Matthew Jordan * /, main/audiohook.c: Only log a failure to get read/write samples from factories if it didn't happen In audiohook_read_frame_both, anytime samples are obtained from the read/write factories a debug statement is logged stating that samples were not obtained from the factories. This statement used to only occur if option_debug was turned on and no samples were obtained; in some refactoring when the option_debug statement was removed, the "else" clause was removed as well. This patch makes it so that those debug log statements only occur if the condition leading up to them actually happened. ........ Merged revisions 364965 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-05-01 23:23 +0000 [r364915] Mark Michelson * channels/chan_sip.c: Remove a function that has been marked unused since Asterisk 1.6.0. The reason I'm removing this is that Coverity reported a STRAY_SEMICOLON issue here. Since the function has been unused for so long, I just elected to remove it altogether. (closes issue ASTERISK-19660) 2012-05-01 23:21 +0000 [r364910] Richard Mudgett * /, main/astobj2.c: Fixed __ao2_ref() validating user_data twice. (closes issue ASTERISK-19755) Reported by: Gunther Kelleter Patches: ao2_ref.patch (license #6372) patch uploaded by Gunther Kelleter ........ Merged revisions 364902 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364903 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-05-01 23:11 +0000 [r364901] Mark Michelson * /, funcs/func_volume.c: Fix Coverity-reported ARRAY_VS_SINGLETON error. As it turned out, this wasn't a huge deal. We were calling ast_app_parse_options() for a set of options of which none took arguments. The proper thing to do for this case is to pass NULL for the "args" parameter here. We were instead passing a seemingly-randomly chosen char * from the function. While this would never get written to, you can rest assured things would have gotten bad had new options (which took arguments) been added to func_volume. (closes issue ASTERISK-19656) ........ Merged revisions 364899 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364900 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-05-01 22:00 +0000 [r364846] Richard Mudgett * channels/chan_local.c, /: * Fix error path resouce leak in local_request(). * Restructure local_request() to reduce indentation. ........ Merged revisions 364840 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364845 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-05-01 21:49 +0000 [r364844] Jason Parker * main/manager.c, /: Prevent a potential crash when using manager hooks. Found by me while poking at DPMA-127. ........ Merged revisions 364841 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364842 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-05-01 19:10 +0000 [r364788] Kinsey Moore * /, apps/app_confbridge.c: Play conf-placeintoconf message to the correct channel Correct the code in app_confbridge to play the conf-placeintoconf message to the marked user entering the bridge instead of to the conference while the marked user hears silence. (closes issue ASTERISK-19641) Reported-by: Mark A Walters ........ Merged revisions 364786 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364787 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-05-01 18:29 +0000 [r364785] Jonathan Rose * /, main/app.c: Fix bad check in voicemail functions for ast_inboxcount2_func Check looks for ast_inboxcount_func instead of ast_inboxcount2_func on ast_inboxcount2_func calls. (closes issue ASTERISK-19718) Reported by: Corey Farrell Patches: ast_app_inboxcount2-null-refcheck.patch uploaded by Corey Farrell (license 5909) ........ Merged revisions 364769 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364777 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-30 19:51 +0000 [r364708] Mark Michelson * /, channels/chan_sip.c: Revert revision 360862. Revision 360862 was intended to improve identities sent in dialog-info NOTIFY requests. Some users reported that hint became broken once this was done. It's not clear exactly what part of the patch has caused this regression, but broken hints are bad. For now, this revision is being reverted so that the next releases of Asterisk do not have bad behavior in them. The original reported issue will have to be fixed differently in the next version of Asterisk. (issue ASTERISK-16735) ........ Merged revisions 364706 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364707 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-30 17:17 +0000 [r364654] Mark Murawki * /, main/logger.c: Merged revisions 364635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r364635 | markm | 2012-04-30 11:51:12 -0400 (Mon, 30 Apr 2012) | 10 lines Sanatize result from bfd_find_nearest_line (BETTER_BACKTRACES) bfd_find_nearest_line can possibly set file to null resulting in a crash when strrchr(file) runs (closes issue ASTERISK-19815) Reported by Mark Murawski Tested by Mark Murawski ........ ........ Merged revisions 364650 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-30 16:59 +0000 [r364652] Alexandr Anikin * /, addons/ooh323cDriver.c: Fix use freed pointer in return value from call thread (issue ASTERISK-19663) Reported by: Matt Jordan Patches: ASTERISK-19663-ooh323.patch (License #5415) ........ Merged revisions 364649 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364651 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-29 19:50 +0000 [r364580] Matthew Jordan * formats/format_ilbc.c, /, formats/format_sln.c, formats/format_vox.c, formats/format_wav.c, formats/format_pcm.c, formats/format_g723.c, formats/format_h263.c, formats/format_h264.c, formats/format_wav_gsm.c, formats/format_siren14.c, formats/format_gsm.c, formats/format_g719.c, formats/format_siren7.c, formats/format_g729.c: Fix error that caused truncate operations to fail Another very inappropriate placement of a ')' (again introduced in r362151) caused the various truncate operations to attempt to truncate the sound file at a position of '0'. (issue ASTERISK-19655) Reported by: Matt Jordan (issue ASTERISK-19810) Reported by: colbec ........ Merged revisions 364578 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364579 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-29 02:23 +0000 [r364537] Michael L. Young * /, apps/confbridge/conf_config_parser.c: Fix configuring custom sound_leader_has_left in confbridge.conf The configuration option to specify a custom sound_leader_has_left file for a conference bridge was not being parsed. This patch fixes it so that a custom sound file will now be used. (closes issue ASTERISK-19771) Reported by: Pawel Kuzak Tested by: Pawel Kuzak, Michael L. Young Patches: leaderhasleft_sound.dpatch uploaded by Pawel Kuzak (license 6380) Review: https://reviewboard.asterisk.org/r/1884/ ........ Merged revisions 364536 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-28 20:24 +0000 [r364500] Joshua Colp * channels/chan_sip.c, configs/sip.conf.sample, CHANGES, channels/sip/include/sip.h: Add support for lightweight NAT keepalive. If enabled using the keepalive option in sip.conf a small packet will be sent at a regular interval to keep the NAT mapping open. This is lightweight as the remote side does not need to parse and handle a SIP message. (closes issue AST-783) Review: https://reviewboard.asterisk.org/r/1756/ 2012-04-28 01:33 +0000 [r364437-364462] Russell Bryant * main/md5.c: md5: supress some compiler warnings. md5.c: In function ‘MD5Final’: md5.c:154:2: error: dereferencing type-punned pointer will break strict-aliasing rules [-Werror=strict-aliasing] md5.c:155:2: error: dereferencing type-punned pointer will break strict-aliasing rules [-Werror=strict-aliasing] There is an md5 unit test and it still passes. * configure, include/asterisk/autoconfig.h.in, res/res_corosync.c, configure.ac: res_corosync: Fix build against corosync 2.0. * apps/app_minivm.c: app_minivm: Fix a couple compiler warnings. The warnings were about argv[0] being used uninitialized, which is correct. Just remove setting username to this value, since username is set again before it actually gets used. * main/features.c, CHANGES: features: Add FEATURE() and FEATUREMAP() functions. Add two new dialplan functions: FEATURE() and FEATUREMAP(). FEATURE() lets you set some of the configuration options from the [general] section of features.conf on a per-channel basis. FEATUREMAP() lets you customize the key sequence used to activate built-in features, such as blindxfer, and automon. See the built-in documentation for details. Review: https://reviewboard.asterisk.org/r/1871/ 2012-04-28 00:31 +0000 [r364436] Richard Mudgett * apps/app_dial.c, CHANGES: PreDial - Ability to run dialplan on callee and caller channels before Dial. Thanks to Mark Murawski for the initial patch and feature definition. (closes issue ASTERISK-19548) Reported by: Mark Murawski Review: https://reviewboard.asterisk.org/r/1878/ Review: https://reviewboard.asterisk.org/r/1229/ 2012-04-27 22:54 +0000 [r364397] Terry Wilson * /, tests/test_config.c (added), main/config.c: Multiple revisions 364365,364369 ........ r364365 | twilson | 2012-04-27 17:31:01 -0500 (Fri, 27 Apr 2012) | 11 lines Fix ast_parse_arg numeric type range checking and add tests ast_parse_arg wasn't checking for strto* parse errors or limiting the results by the actual range of the numeric types. This patch fixes that and adds unit tests as well. Review: https://reviewboard.asterisk.org/r/1879/ ........ Merged revisions 364340 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r364369 | twilson | 2012-04-27 17:33:10 -0500 (Fri, 27 Apr 2012) | 2 lines Add missing test_config.c ........ Merged revisions 364365,364369 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-27 22:11 +0000 [r364343] Mark Michelson * /, channels/chan_sip.c: Don't attempt to make use of the dynamic_exclude_static ACL if DNS lookup fails. (closes issue ASTERISK-18321) Reported by Dan Lukes Patches: ASTERISK-18321.patch by Mark Michelson (license #5049) ........ Merged revisions 364341 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364342 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-27 19:30 +0000 [r364287] Matthew Jordan * /, include/asterisk/time.h: Prevent overflow in calculation in ast_tvdiff_ms on 32-bit machines The method ast_tvdiff_ms attempts to calculate the difference, in milliseconds, between two timeval structs, and return the difference in a 64-bit integer. Unfortunately, it assumes that the long tv_sec/tv_usec members in the timeval struct are large enough to hold the calculated values before it returns. On 64-bit machines, this might be the case, as a long may be 64-bits. On 32-bit machines, however, a long may be less (32-bits), in which case, the calculation can overflow. This overflow caused significant problems in MixMonitor, which uses the method to determine if an audio factory, which has not presented audio to an audiohook, is merely late in providing said audio or will never provide audio. In an overflow situation, the audiohook would incorrectly determine that an audio factory that will never provide audio is merely late instead. This led to situations where a MixMonitor never recorded any audio. Note that this happened most frequently when that MixMonitor was started by the ConfBridge application itself, or when the MixMonitor was attached to a Local channel. (issue ASTERISK-19497) Reported by: Ben Klang Tested by: Ben Klang Patches: 32-bit-time-overflow-10-2012-04-26.diff (license #6283) by mjordan (closes issue ASTERISK-19727) Reported by: Mark Murawski Tested by: Michael L. Young Patches: 32-bit-time-overflow-2012-04-27.diff (license #6283) by mjordan) (closes issue ASTERISK-19471) Reported by: feyfre Tested by: feyfre (issue ASTERISK-19426) Reported by: Johan Wilfer Review: https://reviewboard.asterisk.org/r/1889/ ........ Merged revisions 364277 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364285 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-27 18:59 +0000 [r364260] Kinsey Moore * /, channels/chan_sip.c: Allow SIP pvts involved in Replaces transfers to fall out of reference sooner Unref the SIP pvt stored in the refer structure as soon as it is no longer needed so that the pvt and associated file descriptors can be freed sooner. This change makes a reference decrement unnecessary in code that handles SIP BYE/Also transfers which should not touch the reference anyway. (Closes issue ASTERISK-19579) Reported by: Maciej Krajewski Tested by: Maciej Krajewski ........ Merged revisions 364258 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364259 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-27 14:45 +0000 [r364205] Matthew Jordan * /, channels/chan_sip.c: Allow for reloading SRTP crypto keys within the same SIP dialog As a continuation of the patch in r356604, which allowed for the reloading of SRTP keys in re-INVITE transfer scenarios, this patch addresses the more common case where a new key is requested within the context of a current SIP dialog. This can occur, for example, when certain phones request a SIP hold. Previously, once a dialog was associated with an SRTP object, any subsequent attempt to process crypto keys in any SDP offer - either the current one or a new offer in a new SIP request - were ignored. This patch changes this behavior to only ignore subsequent crypto keys within the current SDP offer, but allows future SDP offers to change the keys. (issue ASTERISK-19253) Reported by: Thomas Arimont Tested by: Thomas Arimont Review: https://reviewboard.asteriskorg/r/1885/ ........ Merged revisions 364203 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364204 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-27 12:58 +0000 [r364164] Stefan Schmidt * res/res_calendar_icalendar.c, /, res/res_calendar_caldav.c: fix a wrong behavior of alarm timezones in caldav and icalendar when an alarm doesnt use utc. This change uses the same timezone from the start time. ........ Merged revisions 364163 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-26 21:11 +0000 [r364082-364110] Richard Mudgett * /, apps/app_directed_pickup.c: Update Pickup application documentation. (With feeling this time.) ........ Merged revisions 364108 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364109 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, main/features.c: Fix DTMF atxfer running h exten after the wrong bridge ends. When party B does an attended transfer of party A to party C, the attending bridge between party B and C should not be running an h exten when the bridge ends. Running an h exten now sets a softhangup flag to ensure that an AGI will run in dead AGI mode. * Set the AST_FLAG_BRIDGE_HANGUP_DONT on the party B channel for the attending bridge between party B and C. (closes issue AST-870) (closes issue ASTERISK-19717) Reported by: Mario (closes issue ASTERISK-19633) Reported by: Andrey Solovyev Patches: jira_asterisk_19633_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett, Andrey Solovyev, Mario ........ Merged revisions 364060 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364065 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-26 19:33 +0000 [r364048] Terry Wilson * /, main/asterisk.c: Add more constness to the end_buf pointer in the netconsole issue ASTERISK-18308 Review: https://reviewboard.asterisk.org/r/1876/ ........ Merged revisions 364046 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364047 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-26 13:59 +0000 [r363989] Olle Johansson * apps/app_queue.c: Code formatting fixes. 2012-04-26 13:31 +0000 [r363988] Kinsey Moore * /, channels/chan_sip.c: Fix reference leaks involving SIP Replaces transfers The reference held for SIP blind transfers using the Replaces header in an INVITE was never freed on success and also failed to be freed in some error conditions. This caused a file descriptor leak since the RTP structures in use at the time of the transfer were never freed. This reference leak and another relating to subscriptions in the same code path have now been corrected. (closes issue ASTERISK-19579) ........ Merged revisions 363986 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363987 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-26 09:48 +0000 [r363936] Alec L Davis * /, channels/chan_sip.c: chan_sip: [general] maxforwards, not checked for a value greater than 255 The peer maxforwards is checked for both '< 1' and '> 255', but the default 'maxforwards' in the [general] section is only checked for '< 1' alecdavis (license 585) Reported by: alecdavis Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1888/ ........ Merged revisions 363934 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363935 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-26 03:12 +0000 [r363689-363877] Richard Mudgett * /, apps/app_directed_pickup.c: Update Pickup application documentation. (Even better) ........ Merged revisions 363875 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363876 from http://svn.asterisk.org/svn/asterisk/branches/10 * apps/app_directed_pickup.c: * Put more information in pickup_exec() LOG_NOTICE. * Delay duplicating a string on the stack in pickup_exec(). * /, apps/app_directed_pickup.c: Update Pickup application documentation. ........ Merged revisions 363788 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363789 from http://svn.asterisk.org/svn/asterisk/branches/10 * channels/chan_dahdi.c, /, channels/sig_pri.c: Make DAHDISendCallreroutingFacility wait 5 seconds for a reply before disconnecting the call. Some switches may not handle the call-deflection/call-rerouting message if the call is disconnected too soon after being sent. Asteisk was not waiting for any reply before disconnecting the call. * Added a 5 second delay before disconnecting the call to wait for a potential response if the peer does not disconnect first. (closes issue ASTERISK-19708) Reported by: mehdi Shirazi Patches: jira_asterisk_19708_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett ........ Merged revisions 363730 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363734 from http://svn.asterisk.org/svn/asterisk/branches/10 * channels/sig_pri.h, channels/chan_dahdi.c, /, channels/sig_pri.c: Clear ISDN channel resetting state if the peer continues to use it. Some ISDN switches occasionally fail to send a RESTART ACKNOWLEDGE in response to a RESTART request. * Made the second SETUP received after sending a RESTART request clear the channel resetting state as if the peer had sent the expected RESTART ACKNOWLEDGE before continuing to process the SETUP. The peer may not be sending the expected RESTART ACKNOWLEDGE. (issue ASTERISK-19608) (issue AST-844) (issue AST-815) Patches: jira_ast_815_v1.8.patch (license #5621) patch uploaded by rmudgett (modified) ........ Merged revisions 363687 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363688 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-25 13:57 +0000 [r363480-363637] Olle Johansson * apps/app_queue.c: Add documentation Thanks Tilghman! * apps/app_queue.c: Formatting changes only * apps/app_followme.c, apps/app_queue.c: Use the DEFINED value for musicclass length. For some reason, features.c has it's own definition. Should propably be fixed too. * main/channel.c, configs/asterisk.conf.sample, CHANGES, include/asterisk/options.h, main/asterisk.c: Make it possible to change the minimum DTMF duration in asterisk.conf Asterisk has a setting for the minimum allowed DTMF. If we get shorter DTMF tones, these will be changed to the minimum on the outbound call leg. (closes issue ASTERISK-19772) Review: https://reviewboard.asterisk.org/r/1882/ Reported by: oej Tested by: oej Patches by: oej Thanks to the reviewers. 1.8 branch for this patch: agave-dtmf-duration-asterisk-conf-1.8 * main/say.c: Formatting fixes Developer guidelines are important. * main/channel.c: Formatting fixes Found a small amount of curly brackets in my hotel room here in Denmark. I hereby donate them to the Asterisk project. 2012-04-25 01:26 +0000 [r363377-363430] Richard Mudgett * /, main/features.c: Fix recalled party B feature flags for a failed DTMF atxfer. 1) B calls A with Dial option T 2) B DTMF atxfer to C 3) B hangs up 4) C does not answer 5) B is called back 6) B answers 7) B cannot initiate transfers anymore * Add dial features datastore to recalled party B channel that is a copy of the original party B channel's dial features datastore. * Extracted add_features_datastore() from add_features_datastores(). * Renamed struct ast_dial_features features_caller and features_callee members to my_features and peer_features respectively. These better names eliminate the need for some explanatory comments. * Simplified code accessing the struct ast_dial_features datastore. (closes issue ASTERISK-19383) Reported by: lgfsantos ........ Merged revisions 363428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363429 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, main/features.c: Hangup affected channel in error paths of bridge_call_thread(). ........ Merged revisions 363375 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363376 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-24 17:52 +0000 [r363335] Terry Wilson * /, main/asterisk.c: OpenBSD doesn't have rawmemchr, use strchr (closes issue ASTERISK-19758) Reported by: Barry Miller Tested by: Terry Wilson Patches: 362758-diff uploaded by Barry Miller (license 5434) ........ Merged revisions 362868 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362869 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-23 17:05 +0000 [r363269] Richard Mudgett * apps/app_dial.c, apps/app_queue.c: Make app_dial and app_queue use new macro and gosub calls. * Simplify some code in app_dial and app_queue by calling ast_app_exec_macro() and ast_app_exec_sub(). * Fix minor locking issue in app_dial for post-answer macro/gosub MACRO/GOSUB_RESULT=GOTO: handling. 2012-04-23 16:08 +0000 [r363215] Tilghman Lesher * /, main/astfd.c: On some platforms, O_RDONLY is not a flag to be checked, but merely the absence of O_RDWR and O_WRONLY. The POSIX specification does not mandate how these 3 flags must be specified, only that one of the three must be specified in every call. ........ Merged revisions 363209 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363212 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-23 14:48 +0000 [r363159] Jonathan Rose * main/manager.c, /: AST-2012-004: Fix an error that allows AMI users to run shell commands sans authorization. As detailed in the advisory, AMI users without write authorization for SYSTEM class AMI actions were able to run system commands by going through other AMI commands which did not require that authorization. Specifically, GetVar and Status allowed users to do this by setting their variable/s options to the SHELL or EVAL functions. Also, within 1.8, 10, and trunk there was a similar flaw with the Originate action that allowed users with originate permission to run MixMonitor and supply a shell command in the Data argument. That flaw is fixed in those versions of this patch. (closes issue ASTERISK-17465) Reported By: David Woolley Patches: 162_ami_readfunc_security_r2.diff uploaded by jrose (license 6182) 18_ami_readfunc_security_r2.diff uploaded by jrose (license 6182) 10_ami_readfunc_security_r2.diff uploaded by jrose (license 6182) ........ Merged revisions 363117 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........ Merged revisions 363141 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363156 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-23 14:10 +0000 [r363105-363108] Matthew Jordan * /, channels/chan_sip.c: AST-2012-006: Fix crash in UPDATE handling when no channel owner exists If Asterisk receives a SIP UPDATE request after a call has been terminated and the channel has been destroyed but before the SIP dialog has been destroyed, a condition exists where a connected line update would be attempted on a non-existing channel. This would cause Asterisk to crash. The patch resolves this by first ensuring that the SIP dialog has an owning channel before attempting a connected line update. If an UPDATE request is received and no channel is associated with the dialog, a 481 response is sent. (closes issue ASTERISK-19770) Reported by: Thomas Arimont Tested by: Matt Jordan Patches: ASTERISK-19278-2012-04-16.diff uploaded by Matt Jordan (license 6283) ........ Merged revisions 363106 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363107 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, channels/chan_skinny.c: AST-2012-005: Fix remotely exploitable heap overflow in keypad button handling When handling a keypad button message event, the received digit is placed into a fixed length buffer that acts as a queue. When a new message event is received, the length of that buffer is not checked before placing the new digit on the end of the queue. The situation exists where sufficient keypad button message events would occur that would cause the buffer to be overrun. This patch explicitly checks that there is sufficient room in the buffer before appending a new digit. (closes issue ASTERISK-19592) Reported by: Russell Bryant ........ Merged revisions 363100 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........ Merged revisions 363102 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363103 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-21 11:45 +0000 [r363045-363046] Russell Bryant * res/res_corosync.c: res_corosync: Recover if corosync gets restarted. If corosync gets restarted while Asterisk is running, automatically recover. * res/res_corosync.c: res_corosync: reimplement "corosync show members" command. Reimplement the "corosync show members" CLI command using a CPG iterator instead of the cpg_membership_get API call. This will also show all CPG members, including those in groups other than 'asterisk', which may be useful at some point for debugging purposes. 2012-04-21 01:46 +0000 [r362920-362999] Richard Mudgett * apps/app_dial.c, /: Update app_dial M and U option GOTO return value documentation. ........ Merged revisions 362997 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362998 from http://svn.asterisk.org/svn/asterisk/branches/10 * include/asterisk/app.h, main/app.c, apps/app_stack.c: Fix connected-line/redirecting interception gosubs executing more than intended. * Redo ast_app_run_sub()/ast_app_exec_sub() to use a known return point so execution will stop after the routine returns there. (s@gosub_virtual_context:1) * Create ast_app_exec_macro() and ast_app_exec_sub() to run the macro and gosub application respectively with the parameter string already created. * main/rtp_engine.c: Move debug message in ast_rtp_instance_early_bridge_make_compatible(). Move debug message in ast_rtp_instance_early_bridge_make_compatible() to be output when what it states has actually happened. 2012-04-20 16:50 +0000 [r362919] Michael L. Young * /, main/event.c: Add missing payload type to events API The Security Events Framework API was changed while adding the generation of security events in chan_sip. A payload type and name was missed from being added to struct ie_maps. (closes issue ASTERISK-19759) Reported by: Michael L. Young Patches: issue-asterisk-19759.diff uploaded by Michael L. Young (license 5026) ........ Merged revisions 362918 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-20 16:23 +0000 [r362867-362888] Richard Mudgett * apps/app_dial.c, channels/chan_dahdi.c, channels/chan_local.c, channels/chan_misdn.c, main/rtp_engine.c: Use ast_channel_lock_both() where it was inlined before. The CHANNEL_DEADLOCK_AVOIDANCE() feature of preserving where the channel lock was originally obtained is overkill where ast_channel_lock_both() was inlined. * main/pbx.c: * Add more information to some messages in __ast_pbx_run(). * Simplify some dialplan priority setting code in ast_explicit_goto() because of opaquification. 2012-04-20 14:50 +0000 [r362817] Terry Wilson * /, apps/app_speech_utils.c: Document Speech* apps hangup on failure and suggest TryExec The Speech API apps return -1 on failure, which will hang up the channel. This may not be desirable behavior for some, but it isn't something that can be changed without breaking people's dialplans or writing an option to all of the Speech apps that does what TryExec already does. This patch documents the hangup behavior of the apps, and suggests TryExec as the solution. (closes issue AST-813) ........ Merged revisions 362815 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362816 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-20 00:57 +0000 [r362779] Richard Mudgett * main/channel.c, UPGRADE.txt, include/asterisk/channel.h, CHANGES, channels/sig_pri.c, funcs/func_callerid.c: Add original party id and reason support. ISDN ETSI PTP and Q.SIG (And SS7 in future) have support for reporting who was the original redirecting party of a call. * Added support for the original redirecting party and reason to the REDIRECTING function and the system core as well as to the stubbed locations in sig_pri.c. Review: https://reviewboard.asterisk.org/r/1829/ 2012-04-19 22:01 +0000 [r362731] Walter Doekes * funcs/func_version.c, /: Fix documentation for ${VERSION(ASTERISK_VERSION_NUM)}. ........ Merged revisions 362729 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362730 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-19 21:14 +0000 [r362682] Michael L. Young * /, tests/test_linkedlists.c, tests/test_poll.c: Add leading and trailing backslashes A couple of unit tests did not have have leading or trailing backslashes when setting their test category resulting in a warning message being displayed. Added the backslash where needed. ........ Merged revisions 362680 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362681 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-19 21:01 +0000 [r362679] Richard Mudgett * /, configs/queues.conf.sample: Update membermacro and membergosub documentation in queues.conf.sample. ........ Merged revisions 362677 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362678 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-19 19:05 +0000 [r362635] Terry Wilson * addons/chan_ooh323.c, apps/app_alarmreceiver.c, channels/iax2-provision.c, res/snmp/agent.c: Convert some strncpys to ast_copy_string Review: https://reviewboard.asterisk.org/r/1732/ 2012-04-19 16:10 +0000 [r362588] Sean Bright * /, apps/app_externalivr.c: Prevent a crash in ExternalIVR when the 'S' command is sent first. If the first command sent from an ExternalIVR client is an 'S' command, we were blindly removing the first element from the play list and deferencing it, even if it was NULL. This corrects that and also locks appropriately in one place. (issue ASTERISK-17889) Reported by: Chris Maciejewski ........ Merged revisions 362586 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362587 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-19 14:35 +0000 [r362538] Terry Wilson * /, main/asterisk.c: Handle multiple commands per connection via netconsole Asterisk would accept multiple NULL-delimited CLI commands via the netconsole socket, but would occasionally miss a command due to the command not being completely read into the buffer. This patch ensures that any partial commands get moved to the front of the read buffer, appended to, and properly sent. (closes issue ASTERISK-18308) Review: https://reviewboard.asterisk.org/r/1876/ ........ Merged revisions 362536 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362537 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-19 02:40 +0000 [r362497] Matthew Jordan * channels/chan_unistim.c, /, main/tdd.c, main/jitterbuf.c, apps/app_sms.c, main/stdtime/localtime.c, utils/extconf.c, addons/chan_mobile.c, main/format_pref.c, main/asterisk.c: Fix a variety of potential buffer overflows * chan_mobile: Fixed an overrun where the cind_state buffer (an integer array of size 16) would be overrun due to improper bounds checking. At worst, the buffer can be overrun by a total of 48 bytes (assuming 4-byte integers), which would still leave it within the allocated memory of struct hfp. This would corrupt other elements in that struct but not necessarily cause any further issues. * app_sms: The array imsg is of size 250, while the array (ud) that the data is copied into is of size 160. If the size of the inbound message is greater then 160, up to 90 bytes could be overrun in ud. This would corrupt the user data header (array udh) adjacent to ud. * chan_unistim: A number of invalid memmoves are corrected. These would move data (which may or may not be valid) into the ends of these buffers. * asterisk: ast_console_toggle_loglevel does not check that the console log level being set is less then or equal to the allowed log levels of 32. * format_pref: In ast_codec_pref_prepend, if any occurrence of the specified codec is not found, the value used to index into the array pref->order would be one greater then the maximum size of the array. * jitterbuf: If the element being placed into the jitter buffer lands in the last available slot in the jitter history buffer, the insertion sort attempts to move the last entry in the buffer into one slot past the maximum length of the buffer. Note that this occurred for both the min and max jitter history buffers. * tdd: If a read from fsk_serial returns a character that is greater then 32, an attempt to read past one of the statically defined arrays containing the values that character maps to would occur. * localtime: struct ast_time and tm are not the same size - ast_time is larger, although it contains the elements of tm within it in the same layout. Hence, when using memcpy to copy the contents of tm into ast_time, the size of tm should be used, as opposed to the size of ast_time. * extconf: this treats ast_timing's minmask array as if it had a length of 48, when it has defined the size of the array as 24. pbx.h defines minmask as having a size of 48. (issue ASTERISK-19668) Reported by: Matt Jordan ........ Merged revisions 362485 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362496 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-18 17:03 +0000 [r362432] Michael L. Young * tests/test_security_events.c: Fix building security events test The Security Events Framework API changed in trunk to support IPv6. This broke the building of the security events test which was based around IPv4. This patches fixes the build by changing the test to conform to the new changes. (related to issue ASTERISK-19447) Review: https://reviewboard.asterisk.org/r/1874/ 2012-04-18 16:41 +0000 [r362430] Richard Mudgett * channels/sig_pri.h, channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /, channels/sig_pri.c: Add ability to ignore layer 1 alarms for BRI PTMP lines. Several telcos bring the BRI PTMP layer 1 down when the line is idle. When layer 1 goes down, Asterisk cannot make outgoing calls. Incoming calls could fail as well because the alarm processing is handled by a different code path than the Q.931 messages. * Add the layer1_presence configuration option to ignore layer 1 alarms when the telco brings layer 1 down. This option can be configured by span while the similar DAHDI driver teignorered=1 option is system wide. This option unlike layer2_persistence does not require libpri v1.4.13 or newer. Related to JIRA AST-598 JIRA ABE-2845 ........ Merged revisions 362428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362429 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-17 21:23 +0000 [r362365-362380] Matthew Jordan * /, main/format_pref.c: Handle case where an unknown format is used to get the preferred codec size In ast_codec_pref_getsize, if an unknown format is passed to the method, no preferred codec will be selected and a negative number will be used to index into the format list. The method now logs an unknown format as a warning, and returns an empty format list. (issue ASTERISK-19655) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863/ ........ Merged revisions 362377 from http://svn.asterisk.org/svn/asterisk/branches/10 * res/res_rtp_asterisk.c, /, res/res_agi.c, res/res_musiconhold.c: Fix places in resources where a negative return value could impact execution This patch addresses a number of modules in resources that did not handle the negative return value from function calls adequately. This includes: * res_agi.c: if the result of the read function is a negative number, indicating some failure, the result would instead be treated as the number of bytes read. This patch now treats negative results in the same manner as an end of file condition, with the exception that it also logs the error code indicated by the return. * res_musiconhold.c: if spawn_mp3 fails to assign a file descriptor to srcfd, and instead assigns a negative value, that file descriptor could later be passed to functions that require a valid file descriptor. If spawn_mp3 fails, we now immediately retry instead of continuing in the logic. * res_rtp_asterisk.c: if no codec can be matched between two RTP instances in a peer to peer bridge, we immediately return instead of attempting to use the codec payload type as an index to determine the appropriate negotiated codec. (issue ASTERISK-19655) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863/ ........ Merged revisions 362362 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362364 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-17 21:10 +0000 [r362363] Jonathan Rose * res/res_config_curl.c, res/res_config_pgsql.c, res/res_config_odbc.c, /: Make use of va_args more appropriate to form in various res_config modules plus utils. A number of va_copy operations weren't matched with a corresponding va_end in res_config_odbc. Also, there was a potential for va_end to be invoked twice on the same va_arg in utils, which would mean invoking va_end on an undefined variable... which is bad. va_end is removed from various functions in config_pgsql and config_curl since they aren't making their own copy. The invokers of those functions are responsible for calling va_end on them. (issue ASTERISK-19451) Reported by: Walter Doekes Review: https://reviewboard.asterisk.org/r/1848/ ........ Merged revisions 362354 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362357 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-17 21:08 +0000 [r362358-362361] Matthew Jordan * main/manager.c, /, main/asterisk.c: Fix places in main where a negative return value could impact execution This patch addresses a number of modules in main that did not handle the negative return value from function calls adequately, or were not sufficiently clear that the conditions leading to improper handling of the return values could not occur. This includes: * asterisk.c: A negative return value from the read function would be used directly as an index into a buffer. We now check for success of the read function prior to using its result as an index. * manager.c: Check for failures in mkstemp and lseek when handling the temporary file created for processing data returned from a CLI command in action_command. Also check that the result of an lseek is sanitized prior to using it as the size of a memory map to allocate. (issue ASTERISK-19655) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863/ ........ Merged revisions 362359 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362360 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, funcs/func_env.c: Fix places where a negative return from ftello could be used as invalid input In a variety of locations in both reading and writing a file, the result from the C library function ftello is used as input to other functions. For the parameters and functions in question, a negative value is invalid input. This patch checks the return value from the ftello function to determine if we were able to determine the current position in the file stream and, if not, fail gracefully. (issue ASTERISK-19655) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863/ ........ Merged revisions 362355 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362356 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-17 18:57 +0000 [r362307] Walter Doekes * channels/chan_unistim.c, cdr/cdr_sqlite3_custom.c, funcs/func_env.c, res/res_phoneprov.c, channels/chan_gtalk.c, cdr/cdr_pgsql.c, res/res_http_post.c, res/res_musiconhold.c, res/res_jabber.c, res/res_format_attr_celt.c, channels/chan_dahdi.c, funcs/func_groupcount.c, apps/app_osplookup.c, funcs/func_odbc.c, main/ast_expr2f.c, apps/app_minivm.c, channels/chan_alsa.c, codecs/codec_resample.c, formats/format_h264.c, res/res_format_attr_silk.c, res/res_config_ldap.c, main/ast_expr2.fl, res/res_config_sqlite3.c, channels/chan_sip.c, channels/vcodecs.c, codecs/codec_g726.c, main/data.c, res/res_corosync.c, channels/chan_h323.c, codecs/codec_dahdi.c, funcs/func_callerid.c, main/asterisk.c, res/res_odbc.c: Avoid cppcheck warnings; removing unused vars and a bit of cleanup. Patch by: junky Review: https://reviewboard.asterisk.org/r/1743/ 2012-04-17 18:29 +0000 [r362306] Matthew Jordan * /, formats/format_sln.c, formats/format_vox.c, formats/format_wav.c, formats/format_pcm.c, formats/format_wav_gsm.c, formats/format_siren14.c, formats/format_gsm.c, formats/format_g719.c, formats/format_siren7.c: Fix error that caused seek format operations to set max file size to '1' or '0' A very inappropriate placement of a ')' (introduced in r362151) caused the maximum size of a file to be set as the result of a comparison operation, as opposed to the result of the ftello operation. This resulted in seeking being restricted to the beginning of the file, or 1 byte into the file. Thanks to the Asterisk Test Suite for properly freaking out about this on at least one test. (issue ASTERISK-19655) Reported by: Matt Jordan ........ Merged revisions 362304 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362305 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-17 15:00 +0000 [r362266] Michael L. Young * /, channels/chan_sip.c: Turn off warning message when bind address is set to any. When a bind address is set to an ANY address (udpbindport=::), a warning message is displayed stating that "Address remapping activated in sip.conf but we're using IPv6, which doesn't need it. Please remove 'localnet' and/or 'externaddr' settings." But if one is running dual stack, we shouldn't be told to turn those settings off. This patch checks if the bind address is an ANY address or not. The warning message will now only be displayed if the bind address is NOT an ANY address and IPv6 is being used. Also, updated the copyright year. (closes issue ASTERISK-19456) Reported by: Michael L. Young Tested by: Michael L. Young Patches: chan_sip_ipv6_message.diff uploaded by Michael L. Young (license 5026) ........ Merged revisions 362253 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362264 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-16 21:58 +0000 [r362203-362206] Matthew Jordan * channels/chan_dahdi.c, /, channels/chan_agent.c: Fix negative return handling in channel drivers In chan_agent, while handling a channel indicate, the agent channel driver must obtain a lock on both the agent channel, as well as the channel the agent channel is using. To do so, it attempts to lock the other channel first, then unlock the agent channel which is locked prior to entry into the indicate handler. If this unlock fails with a negative return value, which can occur if the object passed to agent_indicate is an invalid ao2 object or is NULL, the return value is passed directly to strerror, which can only accept positive integer values. In chan_dahdi, the return value of dahdi_get_index is used to directly index into the sub-channel array. If dahd_get_index returns a negative value, it would use that value to index into the array, which could cause an invalid memory access. If dahdi_get_index returns a negative number, we now default to SUB_REAL. (issue ASTERISK-19655) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863/ ........ Merged revisions 362204 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362205 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, apps/app_voicemail.c: Fix handling of negative return code when storing voicemails in ODBC storage When storing a voicemail message using an ODBC connection to a database, the voicemail message is first stored on disk. The sound file associated with the message is read into memory before being transmitted to the database. When this occurs, a failure in the C library's lseek function would cause a negative value to be passed to the mmap as the size of the memory map to create. This would almost certainly cause the creation of the memory map to fail, resulting in the message being lost. (issue ASTERISK-19655) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863 ........ Merged revisions 362201 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362202 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-16 21:20 +0000 [r362200] Michael L. Young * main/manager.c, main/security_events.c, channels/sip/security_events.c, CHANGES, include/asterisk/security_events_defs.h: Add IPv6 address support to security events framework. The current Security Events Framework API only supports IPv4 when it comes to generating security events. This patch does the following: * Changes the Security Events Framework API to support IPV6 and updates the components that use this API. * Eliminates an error message that was being generated since the current implementation was treating an IPv6 socket address as if it was IPv4. * Some copyright dates were updated on files touched by this patch. (closes issue ASTERISK-19447) Reported by: Michael L. Young Tested by: Michael L. Young Patches: security_events_ipv6v3.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/1777/ 2012-04-16 20:17 +0000 [r362153] Matthew Jordan * formats/format_ilbc.c, /, formats/format_sln.c, formats/format_vox.c, formats/format_wav.c, formats/format_pcm.c, formats/format_g723.c, formats/format_h263.c, formats/format_h264.c, formats/format_wav_gsm.c, formats/format_siren14.c, formats/format_gsm.c, formats/format_g719.c, formats/format_siren7.c, formats/format_g729.c: Check for IO stream failures in various format's truncate/seek operations For the formats that support seek and/or truncate operations, many of the C library calls used to determine or set the current position indicator in the file stream were not being checked. In some situations, if an error occurred, a negative value would be returned from the library call. This could then be interpreted inappropriately as positional data. This patch checks the return values from these library calls before using them in subsequent operations. (issue ASTERISK-19655) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863/ ........ Merged revisions 362151 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362152 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-13 16:12 +0000 [r362081-362085] Jonathan Rose * apps/app_forkcdr.c, /: Make ForkCDR e option not set end time of the newly forked CDR log Prior to this patch, ForkCDR's e option would immediately set the end time of the forked CDR to that of the CDR that is being terminated. This resulted in the new CDR's end time being roughly the same as it's beginning time (which is in turn roughly the same as the original's end time). (closes issue ASTERISK-19164) Reported by: Steve Davies Patches: cdr_fork_end.v10.patch uploaded by Steve Davies (license 5012) ........ Merged revisions 362082 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362084 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, apps/app_meetme.c: Send relative path named recordings to the meetme directory instead of sounds Prior to this patch, no effort was made to parse the path name to determine a proper destination for recordings of MeetMe's r option. This fixes that. Review: https://reviewboard.asterisk.org/r/1846/ ........ Merged revisions 362079 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362080 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-12 20:08 +0000 [r362043] Paul Belanger * main/srv.c: Convert SRV lookup message to debug level This helps clean up the Asterisk CLI by converting the log message from verbose to debug 2012-04-12 16:29 +0000 [r361998] Richard Mudgett * configs/asterisk.conf.sample, UPGRADE.txt, pbx/pbx_config.c, include/asterisk/options.h, main/asterisk.c: Add option to invoke the extensions.conf stdexten using the legacy macro method. ASTERISK-18809 eliminated the legacy macro invocation of the stdexten in favor of the Gosub method without a means of backwards compatibility. (issue ASTERISK-18809) (closes issue ASTERISK-19457) Reported by: Matt Jordan Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1855/ 2012-04-12 16:25 +0000 [r361968-361987] Kinsey Moore * /, channels/chan_iax2.c: Make trunkfreq take effect when set Previously, setting trunkfreq had no effect on initial load or on reload and only ever used the default value. This causes trunkfreq to be used appropriately on initial load and reload. (closes issue ASTERISK-19521) Patch-by: Jaco Kroon ........ Merged revisions 361972 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 361981 from http://svn.asterisk.org/svn/asterisk/branches/10 * Makefile, build_tools/cflags.xml, /, build_tools/menuselect-deps.in, codecs/gsm/src/k6opt.s, configure, codecs/gsm/Makefile, configure.ac, Makefile.rules, makeopts.in, codecs/lpc10/Makefile: Simplify build system architecture optimization This change to the build system rips out any usage of PROC along with architecture-specific optimizations in favor of using -march=native where it is supported. This fixes broken builds on 64bit Intel systems and results in better optimized code on systems running GCC 4.2+. Review: https://reviewboard.asterisk.org/r/1852/ (closes issue ASTERISK-19462) ........ Merged revisions 361955 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 361956 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-11 17:20 +0000 [r361909] Jonathan Rose * /, configs/queues.conf.sample, apps/app_queue.c: Change default value of 'ignorebusy' on Queue members so that behavior is more like 1.8 Prior to this patch, in order to restore that behavior, a function would have to be used on the QueueMember to make the ringinuse option do anything, which is pretty unreasonable. (closes issue ASTERISK-19536) reported by: Philippe Lindheimer Review: https://reviewboard.asterisk.org/r/1860/ ........ Merged revisions 361907 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-10 21:50 +0000 [r361856] Richard Mudgett * channels/chan_dahdi.c, /: Prevent invalid access of free'd memory if DAHDI channel during an MWI event In the MWI processing loop, when a valid event occurs the temporary caller ID information is deallocated. If a new DAHDI channel is successfully created, the event is passed up to the analog_ss_thread without error and the loop exits. If, however, the DAHDI channel is not created, then the caller ID struct has been free'd, and the gains reset to their previous level. This will almost certainly cause an invalid access to the free'd memory, either in subsequent calls to callerid_free or calls to callerid_feed. * Rework the -r361705 patch to better manage the cs and mtd allocated resources. * Fixed use of mwimonitoractive flag to be correct if the mwi_thread() fails to start. ........ Merged revisions 361854 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 361855 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-10 19:58 +0000 [r361659-361805] Matthew Jordan * /, main/http.c: Fix crash caused by unloading or reloading of res_http_post When unlinking itself from the registered HTTP URIs, res_http_post could inadvertently free all URIs registered with the HTTP server. This patch modifies the unregister method to only free the URI that is actually being unregistered, as opposed to all of them. ........ Merged revisions 361803 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 361804 from http://svn.asterisk.org/svn/asterisk/branches/10 * funcs/func_curl.c, /: Allow func_curl to exit gracefully if list allocation fails during write If the global_curl_info data structure could not be allocated, the datastore associated with the operation would be free'd, but the function would not return. This would later dereference the datastore, almost certainly causing Asterisk to crash. With this patch, if the data structure is not allocated the method will return an error code, and not attempt any further operation. ........ Merged revisions 361753 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 361754 from http://svn.asterisk.org/svn/asterisk/branches/10 * channels/chan_dahdi.c, /: Prevent invalid access of free'd memory if DAHDI channel during an MWI event In the MWI processing loop, when a valid event occurs the temporary caller ID information is deallocated. If a new DAHDI channel is successfully created, the event is passed up to the analog_ss_thread without error and the loop exits. If, however, the DAHDI channel is not created, then the caller ID struct has been free'd, and the gains reset to their previous level. This will almost certainly cause an invalid access to the free'd memory, either in subsequent calls to callerid_free or calls to callerid_feed. This patch makes it so that we only free the caller ID structure if a DAHDI channel is successfully created, and we bump the gains back up if we fail to make a DAHDI channel. ........ Merged revisions 361705 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 361706 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, funcs/func_global.c: Change SHARED function to use a safe traversal when modifying a variable When the SHARED function modifies a variable, it removes it from its list of variables and reinserts the new value at the head of the list of variables. Doing this inside a standard list traversal can be dangerous, as the standard list traversal does not account for the list being changed. While the code in question should not cause a use after free violation due to its breaking out of the loop after freeing the variable, it could lead to a maintenance issue if the loop was modified. This also fixes a violation reported by a static analysis tool, which also makes this code easier to maintain in the future. ........ Merged revisions 361657 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 361658 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-06 22:00 +0000 [r361561-361608] Matthew Jordan * /, res/res_calendar_ews.c: Fix memory leak in res_calendar_ews when event email address node is empty If the XML calendar data returned by a Microsoft Exchange Web Service specifies an XML Event E-Mail Address ("EmailAddress"), and no e-mail address is provided, a condition existed where an ast_calendar_attendee struct would be allocated but not appended to the list of attendees. Because of that, the memory associated with the attendee would never be freed. This patch frees the memory if no e-mail address is provided. ........ Merged revisions 361606 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 361607 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, apps/app_meetme.c: Fix memory leak when using MeetMeAdmin 'e' option with user specified A memory leak/reference counting leak occurs if the MeetMeAdmin 'e' command (eject last user that joined) is used in conjunction with a specified user. Regardless of the command being executed, if a user is specified for the command, MeetMeAdmin will look up that user. Because the 'e' option kicks the last user that joined, as opposed to the one specified, the reference to the user specified by the command would be leaked when the user variable was assigned to the last user that joined. ........ Merged revisions 361558 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 361560 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-06 19:58 +0000 [r361523] Richard Mudgett * /, main/message.c: Don't add an empty MESSAGE_DATA(key) header if it doesn't already exist. Doing Set(MESSAGE_DATA(key)=) would add an empty key header if the key header did not already exist. If it already existed it would delete it. * Made msg_set_var_full() exit early if the named variable did not already exist and the value to set is empty. ........ Merged revisions 361522 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-06 18:19 +0000 [r361476] Kinsey Moore * channels/chan_unistim.c, main/pbx.c, /, channels/chan_sip.c, funcs/func_strings.c, formats/format_ogg_vorbis.c, channels/console_video.c, apps/app_ices.c, channels/chan_gtalk.c, channels/chan_iax2.c, res/res_config_sqlite.c, res/res_srtp.c, main/cdr.c, main/tcptls.c, channels/console_gui.c, funcs/func_channel.c, apps/app_sms.c, addons/chan_mobile.c, apps/app_chanspy.c, main/xmldoc.c, channels/chan_mgcp.c, res/res_config_sqlite3.c, res/res_clioriginate.c, apps/app_voicemail.c: Add missing newlines to CLI logging ........ Merged revisions 361471 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 361472 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-06 16:33 +0000 [r361429] Paul Belanger * bridges/bridge_builtin_features.c, /, funcs/func_sysinfo.c, bridges/bridge_multiplexed.c: Multiple revisions 361403,361412 ........ r361403 | pabelanger | 2012-04-06 12:24:36 -0400 (Fri, 06 Apr 2012) | 2 lines Fix typo in svn:keywords ........ r361412 | pabelanger | 2012-04-06 12:27:30 -0400 (Fri, 06 Apr 2012) | 2 lines Fix typo in svn:keywords ........ Merged revisions 361403,361412 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 361422 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-06 15:50 +0000 [r361382] Russell Bryant * /, configs/rpt.conf.sample (removed), configs/usbradio.conf.sample (removed), apps/rpt_flow.pdf (removed): Remove a few more files related to chan_usbradio and app_rpt. ........ Merged revisions 361380 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 361381 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-06 14:02 +0000 [r361334] Matthew Jordan * /, channels/chan_sip.c: Fix a typo in the warning messages for an ignored media stream Added a '\n' to the warning messages when we ignore a media stream due to the port number being '0'. (closes issue ASTERISK-19646) Reported by: Badalian Vyacheslav ........ Merged revisions 361332 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 361333 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-06 13:32 +0000 [r361331] Kinsey Moore * apps/app_dial.c, /: Remove unnecessary error message in app_dial.c The error message for failure to stop autoservice after a gosub or macro call during a dial was removed for macro while Asterisk 1.4 was still being actively developed. The corresponding gosub error message was never removed. (closes issue ASTERISK-19551) ........ Merged revisions 361329 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 361330 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-05 17:22 +0000 [r361092-361279] Jonathan Rose * /, apps/app_meetme.c: Fix MusicOnHold in MeetMe so that it always uses the class if it's been defined There were a few instances of restarting music on hold in meetme that would cause Asterisk to revert to the default class of music on hold for no adequate reason. Review: https://reviewboard.asterisk.org/r/1844/ ........ Merged revisions 361269 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 361270 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, addons/ooh323cDriver.c: Fix some stuff involving calls to memcpy and memset The important parts of the patch were already applied through other updates. (closes issue ASTERISK-19445) Reported by: Makoto Dei Patches: memset-memcpy-length.patch uploaded by Makoto Dei (license 5027) ........ Merged revisions 361210 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 361211 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, funcs/func_devstate.c: Make 'help devstate change' display properly (get rid of excess comma) (closes issue ASTERISK-19444) Reported by: Makoto Dei Patches: devstate-change-usage-truncate.patch uploaded by Makoto Dei (license 5027) ........ Merged revisions 361201 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 361208 from http://svn.asterisk.org/svn/asterisk/branches/10 * main/channel.c, pbx/pbx_loopback.c, addons/chan_ooh323.c, /, channels/chan_sip.c, main/app.c, pbx/pbx_realtime.c, apps/app_externalivr.c, channels/chan_iax2.c, res/res_fax_spandsp.c, apps/app_milliwatt.c: Replace GNU old-style field designator extensions to fix clang warnings (issue ASTERISK-19540) Reported by: Makoto Dei Patches: clang-gnu-designator.patch uploaded by Makoto Dei (license 5027) ........ Also add from the patch the portion in res_fax_spandsp that didn't apply to 1.8 Merged revisions 361142 from http://svn.asterisk.org/svn/asterisk/branches/1.8 (closes issue ASTERISK-19540) ........ Merged revisions 361143 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, apps/app_meetme.c: Make the MeetMeAdmin N command (mute all nonadmins) not mute admins (Closes Issue ASTERISK-19335) Reported by: Johan Wilfer Review: https://reviewboard.asterisk.org/r/1843/ ........ Merged revisions 361090 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 361091 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-03 20:14 +0000 [r361042] Kinsey Moore * /, apps/app_transfer.c: Fix the display of documentation for Transfer This came up while fixing documentation generation for many other cases where the argument separator was not being displayed properly. Now that it is displayed properly, it shows up in the wrong place for Transfer since the '/' is only required if Tech is present. (related to issue ASTERISK-18168) ........ Merged revisions 361040 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 361041 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-04-03 20:03 +0000 [r361038-361039] Mark Murawki * include/asterisk/manager.h: Fix dev-mode compiler warning about gnu_printf (related to ASTERISK-19575) * main/channel.c, main/manager.c, main/utils.c, include/asterisk/channel.h, include/asterisk/strings.h, CHANGES, include/asterisk/manager.h: Allow the Hangup manager action to match channels by regex * Hangup now can take a regular expression as the Channel option. If you want to hangup multiple channels, use /regex/ as the Channel option. Existing behavior to hanging up a single channel is unchanged, but if you pass a regex, the manager will send you a list of channels back that were hung up. (closes issue ASTERISK-19575) Reported by: Mark Murawski Tested by: Mark Murawski 2012-04-02 22:27 +0000 [r360994] Kinsey Moore * /, channels/chan_sip.c: Stop sending out RTCP if RTP is inactive This change prevents Asterisk from sending RTCP receiver reports during a remote bridge since it is no longer receiving media and should not be reporting anything. (related to ASTERISK-19366) ........ Merged revisions 360987 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360993 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-03-30 21:38 +0000 [r360935] Richard Mudgett * /, main/logger.c: Fix logger deadlock on Asterisk shutdown. The logger_thread() had an exit path that failed to release the logmsgs list lock. * Make logger_thread() exit path unlock the logmsgs list lock. * Made ast_log() not queue any messages to the logmsgs list if the close_logger_thread flag is set. (issue ASTERISK-19463) Reported by: Matt Jordan ........ Merged revisions 360933 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360934 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-03-29 23:36 +0000 [r360872-360886] Mark Michelson * /, main/features.c: Fix potential race condition during call pickup. Prior to this patch, a connected line update was queued during call pickup and then an answer frame was queued. The original caller would presumably then have his connected line updated and then the call would be answered. In actuality, the answer frame was not how the call ended up being answered. Rather, an odd section in app_dial that checks if the called channel's state is up. The result is that the order of the connected line update and the answer were variable. In most cases, this wasn't actually a bad thing. However, if the 'I' option was passed to dial, the connected line update would be inhibited. The fix is to queued the connected line after the answer frame is queued. This way the race in app_dial is between two conditions resulting in an answer. This way the connected line update occurs after the answer every time. (closes issue ASTERISK-19183) Reported by: Thomas Arimont Tested by: Thomas Arimont Mark Michelson Patches: ASTERISK-19183.patch uploaded by Mark Michelson (license 5049) ........ Merged revisions 360884 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360885 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, channels/chan_sip.c: Improve accuracy of identifying information sent in dialog-info SIP NOTIFY requests. This change makes use of connected party information in addition to caller ID in order to populate local and remote XML elements in the dialog-info NOTIFYs. (closes issue ASTERISK-16735) Reported by: Maciej Krajewski Tested by: Maciej Krajewski Patches: local_remote_hint2.diff uploaded by Mark Michelson (license 5049) ........ Merged revisions 360862 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360863 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-03-29 21:57 +0000 [r360827] Richard Mudgett * include/asterisk/astobj2.h, main/astobj2.c: Misc changes to make astobj2 enhancement diffs easier to follow. * Rename astobj2 API parameter funcname to func. * Rename astobj2 API iterator parameter to iter. * Update some documentation for OBJ_MULTIPLE. 2012-03-29 20:01 +0000 [r360785-360787] Jonathan Rose * include/asterisk/logger.h, main/dial.c, main/pbx.c, include/asterisk/bridging.h, main/features.c, main/logger.c, CHANGES, apps/app_mixmonitor.c, configs/logger.conf.sample: Introducing the log message unique call identifiers feature Log messages will now display a call number that they are tied to (ordered for calls based on when they started). This feature is made to be minimally invasive without requiring changes to many of the existing log messages. These IDs won't show up for verbose messages on CLI (but they will in log files) This is currently in phase II of production, see more about this feature on the wiki -- https://wiki.asterisk.org/wiki/display/AST/Unique+Call-ID+Logging Review: https://reviewboard.asterisk.org/r/1823/ * include/asterisk/logger.h, main/dial.c, main/pbx.c, /, include/asterisk/bridging.h, main/features.c, main/logger.c, CHANGES, apps/app_mixmonitor.c, configs/logger.conf.sample: undoing 360785 due to merging mistake * include/asterisk/logger.h, main/dial.c, main/pbx.c, /, include/asterisk/bridging.h, main/features.c, main/logger.c, CHANGES, apps/app_mixmonitor.c, configs/logger.conf.sample: Introducing the log message unique call identifiers feature Log messages will now display a call number that they are tied to (ordered for calls based on when they started). This feature is made to be minimally invasive without requiring changes to many of the existing log messages. These IDs won't show up for verbose messages on CLI (but they will in log files) This is currently in phase II of production, see more about this feature on the wiki -- https://wiki.asterisk.org/wiki/display/AST/Unique+Call-ID+Logging Review: https://reviewboard.asterisk.org/r/1823/ 2012-03-28 19:39 +0000 [r360724] Terry Wilson * channels/chan_jingle.c, addons/chan_ooh323.c, cdr/cdr_adaptive_odbc.c, addons/cdr_mysql.c, channels/chan_gtalk.c, apps/confbridge/conf_config_parser.c: Fix setting CDR variables in the hangup extension A previous CDR fix for setting CDR variables during a bridge via custom dialplan features broke setting CDR variables in the hangup extension. This patch fixes the issue. Review: https://reviewboard.asterisk.org/r/1794/ ........ Merged revisions 358978 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 358989 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-03-27 18:44 +0000 [r360673] Mark Michelson * /, channels/chan_sip.c: Make a debug message regarding subscription changes more accurate. I was getting confused during some testing why Asterisk was saying that a subscription was being added when it was clearly being removed. This fixes that confusion. ........ Merged revisions 360625 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360672 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-03-27 17:13 +0000 [r360626-360627] Richard Mudgett * include/asterisk/astobj2.h, tests/test_astobj2.c, main/astobj2.c: Add global ao2 array container. Global ao2 objects must always exist after initialization because there is no access control to obtain another reference to the global object. It is expected that module configuration could use these new API calls to replace an active configuration parameter object with an updated configuration parameter object. With these new API calls, the global object could be replaced, removed, or referenced without the risk of someone using a stale global object pointer. Review: https://reviewboard.asterisk.org/r/1824/ * main/astobj2.c: Attempt to be more helpful when using a bad ao2 object pointer. 2012-03-27 14:43 +0000 [r360576] Jonathan Rose * /, configure: Updates config with bootstrap where I changed configure.ac in r360488 (issue ASTERISK-17842) Reported by: Bryon Clark ........ Merged revisions 360574 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360575 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-03-26 21:22 +0000 [r360536] Paul Belanger * main/dnsmgr.c, /: Convert ast_verb() to ast_debug() and increase log level Rather then flood the CLI with verbose messages, we've changed the level to debug. This will help keep the CLI clean. 2012-03-26 19:49 +0000 [r360490] Jonathan Rose * /, configure.ac: Fix BETTER_BACKTRACES library detection for Fedora/RedHat/CentOS (closes ASTERISK-17842) Reported by: Bryon Clark Patches: 20110512__issue19278.diff.txt uploaded by Tilghman Lesher (license 5003) configure_bfd_with_dl_and_iberty.patch uploaded by Bryon Clark (license 6157) ........ Merged revisions 360488 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360489 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-03-24 23:49 +0000 [r360359-360415] Russell Bryant * funcs/func_curl.c, /: func_curl: Fix leak of an ast_str in error handling code path. ........ Merged revisions 360413 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360414 from http://svn.asterisk.org/svn/asterisk/branches/10 * channels/chan_iax2.c: chan_iax2: Use OBJ_NODATA to be a bit more explicit. This is just a minor code cleanup change. These uses of ao2_callback() would never return anything since the callbacks always returned 0. However, be more explicit that no returned results are wanted by specifying OBJ_NODATA. * /, apps/app_page.c: app_page: Fix a memory leak on every Page(). dial_list is a dynamically allocated array that is allocated at the beginning of Page() based on how many devices will be dialed. This was never being freed. ........ Merged revisions 360363 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360364 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, apps/app_jack.c: app_jack: fix datastore memory leak in error handling path. ........ Merged revisions 360360 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360361 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, main/ast_expr2.h, res/ael/ael.tab.c, main/ast_expr2.y, main/ast_expr2f.c, res/ael/ael_lex.c, res/ael/ael.tab.h, main/ast_expr2.c: Multiple revisions 360356-360357 ........ r360356 | russell | 2012-03-23 22:33:36 -0400 (Fri, 23 Mar 2012) | 6 lines expression parser: Fix (theoretical) memory leak. Fix a memory leak that is very unlikely to actually happen. If a malloc() succeeded, but the following strdup() failed, the memory from the original malloc() would be leaked. ........ r360357 | russell | 2012-03-23 22:34:39 -0400 (Fri, 23 Mar 2012) | 6 lines Rebuild parsers. This is needed to include the last fix to main/ast_expr2.y. The changes look much bigger as this regeneration of the code was done with newer versions of flex and bison. ........ Merged revisions 360356-360357 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360358 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-03-24 00:40 +0000 [r360264-360311] Richard Mudgett * main/channel.c, /, channels/sig_pri.c: Make number not available presentation also set screening to network provided. Q.951 indicates that when the presentation indicator is "Number not available due to interworking" for a number then the screening indicator field should be "Network provided". * Made ast_party_id_presentation() return AST_PRES_NUMBER_NOT_AVAILABLE when the presentation is "Number not available due to interworking". This fix makes Asterisk consistent and it also makes it consistent with earlier branches as far as this presentation value is concerned. * Made pri_to_ast_presentation() and ast_to_pri_presentation() conversions handle the "Number not available due to interworking" case better in sig_pri.c. This change is possible because the minimum required libpri version (v1.4.11) has the necessary defines in libpri.h. ........ Merged revisions 360309 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360310 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, channels/chan_sip.c: Add missing initialization of update_redirecting in chan_sip.c ........ Merged revisions 360262 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360263 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-03-22 21:25 +0000 [r360227] Jonathan Rose * apps/app_dial.c, include/asterisk/utils.h, main/features.c, main/utils.c, CHANGES, apps/app_queue.c: Adds F option to Bridge application Similar to dial and queue F option. (Closes issue ASTERISK-19282) Reported by: To Patches: bridge_f-v3.diff uploaded by To (license 6347) Review: https://reviewboard.asterisk.org/r/1825/ 2012-03-22 19:51 +0000 [r360190] Kinsey Moore * main/udptl.c, main/stdtime/test.c, main/autoservice.c, main/rtp_engine.c, main/frame.c, main/fskmodem_float.c, main/sha1.c, main/say.c, main/ecdisa.h, main/utils.c, main/devicestate.c, main/taskprocessor.c, main/indications.c, main/enum.c, main/config.c, main/loader.c, main/term.c, main/cli.c, main/io.c, main/ulaw.c, main/channel.c, main/dial.c, main/manager.c, main/tdd.c, main/strcompat.c, main/plc.c, main/features.c, main/logger.c, main/fskmodem_int.c, main/app.c, main/stdtime/localtime.c, main/image.c, main/dns.c, main/message.c, main/md5.c, main/sched.c, main/lock.c, main/pbx.c, main/dnsmgr.c, main/slinfactory.c, main/translate.c, main/jitterbuf.c, main/cel.c, main/chanvars.c, main/netsock.c, main/srv.c, main/privacy.c, main/fixedjitterbuf.c, main/file.c, main/callerid.c, main/event.c, main/astmm.c, main/audiohook.c, main/cygload.c, main/fixedjitterbuf.h, main/asterisk.c, main/xmldoc.c, main/dsp.c, main/timing.c: Kill off red blobs in most of main/* Everything still compiled after making these changes, so I assume these whitespace-only changes didn't break anything (and shouldn't have). 2012-03-21 14:55 +0000 [r360140] Jonathan Rose * /, contrib/scripts/install_prereq: Update install_prereq script to include missing GSM library for debian amd move SQLite3. (closes issue ASTERISK-19367) Reported by: Andrew Latham Patches: debian_install_prereq.diff uploaded by Andrew Latham (license 5985) ........ Merged revisions 360138 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360139 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-03-21 14:47 +0000 [r360137] Tzafrir Cohen * /, configure, configure.ac: Also detect gmime 2.6 Also detect gmime version 2.6 (Michael Biebl) Signed-off-by: Tzafrir Cohen (License #5035) ........ Merged revisions 360087 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360098 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-03-21 13:31 +0000 [r360089] Matthew Jordan * /, channels/chan_sip.c: Ensure Asterisk sends a BYE when pending on the final response to a re-INVITE When Asterisk detects a hangup and cannot send a BYE due to a pending INVITE, it sets the pendingbye flag and waits for the final response to that INVITE. When the response is received, it transmits the BYE. If, however, that INVITE request is a pending re-INVITE, it needs to first send a CANCEL request to terminate the pending re-INVITE. In that circumstance, Asterisk was, in some scenarios, clearing the pendingbye flag after processing the CANCEL request and not checking for a pending BYE when receiving the final 487 response to the INVITE. This patch ensures that if the pendingbye flag is set, it is honored regardless of the nature of the INVITE request currently in flight. (closes issue ASTERISK-19365) Reported by: Thomas Arimont Tested by: Thomas Arimont Patches: bugASTERISK-19365_2012_03_08.patch uploaded by mjordan (license 6283) Review: https://reviewboard.asterisk.org/r/1807 ........ Merged revisions 360086 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360088 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-03-20 20:42 +0000 [r360036] Kinsey Moore * /, apps/app_echo.c: Prevent Echo() from relaying control, null, and modem frames Echo()'s description states that it echoes audio, video, and DTMF except for # while it actually echoes any frame that it receives other than DTMF #. This was causing frame storms in the test suite in some circumstances where Echo() was attached to both ends of a pair of local channels and control frames were being periodically generated. Echo()'s behavior and description have been modifed so that it only echoes media and non-# DTMF frames. ........ Merged revisions 360033 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360034 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-03-20 18:17 +0000 [r359983] Sean Bright * /, UPGRADE.txt, channels/chan_iax2.c, include/asterisk/manager.h: chan_iax2: Correct spelling of 'Port' header in IAX2 PeerStatus AMI Events The PeerStatus event for IAX2 channels currently includes a header named Post which should have been Port. Post was removed and the AMI version has been updated to 1.3. ........ Merged revisions 359982 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-03-20 17:31 +0000 [r359942-359981] Richard Mudgett * main/data.c, main/pbx.c, main/manager.c, /, main/features.c, include/asterisk/manager.h, main/db.c: Allow AMI action callback to be reentrant. Fix AMI module reload deadlock regression from ASTERISK-18479 when it tried to fix the race between calling an AMI action callback and unregistering that action. Refixes ASTERISK-13784 broken by ASTERISK-17785 change. Locking the ao2 object guaranteed that there were no active callbacks that mattered when ast_manager_unregister() was called. Unfortunately, this causes the deadlock situation. The patch stops locking the ao2 object to allow multiple threads to invoke the callback re-entrantly. There is no way to guarantee a module unload will not crash because of an active callback. The code attempts to minimize the chance with the registered flag and the maximum 5 second delay before ast_manager_unregister() returns. The trunk version of the patch changes the API to fix the race condition correctly to prevent the module code from unloading from memory while an action callback is active. * Don't hold the lock while calling the AMI action callback. (closes issue ASTERISK-19487) Reported by: Philippe Lindheimer Review: https://reviewboard.asterisk.org/r/1818/ Review: https://reviewboard.asterisk.org/r/1820/ ........ Merged revisions 359979 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359980 from http://svn.asterisk.org/svn/asterisk/branches/10 * res/res_mutestream.c: Convert MuteAudio documentation to XML. * Added missing error exits with cause in manager_mutestream(). * Cleaned up manager_mutestream() and func_mute_write(). * Some whitespace and comment cleanup. 2012-03-16 21:00 +0000 [r359905] Jonathan Rose * /, apps/app_chanspy.c: Prevent chanspy from binding to zombie channels This patch addresses a bug with chanspy on local channels which roughly 50% of the time would create a situation where chanspy can latch onto a zombie channel, keeping the zombie alive forever and causing the channel doing the spying to never be able to hang up. (closes issue ASTERISK-19493) Reported by: lvl Review: https://reviewboard.asterisk.org/r/1819/ ........ Merged revisions 359892 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359898 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-03-16 20:37 +0000 [r359904] Richard Mudgett * include/asterisk/app.h, main/app.c: Simplify some code in ast_app_run_sub(). * Remove unnnecessary const from const char * const var declaration in the ast_app_run_macro() and ast_app_run_sub() prototypes. The second const is unnecessary. 2012-03-16 15:38 +0000 [r359857] Mark Michelson * apps/app_dial.c, main/pbx.c, include/asterisk/pbx.h, CHANGES: Revert the pre-dial addition. The code may be just fine, but it had not received a "ship it!" on review board yet. 2012-03-16 08:27 +0000 [r359811] Alec L Davis * /, channels/sip/include/sip.h: Missed lastinvite CSeq int to uint32_t change from Review: https://reviewboard.asterisk.org/r/1699/ ........ Merged revisions 359809 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359810 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-03-15 20:11 +0000 [r359772] Mark Murawki * main/pbx.c: Fix warning from commit r359705 (predial options for app_dial) 2012-03-15 19:11 +0000 [r359708] Matthew Jordan * /, main/utils.c: Fix remotely exploitable stack overflow in HTTP manager There exists a remotely exploitable stack buffer overflow in HTTP digest authentication handling in Asterisk. The particular method in question is only utilized by HTTP AMI. When parsing the digest information, the length of the string is not checked when it is copied into temporary buffers allocated on the stack. This patch fixes this behavior by parsing out pre-defined key/value pairs and avoiding unnecessary copies to the stack. (closes issue ASTERISK-19542) Reported by: Russell Bryant Tested by: Matt Jordan ........ Merged revisions 359706 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359707 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-03-15 18:58 +0000 [r359705] Mark Murawki * apps/app_dial.c, main/pbx.c, include/asterisk/pbx.h, CHANGES: Add options PreDial options 'b' and 'B' to app_dial * Added 'b' and 'B' options to Dial. These options will allow you to run last-minute dialplan on the caller and callee channels while the Dial application is executing, but before the call is started. For example you can use the 'b' option to run dialplan on the callee channel to get the name of the newly created channel right away. Review: https://reviewboard.asterisk.org/r/1229/ (closes issue: ASTERISK-19548) Reported by: Mark Murawski Tested by: Mark Murawski, Stefan Schmidt 2012-03-15 18:55 +0000 [r359704] Matthew Jordan * /, apps/app_milliwatt.c: Fix remotely exploitable stack overrun in Milliwatt Milliwatt is vulnerable to a remotely exploitable stack overrun when using the 'o' option. This occurs due to the milliwatt_generate function not accounting for AST_FRIENDLY_OFFSET when calculating the maximum number of samples it can put in the output buffer. This patch resolves this issue by taking into account AST_FRIENDLY_OFFSET when determining the maximum number of samples allowed. Note that at no point is remote code execution possible. The data that is written into the buffer is the pre-defined Milliwatt data, and not custom data. (closes issue ASTERISK-19541) Reported by: Russell Bryant Tested by: Matt Jordan Patches: milliwatt_stack_overrun.rev1.txt by Russell Bryant (license 6283) Note that this patch was written by Russell, even though Matt uploaded it ........ Merged revisions 359645 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........ Merged revisions 359656 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359694 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-03-15 18:34 +0000 [r359651] Paul Belanger * channels/chan_sip.c: Remove unused variable ‘srch’ Missed on the previous commit 2012-03-15 18:32 +0000 [r359644] Richard Mudgett * apps/app_dial.c, /, apps/app_queue.c: Add missing connected line macro calls to initial dial for Dial and Queue apps. The connected line interception macros do not get executed when the outgoing channel is initially created and that channel's caller-id is implicitly imported into the incoming channel's connected line data. If you are using the interception macros, you would expect that they get run for every change to a channel's connected line information outside of normal dialplan execution. Review: https://reviewboard.asterisk.org/r/1817/ ........ Merged revisions 359609 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359620 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-03-15 17:36 +0000 [r359607] Paul Belanger * channels/chan_sip.c: Remove some dead code found in _sip_show_peers() Review: https://reviewboard.asterisk.org/r/1696/ 2012-03-15 00:54 +0000 [r359456-359560] Russell Bryant * /, channels/chan_iax2.c: chan_iax2: Fix use of uninitialized sockaddr_in in try_transfer(). Initialize a struct sockaddr_in in try_transfer() so that the code isn't (potentially) trying to read from it while uninitialized. ........ Merged revisions 359558 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359559 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, channels/chan_gtalk.c: chan_gtalk: Fix potential use of uninitialized variable. Avoid potential use of idroster in gtalk_alloc() before it has been initialized. ........ Merged revisions 359508 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359509 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, apps/app_chanisavail.c: app_chanisavail: Fix use of uninitialized variable. Ensure that status is set before it is used by resetting it during each loop iteration. This could have resulted in incorrect results from this app. ........ Merged revisions 359486 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359491 from http://svn.asterisk.org/svn/asterisk/branches/10 * main/udptl.c, /: udptl: Ensure fec[] in udptl_build_packet() is initialized. Scan results indicated that this array could be used uninitialized. At a quick look, it looks correct. In any case, initializing it is a Good Thing (tm). ........ Merged revisions 359457 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359458 from http://svn.asterisk.org/svn/asterisk/branches/10 * include/asterisk/app.h, /: app.h: Always initialize AST_DECLARE_APP_ARGS(). This patch ensures that the struct defined by AST_DECLARE_APP_ARGS() is always fully initialized. I'm not sure if this fixes any real bugs, but it silences a bunch of warnings from coverity, and is generally a good thing to do anyway. ........ Merged revisions 359452 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359454 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-03-14 22:38 +0000 [r359455] Richard Mudgett * main/channel.c, /, channels/chan_agent.c, include/asterisk/channel.h: Fix deadlock potential with some ast_indicate/ast_indicate_data calls. Calling ast_indicate()/ast_indicate_data() with the channel lock held can result in a deadlock with a local channel because of how local channels need to avoid deadlock. ........ Merged revisions 359451 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359453 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-03-14 18:56 +0000 [r359406] Matthew Jordan * tests/test_jitterbuf.c (added): Add tests for main/jitterbuf.c This patch adds unit tests for main/jitterbuf.c. This includes checking for the following: * Nominal insertion and retrieval of frames * Insertion and retrieval of frames where the frames are inserted out of order with respect to the previous frame * Insertion and retrieval of frames where some number of frames that would occur in the expected sequence are instead dropped * Insertion and retrieval of frames with an arrival time that does not occur at the same rate as the surrounding frames * Resynchronization of the jitter buffer when an inserted frame breaks the resynchronization threshold * Overfilling of the jitter buffer For each of the tests, both JB_TYPE_VOICE and JB_TYPE_CONTROL permutations exist. Review: https://reviewboard.asterisk.org/r/1815 (issue: ASTERISK-18964) Reported by: Kris Shaw Tested by: Kris Shaw, Matt Jordan 2012-03-14 18:12 +0000 [r359360] Richard Mudgett * include/asterisk/channel_internal.h: Three copies of the file contents in channel_internal.h are a bit excessive. 2012-03-14 17:48 +0000 [r359359] Matthew Jordan * /, main/jitterbuf.c: Fix incorrect jitter buffer overflow due to missed resynchronizations When a change in time occurs, such that the timestamps associated with frames being placed into an adaptive jitter buffer (implemented in jitterbuf.c) are significantly different then the previously inserted frames, the jitter buffer checks to see if it needs to be resynched to the new time frame. If three consecutive packets break the threshold, the jitter buffer resynchs itself to the new timestamps. This currently only occurs when history is calculated, and hence only on JB_TYPE_VOICE frames. JB_TYPE_CONTROL frames, on the other hand, are never passed to the history calculations. Because of this, if the jump in time is greater then the maximum allowed length of the jitter buffer, the JB_TYPE_CONTROL frames are dropped and no resynchronization occurs. Alterntively, if the overfill logic is not triggered, the JB_TYPE_CONTROL frame will be placed into the buffer, but with a time reference that is not applicable. Subsequent JB_TYPE_VOICE frames will quickly trigger the overflow logic until reads from the jitter buffer reach the errant JB_TYPE_CONTROL frame. This patch allows JB_TYPE_CONTROL frames to resynch the jitter buffer. As JB_TYPE_CONTROL frames are unlikely to occur in multiples, it perform the resynchronization on any JB_TYPE_CONTROL frame that breaks the resynch threshold. Note that this only impacts chan_iax2, as other consumers of the adaptive jitter buffer use the abstract jitter buffer API, which does not use JB_TYPE_CONTROL frames. Review: https://reviewboard.asterisk.org/r/1814/ (closes issue ASTERISK-18964) Reported by: Kris Shaw Tested by: Kris Shaw, Matt Jordan Patches: jitterbuffer-2012-2-26.diff uploaded by Kris Shaw (license 5722) ........ Merged revisions 359356 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359358 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-03-14 17:39 +0000 [r359357] Richard Mudgett * apps/app_dial.c, main/channel.c, /: Fix Dial m and r options and forked calls generating warnings for voice frames. When connected line support was added, the wait_for_answer() variable single changed its meaning slightly. Unfortunately, the places where single was used did not necessarily get updated to reflect that change. Also audio/video frames were sent to all forked calls when the endpoints were never made compatible. * Don't pass audio/video media frames when the channels have not been made compatible. * Added handling of AST_CONTROL_SRCCHANGE to app_dial.c. * Fixed app_dial.c passing on AST_CONTROL_HOLD because that frame can also pass a requested MOH class. (closes issue ASTERISK-16901) Reported by: Chris Gentle (closes issue ASTERISK-17541) Reported by: clint Review: https://reviewboard.asterisk.org/r/1805/ ........ Merged revisions 359344 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359355 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-03-14 14:40 +0000 [r359306] Matthew Jordan * include/asterisk/astobj2.h: Force non-inlining of ao2_iterator_destroy when TEST_FRAMEWORK is enabled In r357272, astobj2 was changed to automatically enable REF_DEBUG when the TEST_FRAMEWORK flag was enabled. Unfortunately, some compilers (gcc 4.5.1 at least) will attempt to inline ao2_iterator_destroy in handle_astobj2_test. This by itself is not a problem; unfortunately, the compiler believes that there is a code path wherein an object allocated on the stack will be free'd. As warnings are treated as errors, this prevents compilation of astobj2. This patch works around that by adding the noinline attribue to ao2_iterator_destroy, but only if the TEST_FRAMEWORK flag is enabled. Preventing inlining is only needed for the test method defined in astobj2, which is also only enabled if TEST_FRAMEWORK is enabled. 2012-03-14 10:56 +0000 [r359052-359261] Russell Bryant * include/asterisk/logger.h, /, main/logger.c: Fix bogus reads/writes of console log levels in asterisk.c This patch updates the NUMLOGLEVELS define in logger.h to 32, to match the fact that logger.c implements 32 log levels (because of the custom log level stuff). asterisk.c uses this define to size an array of levels per remote console. This array is modified in ast_console_toggle_loglevel(), which is called by the "logger set level" CLI command. While the documentation for the CLI command doesn't make it terribly obvious, you can use this CLI command to toggle a custom log level on a remote console, as well. However, doing so led to an invalid array index in asterisk.c. This array is read from any time a log message is written to a console. So, all custom log level messages resulted in a bogus read if a remote console was connected. ........ Merged revisions 359259 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359260 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, apps/app_externalivr.c, channels/chan_iax2.c: Fix invalid reads/writes due to incorrect sizeof(). These few places in the code used sizeof() on h_addr in struct hostent. This is sizeof(char *). The correct way to get the size of this address is to use h_length. This error would result in reads/writes of 8 bytes instead of 4 on 64-bit machines. ........ Merged revisions 359211 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359212 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, main/sched.c: Fix inaccurate sizeof() in sched.c. This code just needed sizeof(int), not sizeof(int *). ........ Merged revisions 359157 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359162 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, utils/astman.c: Fix incorrect sizeof() in astman. ........ Merged revisions 359116 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359117 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, res/res_crypto.c: Fix incorrect usage of sizeof() in res_crypto. In this case, just remove the memset(). There was a redundant memset that is done correctly just 2 lines later. ........ Merged revisions 359110 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359114 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, res/res_adsi.c: Fix broken usage of sizeof() in res_adsi. ........ Merged revisions 359088 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359091 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, main/features.c: Fix incorrect sizeof() usage in features.c. This didn't actually result in a bug anywhere, luckily. The only place where the result of these memcpys was used is in app_dial, and the only field that it read out of ast_call_feature was the first one, which is an int, so these memcpys always copied just enough to avoid a problem. ........ Merged revisions 359069 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359072 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, main/md5.c: Fix incorrect sizeof() on a pointer in MD5Final(). ........ Merged revisions 359059 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359060 from http://svn.asterisk.org/svn/asterisk/branches/10 * main/pbx.c, /: Don't use a buffer after it goes out of scope. 's' is set to 'workspace'. Make sure 'workspace' doesn't go out of scope while the reference to it via 's' is still used. ........ Merged revisions 359056 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359057 from http://svn.asterisk.org/svn/asterisk/branches/10 * channels/chan_usbradio.c (removed), /, channels/xpmr (removed), build_tools/menuselect-deps.in, configure, include/asterisk/autoconfig.h.in, configure.ac, makeopts.in, apps/app_rpt.c (removed): Remove chan_usbradio and app_rpt. These modules are being maintained outside of the tree and have been for a long time now, so it doesn't make sense to keep them here. Review: https://reviewboard.asterisk.org/r/1764/ ........ Merged revisions 359050 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359051 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-03-13 21:24 +0000 [r359011] Terry Wilson * include/asterisk/channel_internal.h (added): Add missing channel_internal.h ...again. 2012-03-13 21:18 +0000 [r358997] Richard Mudgett * channels/sig_pri.h, channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, channels/sig_pri.c: Add ability for chan_dahdi ISDN to block connected line updates per span. Added new chan_dahdi.conf colp_send option parameter to block connected line updates per span. (closes issue ASTERISK-17025) Reported by: Michael Smith 2012-03-13 20:43 +0000 [r358907-358993] Terry Wilson * /, main/features.c: Fix setting CDR variables in the hangup extension A previous CDR fix for setting CDR variables during a bridge via custom dialplan features broke setting CDR variables in the hangup extension. This patch fixes the issue. Review: https://reviewboard.asterisk.org/r/1794/ ........ Merged revisions 358978 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 358989 from http://svn.asterisk.org/svn/asterisk/branches/10 * include/asterisk/devicestate.h, /, channels/chan_sip.c, tests/test_devicestate.c, main/devicestate.c: Make hints for invalid SIP devices return Unavail, not idle This patch drastically simplifies the device state aggegation code. The old method was not only overly complex, but also made it impossible to return AST_DEVICE_INVALID from the aggregation code. The unit test update is as a result of fixing that bug. The SIP change stems from a bug introduced by removing a DNS lookup for hostname-based SIP channels. (closes issue ASTERISK-16702) Review: https://reviewboard.asterisk.org/r/1808/ ........ Merged revisions 358943 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 358944 from http://svn.asterisk.org/svn/asterisk/branches/10 * apps/app_voicemail.c: Fix IMAP storage compilation after opaquification changes (closes issue ASTERISK-19513) * channels/chan_unistim.c, main/autoservice.c, channels/chan_vpb.cc, channels/chan_local.c, main/rtp_engine.c, res/res_musiconhold.c, bridges/bridge_multiplexed.c, apps/app_followme.c, main/indications.c, main/cli.c, main/channel.c, channels/chan_phone.c, channels/chan_dahdi.c, channels/sig_analog.c, main/manager.c, main/features.c, apps/app_dumpchan.c, res/res_agi.c, main/app.c, apps/app_confbridge.c, apps/app_externalivr.c, main/bridging.c, apps/app_parkandannounce.c, apps/app_dial.c, main/pbx.c, channels/chan_sip.c, channels/chan_bridge.c, main/channel_internal_api.c, channels/chan_agent.c, apps/app_disa.c, include/asterisk/channel.h, apps/app_talkdetect.c, apps/app_queue.c, apps/app_speech_utils.c, apps/app_channelredirect.c, main/file.c, res/snmp/agent.c, apps/app_macro.c, apps/app_stack.c, apps/app_chanspy.c, apps/app_mixmonitor.c: Finalize ast_channel opaquification Review: https://reviewboard.asterisk.org/r/1786/ 2012-03-13 17:01 +0000 [r358858-358861] Richard Mudgett * main/channel.c: Fix crash caused by opaquification change -r356042. The set_format() function was more subtle in how it modified the struct ast_channel readtrans/writetrans values. * Fixed ast_activate_generator() conversion correctly. (closes issue ASTERISK-19434) Reported by: Birger Harzenetter Tested by: rmudgett * main/format.c: Use struct copy instead of memcpy(). 2012-03-13 08:06 +0000 [r358812] Tilghman Lesher * res/ael/pval.c, funcs/func_dialplan.c, /, tests/test_gosub.c, utils/ael_main.c, apps/app_stack.c, utils/conf2ael.c: Enable macros in 1.8 to find the next highest "h" extension in a context, like in 1.4. This change restores functionality that was present in 1.4, when AEL macros were implemented with the Macro dialplan application. Macros are fraught with functionality issues, because they consume a large portion of the underlying application stack. This limits the ability of AEL users to call many layers of subroutines, an issue which Gosub does not have (originally tested to 100,000 levels deep). Therefore, starting in 1.6.0, AEL macros were implemented with Gosub. However, there were some implicit behaviors of Macro, which were not replicated at the same time as with the transition to Gosub, one of which is documented in the related issue. In particular, the "h" extension is designed to execute not in the Macro context, but in the topmost calling context. Due to legacy issues with a misapplied bugfix many years ago, when a macro exited in 1.4, it looks in all calling contexts, bubbling up from the deepest level until it finds an "h" extension. Since AEL hides the complexity of the underlying dialplan logic from the AEL programmer, it's reasonable to assume that this behavior should not change in the transition from Asterisk 1.4 LTS to Asterisk 1.8 LTS, lest we break working AEL configurations in the transition to Asterisk 1.8 LTS. This fix is the result, which implements a search for the "h" extension in all calling Gosub contexts. Fixes ASTERISK-19336 Patch: 20120308__ael_bugfix_for_trunk__2.diff (License #5003) by Tilghman Lesher (with slight modifications for 1.8) Tested by: Johan Wilfer Review: https://reviewboard.asterisk.org/r/1776/ ........ Merged revisions 358810 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 358811 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-03-12 17:01 +0000 [r358766] Igor Goncharovskiy * channels/chan_unistim.c, contrib/unistimLang/ru.po (added), contrib/unistimLang/ru.po.utf8 (added), configs/unistim.conf.sample, UPGRADE.txt, CHANGES, contrib/unistimLang/en.po (added), contrib/unistimLang (added): Massive changes in chan_unistim channel driver. Include many fixes in channel driver operation and add additional functionality: * Added ability to use multiple lines on phone, so for one device in configuration multiple lines can be defined, it allows to have multiple calls on one phone, callwaiting and switching between calls. * Added ability for translation on-screen menu to multiple languages. Tested on Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4, ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen menu of phone * Other described in CHANGES file Testing done by issue tracker users: ibercom, scsiborg, idarwin, TeknoJuce, c0rnoTa. Tested on production system by Jonn Taylor (jonnt) using phone models: Nortel i2004, 1120E and 1140E. (closes issue ASTERISK-16890) Review: https://reviewboard.asterisk.org/r/1243/ 2012-03-10 20:06 +0000 [r358730] Joshua Colp * configs/confbridge.conf.sample, main/dial.c, apps/app_page.c, apps/confbridge/include/confbridge.h, apps/app_confbridge.c, include/asterisk/dial.h, CHANGES, apps/confbridge/conf_config_parser.c: Transition app_page to using app_confbridge internally for the conference bridge portion of paging. This also adds a new 'announcement' option to ConfBridge user profiles. Review: https://reviewboard.asterisk.org/r/1754/ 2012-03-08 17:48 +0000 [r358646-358691] Sean Bright * apps/app_dial.c, apps/app_directory.c, apps/app_queue.c: Resolve a few more cases of variable shadowing. * channels/chan_phone.c, channels/chan_skinny.c, channels/chan_agent.c, pbx/pbx_lua.c, pbx/pbx_dundi.c, channels/chan_gtalk.c, pbx/pbx_config.c, channels/chan_oss.c, apps/confbridge/conf_config_parser.c: Eliminate a bunch of shadow warnings. * include/asterisk/linkedlists.h: Add some underscores in a few of our llist macros to reduce name collisions. 2012-03-08 16:59 +0000 [r358645] Jonathan Rose * /, channels/chan_sip.c: Make transfer not ignore port information with SIP. Attempting to transfer with SIP to an address like 1XXXXX@ip.ad.re.ss:5061 would fail because port would be cut from the host string and ignored. This simply keeps chan_sip from cutting off the port number during these kinds of transfers. (closes issue ASTERISK-19321) Reported by: Federico Alves Review: https://reviewboard.asterisk.org/r/1790/diff/#index_header ........ Merged revisions 358643 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 358644 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-03-08 16:21 +0000 [r358609-358622] Sean Bright * Makefile, configure, configure.ac, makeopts.in: Add --enable-dev-mode=strict to configure. Passing -Wshadow to gcc enables shadow warnings. From the gcc manual: Warn whenever a local variable or type declaration shadows another variable, parameter, type, or class member (in C++), or whenever a built-in function is shadowed. Asterisk will not currently compile with this option set, but a number of bugs have been discovered by enabling this flag on specific files. The long-term goal is to eliminate all of the suspect code that causes this warning to be emitted. * Makefile: Whitespace only change to the Makefile 2012-03-07 21:28 +0000 [r358576] Terry Wilson * cel/cel_odbc.c, configs/cel_odbc.conf.sample: Handle numeric columns for eventtype properly in cel_odbc Patch also implements correct handling of datetime2 and datetimeoffset new datatypes in SQL Server 2008 and 2008 R2. (closes issue ASTERISK-17548) Review: https://reviewboard.asterisk.org/r/1160/ Review: https://reviewboard.asterisk.org/r/1804/ 2012-03-07 18:33 +0000 [r358532] Richard Mudgett * /, channels/sig_ss7.c: Change directly setting _softhangup in sig_ss7.c to use ast_softhangup_nolock(). Update to: (issue ASTERISK-19372) ........ Merged revisions 358530 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 358531 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-03-07 16:16 +0000 [r358486] Sean Bright * /, codecs/codec_dahdi.c: Return g729 and g723.1 frames with the number of samples set properly. If the wctc4xxp returns more than a single packet, we need to update the number of samples in the returned frame accordingly. Acked-by: Shaun Ruffell ........ Merged revisions 358484 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 358485 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-03-07 15:19 +0000 [r358437-358444] Terry Wilson * /, configs/cdr_adaptive_odbc.conf.sample: Set snarkiness = 0 in cdr_adaptive_odbc.conf.sample ........ Merged revisions 358438 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 358441 from http://svn.asterisk.org/svn/asterisk/branches/10 * cel/cel_odbc.c, /, cdr/cdr_adaptive_odbc.c: Add detection for ODBC WCHAR fields Without detecting these types, cel_odbc blows up when the character set for the table is utf8. This also wraps cdr_adaptive_odbc's use of those types in the HAVE_ODBC_WCHAR #ifdef seen in other parts of the code. ........ Merged revisions 358435 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 358436 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-03-06 17:47 +0000 [r358262-358379] Richard Mudgett * channels/chan_dahdi.c, /: Fix ring cadance setup for outgoing calls on FXS ports. * Fix referencing the wrong variable in chan_dahdi.c:my_set_cadence(). Thanks to Sean Bright for compiling with -Wshadow and finding this bug. ........ Merged revisions 358377 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 358378 from http://svn.asterisk.org/svn/asterisk/branches/10 * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, CHANGES: Add dialtone_detect option for analog incoming calls. For analog lines, enables Asterisk to use dialtone detection per channel if an incoming call was hung up before it was answered. If dialtone is detected, the call is hung up. no: Disabled. (Default) yes: Look for dialtone for 10000 ms after answer. : Look for dialtone for the specified number of ms after answer. always: Look for dialtone for the entire call. Dialtone may return if the far end hangs up first. dialtone_detect=yes dialtone_detect=5000 dialtone_detect=always (closes issue ASTERISK-19316) Reported by: Jeremy Pepper Patch by: Jeremy Pepper Tested by: rmudgett,Jeremy Pepper Review: https://reviewboard.asterisk.org/r/1737/ * /, channels/sig_ss7.c: Drop SS7 call if not connected yet when INCOMPLETE/BUSY/CONGESTION. SS7 is a trunk protocol and should clear a failed call as soon as possible. * Made SS7 hangup a call immediately if it has not connected yet for INCOMPLETE/BUSY/CONGESTION causes. Otherwise, play an appropriate inband tone. (closes issue ASTERISK-19372) Reported by: Igor Nikolaev ........ Merged revisions 358278 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 358284 from http://svn.asterisk.org/svn/asterisk/branches/10 * include/asterisk/channel.h: Make usage of DECLARE_STRINGFIELD_SETTERS_FOR() not look so odd. * channels/chan_dahdi.c, channels/sig_ss7.h, /, channels/sig_ss7.c: Setup DSP when SS7 call is connected or early media is available. Outgoing SS7 calls fail to detect incoming DTMF so any bridged channel that requires out-of-band DTMF will not work. * Added sig_ss7_open_media() calls at appropriate places in sig_ss7.c. The new call converts conditionaled out unconverted code and shows that the code really did something useful. * Improved some chan_dahdi DTMF debug messages to help track DTMF handling. (closes issue ASTERISK-19312) Reported by: Igor Nikolaev ........ Merged revisions 358260 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 358261 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-03-05 19:06 +0000 [r358216] Jonathan Rose * main/manager.c, /: Eliminate double close of file descriptor in manager.c The process_output function in manager.c attempted to call fclose and close immediately afterwards. Since fclose implies close, this resulted in a potential double free on file descriptors. This patch changes that behavior and also adds error checking to fclose and close depending on which was deemed necessary. Also error messages. Thanks to Rosen Iliev for pointing out the location of the problem. (closes issue ASTERISK-18453) Reported By: Jaco Kroon Review: https://reviewboard.asterisk.org/r/1793/ ........ Merged revisions 358214 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 358215 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-03-05 16:44 +0000 [r358164] Joshua Colp * /, channels/chan_sip.c: Defer sending the connected line reinvite if a reinvite is already in progress. (issue ASTERISK-19355) Reported by: tomaso (closes issue AST-825) ........ Merged revisions 358162 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 358163 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-03-05 16:00 +0000 [r358117] Kinsey Moore * /, channels/chan_sip.c: Ensure Asterisk acknowledges ACKs to 4xx on Replaces errors Asterisk was not setting pendinginvite in the upper half of handle_request_invite such that the 4xx was retransmitted repeatedly even though an ack was received for every retransmission. (closes issue ASTERISK-19303) Reported by: Jon Tsiros Patches: fix-19303.patch uploaded by Jeremiah Gowdy (license 6358) ........ Merged revisions 358115 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 358116 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-03-05 11:20 +0000 [r358082] Sean Bright * configs/iax.conf.sample: Tab to spaces and text change. 2012-03-02 23:29 +0000 [r357999-358038] Terry Wilson * channels/chan_usbradio.c, /, channels/xpmr/xpmr.c: Fix unused-but-set-variable warnings All of these were pretty obviously unused. Some were unused because the code that used them was #if 0'd. In those cases, I just commented out the unused-but-set variables. ........ Merged revisions 358029 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 358033 from http://svn.asterisk.org/svn/asterisk/branches/10 * /: Correct some set-but-unused variable warnings in the mISDN library. (from kpfleming's commit to trunk r356292) ........ Merged revisions 358011 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 358017 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, channels/xpmr/xpmr.c: Make chan_usbradio compile under dev mode x=++x and x=x=1? Really? ........ Merged revisions 357986 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 357987 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-03-02 21:06 +0000 [r357942] Kinsey Moore * /, main/ccss.c, tests/test_event.c, main/event.c, include/asterisk/strings.h: Fix case-sensitivity for device-specific event subscriptions and CCSS This change fixes case-sensitivity for device-specific subscriptions such that the technology identifier is case-insensitive while the remainder of the device string is still case-sensitive. This should also preserve the original case of the device string as passed in to the event system. CCSS is the only feature affected as it is the only consumer of device-specific event subscriptions. The second part of this patch addresses similar case-sensitivity issues within CCSS itself that prevented it from functioning correctly after the fix to the events system. This adds a unit test to verify that the event system works as expected. (closes issue ASTERISK-19422) Review: https://reviewboard.asterisk.org/r/1780/ ........ Merged revisions 357940 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 357941 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-03-02 18:38 +0000 [r357896] Richard Mudgett * main/channel.c, /, channels/sig_pri.c: Remove ISDN hold restriction for non-bridged calls. The check if an ISDN call is bridged before it could be placed on hold is not necessary and is overly restrictive. The check was originally done to prevent problems with call transfers in case a user tried to transfer a call connected to an application to another call connected to an application. The ISDN transfer code has not required this restriction for quite some time because ECT could transfer any two active calls to each other. * Remove ISDN hold restriction for calls connected to applications. * Made ast_waitfordigit_full() ignore AST_CONTROL_HOLD and AST_CONTROL_UNHOLD instead of generating a warning message. (closes issue ASTERISK-19388) Reported by: Birger Harzenetter Tested by: rmudgett ........ Merged revisions 357894 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 357895 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-03-02 16:57 +0000 [r357861] Jonathan Rose * apps/app_queue.c: Adds a transfer callee on hangup option (like with Dial option F) to queues. This should (and does in my testing) act just like the Dial option of the same name. This allows a queue member to be transfered to the next priority (no args), or to a context/extension/priority similar to goto (with args context^extension^priority) when a caller hangs up on them. (closes issue ASTERISK-19283) Reported by: To Patches: queue_f-v3.diff uploaded by To (license 6347) Review: https://reviewboard.asterisk.org/r/1785/ 2012-03-02 16:26 +0000 [r357834] Richard Mudgett * apps/app_chanspy.c: Remove bad usage of goto in ChanSpy next_channel(). 2012-03-02 16:19 +0000 [r357821] Sean Bright * configs/iax.conf.sample: Beef up the IAX2 sample configuration a bit and fix some formatting issues. 2012-03-02 16:03 +0000 [r357814-357815] Richard Mudgett * /, apps/app_chanspy.c: Fix channel reference leak in ChanSpy. * Fix next_channel() channel reference leak in ChanSpy. (closes issue ASTERISK-19461) Reported by: Irontec Patches: app_chanspy_iteartor_next_unref.patch (license #6213) patch uploaded by Irontec (issue ASTERISK-17515) ........ Merged revisions 357809 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 357810 from http://svn.asterisk.org/svn/asterisk/branches/10 * channels/chan_usbradio.c: Fix compile error from latest channel opaquification change. 2012-03-02 16:00 +0000 [r357813] Sean Bright * /, channels/chan_iax2.c: The default value for mohinterpret is the empty string, so when resetting to default values don't explicitly set the value to "default." ........ Merged revisions 357811 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 357812 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-03-02 01:33 +0000 [r357774-357775] Mark Michelson * main/channel.c, /: Fix race condition that can cause important control frames (such as a hangup) to be missed. This takes two actions. 1. Move the reading of the alertpipe in __ast_read() to immediately before the removal of frames from the readq. This means we won't do something silly like read from the alertpipe, then ignore the fact that there's a frame to get from the readq since channel's fdno is the AST_TIMING_FD. 2. When ast_settimeout() sets the rate to 0 and the timingfunc to NULL, if the channel's fdno is the AST_TIMING_FD, then set the fdno to -1. This is because if the rate is 0 and the timingfunc is NULL, it means that the channel's timing fd is being invalidated, so any pending reads should not occur. This may actually solve more issues than the referenced one below, but it's not known at this time for sure. (closes issue ASTERISK-19223) reported by Frank-Michael Wittig Review: https://reviewboard.asterisk.org/r/1779 ........ Merged revisions 357761 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 357762 from http://svn.asterisk.org/svn/asterisk/branches/10 * channels/chan_dahdi.c: Fix compilation error due to typo during channel opaquification. s/ast_channel_fd_set/ast_channel_internal_fd_set/g 2012-03-01 22:09 +0000 [r357721] Terry Wilson * channels/chan_unistim.c, apps/app_dahdibarge.c, main/autoservice.c, addons/chan_ooh323.c, channels/chan_vpb.cc, apps/app_meetme.c, channels/console_video.c, channels/chan_gtalk.c, channels/chan_iax2.c, main/cli.c, main/channel.c, channels/chan_phone.c, channels/chan_dahdi.c, channels/sig_analog.c, channels/chan_skinny.c, main/features.c, apps/app_dumpchan.c, channels/sig_ss7.c, channels/chan_mgcp.c, main/pbx.c, channels/chan_sip.c, main/channel_internal_api.c, channels/chan_agent.c, apps/app_dahdiras.c, include/asterisk/channel.h, apps/app_queue.c, channels/sig_pri.c, channels/chan_jingle.c, channels/chan_misdn.c, apps/app_flash.c, funcs/func_channel.c, apps/app_directed_pickup.c, main/file.c, channels/chan_h323.c, res/snmp/agent.c, main/dsp.c: Opaquify ast_channel typedefs, fd arrays, and softhangup flag Review: https://reviewboard.asterisk.org/r/1784/ 2012-03-01 14:22 +0000 [r357673] Kinsey Moore * /, main/acl.c: Prevent outbound SIP NOTIFY packets from displaying a port of 0 In the change from 1.6.2 to 1.8, ast_sockaddr was introduced which changed the behavior of ast_find_ourip such that port number was wiped out. This caused the port in internip (which is used for Contact and Call-ID on NOTIFYs) to be 0. This change causes ast_find_ourip to be port-preserving again. (closes issue ASTERISK-19430) ........ Merged revisions 357665 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 357667 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-29 20:41 +0000 [r357621] Walter Doekes * /, main/utils.c, include/asterisk/stringfields.h: Update stringfield documentation for removed second va_list in favor of va_copy. In r320946, the second va_list that was passed to ast_string_field_build_va and friends, was removed. This patch updates the documentation to reflect that. ........ Merged revisions 357620 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-29 20:31 +0000 [r357610] Sean Bright * res/res_agi.c, CHANGES: Add IPv6 support to FastAGI. Review: https://reviewboard.asterisk.org/r/1774/ Reviewed by: Simon Perreault, Mark Michelson 2012-02-29 19:48 +0000 [r357577] Walter Doekes * apps/app_dial.c, /: Fix copying of CDR(accountcode) to local channels. In r203638, during the addition of the Channel Event Logging, in mid-2009, this got broken in trunk and ended up in asterisk 1.8 and higher. This fixes so the CDR(accountcode) from the calling channel is available to dialed channels again as well as showing up properly in the CDR's. (closes issue ASTERISK-19384) Reported by: jamicque Patches: accountcode.patch (License #6033) by jamicque Review: https://reviewboard.asterisk.org/r/1775/ Reviewed by: Richard Mudgett ........ Merged revisions 357575 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 357576 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-29 16:52 +0000 [r357542] Terry Wilson * channels/chan_local.c, addons/chan_ooh323.c, funcs/func_strings.c, channels/console_video.c, apps/app_alarmreceiver.c, channels/chan_iax2.c, main/cli.c, channels/chan_dahdi.c, channels/sig_analog.c, channels/chan_skinny.c, apps/app_dumpchan.c, main/features.c, apps/app_amd.c, channels/sig_ss7.c, apps/app_dial.c, main/pbx.c, include/asterisk/utils.h, funcs/func_timeout.c, apps/app_privacy.c, apps/app_fax.c, channels/chan_agent.c, apps/app_disa.c, include/asterisk/channel.h, apps/app_talkdetect.c, main/cel.c, channels/chan_misdn.c, apps/app_macro.c, apps/app_zapateller.c, apps/app_mixmonitor.c, apps/app_voicemail.c, channels/chan_unistim.c, tests/test_substitution.c, channels/chan_vpb.cc, apps/app_meetme.c, main/ccss.c, apps/app_readexten.c, channels/chan_gtalk.c, main/autochan.c, apps/app_followme.c, main/cdr.c, main/channel.c, main/dial.c, channels/chan_phone.c, apps/app_osplookup.c, apps/app_setcallerid.c, main/manager.c, bridges/bridge_builtin_features.c, apps/app_minivm.c, res/res_agi.c, main/app.c, apps/app_confbridge.c, apps/app_rpt.c, main/message.c, channels/chan_mgcp.c, apps/app_parkandannounce.c, apps/app_while.c, funcs/func_dialplan.c, channels/chan_sip.c, res/res_fax.c, main/channel_internal_api.c, pbx/pbx_lua.c, channels/chan_console.c, channels/sig_pri.c, apps/app_queue.c, channels/chan_oss.c, channels/chan_jingle.c, channels/chan_usbradio.c, funcs/func_blacklist.c, main/abstract_jb.c, channels/chan_h323.c, main/file.c, res/snmp/agent.c, apps/app_sms.c, apps/app_stack.c, funcs/func_callerid.c: Opaquify ast_channel structs and lists Review: https://reviewboard.asterisk.org/r/1773/ 2012-02-28 22:31 +0000 [r357460-357503] Jonathan Rose * /, configs/sip.conf.sample, UPGRADE-1.8.txt: Adding transport=udp to sample sip.conf - Also changes version of Asterisk 1.8 in UPGRADE (issue ASTERISK-19352) Reported by: jamicque Patches: asterisk-19352-transport-warning-message-v1.patch uploaded by Michael L. Young (license 5026) ........ Merged revisions 357490 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 357497 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, cdr/cdr_adaptive_odbc.c: Add additional character type types to supported data types for cdr_adaptive_odbc The reporter was uable to use varchar utf8_unicode_ci with cdr_adaptive_odbc, so this patch adds those along with some other character types to the list of types cdr_adaptive_odbc will work using the varchar conditions. The problem wasn't really UTF8 characters as much as it was a failure to respond to the exact type that was declared/in use on that database. (closes issue ASTERISK-19334) Reported By: Igor Nikolaev Patches: cdr_adaptive_odbc.patch uploaded by Igor Nikolaev (license 6236) ........ Merged revisions 357455 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 357458 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-28 21:26 +0000 [r357436] Tilghman Lesher * /, apps/app_stack.c: Correctly reset the dialplan priority. When the stack frame is allocated, we save the address to which we should return, when the Gosub returns. However, if we just want to restore the priority, then we need to subtract 1 before setting it. Otherwise, when a Gosub goes to a nonexistent address, it will skip a priority in the dialplan. This is because when we return from an application, the PBX increments the priority for us. ........ Merged revisions 357416 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 357421 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-28 21:01 +0000 [r357409] Richard Mudgett * /, channels/sig_pri.c: Use more reasonable cause code when rejecting incoming call waiting calls. (closes issue ASTERISK-19397) Reported by: Birger Harzenetter Patches: nochannel-cause.patch (license #5870) patch uploaded by Birger Harzenetter ........ Merged revisions 357407 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 357408 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-28 20:43 +0000 [r357406] Jonathan Rose * /, UPGRADE-10.txt: revision 357386 -- oops, accidentally made it 10.3 to 10.4 instead of 10.2 to 10.3 (issue ASTERISK-19352) reported by: jamicque ........ Merged revisions 357405 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-28 20:34 +0000 [r357404] Richard Mudgett * main/channel.c, res/res_musiconhold.c, apps/app_queue.c: Fix REF_DEBUG compile errors. 2012-02-28 20:33 +0000 [r357358-357403] Jonathan Rose * /, UPGRADE-10.txt, UPGRADE-1.8.txt: Moves UPGRADE.txt notes from r357356 to a new section specific to 1.8.12 (issue ASTERISK-19352) reported by: jamicque ........ Merged revisions 357386 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 357400 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, UPGRADE-1.8.txt: Adds UPGRADE.txt notes to r357266 indicating changes to transport option (issue ASTERISK-19352) Reported by: jamicque ........ Merged revisions 357356 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 357357 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-28 19:55 +0000 [r357355] Sean Bright * include/asterisk/netsock2.h: Documentation update. There is no AST_SOCKADDR_UNSPEC. 2012-02-28 19:37 +0000 [r357354] Richard Mudgett * /, apps/app_page.c: Remove dupliate 'i' option table entry in app_page.c. (closes issue ASTERISK-19310) Reported by: Makoto Dei Patches: app_page-duplicate-i-option.patch (license #5027) patch uploaded by Makoto Dei ........ Merged revisions 357352 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 357353 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-28 18:52 +0000 [r357319] Mark Michelson * /, channels/sip/security_events.c: Add a security event for the case where fake authentication challenge is sent. ........ Merged revisions 357318 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-28 18:46 +0000 [r357317] Richard Mudgett * main/tcptls.c, channels/chan_sip.c, include/asterisk/tcptls.h: Convert struct ast_tcptls_session_instance to finally use the ao2 object lock. 2012-02-28 18:23 +0000 [r357288] Jonathan Rose * /, channels/chan_sip.c: Changes transport option in sip.conf so that using multiple instances doesn't stack. Prior to this patch, Using "transport=" multiple times would cause them to add to one another like allow/deny. This patch changes that behavior to simply use the transport option specified last. Also, if no transport option is applied now, the default will automatically be UDP. (closes ASTERISK-19352) Reported by: jamicque Patches: asterisk-19352-transport-warning-message-v1.patch uploaded by Michael L. Young (license 5026) issueA19352_no_transport_is_udp.patch uploaded by Walter Doekes (license 5674) Review: https://reviewboard.asterisk.org/r/1745/diff/#index_header ........ Merged revisions 357266 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 357271 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-28 18:15 +0000 [r357272] Richard Mudgett * main/format.c, main/format_cap.c, include/asterisk/astobj2.h, include/asterisk/lock.h, main/astobj2.c: Astobj2 locking enhancement. Add the ability to specify what kind of locking an ao2 object has when it is allocated. The locking could be one of: MUTEX, RWLOCK, or none. New API: ao2_t_alloc_options() ao2_alloc_options() ao2_t_container_alloc_options() ao2_container_alloc_options() ao2_rdlock() ao2_wrlock() ao2_tryrdlock() ao2_trywrlock() The OBJ_NOLOCK and AO2_ITERATOR_DONTLOCK flags have a slight meaning change. They no longer mean that the object is protected by an external mechanism. They mean the lock associated with the object has already been manually obtained by one of the ao2_lock calls. This change is necessary for RWLOCK support since they are not reentrant. Also an operation on an ao2 container may require promoting a read lock to a write lock by releasing the already held read lock to re-acquire as a write lock. Replaced API calls: ao2_t_link_nolock() ao2_link_nolock() ao2_t_unlink_nolock() ao2_unlink_nolock() with the respective ao2_t_link_flags() ao2_link_flags() ao2_t_unlink_flags() ao2_unlink_flags() API calls to be more flexible and to allow an anticipated enhancement to control linking duplicate objects into a container. The changes to format.c and format_cap.c are taking advantange of the new ao2 locking options to simplify the use of the format capabilities containers. Review: https://reviewboard.asterisk.org/r/1554/ 2012-02-28 14:47 +0000 [r357178-357214] Kevin P. Fleming * /, Makefile.rules: Make COMPILE_DOUBLE magic actually work. The build system has some special magic to ensure that if Asterisk is built with --enable-dev-mode *and* DONT_OPTIMIZE, that all the source is still compiled with the optimizer enabled (even though the result will be thrown away), because the compiler is able to find a great deal of coding errors and bugs as a result of running its optimizers. Unfortunately at some point this mode got broken, and the 'throwaway' compile of the code was no longer done with the optimizer enabled. This patch corrects that problem. ........ Merged revisions 357212 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 357213 from http://svn.asterisk.org/svn/asterisk/branches/10 * main/astobj2.c: Trailing whitespace cleanup. 2012-02-28 00:42 +0000 [r357096-357145] Richard Mudgett * include/asterisk/astobj2.h, tests/test_astobj2.c, main/astobj2.c: Add ability to clone ao2 containers. Occasionally there is a need to put all objects in one container also into another container. Some reasons you might need to do this: 1) You need to reconfigure a container. You would do this by creating a new container with the new configuration and ao2_container_dup the old container into it. Then replace the old container with the new. Then destroy the old container. 2) You need the contents of a container to remain stable while operating on all of the objects. You would do this by creating a cloned container of the original with ao2_container_clone. The cloned container is a snapshot of the objects at the time of the cloning. When done, just destroy the cloned container. Review: https://reviewboard.asterisk.org/r/1746/ * main/channel.c: Fix ast_channel allocation init setting priority to -1 instead of 1. * Fix opaquification conversion error. (closes issue ASTERISK-19424) Reported by: Jeremy Pepper Patches: asterisk-19424-initialize_priority_regression.diff (license #5026) patch uploaded by Michael L. Young * main/channel.c, /: Fix callerid of Originated calls. Thanks to Matt Riddell for tracking this down. (closes issue ASTERISK-19385) Reported by: ornix ........ Merged revisions 357093 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 357095 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-27 19:55 +0000 [r357051] Jonathan Rose * include/asterisk/res_odbc.h, res/res_odbc.c: Converts locking for odbc containers from ast_mutex_lock to ao2_locks. 2012-02-27 17:03 +0000 [r357014] Sean Bright * channels/chan_iax2.c, main/netsock.c: Address comments from Mark Michelson 2012-02-27 16:50 +0000 [r357013] Kinsey Moore * apps/app_dial.c, main/channel.c, include/asterisk/app.h, main/dial.c, main/rtp_engine.c, main/ccss.c, main/features.c, UPGRADE.txt, main/app.c, include/asterisk/channel.h, configs/ccss.conf.sample, apps/app_followme.c, apps/app_queue.c, include/asterisk/ccss.h: Deprecated macro usage for connected line, redirecting, and CCSS This commit adds GoSub alternatives to connected line, redirecting, and CCSS macro hooks so that macro can finally be deprecated. This also adds deprecation warnings for those features when used and in documentation. Review: https://reviewboard.asterisk.org/r/1760/ (closes issue SWP-4256) 2012-02-27 16:31 +0000 [r357005] Sean Bright * include/asterisk/netsock.h, channels/chan_iax2.c, main/netsock.c: Convert netsock.h over to use ast_sockaddrs rather than sockaddr_in and update chan_iax2 to pass in the correct types. chan_iax2 is the only consumer for the various ast_netsock_* functions in trunk at this point, so this feels like a safe change to make. 2012-02-27 16:24 +0000 [r356987] Jonathan Rose * channels/chan_sip.c, configs/sip.conf.sample, CHANGES, channels/sip/include/sip.h: Adds an option to sip.conf that prevents diversion headers from being added. send_diversion=no will prevent Diversion headers from being added to SIP requests. This doesn't prevent Diversion from being added with dialplan such as with SIPAddHeader. (closes issue ASTERISK-16862) Reported by: rsw686 Review: https://reviewboard.asterisk.org/r/1769/ 2012-02-27 16:12 +0000 [r356966] Sean Bright * channels/chan_iax2.c: There isn't much point in saving off and restoring a value that we never use again. 2012-02-27 16:08 +0000 [r356965] Terry Wilson * /, main/features.c: Copy CDR variables when set during a bridge This patch makes sure amaflags, accountcode, and userfield get copied to the bridge CDR when set during a bridge (like via a custom feature). (closes issue ASTERISK-16990) Review: https://reviewboard.asterisk.org/r/1721/ ........ Merged revisions 356963 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 356964 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-27 15:35 +0000 [r356962] Jonathan Rose * /, res/res_odbc.c: Remove possible segfaults from res_odbc by adding locks around usage of odbc handle (closes issue ASTERISK-19011) Reported by: Walter Doekes Patches: issueA19011_combine_read_and_write_locks_WORK_IN_PROGRESS.patch uploaded by Walter Doekes (license 5674) review: https://reviewboard.asterisk.org/r/1719/ review: https://reviewboard.asterisk.org/r/1622/ ........ Merged revisions 356917 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 356961 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-27 14:57 +0000 [r356881-356916] Sean Bright * include/asterisk/netsock.h, main/netsock.c: Make ast_netsock_set_qos() delegate to ast_set_qos(). * include/asterisk/netsock.h: Correct typo in deprecation comment. * channels/chan_unistim.c, main/udptl.c, channels/chan_skinny.c, include/asterisk/netsock.h, pbx/pbx_dundi.c, channels/chan_mgcp.c: Prefer ast_set_qos() over ast_netsock_set_qos() * main/netsock.c: Remove trailing whitespace 2012-02-26 18:25 +0000 [r356848] Alexandr Anikin * addons/ooh323c/src/ooGkClient.c, addons/chan_ooh323.c: Add support change gatekeeper mode or ip per ooh323 reload command (issue ASTERISK-19298) Reported by: Dmitry Melekhov Patches: change_gk_on_reload-1.patch (License #5415) 2012-02-25 17:22 +0000 [r356799] Matthew Jordan * /, apps/app_voicemail.c: Fix crash in app_voicemail during close_mailbox In r354890, a memory leak in app_voicemail was fixed by properly disposing of the allocated heard/deleted pointers. However, there are situations, particularly when no messages are found in a folder, where these pointers are not allocated and not NULL. In that case, an invalid free would be attempted, which could crash app_voicemail. As there are a number of code paths where this could occur, this patch uses the number of messages detected in the folder before it attempts to free the pointers. This resolves the crash detected in the Asterisk Test Suite's check_voicemail_nominal test. ........ Merged revisions 356797 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 356798 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-24 23:40 +0000 [r356697-356765] Richard Mudgett * include/asterisk/astobj2.h: astobj2.h comment tweaks. * include/asterisk/astobj2.h, main/astobj2.c: astobj2.h documentation updates. * /, channels/chan_sip.c, include/asterisk/tcptls.h, channels/sip/include/sip.h: Fix worker thread resource leak in SIP TCP/TLS. The SIP TCP/TLS worker threads were created joinable but noone could join them if they died on their own. * Fix the SIP TCP/TLS worker threads to not be created joinable. * _sip_tcp_helper_thread() only needs one parameter since the pvt parameter is only passed in as NULL and never used. (closes issue ASTERISK-19203) Reported by: Steve Davies Review: https://reviewboard.asterisk.org/r/1714/ ........ Merged revisions 356677 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 356690 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-24 17:43 +0000 [r356606-356652] Matthew Jordan * /, res/res_srtp.c: Remove srtp_shutdown from res_srtp The patch for ASTERISK-19253 included properly shutting down the libsrtp library in the case of module unload. Unfortunately, not all distributions have the srtp_shutdown call. As such, this patch removes calling srtp_shutdown. ........ Merged revisions 356650 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 356651 from http://svn.asterisk.org/svn/asterisk/branches/10 * channels/sip/sdp_crypto.c, include/asterisk/res_srtp.h, main/rtp_engine.c, /, include/asterisk/rtp_engine.h, res/res_srtp.c: Allow SRTP policies to be reloaded Currently, when using res_srtp, once the SRTP policy has been added to the current session the policy is locked into place. Any attempt to replace an existing policy, which would be needed if the remote endpoint negotiated a new cryptographic key, is instead rejected in res_srtp. This happens in particular in transfer scenarios, where the endpoint that Asterisk is communicating with changes but uses the same RTP session. This patch modifies res_srtp to allow remote and local policies to be reloaded in the underlying SRTP library. From the perspective of users of the SRTP API, the only change is that the adding of remote and local policies are now added in a single method call, whereas they previously were added separately. This was changed to account for the differences in handling remote and local policies in libsrtp. Review: https://reviewboard.asterisk.org/r/1741/ (closes issue ASTERISK-19253) Reported by: Thomas Arimont Tested by: Thomas Arimont Patches: srtp_renew_keys_2012_02_22.diff uploaded by Matt Jordan (license 6283) (with some small modifications for this check-in) ........ Merged revisions 356604 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 356605 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-24 00:32 +0000 [r356573] Terry Wilson * channels/chan_unistim.c, channels/chan_local.c, addons/chan_ooh323.c, channels/chan_multicast_rtp.c, channels/chan_vpb.cc, main/rtp_engine.c, apps/app_meetme.c, apps/app_dictate.c, apps/app_record.c, apps/app_test.c, bridges/bridge_softmix.c, channels/chan_gtalk.c, apps/app_ices.c, res/res_musiconhold.c, channels/chan_iax2.c, bridges/bridge_multiplexed.c, main/indications.c, main/cli.c, main/channel.c, channels/chan_phone.c, channels/chan_dahdi.c, channels/chan_skinny.c, res/res_agi.c, main/features.c, apps/app_mp3.c, apps/app_dumpchan.c, main/app.c, apps/app_amd.c, channels/chan_alsa.c, apps/app_confbridge.c, addons/chan_mobile.c, main/bridging.c, channels/chan_mgcp.c, apps/app_nbscat.c, main/pbx.c, channels/chan_sip.c, res/res_fax.c, apps/app_festival.c, channels/chan_bridge.c, main/channel_internal_api.c, apps/app_fax.c, apps/app_waitforsilence.c, res/res_adsi.c, channels/chan_agent.c, bridges/bridge_simple.c, include/asterisk/channel.h, channels/chan_console.c, apps/app_talkdetect.c, channels/chan_oss.c, apps/app_speech_utils.c, channels/chan_usbradio.c, channels/chan_jingle.c, channels/chan_misdn.c, funcs/func_channel.c, main/file.c, channels/chan_nbs.c, apps/app_chanspy.c, apps/app_voicemail.c, res/res_calendar.c: Opaquification for ast_format structs in struct ast_channel Review: https://reviewboard.asterisk.org/r/1770/ 2012-02-23 20:14 +0000 [r356523] Richard Mudgett * /, channels/chan_sip.c, main/features.c: Fix blind transfer parking issues if the dialed extension is not recognized as a parking extension. Custom parking extensions may not be coded such that the first and only extension priority is the Park application. These custom parking extensions will not be recognized as parking extensions. When a call is blind transferred to an extension that is not recognized as a parking extension, the normal blind transfer code causes the transferred channel to start executing dialplan. Calls that get parked in this manner do not know the original channel name that parked the call so the original parker could never be called back if the parked call is not retrieved before the timeout time. The parking space is also announced to the call being parked as a side effect of not knowing the original parking channel. * Fix handling of BLINDTRANSFER channel variable for call parking. * Fixed SIP blind transfer using the wrong dialplan context variable to check for the parking extension. (closes issue ASTERISK-19322) Reported by: aragon Tested by: rmudgett, jparker Review: https://reviewboard.asterisk.org/r/1730/ JIRA AST-766 ........ Merged revisions 356521 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 356522 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-23 15:49 +0000 [r356477] Mark Michelson * /, channels/chan_sip.c: Fix ACK routing for non-2xx responses. When we send an ACK for a 2xx response to an INVITE, we are supposed to use the learned route set. However, when we receive a non-2xx final response to an INVITE, we are supposed to send the ACK to the same place we initially sent the INVITE. We had been doing this up until the changes went in that would build a route set from provisional responses. That introduced a regression where we would use the learned route set under all circumstances. With this change, we now will set the destination of our ACK based on the invitestate. If it is INV_COMPLETED then that means that we have received a non-2xx final response (INV_TERMINATED indicates a 2xx response was received). If it is INV_CANCELLED, then that means the call is being canceled, which means that we should be ACKing a 487 response. The other change introduced here is setting the invitestate to INV_CONFIRMED when we send an ACK *after* the reqprep instead of before. This way, we can tell in reqprep more easily what the invitestate is prior to sending the ACK. (closes issue ASTERISK-19389) reported by Karsten Wemheuer patches: ASTERISK-19389v2.patch uploaded by Mark Michelson (license #5049) (with some slight modifications prior to commit) ........ Merged revisions 356475 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 356476 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-23 03:27 +0000 [r356429] Paul Belanger * /, apps/app_rpt.c: Multiple revisions 356290,356335,356337 ........ r356290 | pabelanger | 2012-02-22 15:20:29 -0500 (Wed, 22 Feb 2012) | 4 lines Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2) Review: https://reviewboard.asterisk.org/r/1763/ ........ r356335 | pabelanger | 2012-02-22 16:29:25 -0500 (Wed, 22 Feb 2012) | 2 lines Add back strsep() function for previous commit ........ r356337 | pabelanger | 2012-02-22 16:36:37 -0500 (Wed, 22 Feb 2012) | 2 lines Missed one strsep() function ........ Merged revisions 356290,356335,356337 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 356428 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-23 01:53 +0000 [r356397] Terry Wilson * tests/test_substitution.c, tests/test_utils.c: Fix some tests that didn't get opaquification changes Review: https://reviewboard.asterisk.org/r/1766/ 2012-02-23 00:56 +0000 [r356366] Richard Mudgett * main/channel_internal_api.c: Revert some apparently accidental spacing changes. 2012-02-22 21:22 +0000 [r356314] Terry Wilson * /, include/asterisk/calendar.h, main/loader.c, res/res_calendar.c: Track module use count for res_calendar If the res_calendar module was followed immediately by one of the calendar tech modules and "core stop gracefully" was run, Asterisk would crash. This patch adds use count tracking for res_calendar so that it is unloaded after the tech modules when shutting down gracefully. It is now not possible to unload all the of the calendar modules via "module unload res_calednar.so", but it is still possible to unload them all via "module unload -h res_calendar.so". Review: https://reviewboard.asterisk.org/r/1752/ ........ Merged revisions 356291 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 356297 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-22 21:10 +0000 [r356292] Kevin P. Fleming * channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c: Correct some set-but-unused variable warnings in the mISDN library. 2012-02-22 17:34 +0000 [r356259] Terry Wilson * channels/chan_misdn.c: Fix chan_misdn after the lastest opaquification changes It now compiles, but there are some unrelated warnings for set but unused variables. 2012-02-22 14:54 +0000 [r356216] Matthew Jordan * /, channels/chan_sip.c: Merged revisions 356215 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r356215 | mjordan | 2012-02-22 08:53:53 -0600 (Wed, 22 Feb 2012) | 32 lines Merged revisions 356214 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r356214 | mjordan | 2012-02-22 08:50:20 -0600 (Wed, 22 Feb 2012) | 27 lines Fix potential buffer overrun and memory leak when executing "sip show peers" The "sip show peers" command uses a fix sized array to sort the current peers in the peers ao2_container. The size of the array is based on the current number of peers in the container. However, once the size of the array is determined, the number of peers in the container can change, as the peers container is not locked. This could cause a buffer overrun when populating the array, if peers were added to the container after the array was created. Additionally, a memory leak of the allocated array would occur if a user caused the _show_peers method to return CLI_SHOWUSAGE. We now create a snapshot of the current peers using an ao2_callback with the OBJ_MULTIPLE flag. This size of the array is set to the number of peers that the iterator will iterate over; hence, if peers are added or removed from the peers container it will not affect the execution of the "sip show peers" command. Review: https://reviewboard.asterisk.org/r/1738/ (closes issue ASTERISK-19231) (closes issue ASTERISK-19361) Reported by: Thomas Arimont, Jamuel Starkey Tested by: Thomas Arimont, Jamuel Starkey Patches: sip_show_peers_2012_02_16.diff uploaded by mjordan (license 6283) ........ ................ 2012-02-22 00:35 +0000 [r356152-356183] Terry Wilson * main/channel.c, main/channel_internal_api.c, include/asterisk/channel.h: Rename ast_channel_emulate_dtmf_digit* funcs The accessors names for the "emulate_dtmf_digit" field on the ast_channel are misleading. Change them to ast_channel_dtmf_digit_to_emulate*. * main/channel.c, main/framehook.c, res/res_monitor.c: Fix some opaquification-related compiler warnings (closes issue ASTERISK-19419) PseudoReview - seanbright on IRC 2012-02-21 11:17 +0000 [r356111] Sean Bright * /, channels/chan_iax2.c: Make 'iax2 show callnumber usage' output make sense when an IP is passed in. ........ Merged revisions 356107 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 356108 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-21 04:31 +0000 [r356075] Kinsey Moore * /, main/ccss.c: Add missing newline to ccss state change notification Move along, nothing to see here... ........ Merged revisions 356074 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-20 23:43 +0000 [r356042] Terry Wilson * main/udptl.c, apps/app_dahdibarge.c, addons/chan_ooh323.c, cdr/cdr_sqlite3_custom.c, channels/chan_local.c, main/rtp_engine.c, apps/app_playtones.c, apps/app_record.c, apps/app_sayunixtime.c, apps/app_test.c, main/devicestate.c, apps/app_alarmreceiver.c, apps/app_chanisavail.c, apps/app_ices.c, channels/chan_iax2.c, bridges/bridge_multiplexed.c, main/cli.c, channels/chan_dahdi.c, channels/sig_analog.c, main/framehook.c, channels/chan_skinny.c, main/features.c, apps/app_dumpchan.c, pbx/pbx_realtime.c, channels/chan_alsa.c, apps/app_externalivr.c, main/bridging.c, channels/sig_ss7.c, apps/app_milliwatt.c, cdr/cdr_manager.c, apps/app_dial.c, main/pbx.c, funcs/func_timeout.c, apps/app_privacy.c, channels/chan_bridge.c, apps/app_echo.c, apps/app_softhangup.c, apps/app_fax.c, apps/app_dahdiras.c, channels/chan_agent.c, apps/app_disa.c, bridges/bridge_simple.c, include/asterisk/channel.h, apps/app_talkdetect.c, apps/app_transfer.c, main/cel.c, res/res_monitor.c, apps/app_playback.c, apps/app_speech_utils.c, channels/chan_misdn.c, apps/app_sendtext.c, funcs/func_channel.c, funcs/func_cdr.c, channels/sip/dialplan_functions.c, apps/app_macro.c, apps/app_zapateller.c, main/audiohook.c, apps/app_chanspy.c, apps/app_voicemail.c, apps/app_cdr.c, res/res_calendar.c, channels/chan_unistim.c, channels/chan_multicast_rtp.c, channels/chan_vpb.cc, apps/app_meetme.c, main/ccss.c, apps/app_dictate.c, apps/app_authenticate.c, apps/app_readexten.c, channels/chan_gtalk.c, res/res_musiconhold.c, apps/app_followme.c, main/channel.c, main/cdr.c, channels/chan_phone.c, main/dial.c, main/manager.c, apps/app_osplookup.c, bridges/bridge_builtin_features.c, res/res_agi.c, apps/app_minivm.c, main/app.c, apps/app_confbridge.c, main/image.c, apps/app_directory.c, main/message.c, apps/app_ivrdemo.c, addons/chan_mobile.c, apps/app_rpt.c, cdr/cdr_custom.c, apps/app_parkandannounce.c, channels/chan_mgcp.c, apps/app_while.c, res/res_rtp_asterisk.c, apps/app_read.c, channels/chan_sip.c, apps/app_festival.c, res/res_fax.c, cdr/cdr_syslog.c, apps/app_waitforsilence.c, main/channel_internal_api.c, res/res_adsi.c, pbx/pbx_lua.c, funcs/func_jitterbuffer.c, channels/chan_console.c, apps/app_queue.c, channels/sig_pri.c, channels/chan_oss.c, channels/chan_jingle.c, channels/chan_usbradio.c, apps/app_channelredirect.c, apps/app_forkcdr.c, apps/app_flash.c, main/abstract_jb.c, main/file.c, channels/chan_h323.c, include/asterisk/sched.h, res/snmp/agent.c, apps/app_sms.c, channels/chan_nbs.c, funcs/func_callerid.c, apps/app_verbose.c, apps/app_stack.c: ast_channel opaquification of pointers and integral types Review: https://reviewboard.asterisk.org/r/1753/ 2012-02-20 18:40 +0000 [r355903-355999] Sean Bright * /, channels/chan_iax2.c: Remove spurious warning when 'qualifyfreqnotok' is set successfully. (closes issue ASTERISK-17176) Reported by: John Covert Tested by: Sean Bright Patches: chan_iax2.c.qualifyfreqnotok.patch uploaded by John Covert (license 5512) ........ Merged revisions 355997 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355998 from http://svn.asterisk.org/svn/asterisk/branches/10 * channels/chan_dahdi.c, /: This was a LOG_NOTICE, so roll it back. ........ Merged revisions 355952 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355953 from http://svn.asterisk.org/svn/asterisk/branches/10 * channels/chan_dahdi.c, /: Change some debug messages from LOG_DEBUG to ast_debug. ........ Merged revisions 355949 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355950 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, channels/chan_iax2.c: Add some boilerplate documentation for IAXVAR and IAXPEER. ........ Merged revisions 355904 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355905 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, channels/chan_iax2.c: Set the length of the ast_sockaddr, so that we can set it's port later. Without this, the call to ast_sockaddr_set_port a few lines later is a noop. ........ Merged revisions 355901 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355902 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-18 08:02 +0000 [r355852] Alec L Davis * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c, channels/sig_ss7.h, /, channels/sig_analog.h, channels/sig_pri.c, channels/sig_ss7.c: push 'outgoing' flag from sig_XXX up to chan_dahdi 'p->outgoing' in chan_dahdi and sig_analog wern't kept in sync, particulary FXS ast_hangup didn't clear the 'outgoing' flag. sig_pri and sig_ss7 were keeping 'outgoing' flag insync. Now provides a callback for all the low level sig_XXX modules. (issue ASTERISK-19316) alecdavis (license 585) Reported by: Jeremy Pepper Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1747/ ........ Merged revisions 355850 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355851 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-17 22:03 +0000 [r355795] Sean Bright * configs/iax.conf.sample, /, channels/chan_iax2.c: Don't allow trunkfreq to be greater than 1000ms. ........ Merged revisions 355793 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355794 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-17 19:56 +0000 [r355749] Tilghman Lesher * main/asterisk.c: Non-verbose output should always go to the remote console, regardless of the previous level. 2012-02-17 19:35 +0000 [r355748] Sean Bright * /, channels/chan_iax2.c: Pass the correct value to ast_timer_set_rate() for IAX2 trunking. IAX2 uses the trunkfreq variable to determine how often to send trunk packets, but this value is in milliseconds while ast_timer_set_rate() expects the rate argument to be ticks per second. So we divide 1000 by trunkfreq and pass that in instead. With a default of 20ms, this change makes IAX2 send trunk packets every 20ms instead of every 50ms. Tracked down by myself and Bob Wienholt. ........ Merged revisions 355746 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355747 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-17 19:22 +0000 [r355745] Mark Michelson * /, channels/chan_sip.c: Fix regressions with regards to route-set creation on early dialogs. This fixes two main issues: 1. Asterisk would send a CANCEL to the route created by the provisional response instead of using the same destination it did in the initial INVITE. 2. If a new route set arrives in a 200 OK than was in the 1XX response (perfectly possible if our outbound INVITE gets forked), then the route set in the 200 OK needs to overwrite the route set in the 1XX response. (closes issue ASTERISK-19358) Reported by: Karsten Wemheuer Tested by: Karsten Wemheuer patches: ASTERISK-19358.patch uploaded by Mark Michelson (license 5049) ASTERISK-19358.patch uploaded by Stefan Schmidt (license 6034) Review: https://reviewboard.asterisk.org/r/1749 ........ Merged revisions 355732 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355733 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-16 22:00 +0000 [r355667] Paul Belanger * apps/app_rpt.c: Fix channel opaquification for app_rpt 2012-02-16 20:03 +0000 [r355624] Sean Bright * /, main/audiohook.c: Revert a change to audio_audiohook_write_list that had no affect. When I made this change initially, I was under the false impression that the audiohooks structure remained on the channel after all of the hooks had been detached. This is not the case, ast ast_read takes care of removing the audiohooks structure if the lists are empty. ........ Merged revisions 355622 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355623 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-16 19:51 +0000 [r355576-355621] Richard Mudgett * /, configure, include/asterisk/autoconfig.h.in, autoconf/ast_c_declare_check.m4 (added), configure.ac, formats/format_ogg_vorbis.c: Fix compile problem when old version of libvorbisfile v1.1.2 is used. The principle difference between libvorbisfile v1.1.2 and newer (at least v1.2.0) is the addition of the predefined callbacks OV_CALLBACKS_xxx in vorbis/vorbisfile.h used for ov_open_callbacks(). * Updated the configure script to detect if libvorbisfile.h declares OV_CALLBACKS_NOCLOSE. * Copied the declaration of OV_CALLBACKS_NOCLOSE from v1.2.0 to allow v1.1.2 to compile. (closes issue ASTERISK-19370) Reported by: Jonn Taylor ........ Merged revisions 355608 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355620 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, res/res_monitor.c: Fix AMI Monitor action without File header converting channel name into filename. * Fix potential Solaris crash if Monitor application has a urlbase and no fname_base option. ........ Merged revisions 355574 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355575 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-15 19:29 +0000 [r355450-355531] Sean Bright * /, channels/chan_iax2.c: When IAX2 debugging is enabled, make sure to log 'apathetic' messages too. ........ Merged revisions 355529 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355530 from http://svn.asterisk.org/svn/asterisk/branches/10 * build_tools/cflags.xml, channels/chan_iax2.c: Remove IAX_OLD_FIND from chan_iax2. * /, channels/chan_iax2.c: Use TRUNK_CALL_START as originally intended. Back in r646, TRUNK_CALL_START was added and defined as 0x4000. That same value was also hard-coded in one part of the IAX2 code instead of using the #define. TRUNK_CALL_START has changed over the years (for dealing with LOW_MEMORY), but the hard-coded usage was never updated to match. This patch fixes that. ........ Merged revisions 355448 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355449 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-14 20:27 +0000 [r355413] Tilghman Lesher * utils/refcounter.c, main/pbx.c, funcs/func_timeout.c, include/asterisk/autoconfig.h.in, utils/hashtest.c, UPGRADE.txt, CHANGES, main/config.c, configs/logger.conf.sample, main/loader.c, include/asterisk/logger.h, main/manager.c, main/logger.c, utils/ael_main.c, utils/hashtest2.c, codecs/codec_dahdi.c, main/stdtime/localtime.c, main/asterisk.c, addons/res_config_mysql.c: Re-commit the verbose branch. This change permits each verbose destination (consoles, logger) to have its own concept of what the verbosity level is. The big feature here is that the logger will now be able to capture a particular verbosity level without condemning each console to need to suffer that level of verbosity. Additionally, a stray 'core set verbose' will no longer change what will go to the log. Review: https://reviewboard.asterisk.org/r/1599/ 2012-02-14 19:29 +0000 [r355321-355376] Richard Mudgett * /, configure, include/asterisk/autoconfig.h.in, configure.ac, formats/format_ogg_vorbis.c: Fix voicemail problems when using ogg/vorbis. Ogg/vorbis was fairly useless as a voicemail file format because it did not implement the seek and tell format callbacks among other problems. Since we were already using the libvorbis and libvorbisenc libraries we can use libvorbisfile as it is also part of the vorbis library package. * Made use the libvorbisfile to handle the ogg/vorbis file stream. The format_ogg_vorbis.c is now mostly a wrapper around libvorbisfile. (closes issue ASTERISK-16926) Reported by: sque Patches: ogg_vorbis_use_libvorbisfile.patch (license #6108) patch uploaded by sque ........ Merged revisions 355365 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355375 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, cel/cel_sqlite3_custom.c: Fix lock typo that should be unlock in cel_sqlite_custom reload. (closes issue ASTERISK-19356) Reported by: Alex Villacis Lasso Patches: asterisk-1.8.9.2-cel_sqlite3_custom-fix-reload-locking-typo.patch (license #5617) patch uploaded by Alex Villacis Lasso Review: https://reviewboard.asterisk.org/r/1740/ ........ Merged revisions 355319 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355320 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-14 16:28 +0000 [r355274] Mark Michelson * /, channels/chan_sip.c: Properly invert the return of a strncmp call. This was causing identification that should have been made private to be public. (closes issue AST-814) reported by Patrick Anderson Patches: chan_sip.c.diff uploaded by Patrick Anderson (license 5430) ........ Merged revisions 355268 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355271 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-14 15:58 +0000 [r355230] Jason Parker * /, configs/cdr_sqlite3_custom.conf.sample: Don't enable sqlite3 CDRs by default in sample configs. ........ Merged revisions 355228 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355229 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-14 13:35 +0000 [r355184] Sean Bright * /, channels/chan_iax2.c: Clear the high order bit from the destination call number before sending. send_apathetic_reply takes the incoming frame's source call number as the destination call number for the outgoing frame. If the incoming frame was a full frame, then the high order bit of the source call number is set and will be interpreted as a retransmit when sent back out as the destination call number. ........ Merged revisions 355182 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355183 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-14 09:58 +0000 [r355138] Alexandr Anikin * addons/chan_ooh323.c, /: call manager_event only if there is not null channel structure (Closes issue ASTERISK-19298) Reported by: robinfood Patches: issue19298.patch uploaded by may213 (License #5415) ........ Merged revisions 355136 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355137 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-14 00:43 +0000 [r355102] Russell Bryant * res/res_agi.c, CHANGES: res_agi: Add AGIEXITONHANGUP variable. This patch adds a variable AGIEXITONHANGUP for res_agi. If this variable is set to "yes" on a channel, AGI() will exit immediately once a channel hangup has been detected. This was the behavior of AGI() in Asterisk 1.4 and earlier and is still desired by some people. Review: https://reviewboard.asterisk.org/r/1734/ 2012-02-13 22:04 +0000 [r355055-355058] Richard Mudgett * pbx/pbx_spool.c, /: Fix occasional incorrectly delayed call-file execution. Since the dir timestamp is available at one second resolution, we cannot know if it was updated within the same second after we scanned it. Therefore, we will force another scan if the dir was just modified. * Changed to force another scan if the directory was just modified. (closes issue ASTERISK-19081) Reported by: Knut Bakke Review: https://reviewboard.asterisk.org/r/1688/ ........ Merged revisions 355056 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355057 from http://svn.asterisk.org/svn/asterisk/branches/10 * channels/chan_misdn.c: Fix compile error from most recent ast_channel opaquification installment. 2012-02-13 19:56 +0000 [r355011] Joshua Colp * /, pbx/pbx_config.c: Only allow one 'dialplan reload' to execute at a time as otherwise they would share the same common local context list. (closes issue AST-758) ........ Merged revisions 355009 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355010 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-13 17:27 +0000 [r354968] Terry Wilson * channels/chan_local.c, addons/chan_ooh323.c, channels/chan_iax2.c, main/cli.c, channels/chan_dahdi.c, channels/sig_analog.c, channels/chan_skinny.c, main/features.c, apps/app_dumpchan.c, pbx/pbx_realtime.c, channels/chan_alsa.c, apps/app_dial.c, main/pbx.c, apps/app_fax.c, channels/chan_agent.c, include/asterisk/channel.h, apps/app_talkdetect.c, main/cel.c, channels/chan_misdn.c, funcs/func_channel.c, apps/app_macro.c, apps/app_chanspy.c, res/res_calendar.c, apps/app_voicemail.c, channels/chan_unistim.c, tests/test_substitution.c, channels/chan_vpb.cc, apps/app_meetme.c, main/ccss.c, apps/app_readexten.c, channels/chan_gtalk.c, main/cdr.c, main/channel.c, main/dial.c, channels/chan_phone.c, main/manager.c, apps/app_osplookup.c, bridges/bridge_builtin_features.c, res/res_agi.c, apps/app_minivm.c, apps/app_confbridge.c, apps/app_directory.c, addons/chan_mobile.c, apps/app_rpt.c, apps/app_parkandannounce.c, channels/chan_mgcp.c, apps/app_while.c, funcs/func_dialplan.c, channels/chan_sip.c, res/res_fax.c, main/channel_internal_api.c, pbx/pbx_lua.c, channels/sig_pri.c, apps/app_queue.c, channels/chan_oss.c, channels/chan_jingle.c, apps/app_directed_pickup.c, main/file.c, channels/chan_h323.c, res/snmp/agent.c, pbx/pbx_dundi.c, channels/chan_nbs.c, apps/app_stack.c, apps/app_verbose.c: Opaquify char * and char[] in ast_channel Review: https://reviewboard.asterisk.org/r/1733/ 2012-02-13 17:25 +0000 [r354964] Richard Mudgett * res/res_config_pgsql.c, /, configs/extconfig.conf.sample: Fix reconnecting to pgsql database after connection loss. There can only be one database connection in res_config_pgsql just like res_config_sqlite. If the connection is lost, the connection may not get reestablished to the same database if the res_pgsql.conf and extconfig.conf files are inconsistent. * Made only use the configured database from res_pgsql.conf. * Fixed potential buffer overwrite of last[] in config_pgsql(). (closes issue ASTERISK-16982) Reported by: german aracil boned Review: https://reviewboard.asterisk.org/r/1731/ ........ Merged revisions 354953 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 354959 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-13 16:42 +0000 [r354939] Joshua Colp * /, apps/app_confbridge.c: Don't try to play sound files that do not exist. (closes issue ASTERISK-19188) Reported by: slesru ........ Merged revisions 354938 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-10 22:44 +0000 [r354903] Jason Parker * /, apps/app_voicemail.c: Fix a voicemail memory leak with heard/deleted messages. open_mailbox() was changed quite a long time ago to allocate this memory. close_mailbox() should have been changed to be responsible for freeing it. ........ Merged revisions 354889 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 354890 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-10 18:08 +0000 [r354837] Richard Mudgett * main/manager.c, /: Fix AMI Redirect ExtraChannel not redirecting to the same exten and context. The astman_get_header() never returns NULL so the check by the code for NULL would never fail. (closes issue ASTERISK-16974) Reported by: Nuno Borges Patches: 0018325.patch (license #6116) patch uploaded by Nuno Borges (modified) ........ Merged revisions 354835 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 354836 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-10 14:51 +0000 [r354799] Matthew Jordan * apps/app_voicemail.c: Fix IMAP app_voicemail compilation issue introduced in r354429 This simply fixes the compilation issue introduced in r354429 by re-adding the 'quote' variable. (closes issue ASTERISK-19337) Reported by: John Taylor 2012-02-09 22:06 +0000 [r354751] Terry Wilson * /, funcs/func_cdr.c: Note that CDRs are immutable once a bridge is torn down CDRs cannot be modified after a bridge is torn down, (e.g. after Dial() returns) even though the CDR() function may be called. Since modifying the CDR code to change this behavior could very easily break all kinds of things, this patch just documents this limitation. (closes issues ASTERISK-16923) Review: https://reviewboard.asterisk.org/r/1720/ ........ Merged revisions 354749 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 354750 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-09 20:52 +0000 [r354657-354704] Kinsey Moore * /, channels/chan_sip.c: Fix parsing of SIP headers where compact and non-compact headers are mixed Change parsing of SIP headers so that compactness of the header no longer influences which header will be chosen. Previously, a non-compact header would be chosen instead of a preceeding compact-form header. (closes issue ASTERISK-17192) Review: https://reviewboard.asterisk.org/r/1728/ ........ Merged revisions 354702 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 354703 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, main/config.c: Make the config parser remove escaping backslashes The config parser in Asterisk does not currently remove a backslash that is used to escape a semicolon which would otherwise be interpreted as the start of a comment. The change here causes that backslash to be removed, but does not create a real escape system in the config parser. The biggest complication with a real escape system would be breaking existing configs everywhere (parsing \\ as \ and breaking on escaped non-semicolon characters) even though it would be the "right" way to do things. (closes issue ASTERISK-17121) Review: https://reviewboard.asterisk.org/r/1724/ ........ Merged revisions 354655 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 354656 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-09 18:14 +0000 [r354597] Terry Wilson * channels/chan_sip.c, channels/sip/include/config_parser.h, channels/sip/utils.c (added), configs/sip.conf.sample, CHANGES, channels/sip/config_parser.c, channels/sip/include/sip.h, channels/sip/include/sip_utils.h: Add auto_force_rport and auto_comedia NAT options This patch adds the auto_force_rport and auto_comedia NAT options. It also converts the nat= setting to a list of comma-separated combinable options: no, force_rport, comedia, auto_force_rport, and auto_comedia. nat=yes remains as an undocumented option equal to "force_rport,comedia". The first instance of 'yes' or 'no' in the list stops parsing and overrides any previously set options. If an auto_* option is specified with its non-auto_ counterpart, the auto setting takes precedence. This patch builds upon the patch posted to ASTERISK-17860 by JIRA user pedro-garcia. (closes issue ASTERISK-17860) Review: https://reviewboard.asterisk.org/r/1698/ 2012-02-09 17:17 +0000 [r354552] Mark Michelson * /, res/res_fax.c: Adding reload support to res_fax.so (closes issue ASTERISK-16712) reported by Frank DiGennaro Review: https://reviewboard.asterisk.org/r/1713 ........ Merged revisions 354545 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 354546 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-09 17:09 +0000 [r354544-354549] Matthew Jordan * /, channels/chan_sip.c: Clean-up of minor formatting issues in r354542/3/4 rmudgett pointed out some formatting issues in the check-in for ASTERISK-19290. This cleans those up. Review: https://reviewboards.asterisk.org/r/1722/ ........ Merged revisions 354547 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 354548 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, channels/chan_sip.c: Fix SIP INFO DTMF handling for non-numeric codes In ASTERISK-18924, SIP INFO DTMF handlingw as changed to account for both lowercase alphatbetic DTMF events, as well as uppercase alphabetic DTMF events. When this occurred, the comparison of the character buffer containing the event code was changed such that the buffer was first compared again '0' and '9' to determine if it was numeric. Unfortunately, since the first character in the buffer will typically be '1' in the case of non-numeric event codes (10-16), this caused those codes to be converted to a DTMF event of '1'. This patch fixes that, and cleans up handling of both application/dtmf-relay and application/dtmf content types. Review: https://reviewboard.asterisk.org/r/1722/ (closes issue ASTERISK-19290) Reported by: Ira Emus Tested by: mjordan ........ Merged revisions 354542 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 354543 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-09 03:09 +0000 [r354497-354498] Richard Mudgett * channels/chan_dahdi.c, channels/chan_misdn.c: Fix some compile problems from the 'cppcheck' patch. * /, apps/app_parkandannounce.c: Fix crash in ParkAndAnnounce. Well, thats embarrasing. I forgot to initialize the caller_id storage. (closes issue ASTERISK-19311) Reported by: tootai Tested by: rmudgett ........ Merged revisions 354495 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 354496 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-09 02:28 +0000 [r354494] Russell Bryant * main/channel.c, /: Remove some unnecessary locking from ast_hangup(). This patch removes some unnecessary locking of the channels container in ast_hangup(). The reason this came up is that this lock can very quickly block the entire system. If any of the channel cleanup code decides to block, it causes a problem for the whole system. For example, when audiohooks get destroyed, if that blocks for a while waiting on the mixmonitor thread to exit because it's busy blocking on some I/O, it causes a problem for many other threads in the meantime. Review: https://reviewboard.asterisk.org/r/1712/ ........ Merged revisions 354492 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 354493 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-08 21:29 +0000 [r354459] Kevin P. Fleming * res/res_ais.c (removed), contrib/scripts/install_prereq: Revision 354046 added res_corosync as a replacement for res_ais, but didn't actually remove res_ais. This commit removes it. In addition, the 'install_prereq' script has been updated to no longer install AIS dependency packages, and instead install Corosync packages instead. 2012-02-08 21:28 +0000 [r354458] Terry Wilson * channels/chan_sip.c, contrib/realtime/postgresql/realtime.sql, CHANGES, channels/sip/include/sip.h: Add callbackextension matching & realtime callbackextensions This patch is based on the one by David Vossel, developer extrodinaire, at https://reviewboard.asterisk.org/r/344/. If multiple peers are defined with the same host/port, but differing callbackextensions, it chooses the peer with the matching callbackextension. Since callbackextension creates an outbound registration with the callbackextension as the Contact address, matching an incoming request by that (in addition to the host/port) makes a lot of sense. This patch also adds support for callbackextension to realtime by querying all peers with callbackextensions on reload and adding registrations for them. (closes issue ASTERISK-13456) Review: https://reviewboard.asterisk.org/r/344/ Review: https://reviewboard.asterisk.org/r/1717/ 2012-02-08 21:25 +0000 [r354450] Kevin P. Fleming * channels/chan_dahdi.c: Restore some variables removed by the 'cppcheck' patch that were actually needed. 2012-02-08 20:49 +0000 [r354429] Walter Doekes * apps/app_dial.c, main/udptl.c, main/pbx.c, addons/chan_ooh323.c, funcs/func_env.c, funcs/func_strings.c, utils/astman.c, main/acl.c, apps/app_disa.c, apps/app_alarmreceiver.c, apps/app_queue.c, channels/chan_iax2.c, addons/ooh323c/src/memheap.c, channels/chan_usbradio.c, channels/chan_dahdi.c, apps/app_osplookup.c, channels/chan_misdn.c, channels/chan_skinny.c, funcs/func_odbc.c, main/ast_expr2f.c, apps/app_minivm.c, formats/format_h263.c, addons/chan_mobile.c, apps/app_chanspy.c, main/ast_expr2.fl, apps/app_voicemail.c: Avoid cppcheck warnings; removing unused vars and a bit of cleanup. Patch by: Clod Patry Review: https://reviewboard.asterisk.org/r/1651 2012-02-08 15:28 +0000 [r354395] Kinsey Moore * CHANGES: Add CHANGES documentation for the "pri set debug" bitmask change (related to ASTERISK-17159) 2012-02-07 21:33 +0000 [r354360] Terry Wilson * /, channels/chan_sip.c, contrib/realtime/postgresql/realtime.sql: Fix multiple SIP realtime issues 1. Set lastms to 0 when clearing instead of "" 2. Don't set ipaddr or port to the string "(null)" when they are empty 3. Add missing required fields, set default for lastms to 0, and modify the length of the ipaddr field to 45 in the Postgresql realtime.sql file. (closes issue ASTERISK-19172) Review: https://reviewboard.asterisk.org/r/1703/ ........ Merged revisions 354348 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 354349 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-07 18:07 +0000 [r354312-354314] Sean Bright * contrib/scripts/live_ast: Continuation of last patch - since LIVE_AST_LD_PATH_EXTRA will now never be empty, don't check for it, instead of check if LD_LIBRARY_PATH is already set and if so, append LIVE_AST_LD_PATH_EXTRA properly. * contrib/scripts/live_ast: Include live/usr/lib in the shared library search path to that we pick up libasteriskssl.so at run time when using live_ast. * contrib/scripts/live_ast: Whitespace only (remove trailing spaces) 2012-02-07 15:29 +0000 [r354275] Jonathan Rose * /, cdr/cdr_pgsql.c: Fix column duplication bug in module reload for cdr_pgsql. Prior to this patch, attempts to reload cdr_pgsql.so would cause the column list to keep its current data and then add a second copy during the reload. This would cause attempts to log the CDR to the database to fail. This patch also cleans up some unnecessary null checks for ast_free and deals with a few potential locking problems. (closes issue ASTERISK-19216) Reported by: Jacek Konieczny Review: https://reviewboard.asterisk.org/r/1711/ ........ Merged revisions 354263 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 354270 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-06 23:15 +0000 [r354174-354218] Richard Mudgett * /, pbx/pbx_config.c: Improved documentation of CLI "dialplan add extension" command. * Documented dialplan add extension ,,)> format. * Allow acceptance of command without the app-data value. There are many applications that do no need any parameters so it is silly to require that field for all commands. * Fixed a couple ast_malloc/ast_free mismatches with ast_add_extension2() calls. (closes issue ASTERISK-19222) Reported by: Andrey Solovyev Tested by: rmudgett ........ Merged revisions 354216 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 354217 from http://svn.asterisk.org/svn/asterisk/branches/10 * channels/sig_pri.h: Restore alternate SIG_PRI_DEBUG_DEFAULT meaning. 2012-02-06 20:18 +0000 [r354165] Kinsey Moore * channels/sig_pri.h, channels/chan_dahdi.c: Allow more control over the output of pri debug This changes the debuglevel of 'pri set debug' to a bit mask allowing the user to independently select bits of output: 1 libpri internals including state machine 2 Decoded Q.931 messages 4 Decoded Q.921 headers 8 raw hex dump of the full frames Additionally, this ensures that the meaning of "on" does not change and intrudces intense and hex to simplify usage. (closes issue ASTERISK-17159) Original-patch-by: wimpy 2012-02-06 17:33 +0000 [r354120] Richard Mudgett * /, main/features.c: Add missing headers to AMI UnParkedCall event to uniquely identify the call. The AMI UnParkedCall event was missing the Parkinglot and Uniqueid headers that the AMI ParkedCall event contains. (closes issue ASTERISK-19240) Reported by: Michael Yara ........ Merged revisions 354116 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 354119 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-06 16:38 +0000 [r354084] Joshua Colp * apps/app_meetme.c, UPGRADE.txt: Make the 'c' option to MeetMe work even if the 'q' option is used. (closes issue ASTERISK-17053) Reported by: justdave 2012-02-05 10:58 +0000 [r354046] Russell Bryant * build_tools/menuselect-deps.in, configure, include/asterisk/autoconfig.h.in, res/res_corosync.c (added), configure.ac, configs/res_corosync.conf.sample (added), res/ais (removed), UPGRADE.txt, configs/ais.conf.sample (removed), CHANGES, makeopts.in: Replace res_ais with a new module, res_corosync. This patch removes res_ais and introduces a new module, res_corosync. The OpenAIS project is deprecated and is now just a wrapper around Corosync. This module provides the same functionality using the same core infrastructure, but without the use of the deprecated components. Technically res_ais could have been used with an AIS implementation other than OpenAIS, but that is the only one I know of that was ever used. Review: https://reviewboard.asterisk.org/r/1700/ 2012-02-03 21:33 +0000 [r354001] Jonathan Rose * /, channels/chan_agent.c: Fixes deadlocks occuring in chan_agent due to r335976 Bad locking order was added to chan_agent to prevent segfaults from having no locking in a patch by irroot. This patch addresses the bad locking order by releasing locks before getting the right locking order to stop deadlocks from occuring when doing multiple interactions with agents. (closes issue ASTERISK-19285) Reported by: Alex Villacis Lasso Review: https://reviewboard.asterisk.org/r/1708/ ........ Merged revisions 353999 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 354000 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-03 16:50 +0000 [r353964] Kinsey Moore * UPGRADE.txt, cdr/cdr_adaptive_odbc.c, configs/cdr_adaptive_odbc.conf.sample: Support schema selection in cdr_adaptive_odbc Asterisk now supports using ODBC with databases where a single schema must be selected. Previously, INSERTs would fail because they did not take into account extra fields cause by having multiple schemas. This also corrects some SQL resource leaks. (closes issue ASTERISK-17106) Patch-by: Alexander Frolkin Patch-by: Tilgnman Lesher 2012-02-03 16:23 +0000 [r353963] Jonathan Rose * /, res/res_fax.c: Fixes a segfault occuring when performing attended transfer with FAXOPT(gateway)=yes (closes issue ASTERISK-19184) Reported by: Alexandr ........ Merged revisions 353962 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-02 22:28 +0000 [r353917] Kinsey Moore * /, channels/chan_sip.c: Ensure entering T.38 passthrough does not cause an infinite loop After R340970 Asterisk was still polling the RTCP file descriptor after RTCP is shut down and removed. If the descriptor happened to have data ready when the removal occured then Asterisk would go into an infinite loop trying to read data that it can never actually access. This change disables the audio RTCP file descriptor for the duration of the T.38 transaction. (closes issue ASTERISK-18951) Reported-by: Kristijan Vrban ........ Merged revisions 353915 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 353916 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-02 20:18 +0000 [r353872] Richard Mudgett * channels/sig_pri.h, channels/chan_dahdi.c, /, channels/sig_pri.c: Restore the 'w' modifier support for ISDN spans. Dial(DAHDI/g0/1234w888) This feature also causes the sending complete ie to be sent for switch types that do not automatically send the ie. (EuroISDN/ETSI) The main difference between dialing Dial(DAHDI/g0/1234w888) and Dial(DAHDI/g0/1234,,D(888)) is the sending of the sending complete ie. (closes issue ASTERISK-19176) Reported by: rmudgett Tested by: rmudgett ........ Merged revisions 353867 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 353868 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-02 18:55 +0000 [r353821] Mark Michelson * main/manager.c, /, main/http.c, configs/manager.conf.sample, include/asterisk/manager.h, configs/http.conf.sample: Fix TLS port binding behavior as well as reload behavior: * Removes references to tlsbindport from http.conf.sample and manager.conf.sample * Properly bind to port specified in tlsbindaddr, using the default port if specified. * On a reload, properly close socket if the service has been disabled. A note has been added to UPGRADE.txt to indicate how ports must be set for TLS. (closes issue ASTERISK-16959) reported by Olaf Holthausen (closes issue ASTERISK-19201) reported by Chris Mylonas (closes issue ASTERISK-19204) reported by Chris Mylonas Review: https://reviewboard.asterisk.org/r/1709 ........ Merged revisions 353770 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 353820 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-02 17:07 +0000 [r353725-353772] Jonathan Rose * /, channels/chan_sip.c: Fix sip show peers port output, align columns, and fix ami port output. A previous patch I committed from ASTERISK-16930 unexpectedly changed some output for the AMI action "sippeers" which this patch changes back. Also, this aligns the output for the cli command "sip show peers" and fixes another issue that patch introduced by using ast_sockaddr_stringify calls multiple times without immediately using the pointer. I also went ahead and did a little janitorial work to clean up whitespace in _sip_show_peers. (issue ASTERISK-16930) (closes issue ASTERISK-19281) Reported by: Patrick El Youssef Patches: ASTERISK-19281.diff uploaded by Walter Doekes (license 5674) ........ Merged revisions 353769 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 353771 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, channels/chan_sip.c: Use ast_sockaddr_stringify_fmt wrappers for various functions in chan_sip There are a number of cleaner looking wrappers for ast_sockaddr_stringify_fmt available which are slightly more readable than using a direct call to ast_sockaddr_stringify_fmt. This patch switches a number of those calls in chan_sip to use those wrappers and is generally harmless. (Closes issue ASTERISK-16930) Reported by: Michael L. Young Patches: chan_sip-broken-registration-1.8.diff uploaded by Michael L. Young (license 5026) ........ Merged revisions 353720 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 353721 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-01 19:53 +0000 [r353647-353685] Richard Mudgett * channels/chan_unistim.c, channels/chan_multicast_rtp.c, channels/chan_local.c, addons/chan_ooh323.c, channels/chan_vpb.cc, channels/chan_gtalk.c, channels/chan_iax2.c, main/channel.c, channels/chan_phone.c, channels/chan_dahdi.c, channels/sig_analog.c, main/manager.c, pbx/pbx_spool.c, channels/chan_skinny.c, main/features.c, channels/sig_analog.h, channels/chan_alsa.c, apps/app_confbridge.c, addons/chan_mobile.c, channels/sig_ss7.c, channels/chan_mgcp.c, main/pbx.c, channels/sig_ss7.h, channels/chan_sip.c, channels/chan_bridge.c, channels/chan_agent.c, include/asterisk/channel.h, channels/chan_console.c, channels/sig_pri.c, channels/chan_oss.c, channels/chan_usbradio.c, channels/chan_jingle.c, channels/sig_pri.h, channels/chan_misdn.c, channels/chan_h323.c, channels/chan_nbs.c, include/asterisk/pbx.h: Constify some more channel driver technology callback parameters. Review: https://reviewboard.asterisk.org/r/1707/ * cel/cel_pgsql.c, configs/cel_sqlite3_custom.conf.sample, cel/cel_odbc.c, configs/cel.conf.sample, cel/cel_manager.c, cel/cel_tds.c, configs/cel_pgsql.conf.sample, configs/cel_odbc.conf.sample, main/cel.c, configs/cel_custom.conf.sample: Remove inconsistency in CEL eventtype for user defined events. The CEL eventtype field for ODBC and PGSQL backends should be USER_DEFINED instead of the user defined event name supplied by the CELGenUserEvent application. If the field is output as a number, the user defined name does not have a value and is always output as 21 for USER_DEFINED and the userdeftype field would be required to supply the user defined name. The following CEL backends (cel_odbc, cel_pgsql, cel_custom, cel_manager, and cel_sqlite3_custom) can be independently configured to remove this inconsistency. * Allows cel_manager, cel_custom, and cel_sqlite3_custom to behave the same way. (closes issue ASTERISK-17189) Reported by: Bryant Zimmerman Review: https://reviewboard.asterisk.org/r/1669/ * main/channel.c, include/asterisk/channel.h: Fix ExtenSpy and simplify the channel search functions. When ast_channel name was opaquified, the channel search functions did not get converted correctly. As a result ExtenSpy which uses a channel iterator search by exten@context could never find anything. * Updated the doxygen documentation for the search functions in channel.h. Review: https://reviewboard.asterisk.org/r/1702/ 2012-02-01 15:59 +0000 [r353600] Sean Bright * /, include/asterisk/audiohook.h: Resolve an overlap in the ast_audiohook_flags values. AST_AUDIOHOOK_TRIGGER_WRITE and AST_AUDIOHOOK_WANTS_DTMF were overlapping which may have caused unintended side effects. This patch moves AST_AUDIOHOOK_TRIGGER_WRITE, and updates AST_AUDIOHOOK_TRIGGER_MODE to reflect the original intention. This will affect existing modules that use these flags, so be sure to recompile as necessary. (closes issue ASTERISK-19246) Reported by: feyfre ........ Merged revisions 353598 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 353599 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-01 15:07 +0000 [r353552] Matthew Jordan * /, contrib/init.d/etc_default_asterisk: Added clarification for the VERBOSITY setting to etc_default_asterisk Clarified that using the VERBOSITY setting in etc_default_asterisk is the same as using the -v command line switch, which causes Asterisk to launch in console mode. (closes issue ASTERISK-17030) Reported by: Jonas ........ Merged revisions 353550 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 353551 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-02-01 00:08 +0000 [r353504] Terry Wilson * /, res/res_calendar.c: Allow res_calendar to be unloaded The calendaring tech modules depend on res_calendar and initially res_calendar just bumped the use count so that it couldn't be unloaded. res_calendar can potentially create many threads and I've seen issues where the Asterisk shutdown has failed where it looked like these threads could be the culprit. This patch adds unload support for res_calendar. Unloading res_calendar will also unload the dependant tech modules as well. (closes issue ASTERISK-16744) Review: https://reviewboard.asterisk.org/r/1657/ ........ Merged revisions 353502 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 353503 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-31 17:26 +0000 [r353466] Richard Mudgett * main/manager.c, /, include/asterisk/channel.h: Fix memory leak in error paths for action_originate(). * Fix memory leak of vars in error paths for action_originate(). * Moved struct fast_originate_helper tech and data members to stringfields. * Simplified ActionID header handling for fast_originate(). * Added doxygen note to ast_request() and ast_call() and the associated channel callbacks that the data/addr parameters should be treated as const char *. Review: https://reviewboard.asterisk.org/r/1690/ ........ Merged revisions 353454 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 353463 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-30 23:58 +0000 [r353418] Terry Wilson * main/dnsmgr.c, /, channels/chan_sip.c, include/asterisk/dnsmgr.h: Re-link peers by IP when dnsmgr changes the IP Asterisk's dnsmgr currently takes a pointer to an ast_sockaddr and updates it anytime an address resolves to something different. There are a couple of issues with this. First, the ast_sockaddr is usually the address of an ast_sockaddr inside a refcounted struct and we never bump the refcount of those structs when using dnsmgr. This makes it possible that a refresh could happen after the destructor for that object is called (despite ast_dnsmgr_release being called in that destructor). Second, the module using dnsmgr cannot be aware of an address changing without polling for it in the code. If an action needs to be taken on address update (like re-linking a SIP peer in the peers_by_ip table), then polling for this change negates many of the benefits of having dnsmgr in the first place. This patch adds a function to the dnsmgr API that calls an update callback instead of blindly updating the address itself. It also moves calls to ast_dnsmgr_release outside of the destructor functions and into cleanup functions that are called when we no longer need the objects and increments the refcount of the objects using dnsmgr since those objects are stored on the ast_dnsmgr_entry struct. A helper function for returning the proper default SIP port (non-tls vs tls) is also added and used. This patch also incorporates changes from a patch posted by Timo Teräs to ASTERISK-19106 for related dnsmgr issues. (closes issue ASTERISK-19106) Review: https://reviewboard.asterisk.org/r/1691/ ........ Merged revisions 353371 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 353397 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-30 22:44 +0000 [r353347-353370] Alec L Davis * /, channels/chan_sip.c: Merged revisions 353369 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r353369 | alecdavis | 2012-01-31 11:42:28 +1300 (Tue, 31 Jan 2012) | 9 lines Merged revisions 353368 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r353368 | alecdavis | 2012-01-31 11:40:40 +1300 (Tue, 31 Jan 2012) | 2 lines prevent debug messsges displaying -ve Cseq numbers. Missed in R353320 ........ ................ * channels/sip/include/dialog.h, /, channels/chan_sip.c, channels/sip/include/sip.h: Merged revisions 353321 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r353321 | alecdavis | 2012-01-31 11:16:22 +1300 (Tue, 31 Jan 2012) | 25 lines Merged revisions 353320 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r353320 | alecdavis | 2012-01-31 10:57:49 +1300 (Tue, 31 Jan 2012) | 18 lines RFC3261 Section 8.1.1.5. The sequence number value MUST be expressible as a 32-bit unsigned integer * fix: use %u instead of %d when dealing with CSeq numbers - to remove possibility of -ve numbers. * fix: change all uses of seqno and friends (ocseq icseq) from 'int' or 'unsigned int' to uint32_t. Summary of CSeq numbers. An initial CSeq number must be less than 2^31 A CSeq number can increase in value up to 2^32-1 An incrementing CSeq number must not wrap around to 0. Tested with Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1699/ ........ ................ 2012-01-30 21:34 +0000 [r353262-353319] Kevin P. Fleming * Makefile: Correct serious flaw in the top-level Makefile. * include/asterisk.h, /, main/Makefile, main/libasteriskssl.c (added), configure.ac, Makefile.moddir_rules, main/ssl.c (removed), addons, CHANGES, include/asterisk/optional_api.h, Makefile, build_tools/mkpkgconfig, configure, main, makeopts.in, build_tools/make_defaults_h, main/libasteriskssl.exports.in (added): Address OpenSSL initialization issues when using third-party libraries. When Asterisk is used with various third-party libraries (CURL, PostgresSQL, many others) that have the ability themselves to use OpenSSL, it is possible for conflicts to arise in how the OpenSSL libraries are initialized and shutdown. This patch addresses these conflicts by 'wrapping' the important functions from the OpenSSL libraries in a new shared library that is part of Asterisk itself, and is loaded in such a way as to ensure that *all* calls to these functions will be dispatched through the Asterisk wrapper functions, not the native functions. This new library is optional, but enabled by default. See the CHANGES file for documentation on how to disable it. Along the way, this patch also makes a few other minor changes: * Changes MODULES_DIR to ASTMODDIR throughout the build system, in order to more closely match what is used during run-time configuration. * Corrects some errors in the configure script where AC_CHECK_TOOLS was used instead of AC_PATH_PROG. * Adds a new variable for linker flags in the build system (DYLINK), used for producing true shared libraries (as opposed to the dynamically loadable modules that the build system produces for 'regular' Asterisk modules). * Moves the Makefile bits that handle installation and uninstallation of the main Asterisk binary into main/Makefile from the top-level Makefile. * Moves a couple of useful preprocessor macros from optional_api.h to asterisk.h. Review: https://reviewboard.asterisk.org/r/1006/ * /, channels/chan_sip.c: Clarify log WARNING message when port-zero SDP 'm' lines received. Previously, if an m-line in an SDP offer or answer had a port number of zero, that line was skipped, and resulted in an 'Unsupported SDP media type...' warning message. This was misleading, as the media type was not unsupported, but was ignored because the m-line indicated that the media stream had been rejected (in an answer) or was not going to be used (in an offer). ........ Merged revisions 353260 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 353261 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-29 22:33 +0000 [r353224] Damien Wedhorn * channels/chan_skinny.c: Allow softkey reject while device onhook. Fixes up softkey endcall. Previous code was a copy of onhook, now allows for endcall softkey to be used while device is still onhook. 2012-01-29 02:45 +0000 [r353177] Russell Bryant * /, main/netsock.c: Find even more network interfaces. The previous change made the code look for emN and pciN in addition to what it did originally, which was search for ethN. However, it needed to be looking for pciN#N, so that's what it does now. This also moves the memset() to be before every ioctl(). ........ Merged revisions 353175 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 353176 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-28 14:52 +0000 [r353128] Kevin P. Fleming * main/rtp_engine.c, /: Add 'L16-256' MIME subtype alias for slin16. Asterisk has supported the 'L16' MIME subtype for 16kHz signed linear (PCM) audio for quite some time, but some endpoints refer to it as 'L16-256'. This commit adds this as an alias for the existing format. ........ Merged revisions 353126 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 353127 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-28 04:31 +0000 [r353079] Russell Bryant * /, main/netsock.c: Update ast_set_default_eid() to find more network interfaces. As of Fedora 15, ethN is not the name of ethernet interfaces. The names are emN or pciN. Update some code that searched for interfaces named ethN to look for the new names, as well. For more information about why this change was made, see this page: http://domsch.com/blog/?p=455 ........ Merged revisions 353077 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 353078 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-27 21:38 +0000 [r352996-353040] Richard Mudgett * /, apps/app_queue.c: Audit of ao2_iterator_init() usage for v10. Missed one. ........ Merged revisions 353039 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, tests/test_format_api.c: Audit of ao2_iterator_init() usage for v10. Fix double format_cap iterator cleanup. ........ Merged revisions 352992 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-27 19:26 +0000 [r352981] Jonathan Rose * /, res/res_monitor.c: Make failed PauseMonitor and UnpauseMonitor with no valid channel not close AMI session. I also went ahead and took a little time to make sure that the manager value AMI_SUCCESS was used instead of just return 0 being thrown around everywhere since that's how we handle this stuff these days. (closes issue ASTERISK-19249) Reporter: Jamuel Starkey Patches: res_monitor.c-ASTERISK-19249.diff uploaded by Jamuel Starkey (license 5766) ........ Merged revisions 352959 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352965 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-27 18:47 +0000 [r352957] Richard Mudgett * main/pbx.c, /, channels/chan_sip.c, include/asterisk/indications.h, res/snmp/agent.c, main/taskprocessor.c, apps/app_queue.c, channels/chan_iax2.c, apps/app_chanspy.c, main/indications.c, res/res_odbc.c, res/res_srtp.c: Audit of ao2_iterator_init() usage for v1.8. Fixes numerous reference leaks and missing ao2_iterator_destroy() calls as a result. Review: https://reviewboard.asterisk.org/r/1697/ ........ Merged revisions 352955 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352956 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-27 15:57 +0000 [r352916] Terry Wilson * res/res_calendar_exchange.c, res/res_calendar_caldav.c, res/res_calendar.c: Add aresult variable for CALENDAR_WRITE This patch adds a CALENDAR_SUCCESS=1/0 variable that is set to show whether or not CALENDAR_WRITE has passed. This patch also adds some debugging for caldav PUT responses and no longer treats responses with no body as an error (as a PUT gets a 201 Created with no body). (closes issue ASTERISK-16903) Reported by: Clod Patry Tested by: Terry Wilson Patches: calendarstatus.diff uploaded by Clod Patry (License #5138), slightly modified by Terry Wilson Review: https://reviewboard.asterisk.org/r/1692/ - This line, and those below, will be ignored-- M res/res_calendar.c M res/res_calendar_exchange.c M res/res_calendar_caldav.c 2012-01-27 00:11 +0000 [r352864] Alec L Davis * /, channels/chan_sip.c, channels/sip/include/sip.h: Merged revisions 352863 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r352863 | alecdavis | 2012-01-27 13:08:03 +1300 (Fri, 27 Jan 2012) | 19 lines Merged revisions 352862 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r352862 | alecdavis | 2012-01-27 13:05:30 +1300 (Fri, 27 Jan 2012) | 12 lines rfc4235 - Section 4.1: Versions MUST be representable using a non-negative 32 bit integer. If a BLF subscription exists for long enough, using %d may print negative version numbers. Unlikely, as 2^32 at 1 update per second is ~137 years, or half that before the versions number started going negative. Tested with Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1694/ ........ ................ 2012-01-26 20:44 +0000 [r352821] Alexandr Anikin * addons/chan_ooh323.c, /: Fix outbound DTMF for inband mode (tell asterisk core to generate DTMF sounds). (Closes issue ASTERISK-19233) Reported by: Matt Behrens Patches: chan_ooh323.c.patch uploaded by Matt Behrens (License #6346) ........ Merged revisions 352807 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352817 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-26 19:09 +0000 [r352757] Jonathan Rose * /, channels/chan_sip.c: Copy amaflags to sip_pvt from peer during create_addr_from_peer For whatever reason, we don't have a single function for copying data like this from SIP peers to the SIP pvt. This patch adds the copying of amaflags to the sip_pvt, but it would probably be worth discussing this function along with the others that essentially just copy some amount of data from a peer to a private. (Closes issue ASTERISK-19029) Reported by: Matt Lehner ........ Merged revisions 352755 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352756 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-26 06:36 +0000 [r352706] Alec L Davis * /, channels/chan_sip.c: Merged revisions 352705 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r352705 | alecdavis | 2012-01-26 19:33:11 +1300 (Thu, 26 Jan 2012) | 27 lines Merged revisions 352704 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r352704 | alecdavis | 2012-01-26 19:27:07 +1300 (Thu, 26 Jan 2012) | 20 lines Cleanup dialog-info+xml Notify dialog Make similar to other Notify messages. sample output: terminated Tested with Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1693/ ........ ................ 2012-01-25 22:25 +0000 [r352659] Paul Belanger * /, apps/app_voicemail.c: Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2) ........ Merged revisions 352643 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352651 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-25 21:31 +0000 [r352626] Kevin P. Fleming * Makefile, include/asterisk/version.h (added), main/test.c, build_tools/make_version_h (removed), include/asterisk: Remove "asterisk/version.h" in favor of "asterisk/ast_version.h". A long time ago, in a land far far away, we added "asterisk/ast_version.h", which provides the ast_get_version() and ast_get_version_num() functions. These were added so that modules that needed the version information for the Asterisk instance they were loaded in could actually get it (as opposed the version that they were compiled against). We changed everything in the tree to use the new mechanism (although later main/test.c was added using the old method). However, the old mechanism was never removed, and as a result, new code is still trying to use it. This commit removes asterisk/version.h and replaces it with a header that will generate a compile-time error if you try to use it (the error message tells you which header you should use instead). It also removes the Makefile and build_tools bits that generated the file, and it updates main/test.c to use the 'proper' method of getting the Asterisk version information. This is an API change and thus is being committed for trunk only, but it's a fairly minor one and definitely improves the situation for out-of-tree modules. 2012-01-25 17:33 +0000 [r352565] Terry Wilson * /, channels/chan_sip.c: Remove some extraneous debugging from registry memleak fix ........ Merged revisions 352551 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352556 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-25 17:23 +0000 [r352538] Richard Mudgett * /, channels/chan_sip.c, CHANGES, main/message.c, channels/sip/include/sip.h: Fixes for sending SIP MESSAGE outside of calls. * Fix authenticate MESSAGE losing custom headers added by the MESSAGE_DATA function in the authorization attempt. * Pass up better From header contents for SIP to use. Now is in the "display-name" format expected by MessageSend. (Note that this is a behavior change that could concievably affect some people.) * Block user from adding standard headers that are added automatically. (To, From,...) * Allow the user to override the Content-Type header contents sent by MessageSend. * Decrement Max-Forwards header if the user transferred it from an incoming message. * Expand SIP short header names so the dialplan and other code only has to deal with the full names. * Documents what SIP expects in the MessageSend(from) parameter. (closes issue ASTERISK-18992) Reported by: Yuri (closes issue ASTERISK-18917) Reported by: Shaun Clark Review: https://reviewboard.asterisk.org/r/1683/ ........ Merged revisions 352520 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-25 17:02 +0000 [r352519] Terry Wilson * /, channels/chan_sip.c: Clean up some SIP registry-related memory leaks 1) Be sure and free at unload the epa_backend we allocate at startup 2) Do the same sip_registry cleanup at unload we do at reload Review: https://reviewboard.asterisk.org/r/1689/ ........ Merged revisions 352514 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352515 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-25 16:54 +0000 [r352517] Kevin P. Fleming * main/format.c, /, main/format_cap.c, main/format_pref.c: Eliminate unnecessary rebuilds of main/format*.c. These files have no need to include "asterisk/version.h", and doing so forces them to be rebuilt each time a Subversion checkout moves between 'modified' and 'unmodified' states. ........ Merged revisions 352516 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-25 16:42 +0000 [r352513] Jonathan Rose * /, configs/sip.conf.sample: Redocuments sip types peer, user, friend in sip.conf.sample There was faulty information in the sample config describing user as a synonym for friend so it has been changed to better elaborate on the differences between the three entity types. (closes issue ASTERISK-15537) Reported by: yarique ........ Merged revisions 352511 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352512 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-25 01:21 +0000 [r352475] Terry Wilson * channels/chan_vpb.cc: Fix channel opaquification of stringfields for chan_vpb 2012-01-24 22:28 +0000 [r352431] Mark Michelson * /, channels/chan_sip.c: Don't do a DNS lookup on an outbound REGISTER host if there is an outbound proxy configured. (closes issue ASTERISK-16550) reported by: Olle Johansson ........ Merged revisions 352424 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352430 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-24 20:37 +0000 [r352377] Jonathan Rose * /, sounds/Makefile: Set core sounds version to 1.4.22. Now that we have the right license for the Russian 1.4.22 sounds as well as the sounds for the Australian English 1.4.22 sounds, we can finally set the sounds to use 1.4.22! (closes issue ASTERISK-18978) Reported by: Cameron Twomey Patches: confbridge.tar.001 uploaded by Cameron Twomey confbridge.tar.002 uploaded by Cameron Twomey ........ Merged revisions 352367 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352373 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-24 20:12 +0000 [r352348] Terry Wilson * channels/chan_local.c, addons/chan_ooh323.c, main/say.c, apps/app_record.c, apps/app_sayunixtime.c, channels/chan_iax2.c, main/cli.c, channels/chan_dahdi.c, channels/sig_analog.c, channels/chan_skinny.c, main/features.c, apps/app_dumpchan.c, channels/chan_alsa.c, pbx/pbx_realtime.c, apps/app_externalivr.c, apps/app_dial.c, main/pbx.c, apps/app_page.c, channels/chan_bridge.c, apps/app_privacy.c, channels/chan_agent.c, apps/app_disa.c, include/asterisk/channel.h, main/aoc.c, apps/app_talkdetect.c, main/cel.c, res/res_monitor.c, apps/app_playback.c, apps/app_speech_utils.c, channels/chan_misdn.c, funcs/func_channel.c, apps/app_chanspy.c, apps/app_voicemail.c, channels/chan_unistim.c, channels/chan_multicast_rtp.c, apps/app_meetme.c, apps/app_dictate.c, apps/app_authenticate.c, apps/app_readexten.c, apps/app_userevent.c, res/res_musiconhold.c, channels/chan_gtalk.c, apps/app_followme.c, main/cdr.c, main/channel.c, channels/chan_phone.c, main/dial.c, main/manager.c, apps/app_minivm.c, res/res_agi.c, main/app.c, apps/app_confbridge.c, main/image.c, apps/app_directory.c, addons/chan_mobile.c, apps/app_rpt.c, channels/chan_mgcp.c, apps/app_parkandannounce.c, channels/chan_sip.c, res/res_fax.c, main/channel_internal_api.c, channels/chan_console.c, channels/sig_pri.c, apps/app_queue.c, channels/chan_oss.c, funcs/func_global.c, channels/chan_jingle.c, channels/chan_usbradio.c, channels/chan_h323.c, main/file.c, res/snmp/agent.c, channels/chan_nbs.c, apps/app_stack.c, addons/app_saycountpl.c: Opaquify channel stringfields Continue channel opaque-ification by wrapping all of the stringfields. Eventually, we will restrict what can actually set these variables, but the purpose for now is to hide the implementation and keep people from adding code that directly accesses the channel structure. Semantic changes will follow afterward. Review: https://reviewboard.asterisk.org/r/1661/ 2012-01-24 17:04 +0000 [r352293] Richard Mudgett * /, funcs/func_odbc.c: Fix locking issues with channel datastores in func_odbc.c. * Fixed a potential memory leak when an existing datastore is manually destroyed by inline code instead of calling ast_datastore_free(). (closes issue ASTERISK-17948) Reported by: Archie Cobbs Review: https://reviewboard.asterisk.org/r/1687/ ........ Merged revisions 352291 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352292 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-23 20:31 +0000 [r352229-352232] Mark Michelson * /, main/features.c: Fix grammar of comment. ........ Merged revisions 352230 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352231 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, main/features.c: Fix blind transfers from failing if an 'h' extension is present. This prevents the 'h' extension from being run on the transferee channel when it is transferred via a native transfer mechanism such as SIP REFER. (closes ASTERISK-19173) Reported by: Ross Beer Tested by: Kristjan Vrban Patches: ASTERISK-19173 by Mark Michelson (license 5049) Review: https://reviewboard.asterisk.org/r/1685 ........ Merged revisions 352199 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352228 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-23 19:22 +0000 [r352166] Matthew Jordan * /, res/res_fax_spandsp.c: Correctly apply FAXOPT settings (V17, V27, V29) before starting spandsp layer While the FAXOPT function could be used to set the modem capabilities, the input to that function was not being applied correctly to the spandsp layer. This patch applies the current model capabilities before starting the spandsp layer. (closes issue: ASTERISK-16409) Reported by: Kristijan Vrban Tested by: Matt Jordan, Matthew Nicholson Patches: spandsp-modems-1.8.diff uploaded by mnicholson (license 5081) spandsp-modems-10.diff uploaded by mnicholson (license 5081) ........ Merged revisions 352144 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352149 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-23 18:34 +0000 [r352093-352134] Jonathan Rose * configs/musiconhold.conf.sample, res/res_musiconhold.c, CHANGES: Add an announcement option to music-on-hold - plays sound when put on hold/between songs This is a feature patch which allows an 'announcement' option to be specified in musiconhold.conf which should be set to the name of a sound. If a valid sound is specified for this option, then it will be played on that music on hold class whenever a channel bound to that class is put on hold as well as when Asterisk is able to detect that a song has ended before starting the next song (excludes external players). (closes ASTERISK-18977) Reported by: Timo Teräs Patches: asterisk-moh-announcement.diff uploaded by Timo Teräs (license 5409) * CHANGES, apps/app_mixmonitor.c: Adds the ability to stop specific mixmonitors by using unique IDs set at monitor launch. MixMonitor receives a new option i(channel_variable) which stores the unique id at said variable. StopMixMonitor now accepts ID as an optional argument, which if included will make StopMixMonitor specifically target the mixmonitor on that particular channel. CLI commands and AMI actions have been ammended to work with the IDs as well. In addition, monitors across a channel can now be listed be listed via CLI command "mixmonitor list " which will display all of the mixmonitors active on that channel along with the files they each have open. Created by Sergio González Martín. (closes issue ASTERISK-19096) Reported by: Sergio González Martín Review: https://reviewboard.asterisk.org/r/1643/ Review: https://reviewboard.asterisk.org/r/1682/ 2012-01-23 17:36 +0000 [r352092] Richard Mudgett * /, channels/chan_sip.c: Fix sip_cfg.notifycid to be set with the defined enum values. The invalid value used when notifycid was enabled was benign. As far as the code was concerned -1 and 1 are equivalent. (closes issue ASTERISK-19232) Reported by: Eike Kuiper ........ Merged revisions 352090 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352091 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-21 00:23 +0000 [r352041] Richard Mudgett * /, funcs/func_timeout.c, main/app.c: Fix ast_app_dtget() time unit inconsistency. Note: Noone calls ast_app_dtget() with the timeout parameter of zero so the bad code normally will never get executed. * Fix unnecessary floating point division in func_timeout.c timeout_write() when all other values are integers. (closes issue ASTERISK-16817) Reported by: Dmitry Andrianov ........ Merged revisions 352029 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352035 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-21 00:11 +0000 [r352018-352019] Mark Michelson * /, channels/chan_sip.c: Remove XXX comment that is not necessary. ........ Merged revisions 352016 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352017 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, channels/chan_sip.c: Fix RTP reference leak. If a blind transfer were initiated using a REFER without a prior reINVITE to place the call on hold, AND if Asterisk were sending RTCP reports, then there was a reference for the RTP instance of the transferer. This fixes the issue by merging two similar but slightly conflicting sections of code into a single area. It also adds a stop_media_flows() call in the case that the transferer's UA never sends a BYE to us like it is supposed to. (issue ASTERISK-19192) Review: https://reviewboard.asterisk.org/r/1681/ ........ Merged revisions 352014 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352015 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-20 23:05 +0000 [r351977] Richard Mudgett * channels/chan_sip.c: Make CLI sip show channel list the complete route set. (closes issue ASTERISK-16877) Reported by: klaus3000 Patches: show-complete-routeset-patch.txt (license #5054) patch uploaded by klaus3000 (modified) 2012-01-20 21:26 +0000 [r351939] Kinsey Moore * channels/chan_sip.c, UPGRADE.txt: SIP session timeout AMI event Add an AMI event in the Call category that is issued when a call is terminated due to either RTP stream inactivity or SIP session timer expiration. Event description: Event: SessionTimeout Source: source Channel: channel-name Uniqueid: channel-unique-id `source` can be either RTPTimeout or SIPSessionTimer (closes issue ASTERISK-16467) Patch-by: Kirill Katsnelson 2012-01-20 20:47 +0000 [r351900-351913] Mark Michelson * main/features.c, UPGRADE.txt, CHANGES, configs/features.conf.sample: Various parking improvements. * Adds per-parking lot options comebackcontext and comebackdialtime * Makes comebacktoorigin settable per parking lot * Sets a PARKER channel variable when comebacktoorigin is disabled (closes issue ASTERISK-16643) Reported by: Mitch Sharp (bluecrow76) Patches: asterisk-1.6.2.17.2-park-features-comebackcontext-consolidated-v3.diff by Mitch Sharp (bluecrow76) license 5231 with updates by me. Review: https://reviewboard.asterisk.org/r/1674 Review: https://reviewboard.asterisk.org/r/963 Reviewed by Richard Mudgett * apps/app_mixmonitor.c: Prevent potential buffer overflow on AMI MixMonitor command. Don't be alarmed. This only affected trunk, and it would have required manager access to your system. 2012-01-20 19:36 +0000 [r351817-351862] Kinsey Moore * /, codecs/ilbc/iLBC_test.c: More corrections for the ilbc code These changes are in a file that is not compiled by default, and so were missed on earlier checks. ........ Merged revisions 351860 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351861 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, codecs/ilbc/LPCencode.c, codecs/ilbc/iLBC_decode.c: Restore LSF_check function calls from set/unused variable removal These functions are not noops and modify the array that is passed in. Thanks for the catch Richard. ........ Merged revisions 351818 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, codecs/ilbc/LPCencode.c, codecs/ilbc/iLBC_decode.c: Remove more set, but unused variables in the ilbc codec GCC 4.6.3 caught these in dev mode as well. ........ Merged revisions 351816 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-20 16:00 +0000 [r351764] Jonathan Rose * /, channels/chan_sip.c: Adds setting of mwi_from field to check_auth_result check_peer_ok (closes ASTERISK-19057) Reported By: Yuri Patches: 348360chan_sip.diff uploaded by Yuri (license 5242) ........ Merged revisions 351759 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351762 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-20 16:00 +0000 [r351763] Matthew Jordan * /, codecs/ilbc/helpfun.c: Remove unused variable 'tmp' from helpfun in ilbc codec gcc version 4.6.2 caught an unused variable in the ilbc codec library. This would prevent compilation with --enable-dev-mode; variable removed. ........ Merged revisions 351760 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351761 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-20 13:12 +0000 [r351709] Stefan Schmidt * /, contrib/asterisk-ng-doxygen: enable doxygen build for files in the channels/sip folder like reqresp_parser.c ........ Merged revisions 351707 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351708 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-19 23:31 +0000 [r351667] Richard Mudgett * /, channels/chan_sip.c, channels/sip/reqresp_parser.c: Misc minor fixes in reqresp_parser.c and chan_sip.c. * Fix corner cases in get_calleridname() parsing and ensure that the output buffer is nul terminated. * Make get_calleridname() truncate the name it parses if the given buffer is too small rather than abandoning the parse and not returning anything for the name. Adjusted get_calleridname_test() unit test to handle the truncation change. * Fix get_in_brackets_test() unit test to check the results of get_in_brackets() correctly. * Fix parse_name_andor_addr() to not return the address of a local buffer. This function is currently not used. * Fix potential NULL pointer dereference in sip_sendtext(). * No need to memset(calleridname) in check_user_full() or tmp_name in get_name_and_number() because get_calleridname() ensures that it is nul terminated. * Reply with an accurate response if get_msg_text() fails in receive_message(). This is academic in v1.8 because get_msg_text() can never fail. ........ Merged revisions 351618 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351646 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-19 22:44 +0000 [r351613] Kinsey Moore * res/res_rtp_asterisk.c, /: Correct output of RTCP jitter statistics in SR and RR reports Change the RTCP RR and SR generation code to convert Asterisk's internal jitter statistics to be represented in RTP timestamp units based on the rate of the codec in use instead of in seconds. (closes issue ASTERISK-14530) ........ Merged revisions 351611 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351612 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-19 21:55 +0000 [r351561] Jonathan Rose * /, channels/chan_sip.c, include/asterisk/netsock2.h: Eliminates doubling the :port part of SIP Notify Message-Account headers. This patch prevents the domain string from getting mangled during the initreqprep step by moving the initialization to before its immediate use. It also documents this pitfall for the ast_sockaddr_stringify functions. (issue ASTERISK-19057) Reported by: Yuri Review: https://reviewboard.asterisk.org/r/1678/ ........ Merged revisions 351559 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351560 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-19 21:13 +0000 [r351506] Joshua Colp * /, channels/chan_sip.c: Prevent crash when an SDP offer is received with an encrypted video stream when support for video is disabled and res_srtp is loaded. (closes issue ASTERISK-19202) Reported by: Catalin Sanda ........ Merged revisions 351504 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351505 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-18 21:06 +0000 [r351452] Matthew Jordan * codecs/ilbc/syntFilter.c (added), /, codecs/ilbc/iCBConstruct.h (added), codecs/ilbc/iLBC_test.c (added), codecs/ilbc/syntFilter.h (added), codecs/ilbc/StateConstructW.c (added), codecs/ilbc/packing.c (added), codecs/ilbc/StateConstructW.h (added), codecs/ilbc/packing.h (added), codecs/ilbc/getCBvec.c (added), codecs/ilbc/LPCdecode.c (added), codecs/ilbc/enhancer.c (added), codecs/ilbc/lsf.c (added), codecs/ilbc/iLBC_encode.c (added), codecs/ilbc/getCBvec.h (added), codecs/ilbc/LPCdecode.h (added), codecs/ilbc/iLBC_define.h (added), codecs/ilbc/FrameClassify.c (added), codecs/ilbc/enhancer.h (added), codecs/ilbc/lsf.h (added), codecs/ilbc/extract-cfile.awk (added), codecs/ilbc/iLBC_encode.h (added), codecs/ilbc/Makefile, codecs/ilbc/FrameClassify.h (added), codecs/ilbc/helpfun.c (added), codecs/ilbc/LICENSE_ADDENDUM (added), codecs/ilbc/doCPLC.c (added), codecs/ilbc/anaFilter.c (added), codecs/ilbc/helpfun.h (added), codecs/ilbc/createCB.c (added), codecs/ilbc/doCPLC.h (added), codecs/ilbc/anaFilter.h (added), codecs/ilbc/constants.c (added), codecs/ilbc/iLBC_decode.c (added), codecs/ilbc/createCB.h (added), codecs/ilbc/constants.h (added), codecs/ilbc/iLBC_decode.h (added), codecs/ilbc/iCBSearch.c (added), codecs/ilbc/filter.c (added), codecs/ilbc/gainquant.c (added), codecs/ilbc/hpInput.c (added), codecs/ilbc/hpOutput.c (added), codecs/ilbc/iCBSearch.h (added), codecs/ilbc/rfc3951.txt (added), codecs/ilbc/filter.h (added), codecs/ilbc/gainquant.h (added), codecs/ilbc/LPCencode.c (added), codecs/ilbc/hpInput.h (added), codecs/ilbc/PATENTS (added), codecs/ilbc/StateSearchW.c (added), codecs/ilbc/hpOutput.h (added), codecs/codec_ilbc.c, contrib/scripts/get_ilbc_source.sh, codecs/ilbc/LICENSE (added), codecs/ilbc/LPCencode.h (added), codecs/ilbc/StateSearchW.h (added), codecs/ilbc/iCBConstruct.c (added): Include iLBC source code for distribution with Asterisk This patch includes the iLBC source code for distribution with Asterisk. Clarification regarding the iLBC source code was provided by Google, and the appropriate licenses have been included in the codecs/ilbc folder. Review: https://reviewboard.asterisk.org/r/1675 Review: https://reviewboard.asterisk.org/r/1649 (closes issue: ASTERISK-18943) Reporter: Leif Madsen Tested by: Matt Jordan ........ Merged revisions 351450 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351451 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-18 16:02 +0000 [r351409] Stefan Schmidt * /, channels/chan_sip.c: The get_pai function in chan_sip.c didn't recognized a proper callerid name and number from a P-Asserted-Identity cause the header parsing logic was wrong. Changing the parsing functions to the sip header parsing APIs in reqresp_parser.h solves this problem. Review: https://reviewboard.asterisk.org/r/1673 Reviewed by: wdoekes2 and Mark Michelson ........ Merged revisions 351396 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351408 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-17 19:45 +0000 [r351360] Walter Doekes * Makefile: Fix support for parallel building with make (-j). Previously make -j would cause a race between doing cleanup of certain files (defaults.h, menuselect, ...) and creating them anew. Add a new target that depends on cleanup only and has a submake doing the rest as command string. This way the cleanup goes first. (closes issue ASTERISK-18751) Tested by: Jeremy Kister Reviewed by: Paul Belanger Review: https://reviewboard.asterisk.org/r/1660 2012-01-17 17:23 +0000 [r351311] Mark Michelson * res/res_rtp_asterisk.c, /: Eliminate odd initialization of probation variable. ........ Merged revisions 351306 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351308 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-17 17:15 +0000 [r351290] Jonathan Rose * res/res_rtp_asterisk.c, /, configs/rtp.conf.sample, CHANGES: Adds pjmedia probation concepts to res_rtp_asterisk's learning mode. In order to better handle RTP sources with strictrtp enabled (which is now default in 10) using the learning mode to figure out new sources when they change is handled by checking for a number of consecutive (by sequence number) packets received to an rtp struct based on a new configurable value called 'probation'. Also, during learning mode instead of liberally accepting all packets received, we now reject packets until a clear source has been determined. Review: https://reviewboard.asterisk.org/r/1663/ ........ Merged revisions 351287 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351289 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-17 16:56 +0000 [r351288] Mark Michelson * /, channels/chan_sip.c: Use built-in parsing functions for Contact and Record-Route headers. If a Contact or a Record-Route header had a quoted string with an item in angle brackets, then we would mis-parse it. For instance, "Bob <1234>" <1234@example.org> would be misparsed as having the URI "1234" The fix for this is to use parsing functions from reqresp_parser.h since they are heavily tested and are awesome. (issue ASTERISK-18990) ........ Merged revisions 351284 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351286 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-17 16:08 +0000 [r351235] Matthew Jordan * /, channels/chan_sip.c: Fix udptl issue with initial INVITE introduced by r351027 When an inital INVITE occurs that contains image media, a channel is not yet associated with the SIP dialog. The file descriptor associated with the udptl session needs to be set in initialize_udptl or in sip_new to account for this scenario. ........ Merged revisions 351233 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351234 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-17 01:48 +0000 [r351184] Russell Bryant * /, channels/chan_sip.c: Merged revisions 351183 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r351183 | russell | 2012-01-16 20:43:19 -0500 (Mon, 16 Jan 2012) | 29 lines Merged revisions 351182 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r351182 | russell | 2012-01-16 20:37:03 -0500 (Mon, 16 Jan 2012) | 22 lines Add some missing locking in chan_sip. This patch adds some missing locking to the function send_provisional_keepalive_full(). This function is called from the scheduler, which is processed in the SIP monitor thread. The associated channel (or pbx) thread will also be using the same sip_pvt and ast_channel so locking must be used. The sip_pvt_lock_full() function is used to ensure proper locking order in a safe manner. In passing, document a suspected reference counting error in this function. The "fix" is left commented out because when the "fix" is present, crashes occur. My theory is that fixing it is exposing a reference counting error elsewhere, but I don't know where. (Or my analysis of this being a problem could have been completely wrong in the first place). Leave the comment in the code for so that someone may investigate it again in the future. Also add a bit of doxygen to transmit_provisional_response(). (closes issue ASTERISK-18979) Review: https://reviewboard.asterisk.org/r/1648 ........ ................ 2012-01-16 21:50 +0000 [r351082-351143] Terry Wilson * /, channels/chan_sip.c: Ensure ACK retransmit & hangup on non-200 response to INVITE When handling a non-2xx final response on an INVITE transaction, we have to keep the transaction around after we send an ACK in case we receive a retransmission of the response so we can re-transmit the ACK, but also tear down the ast_channel as soon as we transmit the ACK. Before this patch, we could fail at both of these things. Calling sip_alreadygone/needdestroy prevented us from keeping the transaction up and retransmitting the ACK, and queueing CONGESTION was not sufficient to cause the channel to be torn down when originating calls via the CLI, for example. This patch queues a hangup with CONGESTION instead of just queueing CONGESTION for these responses and removes the sip_alreadygone and sip_needdestroy calls from handle_response_invite on non-2xx responses. It relies on the hangup calling sip_scheddestroy. For more information, see section 17.1.1.1 of RFC 3261. (closes issue ASTERISK-17717) Review: https://reviewboard.asterisk.org/r/1672/ ........ Merged revisions 351130 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351131 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, channels/chan_sip.c: Don't prematurely stop SIP session timer When Asterisk is the UAS (incoming call, endpoint is re-inviting) the SIP session timer expires after half the time the sip endpoint indicates in the Session-expires header in proc_session_timer(). The session timer was being stopped totally and being handled as an error case instead of running again until the second expiry. This patch treats the half-time expiry as a non-error case and continues the timer until the true expiry. (closes issue ASTERISK-18996) Reported by: Thomas Arimont Tested by: Thomas Arimont Patches: session_timer_fix.diff by Terry Wilson (License #5357) based on session_timer.patch by Thomas Arimont (License #5525) ........ Merged revisions 351080 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351081 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-16 19:49 +0000 [r351079] Tilghman Lesher * main/ast_expr2.y, CHANGES, main/ast_expr2.c: Add ABS() absolute value function to the expression parser. 2012-01-16 19:13 +0000 [r351029] Matthew Jordan * /, channels/chan_sip.c: Create and initialize udptl only when dialog negotiates for image media Prior to this patch, the udptl struct was allocated and initialized when a dialog was associated with a peer that supported T.38, when a new SIP channel was allocated, or what an INVITE request was received. This resulted in any dialog associated with a peer that supported T.38 having udptl support assigned to it, including the UDP ports needed for communication. This occurred even in non-INVITE dialogs that would never send image media. This patch creates and initializes the udptl structure only when the SDP for a dialog specifies that image media is supported, or when Asterisk indicates through the appropriate control frame that a dialog is to support T.38. (closes issue ASTERISK-16698) Reported by: under Tested by: Stefan Schmidt Patches: udptl_20120113.diff uploaded by mjordan (License #6283) (closes issue ASTERISK-16794) Reported by: Elazar Broad Tested by: Stefan Schmidt review: https://reviewboard.asterisk.org/r/1668/ ........ Merged revisions 351027 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351028 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-16 17:12 +0000 [r350979] Sean Bright * /, main/db.c: Sort the output of 'database showkey' as well. You can pass wildcards (%) to the database CLI commands, so this will sort the returned list of matches. ........ Merged revisions 350978 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-16 17:07 +0000 [r350977] Joshua Colp * main/rtp_engine.c, /: Add missing code to set direct RTP setup information during dialing. ........ Merged revisions 350975 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350976 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-16 14:31 +0000 [r350939] Sean Bright * /, main/db.c: Sort the output of 'database show' by key. This more closely mimics the behavior of 'database show' before the conversion to sqlite3. ........ Merged revisions 350938 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-15 20:16 +0000 [r350887-350890] Walter Doekes * /, main/asterisk.c: Allow only one thread at a time to do asterisk cleanup/shutdown. Add locking around the really-really-quit part of the core stop/restart part. Previously more than one thread could be called to do cleanup, causing atexit handlers to be run multiple times, in turn causing segfaults. (issue ASTERISK-18883) Reviewed by: Terry Wilson Review: https://reviewboard.asterisk.org/r/1662/ Review: https://reviewboard.asterisk.org/r/1658/ ........ Merged revisions 350888 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350889 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, utils/extconf.c: Fix -Werror=unused-but-set-variable compile error in utils/extconf.c. Note that I'm not confirming legitimacy of having that file in tree at all. Is anyone using aelparse/conf2ael? (issue ASTERISK-15350) ........ Merged revisions 350885 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350886 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-14 16:43 +0000 [r350791-350839] Kevin P. Fleming * /, configure, autoconf/ast_gcc_attribute.m4, configure.ac, autoconf/libcurl.m4: Ensure that all AC_LANG_PROGRAM calls in the configure script are properly quoted. Recent versions of autoconf (2.68 on my system) won't properly process the configure script unless every call to AC_LANG_PROGRAM is m4-quoted. Many calls in the script were, but many were not. This patch corrects the unquoted calls. ........ Merged revisions 350837 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350838 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, channels/chan_h323.c, addons/chan_mobile.c, res/res_pktccops.c, contrib/scripts/install_prereq: Multiple revisions 350788-350789 ........ r350788 | kpfleming | 2012-01-14 09:22:33 -0600 (Sat, 14 Jan 2012) | 8 lines Ensure that two prerequisites are properly installed on Debian-style distributions. * Don't specify a specific version of libgmime; newer versions are available now and acceptable. * Install libsrtp so that res_srtp can be built. ........ r350789 | kpfleming | 2012-01-14 09:23:32 -0600 (Sat, 14 Jan 2012) | 3 lines Correct some 'set-but-not-used' variable warnings. ........ Merged revisions 350788-350789 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350790 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-13 22:17 +0000 [r350738] Kinsey Moore * /, include/asterisk/autoconfig.h.in: Run bootstrap.sh for the for the ASTERISK-18929 fix configure and autoconfig.h.in were not regenerated when the fix was committed. ........ Merged revisions 350736 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350737 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-13 21:52 +0000 [r350735] Richard Mudgett * /, configs/cel_pgsql.conf.sample, configs/cel_odbc.conf.sample: Correct eventtype names in cel_odbc and cel_pgsql sample files ........ Merged revisions 350733 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350734 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-13 21:42 +0000 [r350732] Kinsey Moore * /, configure.ac, bootstrap.sh, main/asterisk.c: Make sure asterisk builds on OpenBSD OpenBSD defines SO_PEERCRED, but it returns a 'struct sockpeercred', not 'struct ucred', which causes compilation of main/asterisk.c to fail in read_credentials(). This allows configure to check for sockpeercred and asterisk to deal with it properly. (closes issue ASTERISK-18929) Reported-by: Barry Miller Patch-by: Barry Miller ........ Merged revisions 350730 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350731 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-13 20:32 +0000 [r350681] Mark Michelson * /, channels/sip/config_parser.c: Set port to a default sane value if a bogus one is provided when parsing hostnames. ........ Merged revisions 350679 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350680 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-13 18:52 +0000 [r350605-350644] Richard Mudgett * main/features.c: Remove some dead code in ast_bridge_call(). None of the parameters to ast_bridge_call() can be NULL for the bridge to work so no need to check for it. * configs/cel_sqlite3_custom.conf.sample, cel/cel_odbc.c, configs/cel.conf.sample, /, cel/cel_manager.c, configs/cel_pgsql.conf.sample, configs/cel_odbc.conf.sample, main/cel.c, configs/cel_custom.conf.sample: Add missing CEL logging fields to various CEL backends. Multiple revisions 350555,350571 ........ r350555 | rmudgett | 2012-01-13 11:12:51 -0600 (Fri, 13 Jan 2012) | 12 lines Add missing CEL logging fields to various CEL backends. * Add missing eventextra to cel_psql.c and cel_odbc.c. * Add missing PeerAccount and EventExtra to cel_manager.c. * Add missing userdeftype support for cel_custom.conf.sample and cel_sqlite3_custom.conf.sample. (closes issue ASTERISK-17190) Reported by: Bryant Zimmerman ........ r350571 | rmudgett | 2012-01-13 11:23:57 -0600 (Fri, 13 Jan 2012) | 8 lines Use compatible names for event extra data for various CEL backends. * Change eventextra to extra in cel_psql.c and cel_odbc.c. * Change EventExtra to Extra in cel_manager.c. (issue ASTERISK-17190) ........ Merged revisions 350555,350571 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350585 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-13 17:00 +0000 [r350551-350554] Matthew Jordan * /, apps/app_queue.c: Realtime queues failed to load queue information without queue member table Previously, realtime queues could be loaded without defining the queue member table. This allowed for queue members to be dynamic, while the realtime queue definitions could exist in some backing storage. Revision 342223 broke this when it changed the return value for realtime_multientry to return NULL when no results are returned. Previously, an empty ast_config object was expected. (closes issue ASTERISK-19170) Reported by: Rene Mendoza Tested by: Rene Mendoza Patches: rt_queue_member_patch.diff uploaded by Matt Jordan (license 6283) ........ Merged revisions 350552 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350553 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, bridges/bridge_builtin_features.c, channels/chan_bridge.c, include/asterisk/bridging.h, apps/app_confbridge.c, main/bridging.c: Fix crash from bridge channel hangup race condition in ConfBridge This patch addresses two issues in ConfBridge and the channel bridge layer: 1. It fixes a race condition wherein the bridge channel could be hung up 2. It removes the deadlock avoidance from the bridging layer and makes the bridge_pvt an ao2 ref counted object Patch by David Vossel (mjordan was merely the commit monkey) (issue ASTERISK-18988) (closes issue ASTERISK-18885) Reported by: Dmitry Melekhov Tested by: Matt Jordan Patches: chan_bridge_cleanup_v.diff uploaded by David Vossel (license 5628) (closes issue ASTERISK-19100) Reported by: Matt Jordan Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1654/ ........ Merged revisions 350550 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-12 16:10 +0000 [r350503] Jonathan Rose * /, main/features.c: Adds peer to CEL report on CEL_BRIDGE_START and CEL_BRIDGE_END (closes issue ASTERISK-17940) Reporter: Nic Colledge Patches: features_18.patch uploaded by Nic Colledge (license 6245) ........ Merged revisions 350501 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350502 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-11 22:53 +0000 [r350416-350454] Richard Mudgett * /, main/cel.c: Remove extraneous BRIDGEPEER AMI VarSet event on a CEL dummy channel. (closes issue ASTERISK-19180) Reported by: Corey Farrell Patches: asterisk_cel_noevent_varset.diff (license #5909) patch uploaded by Corey Farrell ........ Merged revisions 350452 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350453 from http://svn.asterisk.org/svn/asterisk/branches/10 * apps/app_dial.c, /, CHANGES, apps/app_followme.c: Make FollowMe optionally update connected line information when the accepting endpoint is bridged. Like Dial and Queue, FollowMe needs to deal with AST_CONTROL_CONNECTED_LINE information so when the parties are initially bridged, the connected line information will be correct. * Added the 'I' option just like the app_dial and app_queue 'I' option. * Made 'N' option ignored if the call is already answered. (closes issue ASTERISK-18969) Reported by: rmudgett Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1656/ ........ Merged revisions 350364 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350415 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-11 19:19 +0000 [r350365] Terry Wilson * main/channel.c: Always treat arguments to get_by_name_cb as strings Initially, support was left in for the old style of searching, even though it wasn't actually used. In the case of name_len != 0, the OBJ_KEY flag isn't passed because we aren't matching on a full key and therefor can't use the hash function to optimize. The code left in to support the old way of searching unfortunately treated a prefix search like this as though an ast_channel struct was passed as an arg and caused a crash. This patch also adds needed parentheses around some matching conditions. (closes issue ASTERISK-19182) 2012-01-10 22:10 +0000 [r350273-350313] Richard Mudgett * /, funcs/func_lock.c: Fix absolute/relative time mismatch in LOCK function. The time passed by the LOCK function to an internal function was relative time when the function expected absolute time. * Don't use C++ keywords in get_lock(). (closes issue ASTERISK-16868) Reported by: Andrey Solovyev Patches: 20101102__issue18207.diff.txt (license #5003) patch uploaded by Andrey Solovyev (modified) ........ Merged revisions 350311 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350312 from http://svn.asterisk.org/svn/asterisk/branches/10 * main/channel.c: Fix compiler warnings reported by gcc v4.2.4. 2012-01-09 22:15 +0000 [r350223] Terry Wilson * main/udptl.c, apps/app_dahdibarge.c, addons/chan_ooh323.c, channels/chan_local.c, main/rtp_engine.c, main/say.c, apps/app_record.c, apps/app_test.c, channels/console_video.c, apps/app_alarmreceiver.c, apps/app_chanisavail.c, bridges/bridge_multiplexed.c, channels/chan_iax2.c, main/indications.c, main/cli.c, channels/chan_dahdi.c, channels/sig_analog.c, channels/chan_skinny.c, main/features.c, apps/app_dumpchan.c, pbx/pbx_realtime.c, apps/app_amd.c, channels/chan_alsa.c, apps/app_externalivr.c, main/bridging.c, apps/app_milliwatt.c, channels/sig_ss7.c, apps/app_dial.c, main/pbx.c, apps/app_page.c, apps/app_softhangup.c, apps/app_fax.c, apps/app_dahdiras.c, channels/chan_agent.c, apps/app_disa.c, include/asterisk/channel.h, main/aoc.c, apps/app_talkdetect.c, main/cel.c, res/res_mutestream.c, res/res_monitor.c, apps/app_playback.c, channels/chan_misdn.c, funcs/func_channel.c, apps/app_macro.c, apps/app_mixmonitor.c, apps/app_chanspy.c, apps/app_voicemail.c, res/res_calendar.c, channels/chan_unistim.c, channels/chan_vpb.cc, main/ccss.c, apps/app_meetme.c, apps/app_readexten.c, res/res_musiconhold.c, main/autochan.c, channels/chan_gtalk.c, apps/app_followme.c, res/res_jabber.c, main/cdr.c, main/channel.c, main/dial.c, channels/chan_phone.c, main/manager.c, funcs/func_groupcount.c, funcs/func_audiohookinherit.c, funcs/func_frame_trace.c, res/res_agi.c, apps/app_minivm.c, main/app.c, apps/app_confbridge.c, apps/app_rpt.c, addons/chan_mobile.c, apps/app_parkandannounce.c, channels/chan_mgcp.c, apps/app_jack.c, apps/app_adsiprog.c, channels/chan_sip.c, res/res_fax.c, apps/app_waitforsilence.c, funcs/func_lock.c, main/channel_internal_api.c (added), res/res_adsi.c, pbx/pbx_lua.c, channels/chan_console.c, apps/app_getcpeid.c, channels/sig_pri.c, apps/app_queue.c, channels/chan_oss.c, funcs/func_global.c, channels/chan_usbradio.c, channels/chan_jingle.c, apps/app_flash.c, apps/app_directed_pickup.c, main/abstract_jb.c, main/file.c, channels/chan_h323.c, res/snmp/agent.c, pbx/pbx_dundi.c, apps/app_sms.c, channels/chan_nbs.c, apps/app_stack.c, main/dsp.c: Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ 2012-01-09 21:56 +0000 [r350222] Richard Mudgett * /, channels/chan_iax2.c: Fix joinable thread terminating without joiner memory leak in chan_iax.c. The iax2_process_thread() can exit without anyone waiting to join the thread. If noone is waiting to join the thread then a large memory leak occurs. * Made iax2_process_thread() deatach itself if nobody is waiting to join the thread. (closes issue ASTERISK-17339) Reported by: Tzafrir Cohen Patches: asterisk-1.8.4.4-chan_iax2-detach-thread-on-non-stop-exit.patch (license #5617) patch uploaded by Alex Villacis Lasso (modified) (closes issue ASTERISK-17825) Reported by: wangjin ........ Merged revisions 350220 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350221 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-09 19:37 +0000 [r350181] Walter Doekes * /, main/db.c: Fix shutdown handling of sqlite3 astdb. If a db_sync was scheduled just before shutdown, the atexit code calling db_sync would have no effect, causing the astdb commit thread to stay alive. This caused the SIP/realtime_sipregs test to fail. (The fallback kill would run the atexit code again and that would wreak havoc.) This fixes that the atexit kill condition is picked up properly. (closes issue ASTERISK-18883) Reviewed by: Terry Wilson Review: https://reviewboard.asterisk.org/r/1659 ........ Merged revisions 350180 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-09 18:58 +0000 [r350077-350130] Richard Mudgett * /, contrib/scripts/valgrind_compare (added): Multiple revisions 350127-350128 ........ r350127 | rmudgett | 2012-01-09 12:40:33 -0600 (Mon, 09 Jan 2012) | 12 lines Update contrib script live_ast to invoke Asterisk with valgrind and suppression file. * Added valgrind_compare script to compare two valgrind log files for differences. (issue ASTERISK-17339) Reported by: Tzafrir Cohen Patches: valgrind_compare (license #5035) script uploaded by Tzafrir Cohen live_ast_valgrind.diff (license #5035) patch uploaded by Tzafrir Cohen live_ast_valgrind_v2.diff (license #5185) patch uploaded by Paul Belanger ........ r350128 | rmudgett | 2012-01-09 12:54:56 -0600 (Mon, 09 Jan 2012) | 11 lines live_ast: valgrind: run asterisk under valgrind Adds a new sub-command, "valgrind" to live_ast. It runs asterisk under valgrind. The extra command-line parameters are passed to Asterisk as usual, and parameters to valgrind are passed through LIVE_AST_VALGRIND_ARGS in live.conf . Review: https://reviewboard.asterisk.org/r/1109/ Merged revisions 326636 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 350127-350128 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350129 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, main/asterisk.c: Make Asterisk -x command line parameter imply -r parameter presence. The Asterisk -x command line parameter is documented inconsistently. * Made the -x documentation and behavior consistent. * Since this is also a new year, updated the copyright notices while here. (closes issue ASTERISK-19094) Reported by: Eugene Patches: issueA19094_correct_asterisk_option_x.patch (license #5674) patch uploaded by Walter Doekes (modified) Tested by: Eugene ........ Merged revisions 350075 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350076 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-09 15:40 +0000 [r350025] Kinsey Moore * /, apps/app_meetme.c: Prevent SLA settings from getting wiped out on reload If SLA was reloaded without the config file being changed, current settings got wiped out before the SLA reload code decided it wasn't going to reload the file since nothing was changed. Moving the settings reset later in the reload process fixes this. (closes issue AST-744) ........ Merged revisions 350023 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350024 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-06 23:31 +0000 [r349978] Terry Wilson * /, channels/chan_sip.c: Don't leak CID in From header when presentation=unavailable When someone does Set(CALLERPRES()=unavailable) (or Set(CALLERID(pres)=unavailable)) when sendrpid=no, the From header shows "Anonymous" . When sendrpid=yes/pai, the From header will still display the callerid info, even though we supply an rpid header with the anonymous info. It seems like we shouldn't leak that info in any case. Skimming http://tools.ietf.org/html/draft-ietf-sip-privacy-04 seems to indicate that one shouldn't send identifying info in the From in this case. This patch anonymizes the From header as well even when sendrpid=yes/pai. (closes issue ASTERISK-16538) Review: https://reviewboard.asterisk.org/r/1649/ ........ Merged revisions 349968 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 349977 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-06 21:26 +0000 [r349929] Kinsey Moore * /, pbx/pbx_lua.c: Fix lua goto detection to prevent unexpected behavior with confbridge A bug in the pbx_lua goto detection was causing the dialplan to hangup unexpectedly after confbridge exited if it had called lua dialplan code during execution. Patch-by: Timo Teras Acked-by: Matt Nicholson (closes issue ASTERISK-18976) ........ Merged revisions 349928 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-06 16:50 +0000 [r349874] Richard Mudgett * /, apps/app_followme.c: Fix memory leaks in app_followme find_realtime(). (closes issue ASTERISK-19055) Reported by: Matt Jordan ........ Merged revisions 349872 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 349873 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-05 23:58 +0000 [r349823] Matthew Jordan * /, res/res_fax.c: Fix premature free'ing of the frame committed in r349608 Even though we set the frame to the ast_null_frame and return that, the caller of the frame hook may still need the frame. This now is a bit more careful about when it frees the frame, i.e., only under the same conditions that applied when we duplicated it in the first place. ........ Merged revisions 349822 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-05 23:47 +0000 [r349782-349821] Richard Mudgett * /, cel/cel_sqlite3_custom.c: Make not assume that the cel_sqlite3_custom SQL table primary key is AcctId. If a table is created by some other application and the primary key is not named "AcctId", cel/cel_sqlite3_custom.c will always try to create the table and fail because it already exists. * Change the SQL table query to not require AcctId as the primary key. (closes issue ASTERISK-18963) Reported by: socketpair Patches: fix.patch (license #6337) patch uploaded by socketpair ........ Merged revisions 349819 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 349820 from http://svn.asterisk.org/svn/asterisk/branches/10 * UPGRADE.txt, pbx/pbx_config.c: Make pbx_config.c use Gosub instead of Macro call for stdexten. Users created by users.conf with hasvoicemail=yes have been documented as using a Gosub to stdexten since v1.6.0. However, the code still generates dialplan to access stdexten as a Macro as documented in v1.4; which does not work with the newer extensions.conf.sample file. * Make generated dialplan access the stdexten dialplan with the documented Gosub instead of the older Macro style. (closes issue ASTERISK-18809) Reported by: Jay Allen Patches: gosub_patch-pbx_config.patch (license #6323) patch uploaded by Jay Allen (modified) Tested by: rmudgett 2012-01-05 22:11 +0000 [r349733] Kinsey Moore * /, main/file.c: Allow playback of formats that don't support seeking ast_streamfile previously did unconditional seeking on files that broke playback of formats that don't support that functionality. This patch avoids the seek that was causing the problem. This regression was introduced in r158062. (closes issue ASTERISK-18994) Patch-by: Timo Teras ........ Merged revisions 349731 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 349732 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-05 22:02 +0000 [r349674-349730] Jonathan Rose * /, main/dsp.c: Fix an issue where dsp.c would interpret multiple dtmf events from a single key press. When receiving calls from a mobile phone into a DISA system on a connection with significant interference, the reporter's Asterisk system would interpret DTMF incorrectly and replicate digits received. This patch resolves that by increasing the number of frames a mismatch has to be detected before assuming the DTMF is over by 1 frame and adjusts dtmf_detect function to reset hits and misses only when an edge is detected. (closes issue ASTERISK-17493) Reported by: Alec Davis Patches: bug18904-refactor.diff.txt uploaded by Alec Davis (license 5546) Review: https://reviewboard.asterisk.org/r/1130/ ........ Merged revisions 349728 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 349729 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, main/asterisk.c: Ensures Asterisk closes when receiving terminal signals in 'no fork' mode. When catching a signal, in no fork mode the console thread is identical to the thread responsible for catching the signal and closing Asterisk, which requires it to first dispense with the console thread. Prior to this patch, if these threads were identical, upon receiving a killing signal, the thread will send an URG signal to itself, which we also catch and then promptly do nothing with. Obviously this isn't useful behavior. (closes issue ASTERISK-19127) Reported By: Bryon Clark Patches: quit_on_signals.patch uploaded by Bryon Clark (license 6157) ........ Merged revisions 349672 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 349673 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-04 22:23 +0000 [r349609-349634] Matthew Jordan * /, apps/confbridge/conf_config_parser.c: Fix for ConfBridge config parser unlocking channel mutex too many times When looking up a ConfBridge profile, the config parser would, if it found a channel datastore on the channel requesting the bridge profile, unlock the channel mutex twice. Since that's a little aggressive, it now only unlocks it once. (closes issue ASTERISK-19042) Reported by: Matt Jordan Tested by: Matt Jordan Patches: 19042 uploaded by David Vossel (license 5628) ........ Merged revisions 349619 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, res/res_fax.c: Free successfully translated frame in fax_gateway_framehook A frame that is translated via ast_translate is also duplicated via ast_frdup. This will allocate a new frame on the heap, which needs to be free'd at the appropriate time. This issue reporter used valgrind to find that this occurred in res_fax's fax_gateway_framehook; a quick search through the code showed that only place this was currently not handling the translatted frame properly. (closes issue ASTERISK-19133) Reported by: Sylvain Rochet ........ Merged revisions 349608 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-04 20:55 +0000 [r349560] Richard Mudgett * channels/chan_dahdi.c, /: Fix segfault in chan_dahdi for CHANNEL(dahdi_span) evaluation on hangup. * Added NULL private pointer checks in the following chan_dahdi channel callbacks: dahdi_func_read(), dahdi_func_write(), dahdi_setoption(), and dahdi_queryoption(). (closes issue ASTERISK-19142) Reported by: Diego Aguirre Tested by: rmudgett ........ Merged revisions 349558 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 349559 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-04 20:24 +0000 [r349506-349535] Kinsey Moore * contrib/init.d/rc.debian.asterisk, /: Make debian init script conform to the LSB standard Previously, this init script would return 1 if Asterisk was already running. This is incorrect behavior according to the LSB standard and has been fixed by returning 0 instead. (closes issue ASTERISK-17958) Reported-by: johnc ........ Merged revisions 349529 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 349532 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, contrib/scripts/autosupport.8, contrib/scripts/autosupport: Update autosupport script and man page Added information collection from the output of the utilities: top, free, uptime, ifconfig Added information collection from the output of the Asterisk command 'dahdi show status' Added option / flag '-n, --non-interactive' Updated man page to reflect new option / flag '-n, --non-interactive' Patch-by: John Bigelow (itzanger) (closes issue AST-749) ........ Merged revisions 349504 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 349505 from http://svn.asterisk.org/svn/asterisk/branches/10 2012-01-04 19:53 +0000 [r349452-349503] Jonathan Rose * /, channels/chan_sip.c: Adds Subscription-State header to notify with call completion. per RFC3265 (Closes issue ASTERISK-17953) Reported by: George Konopacki Patches: 19400.patch uploaded by mmichelson (license 5049) ........ Merged revisions 349482 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 349502 from http://svn.asterisk.org/svn/asterisk/branches/10 * main/pbx.c, /: Fix documentation for SayNumber to reflect the fact that language is changed in CHANNEL() (closes issue ASTERISK-18962) reported by: Nir Simionovich ........ Merged revisions 349450 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 349451 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-12-31 15:48 +0000 [r349409-349410] Russell Bryant * channels/chan_sip.c: Fix some minor formatting issues based on coding guidelines. * channels/sip/include/dialog.h, channels/chan_sip.c, include/asterisk/astobj2.h, main/astobj2.c: Constify tag argument in REF_DEBUG related code. 2011-12-29 15:16 +0000 [r349341] Matthew Jordan * main/rtp_engine.c, /: Handle AST_CONTROL_UPDATE_RTP_PEER frames in local bridge loop Failing to handle AST_CONTROL_UPDATE_RTP_PEER frames in the local bridge loop causes the loop to exit prematurely. This causes a variety of negative side effects, depending on when the loop exits. This patch handles the frame by essentially swallowing the frame in the local loop, as the current channel drivers expect the RTP bridge to handle the frame, and, in the case of the local bridge loop, no additional action is necessary. (issue ASTERISK-19040) (issue ASTERISK-19128) (issue ASTERISK-17725) (issue ASTERISK-18340) (closes issue ASTERISK-19095) Reported by: Stefan Schmidt Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1640/ ........ Merged revisions 349339 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 349340 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-12-28 21:39 +0000 [r349291] Sean Bright * /, main/audiohook.c: Use ast_audiohook_write_list_empty to determine if our lists are empty instead of duplicating that logic. ........ Merged revisions 349289 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 349290 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-12-28 19:00 +0000 [r349249-349251] Kevin P. Fleming * utils, /: Tell Subversion to gnore the 'astdb2bdb' binary file if it exists. ........ Merged revisions 349250 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, res/res_fax.c, include/asterisk/dsp.h, include/asterisk/res_fax.h, res/res_fax_spandsp.c, main/dsp.c: Improve T.38 gateway V.21 preamble detection. This commit removes the V.21 preamble detection code previously added to the generic DSP implementation in Asterisk, and instead enhances the res_fax module to be able to utilize V.21 preamble detection functionality made available by FAX technology modules. This commit also adds such support to res_fax_spandsp, which uses the Spandsp modem tone detection code to do the V.21 preamble detection. There should be no functional change here, other than much more reliable V.21 preamble detection (and thus T.38 gateway initiation). ........ Merged revisions 349248 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-12-27 20:55 +0000 [r349196] Matthew Jordan * /, res/res_timing_pthread.c, include/asterisk/module.h, res/res_timing_dahdi.c, res/res_timing_timerfd.c, res/res_musiconhold.c: Fix timing source dependency issues with MOH Prior to this patch, res_musiconhold existed at the same module priority level as the timing sources that it depends on. This would cause a problem when music on hold was reloaded, as the timing source could be changed after res_musiconhold was processed. This patch adds a new module priority level, AST_MODPRI_TIMING, that the various timing modules are now loaded at. This now occurs before loading other resource modules, such that the timing source is guaranteed to be set prior to resolving the timing source dependencies. (closes issue ASTERISK-17474) Reporter: Luke H Tested by: Luke H, Vladimir Mikhelson, zzsurf, Wes Van Tlghem, elguero, Thomas Arimont Patches: asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-1.8.diff uploaded by elguero (License #5026) asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-10.diff uploaded by elguero (License #5026) asterisk-17474-dahdi_timing-infinite-wait-fix_v3.diff uploaded by elguero (License #5026) Review: https://reviewboard.asterisk.org/r/1578/ ........ Merged revisions 349194 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 349195 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-12-27 17:17 +0000 [r349146] Sean Bright * /, main/audiohook.c: Once an audiohook is attached to a channel, we continue to transcode all of the frames, even after all of the hooks are detached. This patch short-cicuits us out before we transcode unnecessarily. ........ Merged revisions 349144 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 349145 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-12-23 21:19 +0000 [r349106] Matthew Jordan * contrib/realtime/mysql/voicemail.sql, configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c: Allow overriding of IMAP server settings on a user by user basis This patch allows the imapserver, imapport, and imapflags settings to be overridden for any voicemail user. It also documents the settings in the sample voicemail.conf file, and updates the voicemail schema to allow storage of those columns. (closes issue ASTERISK-16489) Reporter: Hubert Mickael Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1614/ 2011-12-23 20:42 +0000 [r349097-349098] Jonathan Rose * channels/chan_sip.c, main/features.c, configs/sip.conf.sample, channels/sip/include/sip.h: INFO/Record request configurable to use dynamic features Adds two new options to SIP peers allowing them to specify features (dynamic or builtin) to use when sending INFO/record requests. Recordonfeature activates whatever feature is specified when recieving a record: on request while recordofffeature activates whatever feature is specified when receiving a record: off request. Both of these features can be disabled by setting the feature to an empty string. (closes issue ASTERISK-16507) Reported by: Jon Bright Review: https://reviewboard.asterisk.org/r/1634/ * channels/chan_sip.c, configs/sip.conf.sample, CHANGES, channels/sip/include/sip.h: chan_sip autocreatepeer=persist option for auto-created peers to survive reload This patch moves destruction of sip peers to immediately after the general section of sip.conf is read so that autocreatepeer setting can be read before deletion of peers. If autocreatepeer=persist at reload, then peers created by the autocreatepeer setting will be skipped when purging the current SIP peer list. (closes ASTERISK-16508) Reported by: Kirill Katsnelson Patches: 017797-kkm-persist-autopeers-1.8.patch uploaded by Kirill Katsnelson (license 5845) 2011-12-23 17:36 +0000 [r349046] Sean Bright * /, apps/app_chanspy.c: Merged revisions 349045 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r349045 | seanbright | 2011-12-23 12:32:33 -0500 (Fri, 23 Dec 2011) | 25 lines Merged revisions 349044 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r349044 | seanbright | 2011-12-23 12:25:01 -0500 (Fri, 23 Dec 2011) | 18 lines In ChanSpy, don't create audiohooks that will never be used. When ChanSpy is initialized it creates and attaches 3 audiohooks: 1) Read audio off of the channel that we are spying on 2) Write audio to the channel that we are spying on 3) Write audio to the channel that is bridged to the channel that we are spying on. The first is always necessary, but the others are used only when specific options are passed to the ChanSpy application (B, d, w, and W to be specific). When those flags are not passed, neither of those audiohooks are ever sent frames, but we still try to process the hooks for each voice frame that we recieve on the channel. So in short - only create and attach audiohooks that we actually need. ........ ................ 2011-12-23 15:26 +0000 [r348994] Kinsey Moore * apps/app_dial.c, /: Fix missing doc tags found while fixing ASTERISK-18689 Add missing tags in app_dial documentation. ........ Merged revisions 348992 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 348993 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-12-23 02:35 +0000 [r348953] Richard Mudgett * main/pbx.c, /, channels/chan_sip.c, include/asterisk/pbx.h: Fix extension state callback references in chan_sip. Chan_sip gives a dialog reference to the extension state callback and assumes that when ast_extension_state_del() returns, the callback cannot happen anymore. Chan_sip then reduces the dialog reference count associated with the callback. Recent changes (ASTERISK-17760) have resulted in the potential for the callback to happen after ast_extension_state_del() has returned. For chan_sip, this could be very bad because the dialog pointer could have already been destroyed. * Added ast_extension_state_add_destroy() so chan_sip can account for the sip_pvt reference given to the extension state callback when the extension state callback is deleted. * Fix pbx.c awkward statecbs handling in ast_extension_state_add_destroy() and handle_statechange() now that the struct ast_state_cb has a destructor to call. * Ensure that ast_extension_state_add_destroy() will never return -1 or 0 for a successful registration. * Fixed pbx.c statecbs_cmp() to compare the correct information. The passed in value to compare is a change_cb function pointer not an object pointer. * Make pbx.c ast_merge_contexts_and_delete() not perform callbacks with AST_EXTENSION_REMOVED with locks held. Chan_sip is notorious for deadlocking when those locks are held during the callback. * Removed unused lock declaration for the pbx.c store_hints list. (closes issue ASTERISK-18844) Reported by: rmudgett Review: https://reviewboard.asterisk.org/r/1635/ ........ Merged revisions 348940 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 348952 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-12-22 22:39 +0000 [r348890] Matthew Jordan * cel/cel_pgsql.c, /: Fix for memory leaks / cleanup in cel_pgsql There were a number of issues in cel_pgsql's pgsql_log method: * If either sql or sql2 could not be allocated, the method would return while the pgsql_lock was still locked * If the execution of the log statement succeeded, the sql and sql2 structs were never free'd * Reconnection successes were logged as ERRORs. In general, the severity of several logging statements was reduced (closes issue ASTERISK-18879) Reported by: Niolas Bouliane Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1624/ ........ Merged revisions 348888 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 348889 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-12-22 21:12 +0000 [r348849] Damien Wedhorn * channels/chan_skinny.c: Fix segfault on answer. Only update/change RTP source if RTP has already been started and connected to the subchannel. 2011-12-22 20:44 +0000 [r348848] Matthew Jordan * /, main/say.c, main/file.c, main/app.c, apps/app_confbridge.c, main/bridging.c: Add Asterisk TestSuite event hooks to support ConfBridge testing This patch adds initial testsuite event hooks so that ConfBridge tests can be executed in the Asterisk TestSuite. (issue ASTERISK-19059) ........ Merged revisions 348846 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-12-22 20:39 +0000 [r348847] Terry Wilson * /, include/asterisk/format_pref.h: Allow packetization vaules > 127 According to the RTP packetization documentation, and the maximum values listed in AST_FORMAT_LIST, we should support values > that the signed char array that ast_codec_pref makes available to store the value. All places in the code treat the framing field as though it were an int array instaead of a char array anyway, so this just fixes the type of the array. (closes issue ASTERISK-18876) Review: https://reviewboard.asterisk.org/r/1639/ ........ Merged revisions 348833 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 348845 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-12-21 20:13 +0000 [r348737-348794] Richard Mudgett * /, codecs/speex: Make codecs/speex ignore *.i files also. ........ Merged revisions 348793 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, apps/confbridge: Make apps/confbridge ignore *.i files also. ........ Merged revisions 348790 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, channels/chan_iax2.c: Fix chan_iax2 to not report an RDNIS number if it is blank. Some ISDN switches complain or block the call if the RDNIS number is empty. * Made chan_iax2 not save a RDNIS number into the ast_channel if the string is blank. This is what other channel drivers do. (closes issue ASTERISK-17152) Reported by: rmudgett ........ Merged revisions 348735 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 348736 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-12-20 20:06 +0000 [r348698] Matthew Nicholson * contrib/scripts/safe_asterisk: This adds support for setting several safe_asterisk parameters using environment variables and also enables a custom run directory for asterisk (instead of defaulting to /tmp). Patch by: Byron Clark (byronclark) (closes ASTERISK-17810) 2011-12-19 21:43 +0000 [r348649] Richard Mudgett * /, configure, configure.ac: Fix crashes on other platforms caused by interference from Darwin weak symbol support. Support weak symbols on a platform specific basis. The Mac OS X (Darwin) support must be isolated from the other platforms because it has caused other platforms to crash. Several other platforms including Linux have GCC versions that define the weak attribute. However, this attribute is only setup for use in the code by Darwin. (closes issue ASTERISK-18728) Reported by: Ben Klang Review: https://reviewboard.asterisk.org/r/1617/ ........ Merged revisions 348647 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 348648 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-12-19 19:55 +0000 [r348606] Leif Madsen * /, main/message.c: Update documentation for MESSAGE_SEND_STATUS variable. (Closes issue ASTERISK-19056) Reported by: Yuri Patches: 348360.diff uploaded by Yuri (license #5242) ........ Merged revisions 348605 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-12-19 01:36 +0000 [r348567] Terry Wilson * /, res/res_srtp.c: Add a separate buffer for SRTCP packets The function ast_srtp_protect used a common buffer for both SRTP and SRTCP packets. Since this function can be called from multiple threads for the same SRTP session (scheduler for SRTCP and channel for SRTP) it was possible for the packets to become corrupted as the buffer was used by both threads simultaneously. This patch adds a separate buffer for SRTCP packets to avoid the problem. (closes issue ASTERISK-18889, Reported/patch by Daniel Collins) ........ Merged revisions 347995 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 347996 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-12-18 18:29 +0000 [r348518] Kevin P. Fleming * /, configs/sip.conf.sample: Correct two flaws in sip.conf.sample related to AST-2011-013. * The sample file listed *two* values for the 'nat' option as being the default. Only 'force_rport' is the default. * The warning about having differing 'nat' settings confusingly referred to both peers and users. ........ Merged revisions 348515 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........ Merged revisions 348516 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 348517 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-12-16 23:58 +0000 [r348466] Richard Mudgett * main/channel.c, /, main/features.c: Clean-up on isle five for __ast_request_and_dial() and ast_call_forward(). * Add locking when a channel inherits variables and datastores in __ast_request_and_dial() and ast_call_forward(). Note: The involved channels are not active so there was minimal potential for problems. * Remove calls to ast_set_callerid() in __ast_request_and_dial() and ast_call_forward() because the set information is for the wrong direction. * Don't use C++ keywords for variable names in ast_call_forward(). * Run the redirecting interception macro if defined when forwarding a call in ast_call_forward(). Note: Currently will never execute because the only callers that supply a calling channel supply a hungup or zombie channel. * Make feature_request_and_dial() put the transferee into autoservice when it calls ast_call_forward() in case a redirection interception macro is run. Note: Currently will never happen because the caller channel (Party B) is always hungup at this time. * Make feature_request_and_dial() ignore the AST_CONTROL_PROCEEDING frame to silence a log message. ........ Merged revisions 348464 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 348465 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-12-16 22:00 +0000 [r348416] Jonathan Rose * configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c: Voicemail with the saycid option will now play a caller's name based on cid if available. In order to check the availability of the caller's name, app_voicemail will check for an audio file in /recordings/callerids/ This change sets a precedent for where to put recordings of names. Currently the idea is that recordings here could also be used for applications like confbridge and meetme to find recorded names in this folder from callerid (when another recording isn't available) (closes issue ASTERISK-18565) Reporter: Russell Brown Patches: r uploaded by Russel Brown (license 6182) 2011-12-16 21:30 +0000 [r348312-348408] Richard Mudgett * main/channel.c, /: Fix cut and past error in ast_call_forward(). (issue ASTERISK-18836) ........ Merged revisions 348401 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 348405 from http://svn.asterisk.org/svn/asterisk/branches/10 * main/channel.c, main/pbx.c, /, apps/app_authenticate.c, funcs/func_cdr.c, main/features.c, include/asterisk/cdr.h, apps/app_followme.c, apps/app_queue.c, res/res_monitor.c: Fix crash during CDR update. The ast_cdr_setcid() and ast_cdr_update() were shown in ASTERISK-18836 to be called by different threads for the same channel. The channel driver thread and the PBX thread running dialplan. * Add lock protection around CDR API calls that access an ast_channel pointer. (closes issue ASTERISK-18836) Reported by: gpluser Review: https://reviewboard.asterisk.org/r/1628/ ........ Merged revisions 348362 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 348363 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, apps/app_parkandannounce.c: Fix ParkAndAnnounce to pass the CallerID to the announcing channel. ParkAndAnnounce tried to pass the CallerID to the announcing channel but the ID was wiped out by the channel masquerade done when parking the call. * Save the CallerID before parking the channel to pass it to the announcing channel. * Fixed a minor memory leak in ParkAndAnnounce. * Updated some ParkAndAnnounce log messages. ........ Merged revisions 348310 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 348311 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-12-14 22:36 +0000 [r348215-348266] Matthew Jordan * /, apps/app_originate.c: Added support for all slin formats to app_originate Previously, app_originate could not originate a call into a non-8kHz conference bridge as the formats for non-8kHz slin codecs were not applied to the created channel. This patch adds all of the formats by default, such that if a created channel has a codec that supports a higher sampling rate, a translation path can be built between it and other channels. ........ Merged revisions 348265 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, apps/app_queue.c: Fixed Asterisk crash when function QUEUE_MEMBER receives invalid input The function QUEUE_MEMBER has two required parameters (queuename, option). It was only checking for the presence of queuename. The patch checks for the existence of the option parameter and provides better error logging when invalid values are provided for the option parameter as well. ........ Merged revisions 348211 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-12-14 22:05 +0000 [r348214] Matthew Nicholson * /, res/res_fax.c: Don't clear LOCALSTATIONID before sending or receiving. The user may set that variable. ASTERISK-18921 ........ Merged revisions 348212 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 348213 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-12-14 21:08 +0000 [r348161] Jonathan Rose * main/features.c, configs/features.conf.sample: Add and document PARKEDCALL variable set during timeout PARKEDCALL variable tracks which parking lot the call was last parked in. This can be used afterwards for flow control when returntoorigin is set to off. I went ahead and documented both this and the existing variable set during timeout (PARKINGSLOT) in the sample features.conf since there was no prior mention of variables being set during timeout. (closes issue ASTERISK-16239) Reported By: Clod Patry Patches: M17503.diff uploaded by Clod Patry (license 5138) 2011-12-14 20:51 +0000 [r348160] Matthew Jordan * apps/app_confbridge.c: Improve error message in CONFBRIDGE_INFO Provided a more descriptive error message when a value supplied for the parameter type is not one of the acceptable values. (closes issue ASTERISK-18717) Reported by: Paul Belanger Patches: __20111103-better-confbridge_info-error-msg.txt (License #4999) 2011-12-14 20:37 +0000 [r348156-348159] Jonathan Rose * /, configs/features.conf.sample: Fix accidental use of tabs instead of spaces from previous features.conf.sample change ........ Merged revisions 348157 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 348158 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, configs/features.conf.sample: Document PARKINGSLOT variable in features.conf.sample (issue ASTERISK-16239) ........ Merged revisions 348154 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 348155 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-12-13 23:10 +0000 [r348103] Richard Mudgett * /, bridges/bridge_builtin_features.c, apps/app_followme.c: Fix FollowMe CallerID on outgoing calls. The addition of the Connected Line support changed how CallerID is passed to outgoing calls. The FollowMe application was not updated to pass CallerID to the outgoing calls. * Fix FollowMe CallerID on outgoing calls. * Restructured findmeexec() to fix several memory leaks and eliminate some duplicated code. * Made check the return value of create_followme_number(). Putting a NULL into the numbers list is bad if create_followme_number() fails. * Fixed a couple uses of ast_strdupa() inside loops. * The changes to bridge_builtin_features.c fix a similar CallerID issue with the bridging API attended and blind transfers. (Not used at this time.) (closes issue ASTERISK-17557) Reported by: hamlet505a Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1612/ ........ Merged revisions 348101 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 348102 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-12-13 15:22 +0000 [r348061] Stefan Schmidt * channels/chan_sip.c: Fix possible misshandling of an incoming SIP response as a peer poke response. Also make sure peer has even qualify enabled when handle a peer poke response. (closes issue ASTERISK-18940) Reported by: Vitaliy Tested by: Vitaliy and UnixDev Review: https://reviewboard.asterisk.org/r/1620 Reviewed by: David Vossel ........ Merged revisions 348048 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 348056 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-12-12 19:35 +0000 [r347997] Matthew Jordan * include/asterisk/logger.h, utils/refcounter.c, main/logger.c, utils/hashtest.c, UPGRADE.txt, utils/ael_main.c, utils/hashtest2.c, CHANGES, main/asterisk.c, main/config.c, configs/logger.conf.sample, main/loader.c, main/cli.c: Backed out core changes from r346391 During testing, it was discovered that there were a number of side effects introduced by r346391 and subsequent check-ins related to it (r346429, r346617, and r346655). This included the /main/stdtime/ test 'hanging', as well as the remote console option failing to receive the appropriate output after a period of time. I only backed out the changes to main/ and utils/, as this was adequate to reverse the behavior experienced. (issue ASTERISK-18974) 2011-12-12 17:34 +0000 [r347954] Richard Mudgett * configs/iax.conf.sample, configs/chan_dahdi.conf.sample, /, configs/chan_ooh323.conf.sample, configs/vpb.conf.sample, configs/extensions.lua.sample, configs/sip.conf.sample, configs/extensions.conf.sample: Update sample configs to put incoming calls into context public. * Add warning about the SIP allowguest option in context public. (closes issue ASTERISK-14122) Reported by: Alec Davis Review: https://reviewboard.asterisk.org/r/719/ ........ Merged revisions 347953 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-12-09 21:47 +0000 [r347866-347903] Jonathan Rose * apps/app_mixmonitor.c: Adds MixMonitor and StopMixMonitor AMI commands to the manager These commands work much like the dialplan applications that would otherwise invoke them. A nice benefit of these is that they can be invoked on a call remotely and at any time during a call. They work much like the Monitor and StopMonitor ami commands. (closes issue ASTERISK-17726) Reported by: Sergio González Martín Patches: mixmonitor_actions.diff uploaded by Sergio González Martín (license 5644) Review: https://reviewboard.asterisk.org/r/1193/ * include/asterisk/file.h, apps/app_sayunixtime.c, CHANGES: Remove autojump extensions from SayUnixTime, make an option to perform automatic jumps. When a caller sends DTMF while the SayUnixTime application is saying the time, The call would jump to the next extension much like it does during Background(). This patch adds option 'j' to SayUnixTime which when used employs the old behavior. Also, this patch allows arguments to sayunixtime to not be used as empty strings in the case of something like 'sayunixtime(,,,j)' or 'sayunixtime(,,pattern). (closes issue ASTERISK-16675) Reported by: jlpedrosa Patches: patch_SayUnixTime_noJump.patch uploaded by jlpedrosa (license 5959) Review: https://reviewboard.asterisk.org/r/956/ 2011-12-09 01:33 +0000 [r347813] Richard Mudgett * main/pbx.c, /: Fix some parsing issues in add_exten_to_pattern_tree(). * Simplify compare_char() and avoid potential sign extension issue. * Fix infinite loop in add_exten_to_pattern_tree() handling of character set escape handling. * Added buffer overflow checks in add_exten_to_pattern_tree() character set collection. * Made ignore empty character sets. * Added escape character handling to end-of-range character in character sets. This has a slight change in behavior if the end-of-range character is an escape character. You must now escape it. * Fix potential sign extension issue when expanding character set ranges. * Made remove duplicated characters from character sets. The duplicate characters lower extension matching priority and prevent duplicate extension detection. * Fix escape character handling when the escape character is trying to escape the end-of-string. We could have continued processing characters after the end of the exten string. We could have added the previous character to the pattern matching tree incorrectly. (closes issue ASTERISK-18909) Reported by: Luke-Jr ........ Merged revisions 347811 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 347812 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-12-08 21:32 +0000 [r347735] Walter Doekes * /, channels/chan_sip.c: Fix regression when using tcpenable=no and tlsenable=yes. The tlsenable settings are tucked away in main/tcptls.c, so I missed them when resolving ASTERISK-18837. This should resolve the test suite breakage of the sip tls tests. Review: https://reviewboard.asterisk.org/r/1615 Reviewed by: Matt Jordan ........ Merged revisions 347718 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 347727 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-12-08 20:55 +0000 [r347658] Jonathan Rose * /, apps/app_queue.c: Fix regressed behavior of queue set penalty to work without specifying 'in ' r325483 caused a regression in Asterisk 10+ that would make Asterisk segfault when attempting to set penalty on an interface without specifying a queue in the queue set penalty CLI command. In addition, no attempt would be made whatsoever to perform the penalty setting on all the queues in the core list with either the cli command or the non-segfaulting ami equivalent. This patch fixes that and also makes an attempt to document and rename some functions required by this command to better represent what they actually do. Oh yeah, and the use of this command without specifying a specific queue actually works now. Review: https://reviewboard.asterisk.org/r/1609/ ........ Merged revisions 347656 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-12-08 17:55 +0000 [r347601] Richard Mudgett * /, main/features.c: Mark channel running the h exten with the soft-hangup flag. When a bridge is broken, ast_bridge_call() might execute the h exten on the calling channel. However, that channel may not have been the channel that broke the bridge by hanging up. The channel executing the h exten must be in a hung up state so things like AGI run in the correct mode. * Make sure ast_bridge_call() marks the channel it is executing the h exten on as hung up. (The AST_SOFTHANGUP_APPUNLOAD flag is used so as to match the pbx.c main dialplan execution loop when it executes the h exten.) (closes issue ASTERISK-18811) Reported by: David Hajek Patches: jira_asterisk_18811_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: David Hajek, rmudgett ........ Merged revisions 347595 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 347600 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-12-08 16:24 +0000 [r347533] Terry Wilson * /, channels/chan_sip.c: Don't crash on INFO automon request with no channel AST-2011-014. When automon was enabled in features.conf, it was possible to crash Asterisk by sending an INFO request if no channel had been created yet. (closes issue ASTERISK-18805) ........ Merged revisions 347530 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........ Merged revisions 347531 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 347532 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-12-08 06:59 +0000 [r347490] Damien Wedhorn * channels/chan_skinny.c: Fix segfault on answer. Fix a segfault if an attempt to answer a call is made between when the inbound call gives up (and the channel is removed) and when the device is notified and removes the call from the device. 2011-12-07 21:42 +0000 [r347440] Richard Mudgett * main/manager.c, /: Update AMI Getvar and Setvar documentation about supplying a channel name. (closes issue ASTERISK-18958) Reported by: Red Patches: jira_asterisk_18958_v1.8.patch (license #5621) patch uploaded by rmudgett ........ Merged revisions 347438 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 347439 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-12-07 20:34 +0000 [r347395] Jonathan Rose * /, apps/app_meetme.c: Fix: Meetme recording variables from realtime DB use null entries over channel variables Meetme would attempt to substitute the realtime values of RECORDING_FILE and RECORDING_FORMAT from the meetme db entry instead of using the channel variable set for those variables in spite of those database entries being NULL or even lacking a column to represent them. (closes issue ASTERISK-18873) Reported by: Byron Clark Patches: ASTERISK-18873-1.patch uploaded by Byron Clark (license 6157) ........ Merged revisions 347369 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 347383 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-12-07 20:15 +0000 [r347345] Terry Wilson * Makefile, include/asterisk/paths.h, /, configs/asterisk.conf.sample, build_tools/make_defaults_h, main/asterisk.c, main/db.c: Add ASTSBINDIR to the list of configurable paths This patch also makes astdb2sqlite3 and astcanary use the configured directory instead of relying on $PATH. (closes issue ASTERISK-18959) Review: https://reviewboard.asterisk.org/r/1613/ ........ Merged revisions 347344 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-12-06 23:58 +0000 [r347294] Richard Mudgett * /, channels/chan_sip.c: Make SIP INFO messages for dtmf-relay signals case insensitive. (closes issue ASTERISK-18924) Reported by: Kevin Taylor ........ Merged revisions 347292 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 347293 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-12-06 22:01 +0000 [r347241] Jonathan Rose * main/pbx.c, /: Documents CHANNEL(musicclass) taking priority over m([x]) in waitExten If waitExten specifies a music class to use with its music on hold option, it will use CHANNEL(musicclass) instead if that channel variable has been set on the initiating channel. This documents that behavior in the waitExten app so that this can be known without checking the documentation of the code in function local_ast_moh_start. (closes issue ASTERISK-18804) ........ Merged revisions 347239 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 347240 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-12-06 20:23 +0000 [r347157-347192] Walter Doekes * UPGRADE.txt, CHANGES, apps/app_voicemail.c: Add VM_INFO() dialplan function to gather information about a mailbox. Deprecates MAILBOX_EXISTS. Provides count, email, exists, fullname, language, locale, pager, password, tz. (closes issue ASTERISK-18634) Patch by: Kris Shaw Review: https://reviewboard.asterisk.org/r/1568 Reviewed by: Walter Doekes * /, channels/chan_sip.c: Don't allow transport=tcp when tcpenable=no. When tcpenable=no, sending to transport=tcp hosts was still allowed. Resolving the source address wasn't possible and yielded the string "(null)" in SIP messages. Fixed that and a couple of not-so-correct log messages. (closes issue ASTERISK-18837) Reported by: Andreas Topp Review: https://reviewboard.asterisk.org/r/1585 Reviewed by: Matt Jordan ........ Merged revisions 347166 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 347167 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, apps/app_voicemail.c: Add regression tests for issue ASTERISK-18838. Review: https://reviewboard.asterisk.org/r/1572 Reviewed by: Matt Jordan ........ Merged revisions 347131 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 347146 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, apps/app_voicemail.c: The voicemail [general] zonetag and locale variables weren't loaded until after the mailboxes were initialized. This caused the settings to be unset for those mailboxes until a reload was performed. (closes issue ASTERISK-18838) Review: https://reviewboard.asterisk.org/r/1570 Reviewed by: Matt Jordan ........ Merged revisions 347111 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 347124 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-12-06 19:09 +0000 [r347110] Richard Mudgett * include/asterisk/dlinkedlists.h, tests/test_linkedlists.c: Doubly linked lists unit test and update to implementation. Update the doubly linked list implementation. Now safe traversing can insert before and after the current node when traversing in either direction. Updated the linked lists unit test test_linkedlist to also test doubly linked lists. The old test_dlinkedlist requires a manual check of results and probably should be removed. Review: https://reviewboard.asterisk.org/r/1569/ 2011-12-06 17:34 +0000 [r347069] Matthew Jordan * /, channels/chan_sip.c: Fixed crash from orphaned MWI subscriptions in chan_sip This patch resolves the issue where MWI subscriptions are orphaned by subsequent SIP SUBSCRIBE messages. When a peer is removed, either by pruning realtime SIP peers or by unloading / loading chan_sip, the MWI subscriptions that were orphaned would still be on the event engine list of valid subscriptions but have a pointer to a peer that no longer was valid. When an MWI event would occur, this would cause a seg fault. (closes issue ASTERISK-18663) Reported by: Ross Beer Tested by: Ross Beer, Matt Jordan Patches: blf_mwi_diff_12_06_11.txt uploaded by Matt Jordan (license 6283) Review: https://reviewboard.asterisk.org/r/1610/ ........ Merged revisions 347058 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 347068 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-12-05 17:44 +0000 [r347008] Richard Mudgett * channels/chan_dahdi.c, channels/sig_analog.c, /, channels/sig_analog.h: Restore call progress code for analog ports. Extracting sig_analog from chan_dahdi lost call progress detection functionality. * Fix analog ports from considering a call answered immediately after dialing has completed if the callprogress option is enabled. (closes issue ASTERISK-18841) Reported by: Richard Miller Patches: chan_dahdi.diff (license #5685) patch uploaded by Richard Miller (Modified by me) sig_analog.c.diff (license #5685) patch uploaded by Richard Miller (Modified by me) sig_analog.h.diff (license #5685) patch uploaded by Richard Miller ........ Merged revisions 347006 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 347007 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-12-05 15:04 +0000 [r346956] Jonathan Rose * main/pbx.c, /: Resolve duplicate label used in multiple priorities for the same extension. Prior to this patch, if labels with the same name were used for different priorities in the same extension, the new label would be accepted, but it would be unusable since attempts to reach that label would just go to the first one. Now pbx.c detects this, generates a warning in logs, and culls the label before adding it to the dialplan. (closes issue ASTERISK-18807) Reported by: Kenneth Shumard Patches: pbx.c.patch uploaded by Kenneth Shumard (License 5077) ........ Merged revisions 346954 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346955 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-12-05 14:47 +0000 [r346953] Kinsey Moore * res/res_jabber.exports.in, /: Fix chan_jingle/gtalk load regression introduced in r346087 Add missing symbol exports for ast_aji_client_destroy and ast_aji_buddy_destroy for usage outside res_jabber. Testing of these changes focused on res_jabber itself, so this problem was missed. Reported-by: Michael Spiceland ........ Merged revisions 346951 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346952 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-12-04 10:08 +0000 [r346901] Walter Doekes * /, channels/chan_sip.c: For SIP REGISTER fix domain-only URIs and domain ACL bypass. The code that allowed admins to create users with domain-only uri's had stopped to work in 1.8 because of the reqresp parser rewrites. This is fixed now: if you have a [mydomain.com] sip user, you can register with useraddr sip:mydomain.com. Note that in that case -- if you're using domain ACLs (a configured domain list) -- mydomain.com must be in the allow list as well. Reviewboard r1606 shows a list of registration combinations and which SIP response codes are returned. Review: https://reviewboard.asterisk.org/r/1533/ Reviewed by: Terry Wilson (closes issue ASTERISK-18389) (closes issue ASTERISK-18741) ........ Merged revisions 346899 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346900 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-12-02 23:30 +0000 [r346857] Matthew Jordan * /, channels/chan_sip.c: Update SIP MESSAGE To parsing to correctly handle URI The previous patch (r346040) incorrectly parsed the URI in the presence of a port, e.g., user@hostname:port would fail as the port would be double appended to the SIP message. This patch uses the parse_uri function to correctly parse the URI into its username and hostname parts, and places them in the correct fields in the sip_pvt structure. (issue ASTERISK-18903) Review: https://reviewboard.asterisk.org/r/1597/ ........ Merged revisions 346856 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-12-02 19:40 +0000 [r346777-346816] Alexandr Anikin * addons/chan_ooh323.c: implement nat option for rtp channels with ooh323 * addons/chan_ooh323.c, /, channels/chan_h323.c: Merged revisions 346763 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r346763 | may | 2011-12-02 20:42:32 +0400 (Fri, 02 Dec 2011) | 14 lines Merged revisions 346762 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r346762 | may | 2011-12-02 20:19:19 +0400 (Fri, 02 Dec 2011) | 7 lines process null frame pointer returned by ast_rtp_instance_read correctly (closes issue ASTERISK-16697) Reported by: under Patches: segfault.diff (License #5871) patch uploaded by under ........ ................ 2011-12-01 21:19 +0000 [r346709] Richard Mudgett * main/stun.c, /, res/res_stun_monitor.c, configs/res_stun_monitor.conf.sample, include/asterisk/stun.h: Re-resolve the STUN address if a STUN poll fails for res_stun_monitor. The STUN socket must remain open between polls or the external address seen by the STUN server is likely to change. However, if the STUN request poll fails then the STUN server address needs to be re-resolved and the STUN socket needs to be closed and reopened. * Re-resolve the STUN server address and create a new socket if the STUN request poll fails. * Fix ast_stun_request() return value consistency. * Fix ast_stun_request() to check the received packet for expected message type and transaction ID. * Fix ast_stun_request() to read packets until timeout or an associated response packet is found. The stun_purge_socket() hack is no longer required. * Reduce ast_stun_request() error messages to debug output. * No longer pass in the destination address to ast_stun_request() if the socket is already bound or connected to the destination. (closes issue ASTERISK-18327) Reported by: Wolfram Joost Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1595/ ........ Merged revisions 346700 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346701 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-12-01 20:46 +0000 [r346699] Jonathan Rose * /, channels/chan_sip.c: Change 183 Ringing in sipfrag body to 180 ringing. 183 Ringing isn't even a thing. 183 is actually a session progress message. (closes issue ASTERISK-18925) Reported by: Sebastian Denz Tested by: jrose Patches: asterisk18-use_180_instead_of_183_in_sipfrag.diff by Sebastian Denz (License #6139) ........ Merged revisions 346697 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346698 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-30 23:38 +0000 [r346617-346655] Tilghman Lesher * channels/chan_unistim.c, main/tcptls.c, channels/chan_sip.c, main/config.c, main/loader.c: Remove the few places where we try to ast_verbose() without a newline. * main/asterisk.c: Fix edge case for overflow buffer. 2011-11-30 22:03 +0000 [r346525-346566] Jonathan Rose * main/tcptls.c, /, channels/chan_sip.c, include/asterisk/tcptls.h: r346525 | jrose | 2011-11-30 15:10:38 -0600 (Wed, 30 Nov 2011) | 18 lines Cleaning up chan_sip/tcptls file descriptor closing. This patch attempts to eliminate various possible instances of undefined behavior caused by invoking close/fclose in situations where fclose may have already been issued on a tcptls_session_instance and/or closing file descriptors that don't have a valid index for fd (-1). Thanks for more than a little help from wdoekes. (closes issue ASTERISK-18700) Reported by: Erik Wallin (issue ASTERISK-18345) Reported by: Stephane Cazelas (issue ASTERISK-18342) Reported by: Stephane Chazelas Review: https://reviewboard.asterisk.org/r/1576/ ........ Merged revisions 346564 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346565 from http://svn.asterisk.org/svn/asterisk/branches/10 * main/tcptls.c, channels/chan_sip.c, include/asterisk/tcptls.h: Reverting 346525 due to accidental patch against trunk instead of 1.8 * main/tcptls.c, channels/chan_sip.c, include/asterisk/tcptls.h: Cleaning up chan_sip/tcptls file descriptor closing. This patch attempts to eliminate various possible instances of undefined behavior caused by invoking close/fclose in situations where fclose may have already been issued on a tcptls_session_instance and/or closing file descriptors that don't have a valid index for fd (-1). Thanks for more than a little help from wdoekes. (closes issue ASTERISK-18700) Reported by: Erik Wallin (issue ASTERISK-18345) Reported by: Stephane Cazelas (issue ASTERISK-18342) Reported by: Stephane Chazelas Review: https://reviewboard.asterisk.org/r/1576/ 2011-11-30 19:37 +0000 [r346474] Leif Madsen * configs/queues.conf.sample: Update queues.conf.sample documentation. Update the documentation surrounding the use of MONITOR_EXEC to make it more clear that it can be used for both Monitor() and MixMonitor() usage. (closes issue ASTERISK-17413) Reported by: David Woolley Patches: issue18817_mixmonitor_queues_doc.diff by Michael L. Young (License #5026) ........ Merged revisions 346472 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346473 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-29 20:32 +0000 [r346391-346429] Tilghman Lesher * utils/refcounter.c, utils/hashtest.c, utils/ael_main.c, utils/hashtest2.c: Fix compilation of utilities (caught by Bamboo). * addons/chan_ooh323.c, channels/chan_sip.c, main/say.c, res/res_fax.c, UPGRADE.txt, res/res_musiconhold.c, res/res_jabber.c, CHANGES, configs/logger.conf.sample, main/cli.c, channels/chan_usbradio.c, include/asterisk/logger.h, main/dial.c, channels/chan_skinny.c, main/logger.c, codecs/codec_dahdi.c, apps/app_rpt.c, apps/app_verbose.c, main/asterisk.c, main/bridging.c, res/res_clialiases.c, addons/res_config_mysql.c, apps/app_voicemail.c: Allow each logging destination and console to have its own notion of the verbosity level. Review: https://reviewboard.asterisk.org/r/1599 2011-11-29 00:03 +0000 [r346350] David Vossel * /, include/asterisk/message.h, main/message.c: Merged revisions 346349 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r346349 | dvossel | 2011-11-28 18:00:11 -0600 (Mon, 28 Nov 2011) | 10 lines Fixes memory leak in message API. The ast_msg_get_var function did not properly decrement the ref count of the var it retrieves. The way this is implemented is a bit tricky, as we must decrement the var and then return the var's value. As long as the documentation for the function is followed, this will not result in a dangling pointer as the ast_msg structure owns its own reference to the var while it exists in the var container. ........ 2011-11-28 14:34 +0000 [r346294] Stefan Schmidt * res/res_rtp_asterisk.c, /: Fix regression that 'rtp/rtcp set debup ip' only works when also a port was specified. (closes issue ASTERISK-18693) Reported by: Davide Dal Fra Review: https://reviewboard.asterisk.org/r/1600/ Reviewed by: Walter Doekes ........ Merged revisions 346292 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346293 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-23 23:03 +0000 [r346241] Richard Mudgett * include/asterisk/acl.h, /, channels/chan_skinny.c, channels/chan_h323.c, main/acl.c, channels/chan_iax2.c: Fix calls to ast_get_ip() not initializing the address family. ........ Merged revisions 346239 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346240 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-23 20:48 +0000 [r346146-346199] Walter Doekes * /, channels/chan_sip.c: Minor cleanup in chan_sip get_msg_text() function. In r116240, get_msg_text() got an extra parameter to fix the unwanted addition of trailing newlines to SIP MESSAGE bodies. This caused all linefeeds to be trimmed, which isn't right either. This is a stop-gap; the right fix is to return the original SIP request body. Review: https://reviewboard.asterisk.org/r/1586 Reviewed by: Matt Jordan ........ Merged revisions 346147 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346198 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, include/asterisk/strings.h: Fix ast_str_truncate signedness warning and documentation. Review: https://reviewboard.asterisk.org/r/1594 ........ Merged revisions 346144 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346145 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-23 17:16 +0000 [r346088] Kinsey Moore * channels/chan_jingle.c, /, include/asterisk/jabber.h, channels/chan_gtalk.c, res/res_jabber.c: Fix res_jabber resource leaks This should fix almost all resource leaks in res_jabber that involve ASTOBJ_CONTAINER_FIND and resolves an ambiguous situation where ast_aji_get_client would sometimes bump an object's refcount and sometimes not. Review: https://reviewboard.asterisk.org/r/1553 ........ Merged revisions 346086 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346087 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-23 16:23 +0000 [r346053] Matthew Jordan * /, channels/chan_sip.c: Fixed SendMessage stripping extension from To: header in SIP MESSAGE When using the MessageSend application to send a SIP MESSAGE to a non-peer, chan_sip attempted to validate the hostname or IP Address. In the process, it stripped off the extension and failed to add it back to the sip_pvt structure before transmitting. This patch adds the full URI passed in from the message core to the sip_pvt structure. (closes issue ASTERISK-18903) Reported by: Shaun Clark Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1597/ ........ Merged revisions 346040 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-23 16:12 +0000 [r346033] Terry Wilson * /, res/res_musiconhold.c: Resume playing existing hold music for cached realtime MOH As a result of the fix for ASTERISK-18039, realtime caching MOH no longer properly resumes playing back a file between different holds in the same call. This is because scanning for new files causes the existing file array to be emptied and we were just comparing that the saved pointer to the filename matched the pointer to the filename in a particular position in the array. An easy fix is to save the filename instead of a pointer to it and then do a strcmp instead of comparing the addresses. (closes issue ASTERISK-18912) Review: https://reviewboard.asterisk.org/r/1596/ ........ Merged revisions 346030 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346031 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-23 16:10 +0000 [r346032] Paul Belanger * /, res/res_format_attr_silk.c, res/res_format_attr_celt.c: Added support level for new modules ........ Merged revisions 346029 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-22 23:06 +0000 [r345978] Richard Mudgett * main/dnsmgr.c, /, include/asterisk/dnsmgr.h: Fix dnsmgr entries to ask for the same address family each time. The dnsmgr refresh would always get the first address found regardless of the original address family requested. So if you asked for only IPv4 addresses originally, you might get an IPv6 address on refresh. * Saved the original address family requested by ast_dnsmgr_lookup() to be used when the address is refreshed. ........ Merged revisions 345976 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345977 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-22 20:32 +0000 [r345925] Walter Doekes * include/asterisk/logger.h, /: Clarify why the AST_LOG_* macros exist next to the LOG_* macros. (issue ASTERISK-17973) ........ Merged revisions 345923 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345924 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-22 16:41 +0000 [r345883] Paul Belanger * /, apps/confbridge/conf_config_parser.c: Add missing sound_only_one config variable (closes issue ASTERISK-18895) Reported by: zvision Patches: conf_config_parser.diff (license #5755) patch uploaded by zvision ........ Merged revisions 345882 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-21 21:09 +0000 [r345831] Terry Wilson * /, channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Default to nat=yes; warn when nat in general and peer differ It is possible to enumerate SIP usernames when the general and user/peer nat settings differ in whether to respond to the port a request is sent from or the port listed for responses in the Via header. In 1.4 and 1.6.2, this would mean if one setting was nat=yes or nat=route and the other was either nat=no or nat=never. In 1.8 and 10, this would mean when one was nat=force_rport and the other was nat=no. In order to address this problem, it was decided to switch the default behavior to nat=yes/force_rport as it is the most commonly used option and to strongly discourage setting nat per-peer/user when at all possible. For more discussion of the issue, please see: http://lists.digium.com/pipermail/asterisk-dev/2011-November/052191.html (closes issue ASTERISK-18862) Review: https://reviewboard.asterisk.org/r/1591/ ........ Merged revisions 345776 from http://svn.asterisk.org/svn/asterisk/branches/1.4 ........ Merged revisions 345800 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........ Merged revisions 345828 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345830 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-21 16:40 +0000 [r345735] Paul Belanger * CHANGES, main/config.c: Add #tryinclude statement This provides the same functionality as #include however an asterisk module will still load if the filename does not exist. Review: https://reviewboard.asterisk.org/r/1476/ 2011-11-19 15:11 +0000 [r345643-345684] Tilghman Lesher * /, main/db.c: Update the documentation to better clarify how the existing commands work. Review: https://reviewboard.asterisk.org/r/1593/ ........ Merged revisions 345682 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345683 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, main/db.c: Fix a change in behavior in 'database show' from 1.8. In 1.8 and previous versions, one could use any fullword portion of the key name, including the full key, to obtain the record. Until this patch, this did not work for the full key. Closes issue ASTERISK-18886 Patch: by tilghman Review: by twilson (http://pastebin.com/7rtu6bpk) on #asterisk-dev ........ Merged revisions 345640 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-17 19:47 +0000 [r345560-345601] Matthew Jordan * contrib/realtime/mysql/sipfriends.sql (removed): Accidentally readded sipfriends.sql in r345560. This was removed in r342871 * configs/confbridge.conf.sample, apps/confbridge/include/confbridge.h, apps/app_confbridge.c, CHANGES, contrib/realtime/mysql/sipfriends.sql (added), apps/confbridge/conf_config_parser.c: Add admin toggle mute all and participant count menu options to app_confbridge This patch adds two new menu features to app_confbridge, admin_toggle_menu_ participants and participant_count. The admin action will globally mute / unmute all non-admin participants on a converence, while the participant count simply exposes the existing participant count function to the conference bridge menu. This also adds configuration options to change the sound played when the conference is globally muted / unmuted, as well as the necessary config hooks to place these functions in the DTMF menus. (closes issue ASTERISK-18204) Reported by: Kevin Reeves Tested by: Matt Jordan Patches: app_confbridge.c.patch.txt, conf_config_parser.c.patch.txt, confbridge.h.patch.txt uploaded by Kevin Reeves (license 6281) Review: https://reviewboard.asterisk.org/r/1518/ 2011-11-17 17:31 +0000 [r345559] Richard Mudgett * /, channels/sig_pri.c: Remove dead code since pri_grab() can never fail. Dead code makes programmers sick. I am sick of looking at it. ........ Merged revisions 345546 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345558 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-16 14:56 +0000 [r345489] Jonathan Rose * /, apps/app_voicemail.c: Guarantee messages go into the right folders with multiple recipients Before, using the U flag in Voicemail with multiple recipients would put urgent messages in the INBOX folder for all users past the first thanks to a bug with the message copying function. This would also cause messages to fail to be sent if the INBOX directory hadn't been created for that mailbox yet. (closes issue ASTERISK-18245) Reported by: Matt Jordan (closes issue ASTERISK-18246) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1589/ ........ Merged revisions 345487 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345488 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-15 20:11 +0000 [r345221-345433] Richard Mudgett * /, res/res_agi.c: Make FastAGI HANGUP show up in AGI debug output. * Change from using send() to ast_agi_send() so the HANGUP shows up in the AGI debug output. (closes issue ASTERISK-18723) Reported by: James Van Vleet Patches: jira_asterisk_18723_v1.8.patch (license #5621) patch uploaded by rmudgett ........ Merged revisions 345431 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345432 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, channels/sig_pri.c: Fix typo in sig_pri using wrong structure name. It is fortunate that the typo does not alter generated code since the e->restart.channel and e->ring.channel members are in the same position. (closes issue ASTERISK-18868) Reported by: zvision Patches: sig_pri.c.diff (License #5755) patch uploaded by zvision ........ Merged revisions 345370 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345371 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, apps/app_queue.c: Make queue log indicate if ADDMEMBER is paused for AMI and realtime. * Add parameter to queue log ADDMEMBER to indicate if the member is paused. (closes issue ASTERISK-18645) Reported by: garlew Patches: paused.diff (License #5337) patch uploaded by garlew Tested by: rmudgett, garlew Review: https://reviewboard.asterisk.org/r/1469/ ........ Merged revisions 345285 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345290 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, channels/chan_sip.c, configs/sip.conf.sample, UPGRADE-1.8.txt, channels/sip/include/sip.h: Restore SIP DTMF overlap dialing method. The recent fix for ASTERISK-17288 to get RFC3578 SIP overlap support working correctly removed a long standing ability to do overlap dialing using DTMF in the early media phase of a call. See ASTERISK-18702 it has a very good description of the issue. I started with Pavel Troller's chan_sip.diff patch on issue ASTERISK-18702. * Added 'dtmf' enum value to sip.conf allowoverlap config option. The new option value causes the Incomplte application to not send anything with chan_sip so the caller can supply more digits via DTMF. * Renames SIP_GET_DEST_PICKUP_EXTEN_FOUND to SIP_GET_DEST_EXTEN_MATCHMORE since that is what it really means. * Fixed get_destination() inconsistency with the pickup extension matching. * Fixed initialization of PAGE3 of global_flags in reload_config(). (closes issue ASTERISK-18702) Reported by: Pavel Troller Review: https://reviewboard.asterisk.org/r/1517/ Review: https://reviewboard.asterisk.org/r/1582/ ........ Merged revisions 345273 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345275 from http://svn.asterisk.org/svn/asterisk/branches/10 * main/pbx.c, /: Fix Progress spelling error in main/pbx.c. (closes issue ASTERISK-18857) Reported by: David M Patches: mainpbx-trivial.patch (License #6326) patch uploaded by David M ........ Merged revisions 345219 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345220 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-14 19:12 +0000 [r345165] Terry Wilson * main/channel.c, /: Don't read past end of input when calling write() int blah = 1; ... write(chan->alertpipe[1], &blah, new_frames * sizeof(blah)) != (new_frames * sizeof(blah))) is only valid when new_frames == 1. Otherwise we start reading into adjacent variables declared on the stack. The read end discards what is read, so the values don't matter but it's not a good idea to read past where we want even though new_frames is almost always 1 and should never be large. This patch is basically taken out of kpfleming's eventfd branch, as he mentioned that he remembered fixing it there when I talked to him about this issue. Review: https://reviewboard.asterisk.org/r/1583/ ........ Merged revisions 345163 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345164 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-14 19:03 +0000 [r345162] Walter Doekes * /, channels/sip/include/reqresp_parser.h: Update reqresp_parser parse_uri doxygen comments. The issue mentioned in the bug report had been fixed recently by twilson. The reporter included this documentation fix. (closes issue ASTERISK-18572) Reported by: Richard Miller Patch by: Richard Miller (modified) ........ Merged revisions 345160 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345161 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-14 16:21 +0000 [r345120] Jonathan Rose * /, apps/app_voicemail.c: Moves voicemail setup password entry to the end of the setup process. This change was made because forcegreeting and forcename settings in voicemail could be circumvented by hanging up after entering a password, because the only way voicemail currently observes whether a mailbox is new or not is by checking to see if the password is the same as the mailbox number or not. (closes issue ASTERISK-18282) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1581/ ........ Merged revisions 345062 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345117 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-14 15:11 +0000 [r345065] Kinsey Moore * /, channels/chan_sip.c: Ensure that a null vmexten does not cause a segfault When sip_send_mwi_to_peer was modified recently to avoid deadlocks, vmexten was not expected to be null. This change handles that situation to avoid a segfault. ........ Merged revisions 345063 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345064 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-14 01:25 +0000 [r345023] TransNexus OSP Development * apps/app_osplookup.c: Increased max number of destinations. 2011-11-12 16:32 +0000 [r344979] Gregory Nietsky * channels/chan_misdn.c, /: mISDN Round Robin break when no channel is available Prevent channels been parsed repetitively. ........ Merged revisions 344965 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344966 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-12 00:36 +0000 [r344901] Terry Wilson * /, res/res_musiconhold.c: Don't forget to rescan MOH files for cached realtime classes Realtime MOH class caching was implemented because without it, you would build a completely new MOH class and would start the music over at the beginning each time hold was pressed in a conversation. Unfortunately, this broke re-scanning for file changes for realtime MOH classes. This patch corrects that issue. (closes issue ASTERISK-18039) Review: https://reviewboard.asterisk.org/r/1579/ ........ Merged revisions 344899 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344900 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-11 22:00 +0000 [r344846] Walter Doekes * include/asterisk/utils.h, /, main/utils.c, include/asterisk/stringfields.h: Use __alignof__ instead of sizeof for stringfield length storage. Kevin P Fleming suggested that r343157 should use __alignof__ instead of sizeof. For most systems this won't be an issue, but better fix it now while it's still fresh. Review: https://reviewboard.asterisk.org/r/1573 ........ Merged revisions 344843 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344845 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-11 21:57 +0000 [r344844] Matthew Jordan * /, main/file.c: Video format was treated as audio when removed from the file playback scheduler This patch fixes the format type check in ast_closestream and filestream_destructor. Previously a comparison operator was used, but since audio formats are no longer contiguous (and AST_FORMAT_AUDIO_MASK includes formats that have a value greater than the video formats), a bitwise AND operation is used instead. Duplicated code was also moved to filestream_close. (closes issue ASTERISK-18682) Reported by: Aldo Bedrij Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1580/ ........ Merged revisions 344823 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344842 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-11 21:37 +0000 [r344838-344840] Walter Doekes * /, channels/sip/reqresp_parser.c: Remove unneeded if(params) checks in reqresp_parser. Nick Lewis added them in https://reviewboard.asterisk.org/r/549/diff/1-2/ for no apparent reason. There is no way that params could become NULL in that piece of code, so I removed these excess checks again. ........ Merged revisions 344837 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344839 from http://svn.asterisk.org/svn/asterisk/branches/10 * main/manager.c, /: Fix bad quoting of multiline mxml opaque_data that caused invalid xml. The opaque_data was added and enclosed in single quotes, assuming it would be only a single line. The rest of the lines were appended after the closing quote. (closes issue ASTERISK-18852) Reported by: peep_ on IRC Review: https://reviewboard.asterisk.org/r/1577 ........ Merged revisions 344835 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344836 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-11 20:15 +0000 [r344771] Kinsey Moore * /, channels/chan_sip.c: Fix regression introduced by SDP fixups If capability is adjusted when switching to UDPTL during fax transmission, fax teardown fails. Make sure capability is only touched if RTP is active. This regression was introduced in R344385. ........ Merged revisions 344769 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344770 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-11 18:37 +0000 [r344663-344717] Richard Mudgett * /, channels/chan_sip.c: Check sip.conf maxforwards parameter for range 1 <= x <= 255. JIRA AST-710 ........ Merged revisions 344715 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344716 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, main/cli.c: Make CLI "core show channel" not hold the channel lock during console output. Holding the channel lock while the CLI "core show channel" command is executing can slow down the system. It could block the system if the console output is halted or paused. * Made capture the CLI "core show channel" output into a buffer to be output after the channel is unlocked. * Removed use of C++ keyword as a variable name. out renamed to obuf. * Checked allocation of obuf for failure so will not crash. (closes issue ASTERISK-18571) Reported by: Pavel Troller Tested by: rmudgett ........ Merged revisions 344661 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344662 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-11 15:47 +0000 [r344610] Jonathan Rose * main/pbx.c, /: Fix a segmentation fault when using an extension with CID matching and no CID. Attempting to call an extension which used Caller ID matching with a channel that has an empty caller id string would result in a segmentation fault. (closes issue ASTERISK-18392 Reported By: Ales Zelenik ........ Merged revisions 344608 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344609 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-10 23:21 +0000 [r344538-344560] Richard Mudgett * /, apps/app_macro.c: Fix app_macro.c MODULEINFO section termination. (closes issue ASTERISK-18848) Reported by: Tony Mountifield ........ Merged revisions 344557 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, apps/app_queue.c: Fix potential deadlock calling ast_call() with channel locks held. Fixed app_queue.c:ring_entry() calling ast_call() with the channel locks held. Chan_local attempts to do deadlock avoidance in its ast_call() callback and could deadlock if a channel lock is already held. ........ Merged revisions 344539 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344540 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, apps/app_queue.c: Make AMI event AgentCalled get CallerID/ConnectedLine info from the incoming channel. It was strange that the AgentCalled AMI event would get most of its information from the incoming channel but then get the CallerID information from the outgoing channel. Before connected line support was added, this information was always the same at this point. (closes issue ASTERISK-18152) Reported by: Thomas Farnham Tested by: rmudgett ........ Merged revisions 344536 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344537 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-10 21:56 +0000 [r344494] David Vossel * /, main/bridging.c: Merged revisions 344493 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r344493 | dvossel | 2011-11-10 15:54:42 -0600 (Thu, 10 Nov 2011) | 12 lines Fixes issue with ConfBridge participants hanging up during DTMF feature menu usage getting stuck in conference forever. When a conference user enters the DTMF menu they are suspended from the bridge while the channel is handed off to the DTMF feature code. If a user entered this state and hungup, there existed a race condition where the channel could not exit the conference because it was waiting on a signal that would never arrive. This patch fixes that, because it would stupid for me to talk about the problem and commit a patch for something else. (closes issue ASTERISK-18829) Reported by: zvision ........ 2011-11-10 21:15 +0000 [r344387-344441] Kinsey Moore * /, apps/app_meetme.c: Fix another incorrect case with meetme's PIN logic and add documentation This fixes an issue where a user of a dynamic conference was asked for a PIN twice. This also adds documentation to assist in future modifications to the piece of code responsible for PIN checking. (closes issue AST-670) ........ Merged revisions 344439 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344440 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, channels/chan_sip.c, channels/sip/include/sip.h: Fix several bugs with SDP parsing and well-formedness of responses Fix bug ASTERISK-16558 which dealt with the order of responses to incoming streams defined by SDP. Fix unreported bug where offering multiple same-type streams would cause Asterisk to reply with an incorrect SDP response missing one or more streams without a proper declination. Fix bugs related to a single non-audio stream being offered with responses requesting codecs that were not offered in the initial invite along with an additional audio stream that was not in the initial invite. Review: https://reviewboard.asterisk.org/r/1516/ ........ Merged revisions 344385 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344386 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-10 16:29 +0000 [r344335] Matthew Nicholson * res/res_rtp_asterisk.c, /: only attempt to do stun handling on ipv4 or ipv4 mapped to ipv6 addresses Patch by: jkonieczny (modified) ASTERISK-18490 ........ Merged revisions 344330 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344334 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-09 20:55 +0000 [r344272] Richard Mudgett * /, channels/chan_sip.c: Fix deadlock during dialplan reload. Another deadlock between the conlock/hints and channels/channel locking orders. * Don't hold the channel and private lock in sip_new() when calling ast_exists_extension(). (closes issue ASTERISK-18740) Reported by: Byron Clark Patches: sip_exists_exten_dlock_3.diff (license #5041) patch uploaded by Gregory Hinton Nietsky ASTERISK-18740.patch (license #6157) patch uploaded by Byron Clark Tested by: Byron Clark ........ Merged revisions 344268 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344271 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-09 20:10 +0000 [r344214-344217] Terry Wilson * /, channels/chan_sip.c, channels/sip/reqresp_parser.c, channels/sip/include/sip.h, channels/sip/include/reqresp_parser.h: Don't treat a host:port string as a domain The domain matching code prior to 1.8 used to manually remove the port from the host:port string when determining if an incoming request matched the list of domains. When switching to the new parsing functions, the documentation implied that the "domain" was being returned by these functions, when instead it was returning the "hostport" as defined by RFC 3261. This led to confusion and resulted in 1.8+ rejecting an incoming request from x.x.x.x:xxxxx when domain=x.x.x.x was set in sip.conf. This patch renames the "domain" variables in the parsing functions to "hostport" to more accurately describe what it is that they are returning and also properly truncates the resulting hostport strings when dealing with domain matching. Review: https://reviewboard.asterisk.org/r/1574/ ........ Merged revisions 344215 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344216 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, tests/test_netsock2.c: Add a unit test for ast_sockaddr_split_hostport Review: https://reviewboard.asterisk.org/r/1575/ ........ Merged revisions 344157 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344175 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-09 19:08 +0000 [r344161] Alexandr Anikin * addons/ooh323c/src/ooh323.c, /, addons/ooh323c/src/ooh245.c, addons/ooh323c/src/ooq931.h, addons/ooh323c/src/ootypes.h, addons/ooh323c/src/oochannels.c, addons/ooh323c/src/ooq931.c: Generate response to Status Enquiry message with Status q.931 message. Some PBXes require this for call status checking (closes issue ASTERISK-18748) Reported by: Fabrizio Lazzaretti Patches: ASTERISK-18748-5.patch (License #5415) patch uploaded by may213 Tested by: Fabrizio Lazzaretti ........ Merged revisions 344158 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344159 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-09 17:15 +0000 [r344104] Kinsey Moore * /, apps/app_meetme.c: Fix pin parameter behavior regression in MeetMe The last time this code was touched (by me), a subtlety was missed based on the difference between needing to check a pin's validity and the need to prompt for a pin. (closes issue ASTERISK-18488) ........ Merged revisions 344102 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344103 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-09 15:28 +0000 [r344050] Matthew Nicholson * /, formats/format_wav.c: don't call ltohl() twice on the same value ASTERISK-18739 Patch by: pawel (modified) ........ Merged revisions 344048 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344049 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-08 22:14 +0000 [r344005] Richard Mudgett * /, channels/chan_sip.c: Residual changes for Asterisk v10 branch from ASTERISK-18747. Residual changes for Asterisk v10 branch from ASTERISK-18747 after https://reviewboard.asterisk.org/r/1564/ commit and associated dialogs callid hash key change fix. * Make check_rtp_timeout() return CMP_MATCH if need to delete dialog from dialogs_rtpcheck. This is an optimization to avoid an unneeded lock/unlock and object search when using ao2_unlink. * Prevent crash in check_rtp_timeout() if dialog->rtp is NULL. Review: https://reviewboard.asterisk.org/r/1557/ ........ Merged revisions 344004 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-08 19:29 +0000 [r343951] Walter Doekes * /, pbx/pbx_config.c: Fix crash when dialplan remove include is called with too few arguments. "dialplan remove include x from y" crashed when the amount of arguments was less than 6. (closes issue ASTERISK-18762) Reported by: Andrey Solovyev Tested by: Andrey Solovyev ........ Merged revisions 343936 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 343944 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-08 18:35 +0000 [r343905] David Vossel * /, channels/chan_sip.c: Merged revisions 343900 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r343900 | dvossel | 2011-11-08 12:29:33 -0600 (Tue, 08 Nov 2011) | 11 lines Fixes regression caused by r343635 There was a missing unlock for a function return that is only present in Asterisk 10 and Asterisk Trunk. (closes issue ASTERISK-18839) Reported by: Michael L. Young Patches: asterisk-18839-missing-lock-trunk-v2.diff (License #5026) patch uploaded by Michael L. Young ........ 2011-11-08 18:02 +0000 [r343853] Richard Mudgett * /, channels/chan_sip.c, main/acl.c: Fixed reference to incorrect variable if unknown host configured crash. * Fixed a LOG_ERROR message referencing the config variable list v that had previously been processed and became NULL. * Added error return value set that was missing in an ast_append_ha() error return path. (closes issue ASTERISK-18743) Reported by: Michele Patches: issueA18743-fix_dynamic_exclude_static_bad_host_log.patch (license #5674) patch uploaded by Walter Doekes Tested by: Michele ........ Merged revisions 343851 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 343852 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-08 13:23 +0000 [r343790] Leif Madsen * /, build_tools/prep_tarball: Fix boo-boo in prep_tarball script. A hardcoded a branch number was in the prep_tarball which could not work. Changed it to the variable. ........ Merged revisions 343789 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-07 22:37 +0000 [r343744] Kinsey Moore * /, channels/chan_sip.c: Make "sip show settings" CLI command get RPID flags from the right global page The "Trust RPID" and "Send RPID" entries in the "sip show settings" CLI command pulled the flags from the incorrect global flags page. These are now read from sip global flags page 0. (closes issue AST-711) ........ Merged revisions 343743 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-07 21:58 +0000 [r343693] Leif Madsen * configs/dundi.conf.sample, pbx/pbx_dundi.c, CHANGES: Allow built in variables to be used with dynamic weights. You can now use the built in variables , , and within a dynamic weight. For example, this could be useful when you want to pass requested lookup number to the SHELL() function which could be used to execute a script to dynamically set the weight of the result. (Closes issue ASTERISK-13657) Reported by: Joel Vandal Tested by: Leif Madsen, Russell Bryant Patches: asterisk-1.6-dundi-varhead.patch uploaded by Joel Vandal (License #5374) 2011-11-07 21:44 +0000 [r343692] Matthew Nicholson * /, channels/chan_sip.c: respect case changes in peer names on sip reload ASTERISK-18669 ........ Merged revisions 343690 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 343691 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-07 21:29 +0000 [r343684] Richard Mudgett * /, channels/chan_sip.c: Fix __sip_subscribe_mwi_do() incorectly changing dialogs hash key callid. Changing an object value used as a container key requires removing the object from the container and reinserting it. * Created change_callid_pvt() to call instead of build_callid_pvt(). The change_callid_pvt() will correctly change the dialog callid so the ao2 conainter can explicitly unlink it. ........ Merged revisions 343637 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 343677 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-07 20:35 +0000 [r343636] Kinsey Moore * /, channels/chan_sip.c: Prevent BLF subscriptions from causing deadlocks Fix a locking inversion in sip_send_mwi_to_peer that was causing deadlocks. This function now requires that both the peer and associated pvt be unlocked before it is called for cases where peer and peer->mwipvt form a circular reference. (closes issue ASTERISK-18663) Review: https://reviewboard.asterisk.org/r/1563/ ........ Merged revisions 343621 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 343635 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-07 19:58 +0000 [r343581] Walter Doekes * main/udptl.c, /, UPGRADE.txt: Correct the default udptl port range. The udptl port range was defined as 4000-4999 in the udptl.conf.sample, as 4500-4599 if you didn't have a config and 4500-4999 if your config was broken. Default is now 4000-4999. (closes issue ASTERISK-16250) Reviewed by: Tilghman Lesher Review: https://reviewboard.asterisk.org/r/1565 ........ Merged revisions 343580 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-07 19:54 +0000 [r343579] Richard Mudgett * /, channels/chan_sip.c: Fix deadlock if peer is destroyed while sending MWI notice. A dialog cannot be destroyed by the ao2_callback dialog_needdestroy because of a deadlock between the dialogs container lock and the RWLOCK of the events subscription list. * Create dialogs_to_destroy container to hold dialogs that will be destroyed. * Ensure that the event subscription callback will never happen with an invalid peer pointer by making the event callback removal the first thing in the peer destructor callback. NOTE: This particular deadlock will not happen with Asterisk 10, but some of the changes still apply. (closes issue ASTERISK-18747) Reported by: Gregory Hinton Nietsky Review: https://reviewboard.asterisk.org/r/1564/ ........ Merged revisions 343577 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 343578 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-07 18:42 +0000 [r343534] Matthew Nicholson * main/format.c, /: list all of the codecs associated with a particular format id for CLI command "core show codec" AST-699 ........ Merged revisions 343533 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-06 09:51 +0000 [r343492] Olle Johansson * main/tcptls.c, include/asterisk/tcptls.h: Formatting and doxygen improvements 2011-11-04 19:50 +0000 [r343448] Alexandr Anikin * addons/ooh323c/src/ooGkClient.c, addons/ooh323c/src/ooTimer.c, addons/ooh323c/src/dlist.c, /, addons/ooh323c/src/dlist.h, addons/ooh323c/src/printHandler.c, addons/ooh323c/src/ooq931.c: Final fix memleaks in GkClient codes, same for Timer codes. (these memleaks stop development of gk codes, now i can continue) Fix printHandler 'Unbalanced Structure' issues with locking printHandler data for single thread. ........ Merged revisions 343281 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 343445 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-03 20:37 +0000 [r343394] Walter Doekes * /, res/res_config_sqlite.c: Fix sqlite config driver segfault and broken queries The sqlite realtime handler assumed you had a static config configured as well. The realtime multientry handler assumed that you weren't using dynamic realtime. (closes issue ASTERISK-18354) (closes issue ASTERISK-18355) Review: https://reviewboard.asterisk.org/r/1561 ........ Merged revisions 343375 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 343393 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-03 19:57 +0000 [r343338] Richard Mudgett * /, funcs/func_dialgroup.c: Remove invalid flag given to iterator in func_dialgroup.c ........ Merged revisions 343336 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 343337 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-03 15:40 +0000 [r343222-343278] Terry Wilson * /, channels/sip/include/sip.h: Make room for the fax detect flags The original REGISTERTRYING flag, in addition to being impossible to check, also encroached on the space for the flag above it. This patch moves the flags that were below REGISTERTRYING back to where they were as though we had just removed the REGISTERTRYING option. ........ Merged revisions 343276 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 343277 from http://svn.asterisk.org/svn/asterisk/branches/10 * contrib/realtime/mysql/sippeers.sql, /, channels/chan_sip.c, channels/sip/include/sip.h: Remove registertrying option in chan_sip This option is not only useless, but has been broken since inception since the flag was never copied from the peer where it is set to the pvt where it was checked. RFC 3261 specificially states that you should not send a provisional response to a non-INVITE request, and if we did fix the code so that it worked, it would cause the same kind of user enumeration vulnerability that we've discussed with the nat= setting. This patch removes registertrying option and any code that would have sent a 100 response to a register. Review: https://reviewboard.asterisk.org/r/1562/ ........ Merged revisions 343220 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 343221 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-02 22:46 +0000 [r343163-343219] Walter Doekes * /, channels/chan_sip.c: Fix improper warning introduced by r342927 and more tweaks Changeset r342927 introduced a warning which was only supposed to be emitted when a found realtime peer had an empty (or no) name. It turned out that there were some inconsistencies left. Now found peers with an empty name are explicitly ignored like before r342927 but better. Reviewed by: Stefan Schmidts, Terry Wilson Review: https://reviewboard.asterisk.org/r/1560 ........ Merged revisions 343181 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 343192 from http://svn.asterisk.org/svn/asterisk/branches/10 * include/asterisk/utils.h, /, main/utils.c, include/asterisk/stringfields.h: Ensure that string field lengths are properly aligned Integers should always be aligned. For some platforms (ARM, SPARC) this is more important than for others. This changeset ensures that the string field string lengths are aligned on *all* platforms, not just on the SPARC for which there was a workaround. It also fixes that the length integer can be resized to 32 bits without problems if needed. (closes issue ASTERISK-17310) Reported by: radael, S Adrian Reviewed by: Tzafrir Cohen, Terry Wilson Tested by: S Adrian Review: https://reviewboard.asterisk.org/r/1549 ........ Merged revisions 343157 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 343158 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-02 19:33 +0000 [r343049-343104] Leif Madsen * apps/app_authenticate.c: Add note about how Authenticate() application with option 'd' works. (closes issue ASTERISK-17422) Reported by: Leif Madsen ........ Merged revisions 343102 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 343103 from http://svn.asterisk.org/svn/asterisk/branches/10 * configs/queues.conf.sample: Update documentation for leastrecent strategy. In queues.conf.sample the leastrecent strategy was incorrectly described. Now updated to reflect how the strategy actually checks peers. (closes issue ASTERISK-17854) Reported by: Sebastian Denz Patches: queues.conf-doc_issue.patch (License #6139) ........ Merged revisions 343047 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 343048 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-02 13:46 +0000 [r342992] Kevin P. Fleming * /, apps/app_meetme.c: Modify comments in MeetMe application documentation about DAHDI. The MeetMe application documentation has some comments about usage of DAHDI, and they were a bit outdated relative to modern DAHDI releases. This patch changes the comment to just tell the user that a functional DAHDI timing source is required, and no longer mention 'dahdi_dummy', since that module does not exist in current DAHDI releases. ........ Merged revisions 342990 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342991 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-11-01 21:02 +0000 [r342871-342930] Walter Doekes * /, channels/chan_sip.c, configs/extconfig.conf.sample, include/asterisk/config.h, main/config.c: Several fixes to the chan_sip dynamic realtime peer/user lookup There were several problems with the dynamic realtime peer/user lookup code. The lookup logic had become rather hard to read due to lots of incremental changes to the realtime_peer function. And, during the addition of the sipregs functionality, several possibilities for memory leaks had been introduced. The insecure=port matching has always been broken for anyone using the sipregs family. And, related, the broken implementation forced those using sipregs to *still* have an ipaddr column on their sippeers table. Thanks Terry Wilson for comprehensive testing and finding and fixing unexpected behaviour from the multientry realtime call which caused the realtime_peer to have a completely unused code path. This changeset fixes the leaks, the lookup inconsistenties and that you won't need an ipaddr column on your sippeers table anymore (when you're using sipregs). Beware that when you're using sipregs, peers with insecure=port will now start matching! (closes issue ASTERISK-17792) (closes issue ASTERISK-18356) Reported by: marcelloceschia, Walter Doekes Reviewed by: Terry Wilson Review: https://reviewboard.asterisk.org/r/1395 ........ Merged revisions 342927 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342929 from http://svn.asterisk.org/svn/asterisk/branches/10 * contrib/realtime/mysql/sippeers.sql (added), configs/res_config_mysql.conf.sample, /, configs/extconfig.conf.sample, configs/res_ldap.conf.sample, res/res_realtime.c, UPGRADE-1.8.txt, configs/dbsep.conf.sample, main/config.c, contrib/realtime/mysql/sipfriends.sql (removed): Cleanup references to sipusers and sipfriends dynamic realtime families Somewhere between 1.4 and 1.8 the sipusers family has become completely unused. Before that, the sipfriends family had been obsoleted in favor of separate sipusers and sippeers families. Apparently, they have been merged back again into a single family which is now called "sippeers". Reviewed by: irroot, oej, pabelanger Review: https://reviewboard.asterisk.org/r/1523 ........ Merged revisions 342869 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342870 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-10-31 17:51 +0000 [r342825] Richard Mudgett * main/format.c, /, main/format_cap.c: Misc format capability fixes. * Fixed typo in format_cap.c:joint_copy_helper() using the wrong variable. * Fix potential race between checking if an interface exists and adding it to the container in format.c:ast_format_attr_reg_interface(). * Fixed double rwlock destroy in format.c:ast_format_attr_init() error exit path. * Simplified format.c:find_interface() and format.c:has_interface(). ........ Merged revisions 342824 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-10-31 16:10 +0000 [r342771] Matthew Jordan * main/pbx.c, /, channels/chan_iax2.c: Fixed invalid memory access when adding extension to pattern match tree When an extension is removed from a context, its entry in the pattern match tree is not deleted. Instead, the extension is marked as deleted. When an extension is removed and re-added, if that extension is also a prefix of another extension, several log messages would report an error and did not check whether or not the extension was deleted before accessing the memory. Additionally, if the extension was already in the tree but previously deleted, and the pattern was at the end of a match, the findonly flag was not honored and the extension would be erroneously undeleted. Additionaly, it was discovered that an IAX2 peer could be unregistered via the CLI, while at the same time it could be scheduled for unregistration by Asterisk. The unregistration method now checks to see if the peer was already unregistered before continuing with an unregistration. (closes issue ASTERISK-18135) Reported by: Jaco Kroon, Henry Fernandes, Kristijan Vrban Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1526 ........ Merged revisions 342769 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342770 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-10-30 02:31 +0000 [r342716] Terry Wilson * /, res/res_calendar.c: Don't crash on empty notify channel ........ Merged revisions 342715 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-10-29 04:41 +0000 [r342663-342664] Richard Mudgett * include/asterisk/linkedlists.h: Whitespace and some better macro variable names. * Renamed AST_LIST_TRAVERSE_SAFE_BEGIN __new_prev to __list_current. * Renamed AST_LIST_MOVE_CURRENT __list_cur to __extracted. * /, include/asterisk/linkedlists.h, tests/test_linkedlists.c: Fix AST_LIST_INSERT_BEFORE_CURRENT() updating the wrong variable. AST_LIST_INSERT_BEFORE_CURRENT() could not be used twice in an iteration or before AST_LIST_REMOVE_CURRENT() without corrupting the list. AST_LIST_INSERT_BEFORE_CURRENT() could also corrupt the list if AST_LIST_INSERT_BEFORE_CURRENT() or AST_LIST_REMOVE_CURRENT() is used on the next iteration. * Fixed cut and paste error using the wrong variable in AST_LIST_INSERT_BEFORE_CURRENT(). * Added linked list unit tests for AST_LIST_INSERT_BEFORE_CURRENT(), AST_LIST_APPEND_LIST(), and AST_LIST_INSERT_LIST_AFTER(). ........ Merged revisions 342661 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342662 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-10-27 20:11 +0000 [r342606] Matthew Nicholson * /, main/dsp.c: tweak the v21 detector to detect an additional pattern of hits and misses ........ Merged revisions 342605 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-10-27 19:48 +0000 [r342557-342604] Jonathan Rose * res/res_rtp_multicast.c, /: Fix sequence number overflow over 16 bits causing codec change in RTP packets. Sequence number was handled as an unsigned integer (usually 32 bits I think, more depending on the architecture) and was put into the rtp packet which is basically just a bunch of bits using an or operation. Sequence number only has 16 bits allocated to it in an RTP packet anyway, so it would add to the next field which just happened to be the codec. This makes sure the sequence number is set to be a 16 bit integer regardless of architecture (hopefully) and also makes it so the incrementing of the sequence number does bitwise or at the peak of a 16 bit number so that the value will be set back to 0 when going beyond 65535 anyway. (closes issue ASTERISK-18291) Reported by: Will Schick Review: https://reviewboard.asterisk.org/r/1542/ ........ Merged revisions 342602 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342603 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, res/res_jabber.c: Cleanup reference leaks in res_jabber res_jabber.c had a number of places where astobjs would be referenced and have their reference counts bumped without having a dereference made before the object lost scope. This patch adds a number of ASTOBJ_UNREFs to resolve that. Review: https://reviewboard.asterisk.org/r/1478/ ........ Merged revisions 342545 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342546 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-10-25 22:06 +0000 [r342486-342489] Richard Mudgett * /, main/astobj2.c: Check fopen return value for ao2 reference debug output. Reported by: wdoekes Patched by: wdoekes Review: https://reviewboard.asterisk.org/r/1539/ ........ Merged revisions 342487 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342488 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, channels/sig_pri.c: Change D-channel warning to be less confusing on non-NFAS setups. The "No D-channels available! Using Primary channel as D-channel anyway!" WARNING message has been confusing on non-NFAS setups. The message refers to things that are NFAS specific. * Changed the warning to several different warnings to be more accurate for the situation and less confusing as a result: "No D-channels up! Switching selected D-channel from X to Y.", "No D-channels up!", and "D-channel is down!". ........ Merged revisions 342484 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342485 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-10-25 21:11 +0000 [r342382-342437] Terry Wilson * /, apps/app_queue.c: Use int for storing ao2_container_count instad of size_t AST-676 ........ Merged revisions 342435 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342436 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, apps/app_queue.c: Simplify queue membercount code Despite an ominous sounding comment stating that membercount was for "logged in" members only and thus we couldn't use ao2_container_count(), I could not find a single place in the code where that seemed to be accurate. The only time we decremented membercount was when we were marking something dead or actually removing it. The only places we incremented it were either after ao2_link(), or trying to correct for having set it to 0 during a reload. In every case where we were correcting the value, it seemed that we were trying to make the count actually match what ao2_container_count() would return. The only place I could find where we made a determination about something being "logged in" or not, we didn't trust the membercount, but instead looked at devicestate, paused, etc. This patch removes membercount, replaces its use with ao2_container_count, and manually adds the results of ao2_container_count to a "membercount" field for ast_data queue query results. This patch also would fix AST-676, but as it is slightly riskier than the previously committed fix, the two commits have been made separately. Reivew: https://reviewboard.asterisk.org/r/1541/ ........ Merged revisions 342383 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342384 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, apps/app_queue.c: Properly update membercount for reloaded members Since q->membercount is set to 0 before reloading, it is important to increment it again for reloaded members as well as added. (closes issue AST-676) Review: https://reviewboard.asterisk.org/r/1541/ ........ Merged revisions 342380 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342381 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-10-25 19:09 +0000 [r342278-342330] Kinsey Moore * pbx/pbx_spool.c, /: Fix compilation on Snow Leopard/FreeBSD for pbx_spool.c One of the changes in the recent spool handling of hardlinks patch was just outside a HAVE_INOTIFY block and caused compilation to fail in some build environments. This has been corrected. ........ Merged revisions 342328 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342329 from http://svn.asterisk.org/svn/asterisk/branches/10 * pbx/pbx_spool.c, /: Merged revisions 342277 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r342277 | kmoore | 2011-10-25 11:08:04 -0500 (Tue, 25 Oct 2011) | 25 lines Merged revisions 342276 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r342276 | kmoore | 2011-10-25 11:06:57 -0500 (Tue, 25 Oct 2011) | 18 lines Fix spool handling to allow call files to be hardlinked into place This fixes the inotify code to handle call files being hardlinked into the spool directory. The smsq utility does this, instead of rename(), to ensure that it cannot accidentally overwrite an existing spool file. A rename() might do that, but link() will definitely not. The inotify code had broken this, because it would wait for an IN_CLOSE_WRITE event on the file... which was never forthcoming, since it was never opened. Now we look for IN_OPEN events following the IN_CREATE event, and only wait for an IN_CLOSE_WRITE if the file was actually opened. Patch-by: dwmw2 (closes issue ASTERISK-18331) Review: https://reviewboard.asterisk.org/r/1391/ ........ ................ 2011-10-25 01:29 +0000 [r342225] Terry Wilson * /, include/asterisk/config.h, main/config.c: Return NULL when no results returned for realtime_multientry It was not documented what the return value should be when no entries were returned with the multientry realtime callback. This change forces consistent behavior even if the backends return an empty ast_config. Review: https://reviewboard.asterisk.org/r/1521/ ........ Merged revisions 342223 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342224 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-10-24 22:37 +0000 [r342184] Richard Mudgett * /, include/asterisk/astobj2.h: Fix ao2obj.h comment typos and add missing link/unlink nolock debug defines. ........ Merged revisions 342183 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-10-24 22:09 +0000 [r342148] Jonathan Rose * main/features.c: Fixes a segfault caused by referencing null frames introduced in r338623 2011-10-24 21:01 +0000 [r342112] Richard Mudgett * apps/app_queue.c: Fix use of OBJ_KEY in Queue application. To use the new OBJ_KEY flag, the container hash and compare callback functions must be updated to support OBJ_KEY. Otherwise, bad things happen. (issue ASTERISK-14769) 2011-10-24 20:01 +0000 [r342063] Jonathan Rose * /, channels/chan_sip.c: Outbound SIP OPTIONS messages will now include fromuser of related peer. This behavior matches up more closely with the way invite/register/etc are handled. This patch also modifies some adjacent code for code style compliance. Pretty minor. (closes issue ASTERISK-17616) Reported by: Jeremy Kister Patches: chan_sip.c-options-fromuser-fix-v1.patch uploaded by Jeremy Kister (license #6232) ........ Merged revisions 342061 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342062 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-10-24 07:40 +0000 [r341923-342018] Gregory Nietsky * /, apps/app_queue.c: queues container needs locking when using the OBJ_NOLOCK flag ........ Merged revisions 342017 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, apps/app_queue.c: Remove some ref leaks and a return without unlock. There some resource leaks introduced in asterisk 10 make sure that locks are not held on return and we release ref's held. ........ Merged revisions 341972 from http://svn.asterisk.org/svn/asterisk/branches/10 * apps/app_queue.c: Whitespace Fixups / Add Braces This janitorial patch is related to work on RB1538 2011-10-22 12:03 +0000 [r341869] Alexandr Anikin * addons/chan_ooh323.c, /: Merged revisions 341313 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r341313 | may | 2011-10-19 03:33:49 +0400 (Wed, 19 Oct 2011) | 10 lines Merged revisions 341312 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r341312 | may | 2011-10-19 03:20:53 +0400 (Wed, 19 Oct 2011) | 3 lines fix issue on channel numbering (calls could have same channel number on heavy loaded system) ........ ................ 2011-10-21 16:42 +0000 [r341808-341811] Matthew Nicholson * /, pbx/pbx_lua.c: only process args that exist ASTERISK-18395 ........ Merged revisions 341809 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341810 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, pbx/pbx_lua.c: don't limit the length of app and function arguments ASTERISK-18395 ........ Merged revisions 341806 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341807 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-10-21 09:16 +0000 [r341769] Gregory Nietsky * res/res_fax.c: White space fixes in res_fax 2011-10-20 22:03 +0000 [r341719] Richard Mudgett * /, main/features.c, res/res_agi.c, include/asterisk/features.h: Fix AGI exec Park to honor the Park application parameters. The fix for ASTERISK-12715 and ASTERISK-12685 added a check for the Park application because the channel needed to be masqueraded to prevent a crash. Since the Park application now always masquerades the channel into the parking lot, the special check is no longer needed. The fix also resulted in AGI exec Park attempting to double park the call and not honor the Park application parameters. * Removed no longer necessary call to ast_masq_park_call() by AGI exec for the Park application. (Reverts -r146923) * Fix Park application to only return 0 or -1. The AGI exec Park was causing broken pipe error messages because the Park application returned 1 on successful park. (closes issue ASTERISK-18737) ........ Merged revisions 341717 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341718 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-10-20 21:28 +0000 [r341666-341713] Paul Belanger * /, funcs/func_callerid.c: Fixed typo from previous commit ........ Merged revisions 341704 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341707 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, funcs/func_callerid.c: Updated documentation for the optional CID parameter with CALLERID ........ Merged revisions 341664 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341665 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-10-20 18:27 +0000 [r341583-341624] Gregory Nietsky * /, configs/queues.conf.sample: Merged revisions 341599 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r341599 | irroot | 2011-10-20 20:20:08 +0200 (Thu, 20 Oct 2011) | 8 lines add documentation for check_state_unknown in configs/queues.conf.sample app_queue allows calls to members in a "Unknown" state to be treated as available setting check_state_unknown = yes will cause app_queue to query the channel driver to better determine the state this only applies to queues with ringinuse or ignorebusy set appropriately. ........ * /, CHANGES, apps/app_queue.c: Merged revisions 341580 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r341580 | irroot | 2011-10-20 19:13:23 +0200 (Thu, 20 Oct 2011) | 15 lines Add option to check state when state is unknown r341486 reverts r325483 this is a rework of the patch. optimize to minimize load. add option check_state_unknown to control whether a member with unknown device state is checked there is a small % chance that calls will be sent to the member when they on a call. app_queue will see a device with unknown state as available and does not try verify the state without this option enabled. Review: https://reviewboard.asterisk.org/r/1535/ ........ 2011-10-20 15:17 +0000 [r341533] Terry Wilson * /, include/asterisk/strings.h: Clean up ast_check_digits The code was originally copied from the is_int() function in the AEL code. wdoekes pointed out that the function should take a const char* and that their was an unneeded variable. This is now fixed. ........ Merged revisions 341529 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341530 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-10-19 21:24 +0000 [r341487] Matthew Nicholson * /, apps/app_queue.c: Merged revisions 341486 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r341486 | mnicholson | 2011-10-19 16:23:17 -0500 (Wed, 19 Oct 2011) | 18 lines Fix a performance regression introduced in r325483. The regression was caused by a call to ast_parse_device_state() in app_queue's ring_entry() function. The ast_parse_device_state() function eventually calls ast_channel_get_full() with a channel name prefix which causes it to walk the channel list causing massive lock contention and slow downs. This patch fixes the regression by removing the call to ast_parase_device_state() which should be unnecessary. Queue member device state should be maintained by device state events. Some users have seen instances where busy agents were called when they shouldn't have, which is the reason the call to ast_parse_device_state() was added. That change appears to have resolved that issue but also causes this performance regression. There may still be issues with queue member status, and if so, alternative methods should be investigated to resolve them. AST-695 ........ 2011-10-19 19:02 +0000 [r341437] Paul Belanger * /, channels/chan_gtalk.c: Outgoing calls with Google Voice Google has recently make some changes (again) to their protocol. Rather then patching asterisk to flip between the two different methods, we now allow both. Lets hope this keeps Google Voice happy for a while. (closes issue ASTERISK-18714) Reported by: Iordan Iordanov Patches: chan_gtalk.patch uploaded by Iordan Iordanov (licenses 6311) ........ Merged revisions 341435 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341436 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-10-19 07:45 +0000 [r341381] Terry Wilson * /, channels/chan_sip.c, include/asterisk/strings.h: Don't use is_int() since it doesn't link well on all platforms Just create an normal API function in strings.h that does the same thing just to be safe. ASTERISK-17146 ........ Merged revisions 341379 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341380 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-10-19 07:27 +0000 [r341378] Stefan Schmidt * /, channels/chan_sip.c: Don't sent in-dialog requests like UPDATE when Asterisk has not yet received a Contact URI from a UAS ........ Merged revisions 341366 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341377 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-10-18 23:45 +0000 [r341316] Terry Wilson * /, channels/chan_sip.c: Don't resolve numeric hosts or contact unresolved hosts If a SIP dial string contains a numeric hostname that is not a peer name, don't try to resolve it as it is unlikely that someone really means Dial(SIP/0.0.4.26) when Dial(SIP/1050) is called. Also, make sure that create_addr returns -1 if an address isn't resolved so that we don't attempt to send SIP requests to an address that doesn't resolve. (closes issue ASTERISK-17146, ASTERISK-17716) Review: https://reviewboard.asterisk.org/r/1532/ ........ Merged revisions 341314 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341315 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-10-18 21:15 +0000 [r341256] Richard Mudgett * channels/chan_dahdi.c, channels/sig_analog.c, /, channels/chan_sip.c, main/features.c, channels/chan_iax2.c, channels/sip/include/sip.h, channels/chan_mgcp.c, include/asterisk/features.h: More parking issues. * Fix potential deadlocks in SIP and IAX blind transfer to parking. * Fix SIP, IAX, DAHDI analog, and MGCP channel drivers to respect the parkext_exclusive option with transfers (Park(,,,,,exclusive_lot) parameter). Created ast_park_call_exten() and ast_masq_park_call_exten() to maintian API compatibility. * Made masq_park_call() handle a failed ast_channel_masquerade() setup. * Reduced excessive struct parkeduser.peername[] size. ........ Merged revisions 341254 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341255 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-10-17 17:58 +0000 [r341198] Tzafrir Cohen * /, pbx/pbx_realtime.c: Remove an unused include of md5.h Unused include of asterisk/md5.h in pbx_realtime.c . A commit needed to test the commit message. Merged-From: http://svn.asterisk.org/svn/asterisk/branches/1.8@341074 Merged-From: http://svn.asterisk.org/svn/asterisk/branches/10@341148 2011-10-17 17:38 +0000 [r341191] Terry Wilson * /, channels/chan_sip.c: Initialize variables before calling parse_uri If parse_uri was called with an empty URI, some pointers would be modified and an invalid read could result. This patch avoids calling parse_uri with an empty contact uri when parsing REGISTER requests. AST-2011-012 (closes issue ASTERISK-18668) ........ Merged revisions 341189 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341190 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-10-17 16:39 +0000 [r341126-341147] Paul Belanger * /, tests/test_format_api.c: Set 'core' support level for test_format_api.c ........ Merged revisions 341146 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, apps/app_voicemail.c: Multiple revisions 341108,341112 ........ r341108 | pabelanger | 2011-10-17 12:22:19 -0400 (Mon, 17 Oct 2011) | 2 lines Voicemail compiler flags are 'core' support ........ r341112 | pabelanger | 2011-10-17 12:23:33 -0400 (Mon, 17 Oct 2011) | 2 lines Fix previous commit ........ Merged revisions 341108,341112 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341122 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-10-17 16:18 +0000 [r341096] Jason Parker * /, CHANGES: Add information about limitations of new codec support in channel drivers. (issue ASTERISK-18680) ........ Merged revisions 341094 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-10-17 15:45 +0000 [r341090] Terry Wilson * /, channels/chan_sip.c: Don't try to remove peers without IPs from peers_by_ip (closes issue ASTERISK-18696) ........ Merged revisions 341088 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341089 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-10-14 21:37 +0000 [r341024] Kevin P. Fleming * /, build_tools/embed_modules.xml, Makefile.moddir_rules: Change the internal name of the menuselect options that are used to control whether modules are embedded or not; using just the bare category name led to accidentally enabling these options when users used the wrong "--enable" operation on the menuselect command line. Now the internal option names are prefixed with "EMBED_", so they won't be the same as the name of the category containing the modules they control the embedding of. ........ Merged revisions 341022 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341023 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-10-14 21:15 +0000 [r340973] Damien Wedhorn * channels/chan_skinny.c: Fix simple switch to not progress a call when call already progressed. If a simple switch was started on a device and then a specific call made (such as redial or speed dial), on timeout of the simple switch the call would be attempted again. This patch only allows the simple switch to make a call if the substate is still in the collecting digits mode. Also added small debug message to dialAndAactivate sub. Tested by snuff and myself. 2011-10-14 20:51 +0000 [r340972] Kinsey Moore * res/res_rtp_asterisk.c, /, channels/chan_sip.c: Merged revisions 340971 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r340971 | kmoore | 2011-10-14 15:50:37 -0500 (Fri, 14 Oct 2011) | 15 lines Merged revisions 340970 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340970 | kmoore | 2011-10-14 15:49:39 -0500 (Fri, 14 Oct 2011) | 8 lines Quiet RTCP Receiver Reports during fax transmission RTCP is now disabled for "inactive" RTP audio streams during SIP T.38 sessions. The ability to disable RTCP streams in res_rtp_asterisk was missing, so this code was added to support the bug fix. (closes issue ASTERISK-18400) ........ ................ 2011-10-14 18:38 +0000 [r340932] Jonathan Rose * utils/utils.xml, /, funcs/func_jitterbuffer.c: Some additional module documentation changes for 10 for the menuselect change. (issue ASTERISK-18268) ........ Merged revisions 340931 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-10-14 16:45 +0000 [r340880] Terry Wilson * main/channel.c, /: Avoid unnecessary WARNING message Add AST_CONTROL_UPDATE_RTP_PEER frame to be ignored here to avoid displaying a WARNING message. (closes issue ASTERISK-18610) Patch by: Kristijan_Vrban ........ Merged revisions 340878 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340879 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-10-13 23:08 +0000 [r340811-340813] Richard Mudgett * /, main/features.c: Fix DTMF blind transfer continuing to execute dialplan after transfer. Party A calls Party B. Party A DTMF blind transfers Party B to Party C. Party A channel continues to execute dialplan. * Fixed the return value of builtin_blindtransfer() to return the correct value after a transfer so the dialplan will not keep executing. * Removed unnecessary connected line update that did not really do anything. * Made access to GOTO_ON_BLINDXFR thread safe in check_goto_on_transfer(). * Fixed leak of xferchan for failure cases in check_goto_on_transfer(). * Updated debug messages in builtin_blindtransfer() and check_goto_on_transfer(). (closes issue ASTERISK-18275) Reported by: rmudgett Tested by: rmudgett ........ Merged revisions 340809 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340810 from http://svn.asterisk.org/svn/asterisk/branches/10 * /: Update 10 merged property. * /: Restore branch 10 merge properties. 2011-10-13 08:53 +0000 [r340771] Gregory Nietsky * /: Merged revisions 339463 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r339463 | irroot | 2011-10-05 08:28:46 +0200 (Wed, 05 Oct 2011) | 9 lines Only change the capabilities on the gateway when the session is been destroyed there is still a race condition that ends in a segfault. if the caps are changed the logic in res_fax_spandsp will run T30 code not gateway code to end the session. this has been experienced on a "slower" under spec system. ........ 2011-10-13 07:05 +0000 [r340720] Stefan Schmidt * channels/chan_sip.c: Merged revisions 340718 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r340718 | schmidts | 2011-10-13 06:59:50 +0000 (Thu, 13 Oct 2011) | 9 lines Merged revisions 340717 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340717 | schmidts | 2011-10-13 06:58:00 +0000 (Thu, 13 Oct 2011) | 3 lines storing the route-set also on a 181 response not only on 180,182 or 183. ........ ................ 2011-10-13 07:02 +0000 [r340665-340719] Terry Wilson * /, channels/chan_sip.c: Initialize ast_sockaddr before calling ast_sockaddr_resolve Avoid possible jump based on unitialized value ........ Merged revisions 340715 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340716 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, res/res_config_sqlite.c: Don't skip the query field on a realtime multi query There is no documented reason to not add the query field to the varlist returned by a realtime multi query, despite the config category being set to its value. Of course, there is no documentation that the category should be set to the value either. There is lots of no documentation when it comes to realtime. But, other engines do not skip this field so I am forcing this backend to follow the convention, because not doing so is very silly. ........ Merged revisions 340662 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340663 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-10-12 21:28 +0000 [r340626] Stefan Schmidt * channels/chan_sip.c: Merged revisions 340577 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r340577 | schmidts | 2011-10-12 20:33:37 +0000 (Mit, 12 Okt 2011) | 9 lines Merged revisions 340576 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340576 | schmidts | 2011-10-12 20:30:37 +0000 (Mit, 12 Okt 2011) | 3 lines Store route-set from provisional SIP responses so early-dialog requests can be routed properly ........ ................ 2011-10-12 21:02 +0000 [r340579] Terry Wilson * /, channels/chan_sip.c: Merged revisions 340578 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r340578 | twilson | 2011-10-12 13:57:19 -0700 (Wed, 12 Oct 2011) | 16 lines Merged revisions 340534 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340534 | twilson | 2011-10-12 13:19:36 -0700 (Wed, 12 Oct 2011) | 9 lines Update SIP realtime fullcontact regardless of caching We should update the fullcontact field in the realtime table whether or not rtcachefriends is set. There is no reason to treat a non-cached realtime entity differently than a cached in this regard. (closes issue ASTERISK-18446) Reported by: wdoekes ........ ................ 2011-10-12 20:09 +0000 [r340472-340524] Richard Mudgett * channels/chan_dahdi.c, /: Initialize the PRI channel alarms properly on startup. The PRI channel alarms were initialized with an inverted sense. (closes issue ASTERISK-18710) Reported by: Tzafrir Cohen ........ Merged revisions 340522 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340523 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, apps/app_meetme.c: Update MeetMe p and X option documentation when interacting with the s option. ASTERISK-12175 changed the p and X options to not interfere with the s option when they are used together. It makes more sense for the s option to have priority for the DTMF '*' key since it cannot change its activation code. Otherwise, you could not use option s with the p or X options. JIRA AST-671 ........ Merged revisions 340470 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340471 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-10-12 16:29 +0000 [r340420] Paul Belanger * /, channels/chan_sip.c: Fix verbose messages when IPv6 logic was added (closes issue ASTERISK-18612) Reported by: Tim Osman ........ Merged revisions 340418 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340419 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-10-11 21:06 +0000 [r340318-340367] Richard Mudgett * channels/chan_dahdi.c, channels/sig_ss7.h, /, channels/sig_ss7.c: Add protection for SS7 channel allocation and better glare handling. * Added a CLI "ss7 show channels" command that might prove useful for future debugging. * Made the incoming SS7 channel event check and gripe message uniform. * Made sure that the DNID string for an incoming call is always initialized. (issue ASTERISK-17966) Reported by: Kenneth Van Velthoven Patches: jira_asterisk_17966_v1.8_glare.patch (license #5621) patch uploaded by rmudgett ........ Merged revisions 340365 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340366 from http://svn.asterisk.org/svn/asterisk/branches/10 * channels/sip/include/dialog.h, /, channels/chan_sip.c: Fix some potential deadlocks pointed out by helgrind. * Fixed deadlock potential calling dialog_unlink_all() in __sip_autodestruct(). Found by helgrind. * Fixed deadlock potential in handle_request_invite() after calling sip_new(). Found by helgrind. * The sip_new() function now returns with the created channel already locked. * Removed the dead code that starts a PBX in in sip_new(). No sip_new() callers caused that code to be executed and it was a bad thing to do anyway. * Removed unused parameters and return value from dialog_unlink_all(). * Made dialog_unlink_all() and __sip_autodestruct() safely obtain the owner and private channel locks without a deadlock avoidance loop. ........ Merged revisions 340284 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340310 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-10-11 19:06 +0000 [r340283] Tzafrir Cohen * main/channel.c, /, main/sha1.c, include/asterisk/sha1.h: Update SHA1 code to RFC 6234 RFC 6234 is an update to RFC 3174 from which the code was originally taken. It has a slightly better code, and a better phrased license (simple 3-clause BSD). * main/sha1.c is sha1.c from RFC 6234 with formatting changes only. * include/asterisk/sha1.h merges sha.h and sha-private.h from RFC 6234. * Removed unused include of asterisk/sha1.h from main/channels.c Review: https://reviewboard.asterisk.org/r/1503/ Merge-From: http://svn.asterisk.org/svn/asterisk/branches/1.8@340263 Merge-From: http://svn.asterisk.org/svn/asterisk/branches/10@340280 2011-10-11 18:57 +0000 [r340282] Richard Mudgett * main/manager.c, /, include/asterisk/manager.h: Convert registered AMI actions to ao2 objects. * Fixed race between calling an AMI action callback and unregistering that action. Refixes ASTERISK-13784 broken by ASTERISK-17785 change. * Fixed potential memory leak if an AMI action failed to get registered because is already was registered. Part of the ao2 conversion. * Fixed AMI ListCommands action not walking the actions list with a lock held. * Fix usage of ast_strdupa() and alloca() in loops. Excess stack usage. * Fix AMI Originate action Variable header requiring a space after the header colon. Reported by Yaroslav Panych on the asterisk-dev list. * Increased the number of listed variables allowed per AMI Originate action Variable header to 64. * Fixed AMI GetConfigJSON action output format. * Fixed usage of res contents outside of scope in append_channel_vars(). * Fixed inconsistency of config file channelvars option. The values no longer accumulate with every channelvars option in the config file. Only the last value is kept to be consistent with the CLI "manager show settings" command. (closes issue ASTERISK-18479) Reported by: Jaco Kroon ........ Merged revisions 340279 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340281 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-10-10 23:10 +0000 [r340221-340224] Terry Wilson * UPGRADE.txt, main/db.c: Return error when no rows are deleted for AMI DBDelTree (closes issue AST-654) * /, main/db.c: Merged revisions 340222 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r340222 | twilson | 2011-10-10 15:55:39 -0700 (Mon, 10 Oct 2011) | 8 lines On astdb conversion, also warn about permissions requirements The user running Asterisk must have permission to the directory the Asterisk database resides in since SQLite 3 needs to be able to create a journal file. (closes issue ASTERISK-18174) ........ * utils/Makefile, utils/utils.xml, /, UPGRADE.txt, utils/astdb2bdb.c (added): Merged revisions 340219-340220 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r340219 | twilson | 2011-10-10 15:38:06 -0700 (Mon, 10 Oct 2011) | 8 lines Add astdb conversion utility for Berkeley to SQLite 3 If someone wants to backtrack from Asterisk 1.8 to 10 they can use the astdb2bdb utility to convert the database back to the Berkeley format that Asterisk 1.8 uses. Review: https://reviewboard.asterisk.org/r/1502/ ........ r340220 | twilson | 2011-10-10 15:39:41 -0700 (Mon, 10 Oct 2011) | 2 lines Add a missing file for the astdb2bdb conversion utility ........ 2011-10-10 20:39 +0000 [r340166] Matthew Jordan * /, channels/chan_sip.c: Merged revisions 340165 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r340165 | mjordan | 2011-10-10 15:30:18 -0500 (Mon, 10 Oct 2011) | 20 lines Merged revisions 340164 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340164 | mjordan | 2011-10-10 15:23:48 -0500 (Mon, 10 Oct 2011) | 13 lines Updated chan_sip to place calls on hold if SDP address in INVITE is ANY This patch fixes the case where an INVITE is received with c=0.0.0.0 or ::. In this case, the call should be placed on hold. Previously, we checked for the address being null; this patch keeps that behavior but also checks for the ANY IP addresses. Review: https://reviewboard.asterisk.org/r/1504/ (closes issue ASTERISK-18086) Reported by: James Bottomley Tested by: Matt Jordan ........ ................ 2011-10-10 14:16 +0000 [r340110] Matthew Nicholson * main/pbx.c, main/manager.c, /, res/res_fax.c, apps/app_fax.c, include/asterisk/module.h, res/res_agi.c, include/asterisk/xmldoc.h, doc/appdocsxml.dtd, main/loader.c, main/xmldoc.c: Merged revisions 340109 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r340109 | mnicholson | 2011-10-10 09:15:41 -0500 (Mon, 10 Oct 2011) | 18 lines Merged revisions 340108 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340108 | mnicholson | 2011-10-10 09:14:48 -0500 (Mon, 10 Oct 2011) | 11 lines Load the proper XML documentation when multiple modules document the same application. This patch adds an optional "module" attribute to the XML documentation spec that allows the documentation processor to match apps with identical names from different modules to their documentation. This patch also fixes a number of bugs with the documentation processor and should make it a little more efficient. Support for multiple languages has also been properly implemented. ASTERISK-18130 Review: https://reviewboard.asterisk.org/r/1485/ ........ ................ 2011-10-10 00:57 +0000 [r339993-340071] Damien Wedhorn * channels/chan_skinny.c: Add skinny version 17 protocol support. Added some data to skinny packet structures to make compatible with v17. Added protocolversion to device, set on registration based on the version provided by device. v17 includes some increased ip space for ip6. This patch increases ip space in the packets but still only uses ip4. Some packet structures duplicated (ip4 and ip6 types). ip4 type used unless version is greater or equal to 17. Tested by snuff and myself on 7961 with recent 8.5 firmware. Also tested compatible with old 7960 and older 30VIPs. * channels/chan_skinny.c: Increase SKINNY_MAX_PACKET and add some logging. Increase SKINNY_MAX_PACKET to 2000 bytes to handle some messages in v17 that are greater than the old 1000 bytes. Also add some useful logging regarding packet and session handling. A device (with protocol v17) was sending a packet with length greater than 1000 which resulted in the TCP session being destroyed and registration being retryed. * /, channels/chan_skinny.c: Merged revisions 340031 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r340031 | wedhorn | 2011-10-10 09:18:27 +1100 (Mon, 10 Oct 2011) | 8 lines Return -1 to skinny_session if register rejected. If device registration is rejected, return -1 so that the session is destroyed immediately. Previously, a segfault would occur on a graceful shutdown if a register is rejected and the skinny_session has not yet timed out. ........ * /, channels/chan_skinny.c: Merged revisions 339992 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r339992 | wedhorn | 2011-10-10 08:09:12 +1100 (Mon, 10 Oct 2011) | 9 lines Remove log message on traverse session list. On destroying a session, a list of sessions is traversed to find the matching session. For each session not matching, skinny erroneously logged that the session was not matched. While technically correct the message was misleading, and tended to indicate errors that were not there. ........ 2011-10-09 01:19 +0000 [r339832-339947] Igor Goncharovskiy * channels/chan_unistim.c, /: Merged revisions 339942 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339942 | igorg | 2011-10-09 08:18:02 +0700 (Вск, 09 Окт 2011) | 12 lines Merged revisions 339938 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339938 | igorg | 2011-10-09 08:16:09 +0700 (Вск, 09 Окт 2011) | 6 lines Fix compilation issue, caused by missed session structure (closes issue ASTERISK-18694) Reported by: alex70 ........ ................ * channels/chan_unistim.c, /: Merged revisions 339885 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339885 | igorg | 2011-10-08 22:46:27 +0700 (Сбт, 08 Окт 2011) | 13 lines Merged revisions 339884 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339884 | igorg | 2011-10-08 22:45:20 +0700 (Сбт, 08 Окт 2011) | 7 lines Fix segfault in Unistim channel (closes issue ASTERISK-18638) Reported by: jonnt ........ ................ * channels/chan_unistim.c, /: Merged revisions 339831 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339831 | igorg | 2011-10-08 22:01:35 +0700 (Сбт, 08 Окт 2011) | 14 lines Merged revisions 339830 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339830 | igorg | 2011-10-08 21:56:35 +0700 (Сбт, 08 Окт 2011) | 8 lines Fix char array cast as short array in send_client() function (for ARM platform) (closes issue ASTERISK-17314) Reported by: jjoshua ........ ................ 2011-10-07 19:37 +0000 [r339721-339778] Richard Mudgett * /, apps/app_url.c: Merged revisions 339777 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339777 | rmudgett | 2011-10-07 14:36:24 -0500 (Fri, 07 Oct 2011) | 12 lines Merged revisions 339776 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339776 | rmudgett | 2011-10-07 14:34:55 -0500 (Fri, 07 Oct 2011) | 5 lines Initialize option flags for SendURL application. (closes issue ASTERISK-18574) Reported by: marcelloceschia ........ ................ * /: Recorded merge of revisions 339681 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r339681 | wedhorn | 2011-10-06 15:47:08 -0500 (Thu, 06 Oct 2011) | 10 lines Fixed segfault on core stop gracefully. There was an issue that the cap and confcap pointers for each line and device were being memcpy'd so they all pointed to the same ast_format_cap. On destroying, a segfault occured on the second call to the same struct. skinny reload now works again as well. Tested by snuff (in trunk) and myself. ........ * /, configure, include/asterisk/autoconfig.h.in, configure.ac, autoconf/ast_ext_lib.m4: Merged revisions 339720 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339720 | rmudgett | 2011-10-06 17:58:40 -0500 (Thu, 06 Oct 2011) | 27 lines Merged revisions 339719 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339719 | rmudgett | 2011-10-06 17:47:50 -0500 (Thu, 06 Oct 2011) | 20 lines Fix regression in configure script for libpri capability checks. JIRA AST-598 added the PRI_L2_PERSISTENCE option to fix BRI PTMP TE layer 2 persistence issues with some telcos. ASTERISK-18535 attempted to fix the unexpected requirement that libpri *must* have that feature to work with Asterisk. The AST_EXT_LIB_SETUP_DEPENDENT lines made the PRI optional features required. Unfortunately, I thought AST_EXT_LIB_SETUP_DEPENDENT didn't do anything useful for libpri and deleted those lines for libpri. The result was the HAVE_PRI_xxx defines that control the ability to use optional libpri features were also deleted. * Created AST_EXT_LIB_SETUP_OPTIONAL configuration macro to allow optional features in a library that the source code could take advantage of if the code supports the feature. (closes issue ASTERISK-18687) Reported by: Norbert Tested by: rmudgett ........ ................ 2011-10-06 20:18 +0000 [r339680] Damien Wedhorn * channels/chan_skinny.c: Fixed segfault on core stop gracefully. There was an issue that the cap and confcap pointers for each line and device were being memcpy'd so they all pointed to the same ast_format_cap. On destroying, a segfault occured on the second call to the same struct. skinny reload now works again as well. Tested by snuff and myself. 2011-10-06 17:54 +0000 [r339627] Richard Mudgett * main/udptl.c, /, channels/chan_sip.c: Merged revisions 339626 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339626 | rmudgett | 2011-10-06 12:53:00 -0500 (Thu, 06 Oct 2011) | 25 lines Merged revisions 339625 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339625 | rmudgett | 2011-10-06 12:49:38 -0500 (Thu, 06 Oct 2011) | 18 lines Fix debugging messages generated by 'udptl debug'. * Makes chan_sip set the tag to the channel name. * Fixes received debug message sequence number. * Removed tx/rx debug message type since it was hard coded to 0. * Made udptl.c logged message header consistent if possible: "UDPTL (%s): ". * Removed unused rx_expected_seq_no from struct ast_udptl. (closes issue ASTERISK-18401) Reported by: Kevin P. Fleming Patches: jira_asterisk_18401_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Matthew Nicholson ........ ................ 2011-10-06 13:43 +0000 [r339587] Leif Madsen * build_tools/prep_tarball: Merged revisions 339586 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339586 | lmadsen | 2011-10-06 08:43:21 -0500 (Thu, 06 Oct 2011) | 16 lines Merged revisions 339566 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339566 | lmadsen | 2011-10-05 16:30:11 -0500 (Wed, 05 Oct 2011) | 8 lines Update prep_tarball script to download pre-exported documentation. I've updated the prep_tarball script to now download the pre-exported documentation from the Asterisk wiki. This will give us more control over what is being included in the tarball releases, and will make both the PDF and HTML exported documentation look much better (especially when viewing from a console). (Closes issue ASTERISK-18677) ........ ................ 2011-10-05 17:02 +0000 [r339510-339513] Richard Mudgett * apps/app_dial.c, /: Merged revisions 339512 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339512 | rmudgett | 2011-10-05 12:01:46 -0500 (Wed, 05 Oct 2011) | 9 lines Merged revisions 339511 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339511 | rmudgett | 2011-10-05 12:01:01 -0500 (Wed, 05 Oct 2011) | 1 line Fix Dial F option notes formatting. ........ ................ * main/manager.c, /: Merged revisions 339508 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339508 | rmudgett | 2011-10-05 11:35:02 -0500 (Wed, 05 Oct 2011) | 18 lines Merged revisions 339504,339506 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339504 | rmudgett | 2011-10-05 11:26:45 -0500 (Wed, 05 Oct 2011) | 7 lines Add missing documentation of required AMI action Challenge AuthType header. (closes issue ASTERISK-18554) Reported by: Vlad Povorozniuc Patches: __20110919-manager-challenge-docs.patch.txt (license #4999) patch uploaded by Leif Madsen ........ r339506 | rmudgett | 2011-10-05 11:32:03 -0500 (Wed, 05 Oct 2011) | 1 line Fix XML error in AMI action Challenge. ........ ................ 2011-10-05 16:35 +0000 [r339509] Matthew Nicholson * /, res/res_fax.c: Merged revisions 339507 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339507 | mnicholson | 2011-10-05 11:32:59 -0500 (Wed, 05 Oct 2011) | 10 lines Merged revisions 339505 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339505 | mnicholson | 2011-10-05 11:31:21 -0500 (Wed, 05 Oct 2011) | 3 lines The app name in the documentation must match what we register the application as. ........ ................ 2011-10-05 06:50 +0000 [r339464-339465] Gregory Nietsky * res/res_fax.c, include/asterisk/res_fax.h, CHANGES: Add generic faxdetect framehook to res_fax Added func FAXOPT(faxdetect)=yes,cng,t38[,timeout]/no to enable dialplan faxdetect allowing more flexibility. as soon as a fax tone is detected the framehook is removed. there is a penalty involved in running this framehook on non G711 channels as they will be transcoded. CNG tone is suppresed using the SQUELCH flag to allow WaitForNoise to be run on the channel to detect Voice. (Closes issue ASTERISK-18569) Reported by: Myself Reviewed by: Matthew Nicholson, Kevin Fleming Review: https://reviewboard.asterisk.org/r/1116/ * /, res/res_fax.c: Merged revisions 339463 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r339463 | irroot | 2011-10-05 08:28:46 +0200 (Wed, 05 Oct 2011) | 9 lines Only change the capabilities on the gateway when the session is been destroyed there is still a race condition that ends in a segfault. if the caps are changed the logic in res_fax_spandsp will run T30 code not gateway code to end the session. this has been experienced on a "slower" under spec system. ........ 2011-10-04 22:59 +0000 [r339408] Richard Mudgett * Makefile, /: Merged revisions 339407 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339407 | rmudgett | 2011-10-04 17:56:25 -0500 (Tue, 04 Oct 2011) | 15 lines Merged revisions 339406 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339406 | rmudgett | 2011-10-04 17:54:15 -0500 (Tue, 04 Oct 2011) | 8 lines Make always create the MOH directory (/var/lib/asterisk/moh). (closes issue ASTERISK-18409) Reported by: abelbeck Patches: asterisk-1.8-makefile-moh.patch (license #5903) patch uploaded by abelbeck Tested by: abelbeck, Michael Keuter ........ ................ 2011-10-04 19:51 +0000 [r339315-339354] Jonathan Rose * /, main/say.c: Merged revisions 339353 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339353 | jrose | 2011-10-04 14:44:02 -0500 (Tue, 04 Oct 2011) | 18 lines Merged revisions 339352 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339352 | jrose | 2011-10-04 14:33:12 -0500 (Tue, 04 Oct 2011) | 12 lines Removes improper use of sound 'and' in German language mode from application saynumber Asterisk would say 'Five hundert und sechs und zwanzig' instead of 'Five hundert sechs und zwanzig'... which is both weird sounding and wrong. This patch makes sure Asterisk will only say the 'and' word between the single digit and double digit places. (closes issue ASTERISK-18212) Reported By: Lionel Elie Mamane Patches: upstream_germand_no_and.diff (License #5402) uploaded by Lionel Elie Mamane ........ ................ * /, res/res_jabber.c: Merged revisions 339298 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339298 | jrose | 2011-10-04 09:09:50 -0500 (Tue, 04 Oct 2011) | 19 lines Merged revisions 339297 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339297 | jrose | 2011-10-04 09:01:05 -0500 (Tue, 04 Oct 2011) | 13 lines Reverting revision 333265 due to component connection problems it introduces. I'm going to attempt some generic res_jabber cleanup and come up with a new fix for this problem, but first it seems prudent to remove this rather broad attempt to fix it and instead approach this problem either from the same angle but looking only at canceling (or possibly rescheduling) the send when we absolutely know it will cause a segfault or, if that can't be easily accomplished, strictly from the devstate side of things. Also, I'm pretty sure a lot of the code in res_jabber isn't thread safe. (issue ASTERISK-18626) (issue ASTERISK-18078) ........ ................ 2011-10-04 12:27 +0000 [r339262] Alexandr Anikin * /, addons/ooh323c/src/memheap.c: Merged revisions 339245 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339245 | may | 2011-10-04 15:49:49 +0400 (Tue, 04 Oct 2011) | 9 lines Merged revisions 339244 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339244 | may | 2011-10-04 15:44:55 +0400 (Tue, 04 Oct 2011) | 2 lines fix forget declaration in previous change ........ ................ 2011-10-04 09:43 +0000 [r339206] Olle Johansson * main/manager.c, CHANGES: Generate error message when AMI action originate extension doesn't exist Review: https://reviewboard.asterisk.org/r/1445/ Is this a bug or a new feature? No responses on Asterisk-dev so I'm committing to trunk only. 2011-10-03 20:13 +0000 [r339146-339149] Leif Madsen * channels/chan_sip.c: Merged revisions 339148 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339148 | lmadsen | 2011-10-03 15:13:16 -0500 (Mon, 03 Oct 2011) | 14 lines Merged revisions 339147 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339147 | lmadsen | 2011-10-03 15:12:43 -0500 (Mon, 03 Oct 2011) | 6 lines Remove duplicated Maxforwards line in AMI output. (Closes issue ASTERISK-18637) Reported by: Jacek Konieczny Patches: asterisk-sipshowpeer.patch (License #6298) uploaded by Jacek Konieczny ........ ................ * apps/app_dial.c: Merged revisions 339145 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339145 | lmadsen | 2011-10-03 14:55:15 -0500 (Mon, 03 Oct 2011) | 13 lines Merged revisions 339144 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339144 | lmadsen | 2011-10-03 14:54:52 -0500 (Mon, 03 Oct 2011) | 6 lines Make documentation for Dial() options 'F' and 'F()' more clear. (Closes issue ASTERISK-18646) Reported by: Physis Heckman Tested by: Richard Mudgett ........ ................ 2011-10-03 19:16 +0000 [r339091] Alexandr Anikin * /, addons/ooh323c/src/memheap.c: Merged revisions 339089 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339089 | may | 2011-10-03 22:52:55 +0400 (Mon, 03 Oct 2011) | 10 lines Merged revisions 339087 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339087 | may | 2011-10-03 22:42:49 +0400 (Mon, 03 Oct 2011) | 4 lines destroy memheap mutex properly before memheap deleted (fix memory leak occured after r304950 changes with DEBUG_THREAD compile option) ........ ................ 2011-10-03 18:58 +0000 [r339090] Terry Wilson * /, channels/chan_sip.c, main/file.c: Merged revisions 339088 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339088 | twilson | 2011-10-03 11:44:27 -0700 (Mon, 03 Oct 2011) | 17 lines Merged revisions 339086 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339086 | twilson | 2011-10-03 11:40:52 -0700 (Mon, 03 Oct 2011) | 10 lines Properly ignore AST_CONTROL_UPDATE_RTP_PEER in more places After the change in r336294, the new AST_CONTROL_UPDATE_RTP_PEER frame is sent when a re-invite happens. If we receive a re-invite from a device the waitstream_core was not aware of the new control frame and would drop the call. (closes issue ASTERISK-18610) Reported by: Kristijan_Vrban ........ ................ 2011-10-03 15:55 +0000 [r339021-339046] Matthew Nicholson * /, res/res_fax.c: Merged revisions 339045 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r339045 | mnicholson | 2011-10-03 10:54:55 -0500 (Mon, 03 Oct 2011) | 4 lines Ported ast_fax_caps_to_str() to 10, not sure why it wasn't already here. This function prints a list of caps instead of a hex bitfield. ........ * /, res/res_fax.c: Merged revisions 339043 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r339043 | mnicholson | 2011-10-03 10:41:36 -0500 (Mon, 03 Oct 2011) | 2 lines Don't clear the AST_FAX_TECH_MULTI_DOC flag right after we set it. ........ * /, res/res_fax.c: Merged revisions 339011 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r339011 | mnicholson | 2011-10-03 10:19:44 -0500 (Mon, 03 Oct 2011) | 2 lines properly remove the AST_FAX_TECH_GATEWAY flag (instead of setting all of the other flags) ........ 2011-10-03 14:40 +0000 [r338905-338998] Gregory Nietsky * /, CHANGES: Merged revisions 338997 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r338997 | irroot | 2011-10-03 16:38:25 +0200 (Mon, 03 Oct 2011) | 1 line Documentation noting the extension of CHANNEL() for chan_ooh323 ........ * addons/chan_ooh323.c, /, funcs/func_channel.c: Merged revisions 338995 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r338995 | irroot | 2011-10-03 16:21:40 +0200 (Mon, 03 Oct 2011) | 6 lines Remove the channel function OOH323() and place its options into CHANNEL() channel drivers should not have there own dialplan functions. ........ * /, res/res_fax.c: Merged revisions 338950 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r338950 | irroot | 2011-10-03 11:37:59 +0200 (Mon, 03 Oct 2011) | 14 lines Fixup a race condition in res_fax.c where FAXOPT(gateway)=no will turn off the gateway but the framehook is not destroyed. this problem happens when a gateway is attempted in the dialplan and the device is not available i may want to do fax to mail in the server it will not be allowed. instead of checking only AST_FAX_TECH_GATEWAY also check gateway_id Reverts 338904 Fix some white space. ........ * /, res/res_fax.c: Merged revisions 338904 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r338904 | irroot | 2011-10-02 16:17:32 +0200 (Sun, 02 Oct 2011) | 8 lines Remove T38 Gateway capability when detaching framehook. SET(FAXOPT(gateway)=no) does not remove the capability when detaching the framehook. small patch to fix this problem. ........ 2011-10-01 01:56 +0000 [r338855] TransNexus OSP Development * configure: Update "configure" based on r338139. 2011-09-30 22:08 +0000 [r338802] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 338801 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r338801 | rmudgett | 2011-09-30 17:06:48 -0500 (Fri, 30 Sep 2011) | 19 lines Merged revisions 338800 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338800 | rmudgett | 2011-09-30 17:05:10 -0500 (Fri, 30 Sep 2011) | 12 lines Fix segfault in analog_ss_thread() not checking ast_read() for NULL. NOTE: The problem was reported against v1.6.2. It is unlikely to ever happen on v1.8 and above since chan_dahdi.c:analog_ss_thread() is unlikely to be used. The version in sig_analog.c has largely replaced it. (closes issue ASTERISK-18648) Reported by: Stephan Bosch Patches: jira_asterisk_18648_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Stephan Bosch ........ ................ 2011-09-30 19:25 +0000 [r338755] Olle Johansson * channels/chan_sip.c: Formatting changes only --Denna och nedanstående rader kommer inte med i loggmeddelandet-- M channels/chan_sip.c 2011-09-30 18:59 +0000 [r338720] Jonathan Rose * /, configs/queues.conf.sample: Merged revisions 338719 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r338719 | jrose | 2011-09-30 13:55:27 -0500 (Fri, 30 Sep 2011) | 9 lines Merged revisions 338718 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338718 | jrose | 2011-09-30 13:54:30 -0500 (Fri, 30 Sep 2011) | 1 line Adds documentation for QueueMemberStatus event generation ........ ................ 2011-09-30 16:40 +0000 [r338665] Richard Mudgett * /, channels/chan_sip.c: Fix formatting of AMI header for SIP show peer. ASTERISK-17486 exposed the problem for AMI parsers. (closes issue ASTERISK-18649) Reported by: Jacek Konieczny Patches: asterisk-sipshowpeer_response_end.patch (license #6298) patch uploaded by Jacek Konieczny ........ Merged revisions 338663 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 338664 from http://svn.asterisk.org/svn/asterisk/branches/10 2011-09-30 13:21 +0000 [r338623] Olle Johansson * main/features.c: Preserve DTMF length in main/features.c Review: https://reviewboard.asterisk.org/r/1463/ A small part of much larger work with DTMF duration in Asterisk, funded by IPvision AS in Denmark. Thanks to irroot for the review! 2011-09-29 21:16 +0000 [r338557] Paul Belanger * tests/test_security_events.c, /, tests/test_locale.c, tests/test_logger.c, tests/test_dlinklists.c, tests/test_linkedlists.c, tests/test_amihooks.c: Merged revisions 338556 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r338556 | pabelanger | 2011-09-29 17:14:34 -0400 (Thu, 29 Sep 2011) | 9 lines Merged revisions 338555 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338555 | pabelanger | 2011-09-29 17:12:21 -0400 (Thu, 29 Sep 2011) | 2 lines Test modules should depend on the TEST_FRAMEWORK flag ........ ................ 2011-09-29 20:55 +0000 [r338553] Jason Parker * /, tests/test_db.c, tests/test_netsock2.c: Merged revisions 338552 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r338552 | qwell | 2011-09-29 15:54:55 -0500 (Thu, 29 Sep 2011) | 9 lines Merged revisions 338551 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338551 | qwell | 2011-09-29 15:54:13 -0500 (Thu, 29 Sep 2011) | 1 line Test modules have a support level of core. ........ ................ 2011-09-29 12:22 +0000 [r338435] Gregory Nietsky * /, channels/chan_sip.c, channels/sip/include/sip.h: Merged revisions 338417 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r338417 | irroot | 2011-09-29 14:16:42 +0200 (Thu, 29 Sep 2011) | 19 lines Merged revisions 338416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338416 | irroot | 2011-09-29 14:13:05 +0200 (Thu, 29 Sep 2011) | 12 lines The rtptimeout setting is ignored on a per peer basis. Not only is the rtptimeout ignored in some cases but rtpkeepalive and rtpholdtimeout is affected. this commit also removes rtptimeout/rtpholdtimeout on text rtp. (closes issue ASTERISK-18559) Review: https://reviewboard.asterisk.org/r/1452 ........ ................ 2011-09-29 12:03 +0000 [r338377-338415] Olle Johansson * cdr/cdr_pgsql.c, CHANGES: Add CLI command "cdr show pgsql status" based on "cdr mysql status" Review: https://reviewboard.asterisk.org/r/923/ Thanks all for the code reviews and feedback. * res/res_agi.c: Just formatting. 2011-09-28 22:38 +0000 [r338284-338324] Richard Mudgett * /, channels/sig_pri.c: Merged revisions 338323 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r338323 | rmudgett | 2011-09-28 17:36:57 -0500 (Wed, 28 Sep 2011) | 12 lines Merged revisions 338322 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338322 | rmudgett | 2011-09-28 17:35:52 -0500 (Wed, 28 Sep 2011) | 5 lines Make duplicate call ptr warning message more helpful. * Adds the value of the call ptr to the duplicate call ptr message to help trace why there is a duplicate call ptr. ........ ................ * include/asterisk/logger.h, /: Merged revisions 338253 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r338253 | rmudgett | 2011-09-28 16:22:05 -0500 (Wed, 28 Sep 2011) | 14 lines Merged revisions 338235 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338235 | rmudgett | 2011-09-28 16:17:45 -0500 (Wed, 28 Sep 2011) | 7 lines Fix inconsistency in LOG_VERBOSE/AST_LOG_VERBOSE declaration. (closes issue ASTERISK-17973) Reported by: Luke H Patches: logger_h.patch (license #6278) patch uploaded by Luke H ........ ................ 2011-09-28 20:55 +0000 [r338229] Jason Parker * build_tools/cflags.xml, channels/chan_usbradio.c, build_tools/cflags-devmode.xml, agi/agi.xml, utils/utils.xml, /, build_tools/embed_modules.xml, tests/test_db.c, tests/test_netsock2.c: Merged revisions 338228 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r338228 | qwell | 2011-09-28 15:54:35 -0500 (Wed, 28 Sep 2011) | 9 lines Merged revisions 338227 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338227 | qwell | 2011-09-28 15:52:47 -0500 (Wed, 28 Sep 2011) | 1 line Add support levels to non-module sections of menuselect (cflags, utils, etc). ........ ................ 2011-09-28 20:28 +0000 [r338226] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 338225 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r338225 | rmudgett | 2011-09-28 15:26:39 -0500 (Wed, 28 Sep 2011) | 12 lines Merged revisions 338224 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338224 | rmudgett | 2011-09-28 15:24:41 -0500 (Wed, 28 Sep 2011) | 5 lines Fix chan_dahd compiling with gcc 4.6 when PRI and SS7 not present. (closes issue ASTERISK-18357) Reported by: Matthew Nicholson ........ ................ 2011-09-28 17:00 +0000 [r338187-338188] Terry Wilson * CHANGES: Update CHANGES to reflect autopausebusy not being in Asterisk 10 * configs/queues.conf.sample, CHANGES, apps/app_queue.c: Add autopausebusy and autopauseunavail queue options Make it possible to autopause on a busy or unavailable response from a device. (closes issue ASTERISK-16112) Reported by: jlpedrosa Patches: autopausebusy.txt by twilson Review: https://reviewboard.asterisk.org/r/1399/ 2011-09-28 07:30 +0000 [r338136-338139] TransNexus OSP Development * configure.ac: Updated for checking OSP Toolkit version 4.0.0. * apps/app_osplookup.c: Updated for OSP Toolkit 4.0.0. 2011-09-27 20:15 +0000 [r338086] Paul Belanger * /, apps/app_macro.c: Merged revisions 338085 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r338085 | pabelanger | 2011-09-27 16:13:14 -0400 (Tue, 27 Sep 2011) | 9 lines Merged revisions 338084 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338084 | pabelanger | 2011-09-27 16:10:13 -0400 (Tue, 27 Sep 2011) | 2 lines Upgrade app_macro to core ........ ................ 2011-09-27 12:45 +0000 [r338042] Olle Johansson * channels/chan_sip.c: Whitespace (red blobs) fixes 2011-09-26 19:40 +0000 [r337975] Richard Mudgett * apps/app_dial.c, main/pbx.c, cdr/cdr_sqlite3_custom.c, /, include/asterisk/cel.h, cdr/cdr_syslog.c, tests/test_gosub.c, include/asterisk/channel.h, main/cel.c, main/manager.c, funcs/func_odbc.c, cel/cel_custom.c, apps/app_minivm.c, main/logger.c, cel/cel_sqlite3_custom.c, cdr/cdr_custom.c, cdr/cdr_manager.c, apps/app_voicemail.c: Merged revisions 337974 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337974 | rmudgett | 2011-09-26 14:35:23 -0500 (Mon, 26 Sep 2011) | 37 lines Merged revisions 337973 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337973 | rmudgett | 2011-09-26 14:30:39 -0500 (Mon, 26 Sep 2011) | 30 lines Fix deadlock when using dummy channels. Dummy channels created by ast_dummy_channel_alloc() should be destoyed by ast_channel_unref(). Using ast_channel_release() needlessly grabs the channel container lock and can cause a deadlock as a result. * Analyzed use of ast_dummy_channel_alloc() and made use ast_channel_unref() when done with the dummy channel. (Primary reason for the reported deadlock.) * Made app_dial.c:dial_exec_full() not call ast_call() holding any channel locks. Chan_local could not perform deadlock avoidance correctly. (Potential deadlock exposed by this issue. Secondary reason for the reported deadlock since the held lock was part of the deadlock chain.) * Fixed some uses of ast_dummy_channel_alloc() not checking the returned channel pointer for failure. * Fixed some potential chan=NULL pointer usage in func_odbc.c. Protected by testing the bogus_chan value. * Fixed needlessly clearing a 1024 char auto array when setting the first char to zero is enough in manager.c:action_getvar(). (closes issue ASTERISK-18613) Reported by: Thomas Arimont Patches: jira_asterisk_18613_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Thomas Arimont ........ ................ 2011-09-23 19:20 +0000 [r337855-337910] Gregory Nietsky * /, contrib/init.d/rc.archlinux.asterisk: Merged revisions 337902 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337902 | irroot | 2011-09-23 21:18:14 +0200 (Fri, 23 Sep 2011) | 10 lines Merged revisions 337898 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337898 | irroot | 2011-09-23 21:14:30 +0200 (Fri, 23 Sep 2011) | 4 lines Spelling fix ........ ................ * /, apps/app_queue.c: Merged revisions 337840 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337840 | irroot | 2011-09-23 10:39:22 +0200 (Fri, 23 Sep 2011) | 17 lines Merged revisions 337839 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337839 | irroot | 2011-09-23 10:34:03 +0200 (Fri, 23 Sep 2011) | 11 lines Make sure a CDR is on the stack for call in the Queue. Only let update_cdr act on the last CDR in the stack. In some circumstances [Attended transfer to queue] a CDR record is not inserted for this call where it should. (closes issue ASTERISK-18567) Review: https://reviewboard.asterisk.org/r/1266 ........ ................ 2011-09-23 00:47 +0000 [r337776] Russell Bryant * /, configs/res_pktccops.conf.sample: Merged revisions 337775 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337775 | russell | 2011-09-22 19:45:35 -0500 (Thu, 22 Sep 2011) | 18 lines Merged revisions 337774 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337774 | russell | 2011-09-22 19:44:19 -0500 (Thu, 22 Sep 2011) | 11 lines Comment out entries in sample res_pktccops.conf. With these options enabled, they can cause Asterisk to freak out by SYN flooding a network and eating the CPU. Obviously it would be good to fix the code so that this can't happen, but we can at least change the default configuration so it doesn't happen. This was reported downstream to the Fedora issue tracker: https://bugzilla.redhat.com/show_bug.cgi?id=658431 ........ ................ 2011-09-22 21:42 +0000 [r337722] Richard Mudgett * /, channels/sig_pri.c: Merged revisions 337721 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337721 | rmudgett | 2011-09-22 16:37:41 -0500 (Thu, 22 Sep 2011) | 25 lines Merged revisions 337720 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337720 | rmudgett | 2011-09-22 16:29:46 -0500 (Thu, 22 Sep 2011) | 18 lines Made ISDN not add numbering plan prefix strings to empty numbers. When the Caller-ID is restricted, the expected behavior is for the Caller-ID to be blank. In chan_dahdi, the national prefix is placed onto the Caller-ID number even if it is restricted (empty) causing the Caller-ID to be the national prefix rather than blank. This behavior was lost when sig_pri was extracted from chan_dahdi. * Made not add prefix strings to empty connected line, calling, and ANI number strings. (closes issue ASTERISK-18577) Reported by: Kris Shaw Patches: jira_asterisk_18577_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Kris Shaw ........ ................ 2011-09-22 16:35 +0000 [r337600] Jonathan Rose * /, channels/chan_sip.c, include/asterisk/event_defs.h, main/security_events.c, channels/sip/security_events.c (added), main/event.c, CHANGES, channels/sip/include/security_events.h (added), channels/sip/include/sip.h, include/asterisk/security_events_defs.h, configs/logger.conf.sample: Merged revisions 337595,337597 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r337595 | jrose | 2011-09-22 10:35:50 -0500 (Thu, 22 Sep 2011) | 12 lines Generate Security events in chan_sip using new Security Events Framework Security Events Framework was added in 1.8 and support was added for AMI to generate events at that time. This patch adds support for chan_sip to generate security events. (closes issue ASTERISK-18264) Reported by: Michael L. Young Patches: security_events_chan_sip_v4.patch (license #5026) by Michael L. Young Review: https://reviewboard.asterisk.org/r/1362/ ........ r337597 | jrose | 2011-09-22 10:47:05 -0500 (Thu, 22 Sep 2011) | 10 lines Forgot to svn add new files to r337595 Part of Generating security events for chan_sip (issue ASTERISK-18264) Reported by: Michael L. Young Patches: security_events_chan_sip_v4.patch (License #5026) by Michael L. Young Reviewboard: https://reviewboard.asterisk.org/r/1362/ ........ 2011-09-22 11:46 +0000 [r337432-337543] Gregory Nietsky * /, res/res_srtp.c: Merged revisions 337542 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337542 | irroot | 2011-09-22 13:44:22 +0200 (Thu, 22 Sep 2011) | 14 lines Merged revisions 337541 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337541 | irroot | 2011-09-22 13:39:49 +0200 (Thu, 22 Sep 2011) | 8 lines Add warned to ast_srtp to prevent errors on each frame from libsrtp The first 9 frames are not reported as some devices dont use srtp from first frame these are suppresed. the warning is then output only once every 100 frames. ........ ................ * /, channels/chan_h323.c: Merged revisions 337487 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337487 | irroot | 2011-09-22 11:26:26 +0200 (Thu, 22 Sep 2011) | 16 lines Merged revisions 337486 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337486 | irroot | 2011-09-22 11:22:26 +0200 (Thu, 22 Sep 2011) | 10 lines If IP address is used in chan_h323 host parameter of peer configuration. module tries to resolve IP address to IP address and fails. Simple fix to set family of socket this is a hangover from ipv6 changes. (closes issue ASTERISK-18237) (issue ASTERISK-17278) (issue ASTERISK-17500) ........ ................ * main/channel.c, /: Merged revisions 337431 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337431 | irroot | 2011-09-22 08:29:09 +0200 (Thu, 22 Sep 2011) | 25 lines Merged revisions 337430 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337430 | irroot | 2011-09-22 08:18:33 +0200 (Thu, 22 Sep 2011) | 19 lines Its possible to loose audio on ast_write when the channel is not transcoded correctly. in the case of DAHDI the channel is hungup. This patch tries to "fix" the problem and make the channel compatiable and warn the user of this problem. Please note there is a underlying problem with codec negotion this does not fix the problem it does try to rectify it and prevent loss of service. Review: https://reviewboard.asterisk.org/r/1442/ (closes issue ASTERISK-17541) (closes issue ASTERISK-18063) (issue ASTERISK-14384) (issue ASTERISK-17502) (issue ASTERISK-18325) (issue ASTERISK-18422) ........ ................ 2011-09-21 21:26 +0000 [r337343-337385] Tilghman Lesher * /, apps/app_voicemail.c: More silly spacing changes ..... Merged revisions 337353 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ..... Merged revisions 337380 from http://svn.asterisk.org/svn/asterisk/branches/10 * /, apps/app_voicemail.c: ................ ........ Dumb little spacing fix. ........ Merged revisions 337344 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ Merged revisions 337345 from http://svn.asterisk.org/svn/asterisk/branches/10 * funcs/func_curl.c, /: ................ ........ Escape commas in keys and values, when keys and values are enumerated by commas. Review: https://reviewboard.asterisk.org/r/1433 ........ Merged revisions 337325 from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ Merged revisions 337342 from https://origsvn.digium.com/svn/asterisk/branches/10 2011-09-21 11:21 +0000 [r337262-337283] Gregory Nietsky * /, configs/sip.conf.sample: Merged revisions 337263 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r337263 | irroot | 2011-09-21 13:15:48 +0200 (Wed, 21 Sep 2011) | 1 line Whitespace fixup from SRTP patch ........ * /, apps/app_originate.c, CHANGES: Merged revisions 337261 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r337261 | irroot | 2011-09-21 12:42:06 +0200 (Wed, 21 Sep 2011) | 10 lines Adds a timeout argument to app_originate the default is 30s this will be used if the timout supplied is invalid or no timeout is supplied. Contributed by: jacco (thank you for the work) Review: https://reviewboard.asterisk.org/r/1310/ ........ 2011-09-21 09:39 +0000 [r337179-337220] Olle Johansson * main/pbx.c, /, CHANGES, configs/extensions.conf.sample: Merged revisions 337219 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r337219 | oej | 2011-09-21 11:32:50 +0200 (Ons, 21 Sep 2011) | 13 lines Make ast_pbx_run() not default to s@default if extension is not found Review: https://reviewboard.asterisk.org/r/1446/ This is a bug - or architecture mistake - that has been in Asterisk for a very long time. It was exposed by the AMI originate action and possibly some other applications. Most channel drivers checks if an extension exists BEFORE starting a pbx on an inbound call, so most calls will not depend on this issue. Thanks everyone involved in the review and on IRC and the mailing list for a quick review and all the feedback. (closes issue ASTERISK-18578) ........ * res/res_rtp_asterisk.c, /, configs/rtp.conf.sample, CHANGES: Merged revisions 337178 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r337178 | oej | 2011-09-21 10:51:41 +0200 (Ons, 21 Sep 2011) | 14 lines Change strictrtp option to default to yes in the RTP module Suggested by Kapejod on Facebook Review: https://reviewboard.asterisk.org/r/1448/ (closes issue ASTERISK-18587) Thanks for quick feedback to kpfleming and Tilghman --Denna och nedanstående rader kommer inte med i loggmeddelandet-- M CHANGES M configs/rtp.conf.sample M res/res_rtp_asterisk.c ........ 2011-09-20 23:02 +0000 [r337124] Matthew Jordan * apps/app_dial.c, include/asterisk/app.h, apps/app_meetme.c, apps/app_minivm.c, main/app.c, apps/app_confbridge.c, apps/app_followme.c, apps/app_voicemail.c: Merged revisions 337120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337120 | mjordan | 2011-09-20 17:49:36 -0500 (Tue, 20 Sep 2011) | 28 lines Merged revisions 337118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337118 | mjordan | 2011-09-20 17:38:54 -0500 (Tue, 20 Sep 2011) | 21 lines Fix for incorrect voicemail duration in external notifications This patch fixes an issue where the voicemail duration was being reported with a duration significantly less than the actual sound file duration. Voicemails that contained mostly silence were reporting the duration of only the sound in the file, as opposed to the duration of the file with the silence. This patch fixes this by having two durations reported in the __ast_play_and_record family of functions - the sound_duration and the actual duration of the file. The sound_duration, which is optional, now reports the duration of the sound in the file, while the actual full duration of the file is reported in the duration parameter. This allows the voicemail applications to use the sound_duration for minimum duration checking, while reporting the full duration to external parties if the voicemail is kept. (issue ASTERISK-2234) (closes issue ASTERISK-16981) Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1443 ........ ................ 2011-09-20 22:54 +0000 [r337121-337123] Richard Mudgett * /, funcs/func_strings.c: Merged revisions 337119 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r337119 | rmudgett | 2011-09-20 17:47:45 -0500 (Tue, 20 Sep 2011) | 16 lines Fix crash with STRREPLACE function. The ast_func_read() function calls the .read2 callback with the len parameter set to zero indicating no size restrictions on the supplied ast_str buffer. The value was used to dimension a local starts[] array with the array subsequently used. * Reworked the strreplace() function to perform the string replacement in a straight forward manner. Eliminated the need for the starts[] array. (closes issue ASTERISK-18545) Reported by: Federico Alves Patches: jira_asterisk_18545_v10.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett, Federico Alves ........ * /: Updated 10 merge property. * /: Restore branch-10 merge properties. 2011-09-20 22:29 +0000 [r337117] Leif Madsen * /, contrib/init.d/rc.redhat.asterisk: Merged revisions 337115 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337115 | lmadsen | 2011-09-20 17:18:25 -0500 (Tue, 20 Sep 2011) | 7 lines Update RedHat Init script to work with Heartbeat. The current RedHat init script was not LSB compatible. This change will make it LSB compatible so that it can work correctly with Heartbeat. (Closes issue ASTERISK-18253) Reported by: c0rnoTa ........ 2011-09-20 21:05 +0000 [r337063] Kinsey Moore * main/pbx.c, /, tests/test_pbx.c: Merged revisions 337062 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337062 | kmoore | 2011-09-20 16:05:01 -0500 (Tue, 20 Sep 2011) | 18 lines Merged revisions 337061 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337061 | kmoore | 2011-09-20 16:04:11 -0500 (Tue, 20 Sep 2011) | 11 lines Make CANMATCH with the new pattern match engine behave more like the old one When checking an extension for E_CANMATCH using the new extension matching algorithm, an exact match was not returned as a possible match resulting in the queue failing to allow a caller to exit on DTMF. This removes the requirement that an extension be longer than acquired digits for an E_CANMATCH operation to succeed. (closes issue ASTERISK-18044) Review: https://reviewboard.asterisk.org/r/1367/ ........ ................ 2011-09-20 19:13 +0000 [r336988-337009] Richard Mudgett * /, channels/sig_ss7.c: Merged revisions 337008 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337008 | rmudgett | 2011-09-20 14:12:24 -0500 (Tue, 20 Sep 2011) | 22 lines Merged revisions 337007 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337007 | rmudgett | 2011-09-20 14:10:30 -0500 (Tue, 20 Sep 2011) | 15 lines Check if a channel was created before using the pointer in sig_ss7_new_ast_channel(). Fixes the crash in ASTERISK-17955 gdb-11918.txt backtrace. * Added some missing libss7 access lock protection. * Prevent cancelling the ss7_linkset() thread at inoportune times just like the pri_dchannel() thread. (issue ASTERISK-17955) Reported by: Ian M Sherman Patches: jira_asterisk_17955_v1.8.patch (license #5621) patch uploaded by rmudgett (attached to related ASTERISK-17966) ........ ................ * /, channels/sig_ss7.c: Merged revisions 336978 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336978 | rmudgett | 2011-09-20 13:14:40 -0500 (Tue, 20 Sep 2011) | 28 lines Merged revisions 336977 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336977 | rmudgett | 2011-09-20 13:12:17 -0500 (Tue, 20 Sep 2011) | 21 lines Fix deadlock from not releasing SS7 linkset lock. sig_ss7_hangup() failed to release the SS7 linkset lock if the call had the alreadyhungup flag set. * Made unlock the SS7 linkset lock in sig_ss7_hangup() if the alreadyhungup flag is set. * Made ss7_start_call() not hold any locks while creating the channel for an incoming call to prevent deadlock. * Made ss7_grab() a void function, since it could never fail, to simplify calling code. * Made obtain the channel lock to do softhangup in some places. Patches: jira_ast_668_v1.8.patch (license #5621) patch uploaded by rmudgett JIRA AST-668 ........ ................ 2011-09-20 16:56 +0000 [r336937] Gregory Nietsky * channels/sip/sdp_crypto.c, /, channels/chan_sip.c, channels/sip/include/sdp_crypto.h, channels/sip/include/srtp.h, configs/sip.conf.sample, CHANGES, channels/sip/include/sip.h: Merged revisions 336936 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r336936 | irroot | 2011-09-20 18:51:59 +0200 (Tue, 20 Sep 2011) | 14 lines Allow Setting Auth Tag Bit length Based on invite or config option Update the SIP SRTP API to allow use of 32 or 80 bit taglen. Curently only 80 bit is supported. The outgoing invite will use the taglen of the incoming invite preventing one-way audio. (Closes issue ASTERISK-17895) Review: https://reviewboard.asterisk.org/r/1173/ ........ 2011-09-20 01:11 +0000 [r336879] Russell Bryant * res/res_rtp_asterisk.c, /: Merged revisions 336878 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336878 | russell | 2011-09-19 20:03:55 -0500 (Mon, 19 Sep 2011) | 43 lines Merged revisions 336877 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336877 | russell | 2011-09-19 19:56:20 -0500 (Mon, 19 Sep 2011) | 36 lines Fix crashes in ast_rtcp_write(). This patch addresses crashes related to RTCP handling. The backtraces just show a crash in ast_rtcp_write() where it appears that the RTP instance is no longer valid. There is a race condition with scheduled RTCP transmissions and the destruction of the RTP instance. This patch utilizes the fact that ast_rtp_instance is a reference counted object and ensures that it will not get destroyed while a reference is still around due to scheduled RTCP transmissions. RTCP transmissions are scheduled and executed from the chan_sip scheduler context. This scheduler context is processed in the SIP monitor thread. The destruction of an RTP instance occurs when the associated sip_pvt gets destroyed (which happens when the sip_pvt reference count reaches 0). However, the SIP monitor thread is not the only thread that can cause a sip_pvt to get destroyed. The sip_hangup function, executed from a channel thread, also decrements the reference count on a sip_pvt and could cause it to get destroyed. While this is being changed anyway, the patch also removes calling ast_sched_del() from within the RTCP scheduler callback. It's not helpful. Simply returning 0 prevents the callback from being rescheduled. (closes issue ASTERISK-18570) Related issues that look like they are the same problem: (issue ASTERISK-17560) (issue ASTERISK-15406) (issue ASTERISK-15257) (issue ASTERISK-13334) (issue ASTERISK-9977) (issue ASTERISK-9716) Review: https://reviewboard.asterisk.org/r/1444/ ........ ................ 2011-09-19 22:28 +0000 [r336837] Terry Wilson * /, channels/chan_sip.c: Merged revisions 336792 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336792 | twilson | 2011-09-19 17:13:34 -0500 (Mon, 19 Sep 2011) | 9 lines Merged revisions 336791 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336791 | twilson | 2011-09-19 17:07:58 -0500 (Mon, 19 Sep 2011) | 2 lines Don't interfere with T.38 reinvites This is an update to the fix for ASTERISK-18340 and ASTERISK-17725 ........ ................ 2011-09-19 21:42 +0000 [r336735-336790] Tilghman Lesher * /, funcs/func_strings.c: Merged revisions 336789 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r336789 | tilghman | 2011-09-19 16:41:16 -0500 (Mon, 19 Sep 2011) | 2 lines Ensure substring will not be found in the previous match. ........ * Makefile, /, configure, include/asterisk/autoconfig.h.in, main/Makefile, codecs/gsm/Makefile, configure.ac, Makefile.rules, include/asterisk/optional_api.h: Merged revisions 336734 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336734 | tilghman | 2011-09-19 15:29:40 -0500 (Mon, 19 Sep 2011) | 18 lines Merged revisions 336733 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336733 | tilghman | 2011-09-19 15:27:03 -0500 (Mon, 19 Sep 2011) | 11 lines Various changes to allow 1.8 to compile on Mac OS X Lion (10.7) * Makefile workaround for 10.6 extended to work on 10.7 and later. * Now uses the 'weak' symbol for Lion systems, which no longer support 'weak_import' Closes ASTERISK-17612. Closes ASTERISK-18213. Tested by: tilghman, oej. ........ ................ 2011-09-19 20:23 +0000 [r336732] Jonathan Rose * /, apps/app_echo.c, apps/app_saycounted.c, apps/app_mp3.c, apps/app_morsecode.c, res/res_musiconhold.c, apps/app_queue.c, apps/app_mixmonitor.c: Merged revisions 336717 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336717 | jrose | 2011-09-19 15:16:23 -0500 (Mon, 19 Sep 2011) | 14 lines Merged revisions 336716 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336716 | jrose | 2011-09-19 15:07:36 -0500 (Mon, 19 Sep 2011) | 7 lines Document applications that play audio and do not answer unanswered calls. This patch is part of an effort to document early media and its usage. If you are interested in contributing to this documentation effort, there are probably other applications worth documenting as well as an Asterisk wiki article at https://wiki.asterisk.org/wiki/display/AST/Early+Media+and+the+Progress+Application ........ ................ 2011-09-19 19:03 +0000 [r336660-336662] Richard Mudgett * apps/app_dial.c, /, UPGRADE-1.8.txt: Merged revisions 336659 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336659 | rmudgett | 2011-09-19 13:51:19 -0500 (Mon, 19 Sep 2011) | 38 lines Merged revisions 336658 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336658 | rmudgett | 2011-09-19 13:46:40 -0500 (Mon, 19 Sep 2011) | 31 lines Made Dial d and H options no longer immediately auto-answer the calling leg. The Dial d and H options break DTMF attended transfer atxferdropcall option. 1) Party A calls party B. 2) Party B does a DTMF attended transfer to Party C. If the dialplan uses the Dial d or H options to call Party C then the Dial application answers the call immediately before initiating the call leg to Party C. The premature answer causes the transfer code to not invoke the atxferdropcall=no behavior for a blonde transfer since Party C has "answered". The transfer code thinks that Party B has "consulted" with Party C when Party B hangs up and completes the transfer to Party A. Party A now hears ringback until Party C actually answers. ASTERISK-13294 Dial d option. ASTERISK-11067 Dial H option to disconnect before answer. The referenced issues made Dial answer with the d and H options because many SIP and ISDN phones cannot send DTMF before the call is connected. * Made require the dialplan to control when or if the call needs to be answered to use the Dial application d and H options. (The call is no longer surprise answered when using the Dial d or H options.) Review: https://reviewboard.asterisk.org/r/1381/ JIRA AST-623 JIRA AST-666 ........ ................ * /: Update merge 10 branch merge propterty. * /: Restore 10 branch merge properties. 2011-09-19 16:22 +0000 [r336600] Jason Parker * cel/cel_odbc.c, configs/cel_odbc.conf.sample, sounds/Makefile: Remove weird mergeinfo props that make merges annoying sometimes. 2011-09-19 15:48 +0000 [r336574] Leif Madsen * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 336572 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336572 | lmadsen | 2011-09-19 10:41:16 -0500 (Mon, 19 Sep 2011) | 7 lines Update get_ilbc_source.sh script to work again. Recently iLBC support in Asterisk has changed after the acquisition of GIPS by Google. More information about how this may affect you is available in a blog post at: http://blogs.asterisk.org/2011/09/19/ilbc-support-in-asterisk-after-googles-acquisition-of-gips/ ........ 2011-09-19 15:36 +0000 [r336571] Richard Mudgett * /, channels/sig_pri.c: Merged revisions 336570 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336570 | rmudgett | 2011-09-19 10:32:00 -0500 (Mon, 19 Sep 2011) | 11 lines Merged revisions 336569 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336569 | rmudgett | 2011-09-19 10:25:34 -0500 (Mon, 19 Sep 2011) | 4 lines Rework sig_pri_hangup() to be simpler and clearer. JIRA AST-675 ........ ................ 2011-09-19 13:57 +0000 [r336505] Olle Johansson * /, channels/chan_sip.c: Merged revisions 336502 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336502 | oej | 2011-09-19 15:38:53 +0200 (Mån, 19 Sep 2011) | 12 lines Merged revisions 336501 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336501 | oej | 2011-09-19 15:33:50 +0200 (Mån, 19 Sep 2011) | 5 lines Add diversion header to a 302 redirect response if we have diversion data (closes issue ASTERISK-18143) patch by oej ........ ................ 2011-09-19 13:41 +0000 [r336503] Gregory Nietsky * /, channels/chan_h323.c: Merged revisions 336500 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336500 | irroot | 2011-09-19 15:31:50 +0200 (Mon, 19 Sep 2011) | 19 lines Merged revisions 336499 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336499 | irroot | 2011-09-19 15:27:52 +0200 (Mon, 19 Sep 2011) | 13 lines A long time ago in a galaxy far far away a IPv6 update was made, chan_h323 was not updated causeing all to flee to chan_ooh323. the brave Jedi [asterisk developers] pondered this miscarrige of justice and restored order to the force for the sake of closing out 2 old issues. (closes issue ASTERISK-17278) (closes issue ASTERISK-17500) Reported by: dread, sybasesql Tested by: irroot Reviewed by: IRC (russellb, kpfleming) ........ ................ 2011-09-19 12:20 +0000 [r336382-336453] Olle Johansson * main/manager.c, /: Merged revisions 336441 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336441 | oej | 2011-09-19 14:15:06 +0200 (Mån, 19 Sep 2011) | 9 lines Merged revisions 336440 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336440 | oej | 2011-09-19 14:06:48 +0200 (Mån, 19 Sep 2011) | 2 lines Make sure manager_debug option is reset at reload ........ ................ * /, channels/chan_sip.c: Merged revisions 336381 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336381 | oej | 2011-09-19 12:05:00 +0200 (Mån, 19 Sep 2011) | 16 lines Merged revisions 336378 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336378 | oej | 2011-09-19 11:40:44 +0200 (Mån, 19 Sep 2011) | 9 lines Add missing unlock at MWI message sending time (closes issue ASTERISK-18573) Patches: sip_mwi_lock.patch (license #5041) by Gregory Hinton Nietsky Thanks to irrot for the reminder, to Gregory for the patch! ........ ................ 2011-09-16 22:12 +0000 [r336315-336317] Terry Wilson * /, funcs/func_frame_trace.c: Merged revisions 336316 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336316 | twilson | 2011-09-16 17:11:39 -0500 (Fri, 16 Sep 2011) | 9 lines Merged revisions 336314 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336314 | twilson | 2011-09-16 17:10:56 -0500 (Fri, 16 Sep 2011) | 2 lines Whitespace fix ........ ................ * /, funcs/func_frame_trace.c: Merged revisions 336313 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336313 | twilson | 2011-09-16 17:07:00 -0500 (Fri, 16 Sep 2011) | 12 lines Merged revisions 336312 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336312 | twilson | 2011-09-16 17:04:25 -0500 (Fri, 16 Sep 2011) | 5 lines Add missing frame types to func_frame_trace Also casts control frames to the proper enum so that the compile will catch new additions. ........ ................ 2011-09-16 21:20 +0000 [r336311] Jonathan Rose * main/channel.c, main/rtp_engine.c, /, channels/chan_sip.c, include/asterisk/frame.h: Merged revisions 336307 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336307 | jrose | 2011-09-16 16:09:20 -0500 (Fri, 16 Sep 2011) | 20 lines Merged revisions 336294 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336294 | jrose | 2011-09-16 14:53:40 -0500 (Fri, 16 Sep 2011) | 13 lines Fix bad RTP media bridges in directmedia calls on peers separated by multiple Asterisk nodes. In a situation involving devices on separate Asterisk trunks, the remote RTP bridge would break when starting a call with directmedia. This patch queues a new type of control frame so that our RTP bridge loop can properly detect when these situations occur and check to see if peers need to be updated in order to send their media to the proper location. (Closes issue ASTERISK-18340) Reported by: Thomas Arimont (Closes issue ASTERISK-17725) Reported by: kwk Tested by: twilson, jrose ........ ................ 2011-09-16 19:11 +0000 [r336236] Sean Bright * /, UPGRADE-1.8.txt: Merged revisions 336235 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336235 | seanbright | 2011-09-16 15:10:39 -0400 (Fri, 16 Sep 2011) | 9 lines Merged revisions 336234 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336234 | seanbright | 2011-09-16 15:06:27 -0400 (Fri, 16 Sep 2011) | 2 lines Make a note that inotify won't work with an NFS mounted spooler directory. ........ ................ 2011-09-16 10:16 +0000 [r336095-336168] Gregory Nietsky * channels/chan_misdn.c, /: Merged revisions 336167 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336167 | irroot | 2011-09-16 12:12:03 +0200 (Fri, 16 Sep 2011) | 22 lines Merged revisions 336166 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336166 | irroot | 2011-09-16 12:09:17 +0200 (Fri, 16 Sep 2011) | 16 lines The round robin routing routine in chan_misdn.c is broken. it rotates between ports but never checks the channels in the ports. i have extensivly tested it and verified it works on 1 upto 4 ports. before the patch only 1 out of each port was used now all are used as expected. (closes issue ASTERISK-18413) Reported by: irroot Tested by: irroot Reviewed by: irroot Review: https://reviewboard.asterisk.org/r/1410/ ........ ................ * /, apps/app_queue.c: Merged revisions 336094 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336094 | irroot | 2011-09-15 17:54:46 +0200 (Thu, 15 Sep 2011) | 26 lines Merged revisions 336093 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336093 | irroot | 2011-09-15 17:46:21 +0200 (Thu, 15 Sep 2011) | 20 lines Locking order in app_queue.c causes deadlocks. a channel lock must never be held with the queues container lock held. the deadlock occured on masquerade. the queues container lock is a relic of the past the old queue module lock. with ao2 there is no need to hold this lock when dealing with members this patch removes unneeded locks. (closes issue ASTERISK-18101) (closes issue ASTERISK-18487) Reported by: Paul Rolfe, Jason Legault Tested by: irroot, Jason Legault, Paul Rolfe Reviewed by: Matthew Nicholson Review: https://reviewboard.asterisk.org/r/1402/ ........ ................ 2011-09-15 15:19 +0000 [r336092] David Vossel * /, main/format_cap.c: Merged revisions 336091 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r336091 | dvossel | 2011-09-15 10:19:10 -0500 (Thu, 15 Sep 2011) | 2 lines Removes some no-op code found in format_cap.c. ........ 2011-09-15 12:50 +0000 [r336043] Olle Johansson * CREDITS, /, apps/app_meetme.c, CHANGES: Merged revisions 336042 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r336042 | oej | 2011-09-15 14:46:38 +0200 (Tor, 15 Sep 2011) | 12 lines Meetme: Introducing a new option "k" to kill a conference if there's only a single member left. When using Meetme as a modular call bridge from third party applications, it's handy to make it behave like a normal call bridge. When the second to last person exists, the last person will be kicked out of the conference when this option is enabled. (closes issue ASTERISK-18234) Review: https://reviewboard.asterisk.org/r/1376/ Patch by oej, sponsored by ClearIT, Solna, Sweden ........ 2011-09-15 08:40 +0000 [r335993] Gregory Nietsky * /, channels/chan_agent.c: Merged revisions 335991 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r335991 | irroot | 2011-09-15 10:29:12 +0200 (Thu, 15 Sep 2011) | 17 lines Merged revisions 335978 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335978 | irroot | 2011-09-15 10:15:22 +0200 (Thu, 15 Sep 2011) | 11 lines lock the channel before calling ast_bridged_channel() to prevent a seg fault. AMI agents list called on shutdown causes a segfault, introducing proper locking will prevent this. (closes issue ASTERISK-18092) Reported by: agustina Patches: chan_agent.patch (License #5041) patch uploaded by irroot ........ ................ 2011-09-14 18:38 +0000 [r335853-335913] Richard Mudgett * /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 335912 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r335912 | rmudgett | 2011-09-14 13:31:15 -0500 (Wed, 14 Sep 2011) | 20 lines Merged revisions 335911 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335911 | rmudgett | 2011-09-14 13:21:35 -0500 (Wed, 14 Sep 2011) | 13 lines Remove unnecessary libpri dependency checks in the configure script. Using the --with-pri option with the configure script generated an error about not having PRI_L2_PERSISTENCE if you did not have the absolute latest libpri SVN checkout installed. The AST_EXT_LIB_SETUP_DEPENDENT macro in the configure.ac script seems to be for libraries that are dependent upon other libraries and not necessarily for optional/added features within a library. (closes issue ASTERISK-18535) Reported by: Michael Keuter ........ ................ * channels/chan_dahdi.c, /: Merged revisions 335852 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r335852 | rmudgett | 2011-09-14 11:00:37 -0500 (Wed, 14 Sep 2011) | 18 lines Merged revisions 335851 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335851 | rmudgett | 2011-09-14 10:53:25 -0500 (Wed, 14 Sep 2011) | 11 lines Fixed cut-n-paste regression using the wrong variable. Fixes the missing DAHDI channels when using the newer chan_dahdi.conf sections for channel configuration. (closes issue ASTERISK-18496) Reported by: Sean Darcy Patches: jira_asterisk_18496_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Sean Darcy, rmudgett ........ ................ 2011-09-14 13:29 +0000 [r335792] Matthew Nicholson * main/manager.c, /: Merged revisions 335791 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r335791 | mnicholson | 2011-09-14 08:28:50 -0500 (Wed, 14 Sep 2011) | 11 lines Merged revisions 335790 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335790 | mnicholson | 2011-09-14 08:28:16 -0500 (Wed, 14 Sep 2011) | 4 lines The tech and data members of fast_originate_helper are not string fields. ASTERISK-17709 ........ ................ 2011-09-13 22:11 +0000 [r335722] Richard Mudgett * /, apps/app_directed_pickup.c: Merged revisions 335721 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r335721 | rmudgett | 2011-09-13 17:10:44 -0500 (Tue, 13 Sep 2011) | 9 lines Merged revisions 335720 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335720 | rmudgett | 2011-09-13 17:10:15 -0500 (Tue, 13 Sep 2011) | 1 line Remove obsolete todo comment about PICKUPRESULT. ........ ................ 2011-09-13 21:52 +0000 [r335719] Paul Belanger * main/dnsmgr.c: Additional updates for parsing dnsmgr.conf Review: https://reviewboard.asterisk.org/r/1432/ 2011-09-13 21:40 +0000 [r335718] Tzafrir Cohen * main/asterisk.c: do parse defaultlanguage from asterisk.conf Do parse the option "defaultlanguage" from the [options] section of asterisk.conf, as in the sample config file. Otherwise the build-time default language (normally "en") is always the default one. Review: https://reviewboard.asterisk.org/r/1342/ Signed-off-by: Tzafrir Cohen (License #5035) Original-Commit: http://svn.digium.com/svn/asterisk/branches/1.8@335716 Original-Commit: http://svn.digium.com/svn/asterisk/branches/10@335717 2011-09-13 18:56 +0000 [r335657] Tilghman Lesher * /, configure, configure.ac: Merged revisions 335656 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r335656 | tilghman | 2011-09-13 13:55:33 -0500 (Tue, 13 Sep 2011) | 11 lines Merged revisions 335655 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335655 | tilghman | 2011-09-13 13:52:38 -0500 (Tue, 13 Sep 2011) | 4 lines Move mandatory checks closer to the beginning of the file. If these are going to fail, they should fail as quickly as possible. ........ ................ 2011-09-13 18:49 +0000 [r335654] Matthew Nicholson * main/pbx.c, main/manager.c, /: Merged revisions 335653 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r335653 | mnicholson | 2011-09-13 13:47:57 -0500 (Tue, 13 Sep 2011) | 12 lines Merged revisions 335618 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335618 | mnicholson | 2011-09-13 13:20:52 -0500 (Tue, 13 Sep 2011) | 5 lines Don't limit the size of appdata for manager originate actions. ASTERISK-17709 Patch by: tilghman (with modifications) ........ ................ 2011-09-13 18:11 +0000 [r335555-335603] Paul Belanger * UPGRADE.txt, main/dsp.c: Clean up dsp.conf parsing Review: https://reviewboard.asterisk.org/r/1434/ * UPGRADE.txt, cdr/cdr_csv.c: Clean up cdr.conf parsing for [csv] section Review: https://reviewboard.asterisk.org/r/1427/ * main/dnsmgr.c, UPGRADE.txt: Clean up dnsmgr.conf parsing Review: https://reviewboard.asterisk.org/r/1432/ 2011-09-13 07:35 +0000 [r335511] Russell Bryant * include/asterisk/event.h, /, res/ais/evt.c, main/event.c: Merged revisions 335510 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r335510 | russell | 2011-09-13 02:24:34 -0500 (Tue, 13 Sep 2011) | 22 lines Merged revisions 335497 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335497 | russell | 2011-09-13 02:11:36 -0500 (Tue, 13 Sep 2011) | 15 lines Fix a crash in res_ais. This patch resolves a crash observed in a load testing environment that involved the use of the res_ais module. I observed some crashes where the event delivery callback would get called, but the length parameter incidcating how much data there was to read was 0. The code assumed (with good reason I would think) that if this callback got called, there was an event available to read. However, if the rare case that there's nothing there, catch it and return instead of blowing up. More specifically, the change always ensure that the size of the received event in the cluster is always big enough to be a real ast_event. Review: https://reviewboard.asterisk.org/r/1423/ ........ ................ 2011-09-12 15:56 +0000 [r335435] Matthew Nicholson * main/channel.c, /: Merged revisions 335434 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r335434 | mnicholson | 2011-09-12 10:55:48 -0500 (Mon, 12 Sep 2011) | 13 lines Merged revisions 335433 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335433 | mnicholson | 2011-09-12 10:54:41 -0500 (Mon, 12 Sep 2011) | 6 lines Properly set caller_warning and callee_warning before we try to use them. ASTERISK-18199 Patch by: elguero Testing by: rtang ........ ................ 2011-09-12 14:33 +0000 [r335385] Olle Johansson * channels/chan_sip.c: Documentation updates 2011-09-12 14:24 +0000 [r335354] Kinsey Moore * apps/app_dial.c, /: Merged revisions 335346 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r335346 | kmoore | 2011-09-12 09:22:15 -0500 (Mon, 12 Sep 2011) | 17 lines Merged revisions 335341 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335341 | kmoore | 2011-09-12 09:21:17 -0500 (Mon, 12 Sep 2011) | 10 lines Ensure frames are not written to dialed channel if ringback is requested When a single channel was dialed and there was media to be forwarded to the calling channel, the media was written without regard for ringback causing silence to be heard in some circumstances. This regression was introduced when the meaning of "single" changed to mean only the number of channels dialed. (closes issue ASTERISK-18083) ........ ................ 2011-09-12 14:22 +0000 [r335324-335349] Olle Johansson * channels/chan_sip.c: Small documentation updates * CREDITS, channels/chan_sip.c, include/asterisk/indications.h, UPGRADE.txt, configs/sip.conf.sample, channels/sip/include/sip.h: New sip.conf option for setting default tonezone for channel or individual devices Review: https://reviewboard.asterisk.org/r/1429/ (closes issue ASTERISK-18497) Thanks to russellb for peer review. * /, channels/chan_sip.c: Merged revisions 335323 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r335323 | oej | 2011-09-12 15:47:13 +0200 (Mån, 12 Sep 2011) | 19 lines Merged revisions 335319 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335319 | oej | 2011-09-12 15:25:30 +0200 (Mån, 12 Sep 2011) | 12 lines Lock the peer->mvipvt to avoid crashes with SIP history enabled After the launch of 1.6 event-based MWI we have two threads handling the peer->mwipvt, which cause issues with SIP history additions in combination with the max limit for number of history entries. Review: https://reviewboard.asterisk.org/r/1373/ (closes issue ASTERISK-18288) Thanks to irrot for peer review. Work with this bug funded by IPvision AS ........ ................ 2011-09-12 13:27 +0000 [r335322] Kinsey Moore * /, channels/chan_iax2.c: Merged revisions 335321 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r335321 | kmoore | 2011-09-12 08:27:04 -0500 (Mon, 12 Sep 2011) | 16 lines Merged revisions 335320 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335320 | kmoore | 2011-09-12 08:25:42 -0500 (Mon, 12 Sep 2011) | 9 lines Prevent IAX2 from getting IPv6 addresses via DNS IAX2 does not support IPv6 and getting such addresses from DNS can cause error messages on the remote end involving bad IPv4 address casts in the presence of IPv6/IPv4 tunnels. This patch ensures that IAX2 will not encounter IPv6 addresses via DNS queries. (closes issue ASTERISK-18090) ........ ................ 2011-09-12 11:15 +0000 [r335261] Stefan Schmidt * /, channels/chan_sip.c: Merged revisions 335260 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r335260 | schmidts | 2011-09-12 11:11:45 +0000 (Mon, 12 Sep 2011) | 12 lines Merged revisions 335259 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335259 | schmidts | 2011-09-12 11:09:19 +0000 (Mon, 12 Sep 2011) | 6 lines build_peer doesnt unlink a peer object from peers_by_ip container which leads to a wrong refcounter value. adding an ao2_unlink from the peers_by_ip container fix it. Review: https://reviewboard.asterisk.org/r/1428/ ........ ................ 2011-09-12 03:10 +0000 [r335170-335212] Paul Belanger * UPGRADE.txt: Be more specific on which section has changed. * main/cdr.c, UPGRADE.txt: Iterate though cdr.conf setting Review: https://reviewboard.asterisk.org/r/1426/ 2011-09-11 17:09 +0000 [r335129] Terry Wilson * configs/res_config_sqlite3.conf.sample (added), res/res_config_sqlite3.c (added): Add SQLite 3 realtime support 2011-09-09 16:28 +0000 [r335079] Matthew Jordan * channels/chan_unistim.c, apps/app_dial.c, main/pbx.c, addons/chan_ooh323.c, channels/chan_sip.c, channels/chan_console.c, channels/sig_pri.c, channels/chan_oss.c, main/channel.c, channels/chan_usbradio.c, main/dial.c, channels/chan_dahdi.c, channels/chan_misdn.c, channels/chan_skinny.c, funcs/func_frame_trace.c, main/features.c, channels/chan_h323.c, channels/chan_alsa.c, include/asterisk/frame.h, channels/sig_ss7.c, channels/chan_mgcp.c: Merged revisions 335078 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r335078 | mjordan | 2011-09-09 11:27:01 -0500 (Fri, 09 Sep 2011) | 29 lines Merged revisions 335064 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335064 | mjordan | 2011-09-09 11:09:09 -0500 (Fri, 09 Sep 2011) | 23 lines Updated SIP 484 handling; added Incomplete control frame When a SIP phone uses the dial application and receives a 484 Address Incomplete response, if overlapped dialing is enabled for SIP, then the 484 Address Incomplete is forwarded back to the SIP phone and the HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete application dialplan logic was automatically triggered; now, explicit dialplan usage of the application is required. Additionally, this patch adds a new AST_CONTOL_FRAME type called AST_CONTROL_INCOMPLETE. If a channel driver receives this control frame, it is an indication that the dialplan expects more digits back from the device. If the device supports overlap dialing it should attempt to notify the device that the dialplan is waiting for more digits; otherwise, it can handle the frame in a manner appropriate to the channel driver. (closes issue ASTERISK-17288) Reported by: Mikael Carlsson Tested by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/1416/ ........ ................ 2011-09-09 07:28 +0000 [r335015] Gregory Nietsky * funcs/func_dialplan.c, /, apps/app_readexten.c, CHANGES: Merged revisions 335014 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r335014 | irroot | 2011-09-09 09:23:53 +0200 (Fri, 09 Sep 2011) | 9 lines Move code for VALID_EXTEN from app_readexten to func_dialplan Mark VALID_EXTEN deprecated. Review: https://reviewboard.asterisk.org/r/1396/ ........ 2011-09-08 22:30 +0000 [r334955] Richard Mudgett * /, main/logger.c: Merged revisions 334954 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r334954 | rmudgett | 2011-09-08 17:28:56 -0500 (Thu, 08 Sep 2011) | 17 lines Merged revisions 334953 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334953 | rmudgett | 2011-09-08 17:27:40 -0500 (Thu, 08 Sep 2011) | 10 lines Fix crash with res_fax when MALLOC_DEBUG and "core stop gracefully" are used. Asterisk crashes if MALLOC_DEBUG is enabled when res_fax tries to unregister its logger level. * Make ast_logger_unregister_level() use ast_free() instead of free(). When MALLOC_DEBUG is enabled, ast_free() does not degenerate into a call to free(). Therefore, if you allocated memory with a form of ast_malloc you must free it with ast_free. ........ ................ 2011-09-08 13:36 +0000 [r334907] Jonathan Rose * main/cdr.c, main/pbx.c: Removes colorful verb statements erroneously commited with r332760 2011-09-07 19:38 +0000 [r334845] Paul Belanger * /, channels/chan_iax2.c: Merged revisions 334844 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r334844 | pabelanger | 2011-09-07 15:37:24 -0400 (Wed, 07 Sep 2011) | 11 lines Merged revisions 334843 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334843 | pabelanger | 2011-09-07 15:35:52 -0400 (Wed, 07 Sep 2011) | 4 lines Cleanup chan_iax2.c log messages Review: https://code.asterisk.org/code/cru/CR-AST-11 ........ ................ 2011-09-07 19:35 +0000 [r334842] Richard Mudgett * /, main/features.c: Merged revisions 334841 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r334841 | rmudgett | 2011-09-07 14:33:38 -0500 (Wed, 07 Sep 2011) | 17 lines Merged revisions 334840 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334840 | rmudgett | 2011-09-07 14:31:44 -0500 (Wed, 07 Sep 2011) | 10 lines Fix AMI action Park crash. * Made AMI action Park not say anything to the parker channel (AMI header Channel2) since the AMI action is a third party parking the call. (This is a change in behavior that cannot be preserved without a lot of effort.) * Made not play pbx-parkingfailed if the Park 's' option is used. JIRA AST-660 ........ ................ 2011-09-07 15:37 +0000 [r334683-334792] Stefan Schmidt * /, main/features.c: Merged revisions 334747 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r334747 | schmidts | 2011-09-07 15:10:37 +0000 (Wed, 07 Sep 2011) | 9 lines Merged revisions 334682 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334682 | schmidts | 2011-09-07 13:26:50 +0000 (Wed, 07 Sep 2011) | 3 lines Adding the Feature to sent a Reason Header in a SIP Cancel message by set the flag AST_FLAG_ANSWERED_ELSEWHERE before doing a masquerade in the pickup function. ........ ................ * main/features.c: clean up wrong merged stuff * /, main/features.c: Merged revisions 334682 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r334682 | schmidts | 2011-09-07 13:26:50 +0000 (Wed, 07 Sep 2011) | 3 lines Adding the Feature to sent a Reason Header in a SIP Cancel message by set the flag AST_FLAG_ANSWERED_ELSEWHERE before doing a masquerade in the pickup function. ........ * main/features.c: Adding the Feature to sent a Reason Header in a SIP Cancel message by set the flag AST_FLAG_ANSWERED_ELSEWHERE before doing a masquerade in the pickup function. 2011-09-07 08:17 +0000 [r334618-334623] Alec L Davis * /, CHANGES, apps/app_queue.c: Merged revisions 334621 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r334621 | alecdavis | 2011-09-07 20:14:50 +1200 (Wed, 07 Sep 2011) | 9 lines Merged revisions 334620 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334620 | alecdavis | 2011-09-07 20:12:49 +1200 (Wed, 07 Sep 2011) | 2 lines peroid typo ........ ................ * main/logger.c: log Asterisk Version number, Build etc into each log file Allow tracking of previous versions through log file records to be tracked. Each time log file is created or opened, log Asterisk Version, Buildinfo. etc. alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1409/ * main/pbx.c, /: Merged revisions 334617 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r334617 | alecdavis | 2011-09-07 19:45:00 +1200 (Wed, 07 Sep 2011) | 17 lines Merged revisions 334616 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334616 | alecdavis | 2011-09-07 19:33:39 +1200 (Wed, 07 Sep 2011) | 10 lines Prevent segfault if call arrives before Asterisk is fully booted. Prevent ast_pbx_start and ast_run_start from starting a new thread unless asterisk is fully booted. alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1407/ ........ ................ 2011-09-07 00:54 +0000 [r334574] Tilghman Lesher * main/frame.c, contrib/realtime/mysql/iaxfriends.sql, contrib/realtime/postgresql/realtime.sql, configs/sip.conf.sample, CHANGES, contrib/realtime/mysql/sipfriends.sql: Implement the '!' negation element to negate codecs directly in the allow keyword. This permits the list of codecs to be specified in one configuration line, instead of two or more, generally with the aim of either allowing all codecs with the exception of a few or disallowing most but permitting a few. Review: https://reviewboard.asterisk.org/r/1411/ 2011-09-06 16:15 +0000 [r334519] Gregory Nietsky * /, apps/app_voicemail.c: Merged revisions 334455 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r334455 | irroot | 2011-09-06 15:58:56 +0200 (Tue, 06 Sep 2011) | 18 lines Merged revisions 334453 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334453 | irroot | 2011-09-06 15:48:03 +0200 (Tue, 06 Sep 2011) | 13 lines Make SQL query in app_voicemail.c portable LIMIT is not portable. Regression from r312212 (closes issue ASTERISK-18255) Reported by: Leif Madsen Tested by: Leif Madsen Review: https://reviewboard.asterisk.org/r/1415/ ........ ................ 2011-09-06 16:08 +0000 [r334517] Paul Belanger * configs/iax.conf.sample, /, CHANGES, channels/chan_iax2.c: Merged revisions 334514 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r334514 | pabelanger | 2011-09-06 11:47:59 -0400 (Tue, 06 Sep 2011) | 6 lines authdebug is now disabled by default To enable this functionaility again set authdebug = yes in iax.conf Review: https://reviewboard.asterisk.org/r/1414/ ........ 2011-09-06 16:04 +0000 [r334472-334515] Gregory Nietsky * /, apps/app_voicemail.c: Revert r334472 due to properties going missing * /, apps/app_voicemail.c: Merged revisions 334455 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r334455 | irroot | 2011-09-06 15:58:56 +0200 (Tue, 06 Sep 2011) | 18 lines Merged revisions 334453 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334453 | irroot | 2011-09-06 15:48:03 +0200 (Tue, 06 Sep 2011) | 13 lines Make SQL query in app_voicemail.c portable LIMIT is not portable. Regression from r312212 (closes issue ASTERISK-18255) Reported by: Leif Madsen Tested by: Leif Madsen Review: https://reviewboard.asterisk.org/r/1415/ ........ ................ 2011-09-02 21:09 +0000 [r334304-334358] Richard Mudgett * /, res/res_musiconhold.c: Merged revisions 334357 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r334357 | rmudgett | 2011-09-02 16:08:16 -0500 (Fri, 02 Sep 2011) | 26 lines Merged revisions 334355 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334355 | rmudgett | 2011-09-02 15:59:49 -0500 (Fri, 02 Sep 2011) | 19 lines MusicOnHold has extra unref which may lead to memory corruption and crash. The problem happens when a call is disconnected and you had started a MOH class that does not use the files mode. If you define REF_DEBUG and recreate the problem, it will announce itself with the following warning: Attempt to unref mohclass 0xb70722e0 (default) when only 1 ref remained, and class is still in a container! * Fixed moh_alloc() and moh_release() functions not handling the state->class reference consistently. (closes issue ASTERISK-18346) Reported by: Mark Murawski Patches: jira_asterisk_18346_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett, Mark Murawski Review: https://reviewboard.asterisk.org/r/1404/ ........ ................ * /, include/asterisk/config.h, main/config.c: Merged revisions 334297 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r334297 | rmudgett | 2011-09-02 12:15:08 -0500 (Fri, 02 Sep 2011) | 46 lines Merged revisions 334296 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334296 | rmudgett | 2011-09-02 12:10:58 -0500 (Fri, 02 Sep 2011) | 39 lines Fix potential memory allocation failure crashes in config.c. * Added required checks to the returned memory allocation pointers to prevent crashes. * Made ast_include_rename() create a replacement ast_variable list node if the new filename is longer than the available space. Fixes potential crash and memory leak. * Factored out ast_variable_move() from ast_variable_update() so ast_include_rename() can also use it when creating a replacement ast_variable list node. * Made the filename stuffed at the end of the struct a minimum allocated size in ast_variable_new() in case ast_include_rename() changes the stored filename. * Constify struct char pointers pointing to strings stuffed at the end of the struct for: ast_variable, cache_file_mtime, and ast_config_map. * Factored out cfmtime_new() to remove inlined code and allow some struct pointers to become const. * Removed the list lock from struct cache_file_mtime that was never used. * Added doxygen comments to several structure elements and better documented what strings are stuffed at the struct end char array. * Reworked ast_config_text_file_save() and set_fn() to handle allocation failure of the include file scratch pad object tracking blank lines. * Made ast_config_text_file_save() fn[] declared with PATH_MAX to ensure it is long enough for any filename with path. Also reduced the number of container fileset buckets from a rediculus 180,000 to 1023. JIRA AST-618 Review: https://reviewboard.asterisk.org/r/1378/ ........ ................ 2011-09-01 17:41 +0000 [r334231-334236] Tilghman Lesher * main/pbx.c, /: Merged revisions 334235 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r334235 | tilghman | 2011-09-01 12:39:32 -0500 (Thu, 01 Sep 2011) | 9 lines Merged revisions 334234 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334234 | tilghman | 2011-09-01 12:38:33 -0500 (Thu, 01 Sep 2011) | 2 lines Remove 1.6 compatibility documentation from 1.8, as it no longer applies. ........ ................ * res/res_config_odbc.c, /: Merged revisions 334230 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r334230 | tilghman | 2011-09-01 12:30:19 -0500 (Thu, 01 Sep 2011) | 25 lines Merged revisions 334229 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334229 | tilghman | 2011-09-01 12:28:09 -0500 (Thu, 01 Sep 2011) | 18 lines Create a local alias for ast_odbc_clear_cache. As a function pointer, the reference has to be resolved at load time irrespective of the RTLD_LAZY flag. Creating a local alias solves this problem, because the structure is initialized with that local function pointer, while the actual function can remain lazily linked until runtime. The reason why this is important is because we lazily load function references during the module loading process, in order to obtain priority values for each module, ensuring that modules are loaded in the correct order. Previous to this change, when this module was initially loaded, the module loader would emit a symbol resolution error, because of the above requirement. Closes ASTERISK-18399 (reported by Mikael Carlsson, fix suggested by Walter Doekes, patch by me) ........ ................ 2011-08-31 18:54 +0000 [r334158] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 334157 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r334157 | mnicholson | 2011-08-31 13:53:40 -0500 (Wed, 31 Aug 2011) | 11 lines Merged revisions 334156 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334156 | mnicholson | 2011-08-31 13:50:33 -0500 (Wed, 31 Aug 2011) | 4 lines Disable T.38 when we get a invite with image media port set to 0 ASTERISK-17678 ........ ................ 2011-08-31 18:11 +0000 [r334115] Richard Mudgett * channels/chan_sip.c: Optimize chan_sip.c check_rtp_timeout() function. * Make check_rtp_timeout() remember the values returned by ast_rtp_instance_get_timeout(), ast_rtp_instance_get_hold_timeout(), and ast_rtp_instance_get_keepalive() instead of repeatedly calling them. (closes issue ASTERISK-18319) Reported by: Rob Gagnon Patches: issue-18319-trunk-r333066.diff (License #6159) patch uploaded by Rob Gagnon Review: https://reviewboard.asterisk.org/r/1377/ 2011-08-31 16:31 +0000 [r334067] Matthew Nicholson * /, res/res_fax.c: Merged revisions 334064 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r334064 | mnicholson | 2011-08-31 11:31:00 -0500 (Wed, 31 Aug 2011) | 4 lines only alter the gateway_timeout when attching the gateway to a channel ASTERISK-18219 ........ 2011-08-31 16:02 +0000 [r334011-334014] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 334013 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r334013 | rmudgett | 2011-08-31 11:00:49 -0500 (Wed, 31 Aug 2011) | 30 lines Merged revisions 334012 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334012 | rmudgett | 2011-08-31 10:57:12 -0500 (Wed, 31 Aug 2011) | 23 lines No DAHDI channel available for conference, user introduction disabled. The following error will consistently occur when trying to dial into a MeetMe conference when the server does not have DAHDI hardware installed: app_meetme.c: No DAHDI channel available for conference, user introduction disabled (is chan_dahdi loaded?) While chan_dahdi is loaded correctly during compilation and install of Asterisk/Dahdi, including associated modules, etc., a chan_dahdi.conf configuration file in /etc/asterisk is not created by FreePBX if hardware does not exist, causing MeetMe to be unable to open a DAHDI pseudo channel. * Allow chan_dahdi to create a pseudo channel when there is no chan_dahdi.conf file to load. (closes issue ASTERISK-17398) Reported by: Preston Edwards Patches: jira_asterisk_17398_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett ........ ................ * main/channel.c, /, channels/chan_agent.c: Merged revisions 334010 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r334010 | rmudgett | 2011-08-31 10:23:11 -0500 (Wed, 31 Aug 2011) | 50 lines Merged revisions 334009 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334009 | rmudgett | 2011-08-31 10:20:31 -0500 (Wed, 31 Aug 2011) | 43 lines Call pickup race leaves orphaned channels or crashes. Multiple users attempting to pickup a call that has been forked to multiple extensions either crashes or fails a masquerade with a "bad things may happen" message. This is the scenario that is causing all the grief: 1) Pickup target is selected 2) target is marked as being picked up in ast_do_pickup() 3) target is unlocked by ast_do_pickup() 4) app dial or queue gets a chance to hang up losing calls and calls ast_hangup() on target 5) SINCE A MASQUERADE HAS NOT BEEN SETUP YET BY ast_do_pickup() with ast_channel_masquerade(), ast_hangup() completes successfully and the channel is no longer in the channels container. 6) ast_do_pickup() then calls ast_channel_masquerade() to schedule the masquerade on the dead channel. 7) ast_do_pickup() then calls ast_do_masquerade() on the dead channel 8) bad things happen while doing the masquerade and in the process ast_do_masquerade() puts the dead channel back into the channels container 9) The "orphaned" channel is visible in the channels list if a crash does not happen. This patch does the following: * Made ast_hangup() set AST_FLAG_ZOMBIE on a successfully hung-up channel and not release the channel lock until that has happened. * Made __ast_channel_masquerade() not setup a masquerade if either channel has AST_FLAG_ZOMBIE set. * Fix chan_agent misuse of AST_FLAG_ZOMBIE since it would no longer work. (closes issue ASTERISK-18222) Reported by: Alec Davis Tested by: rmudgett, Alec Davis, irroot, Karsten Wemheuer (closes issue ASTERISK-18273) Reported by: Karsten Wemheuer Tested by: rmudgett, Alec Davis, irroot, Karsten Wemheuer Review: https://reviewboard.asterisk.org/r/1400/ ........ ................ 2011-08-31 15:20 +0000 [r334008] Kinsey Moore * /, channels/chan_sip.c: Merged revisions 334007 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r334007 | kmoore | 2011-08-31 10:19:30 -0500 (Wed, 31 Aug 2011) | 14 lines Merged revisions 334006 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334006 | kmoore | 2011-08-31 10:18:37 -0500 (Wed, 31 Aug 2011) | 7 lines Correct an AMI protocol violation with SIPshowpeer The response of SIPshowpeer ends with "\r\n\r\n". Since other commands are ended by using \r\n this confuses any interfacing script. (closes issue ASTERISK-17486) ........ ................ 2011-08-30 22:16 +0000 [r333963] Alexandr Anikin * addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooGkClient.c, /, addons/ooh323c/src/ooCalls.h, addons/ooh323c/src/oochannels.c, addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooCalls.c: Merged revisions 333961-333962 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r333961 | may | 2011-08-31 01:21:53 +0400 (Wed, 31 Aug 2011) | 11 lines Merged revisions 333947 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333947 | may | 2011-08-31 01:16:30 +0400 (Wed, 31 Aug 2011) | 5 lines cleanups in ACF/ARJ GK replies processing fixed long (24 sec) pause if acf/arj proccessed before ast_cond_wait called to wait this ........ ................ r333962 | may | 2011-08-31 01:53:42 +0400 (Wed, 31 Aug 2011) | 3 lines security fix. really drop call if signalling addr is not same as socket addr ................ 2011-08-30 14:03 +0000 [r333896] Matthew Nicholson * /, res/res_fax.c: Merged revisions 333895 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r333895 | mnicholson | 2011-08-30 09:01:31 -0500 (Tue, 30 Aug 2011) | 6 lines Replaced FAXOPT(gwtimeout) with a second parameter to FAXOPT(gateway). Patch by: irroot Review: https://reviewboard.asterisk.org/r/1385/ ASTERISK-18219 ........ 2011-08-29 21:43 +0000 [r333838] Terry Wilson * /, channels/chan_sip.c: Merged revisions 333837 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r333837 | twilson | 2011-08-29 16:41:13 -0500 (Mon, 29 Aug 2011) | 22 lines Merged revisions 333836 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333836 | twilson | 2011-08-29 16:38:31 -0500 (Mon, 29 Aug 2011) | 15 lines Refresh peer address if DNS unavailable at peer creation If Asterisk starts and no DNS is available, outbound registrations will fail indefinitely. This patch copies the address from the sip_registry struct, which will be updated, to the peer->addr when necessary. If dnsmgr is enabled, the registration fails without the patch because even though the address on the registry is updated via dnsmgr, the address is just copied on the first try. Since we use ast_sockaddr_copy, dnsmgr can't update the address that is copied to the sip_pvt or peers. Closes issue ASTERISK-18000 Review: https://reviewboard.asterisk.org/r/1335/ ........ ................ 2011-08-29 21:17 +0000 [r333789] Richard Mudgett * /, include/asterisk/channel.h, addons/chan_mobile.c: Merged revisions 333786 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r333786 | rmudgett | 2011-08-29 16:12:29 -0500 (Mon, 29 Aug 2011) | 13 lines Merged revisions 333784-333785 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333784 | rmudgett | 2011-08-29 16:05:43 -0500 (Mon, 29 Aug 2011) | 2 lines Fix deadlock potential of chan_mobile.c:mbl_ast_hangup(). ........ r333785 | rmudgett | 2011-08-29 16:06:16 -0500 (Mon, 29 Aug 2011) | 1 line Add some do not hold locks notes to channel.h ........ ................ 2011-08-29 18:28 +0000 [r333736] Matthew Nicholson * /, res/res_fax_spandsp.c: Merged revisions 333716 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r333716 | mnicholson | 2011-08-29 13:22:58 -0500 (Mon, 29 Aug 2011) | 5 lines It is possible for the gateway to be attached when the channel is still negotiating T.38. This change handles that case. ASTERISK-18329 ........ 2011-08-29 17:31 +0000 [r333689] Terry Wilson * main/channel.c, /, CHANGES: Merged revisions 333681 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r333681 | twilson | 2011-08-29 12:28:59 -0500 (Mon, 29 Aug 2011) | 7 lines Use realtime text when it is negotiated This patch make use of wirte_text() realtime text instead of send_text() if T.140 is in native formats. ASTERISK-17937 Review: https://reviewboard.asterisk.org/r/1356/ ........ 2011-08-29 17:14 +0000 [r333632] Matthew Jordan * apps/app_voicemail.c: Merged revisions 333631 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r333631 | mjordan | 2011-08-29 12:12:55 -0500 (Mon, 29 Aug 2011) | 9 lines Merged revisions 333630 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333630 | mjordan | 2011-08-29 12:11:15 -0500 (Mon, 29 Aug 2011) | 1 line Fixed improperly formatted TestEvent AMI message in app_voicemail ........ ................ 2011-08-29 15:58 +0000 [r333571] Jonathan Rose * /, res/res_jabber.c: Merged revisions 333570 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r333570 | jrose | 2011-08-29 10:56:56 -0500 (Mon, 29 Aug 2011) | 11 lines Merged revisions 333569 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333569 | jrose | 2011-08-29 10:55:34 -0500 (Mon, 29 Aug 2011) | 4 lines Accidental use of variable client->status instead of client->state in from ASTERISK-18078 (issue ASTERISK-18078) ........ ................ 2011-08-28 09:57 +0000 [r333509] Tzafrir Cohen * channels/chan_vpb.cc: chan_vpb: remove unused variables (gcc4.6) GCC 4.6 detects variables that get assined to, but never used later. Also removes some remmed-out lines that become invalid. (closes issue ASTERISK-18336) Signed-off-by: Tzafrir Cohen (License #5035) , 2011-08-26 16:38 +0000 [r333428] Jonathan Rose * /, res/res_jabber.c: Merged revisions 333410 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r333410 | jrose | 2011-08-26 11:28:03 -0500 (Fri, 26 Aug 2011) | 19 lines Merged revisions 333378 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333378 | jrose | 2011-08-26 11:19:07 -0500 (Fri, 26 Aug 2011) | 13 lines [patch] Buddies are always auto-registered when processing the roster Reporter said autoregister flag was ignored for registering 'buddies' which had a subscription to us. Verified that this was the case and observed how the patch addressed this and made sure it didn't break anything. (closes issue ASTERISK-14233) Reported by: Simon Arlott Patches: asterisk-0015229.patch (license #5756) patch uploaded by Simon Arlott Tested by: Jonathan Rose ........ ................ 2011-08-26 16:12 +0000 [r333371] Matthew Jordan * /, apps/app_voicemail.c: Merged revisions 333370 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r333370 | mjordan | 2011-08-26 10:58:37 -0500 (Fri, 26 Aug 2011) | 26 lines Merged revisions 333339 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333339 | mjordan | 2011-08-26 08:36:36 -0500 (Fri, 26 Aug 2011) | 20 lines Bug fixes for voicemail user emailsubject / emailbody. This code change fixes a few issues with the voicemail user override of emailbody and emailsubject, including escaping the strings, potential memory leaks, and not overriding the voicemail defaults. Revision 325877 fixed this for ASTERISK-16795, but did not fix it for ASTERISK-16781. A subsequent check-in prevented 325877 from being applied to 10. This check-in resolves both issues, and applies the changes to 1.8, 10, and trunk. (closes issue ASTERISK-16781) Reported by: Sebastien Couture Tested by: mjordan (closes issue ASTERISK-16795) Reported by: mdeneen Tested by: mjordan Review: https://reviewboard.asterisk.org/r/1374 ........ ................ 2011-08-25 19:13 +0000 [r333276] Jonathan Rose * /, res/res_jabber.c: Merged revisions 333266 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r333266 | jrose | 2011-08-25 14:00:05 -0500 (Thu, 25 Aug 2011) | 20 lines Merged revisions 333265 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333265 | jrose | 2011-08-25 13:47:42 -0500 (Thu, 25 Aug 2011) | 14 lines Segfault when publishing device states via XMPP and not connected When using publishing device state with res_jabber, Asterisk will attempt to send a device state using the unconnected client using iks_send_raw and crash. This patch checks the validity of the connection before attempting to send the device state. (closes issue ASTERISK-18078) Reported by: Michael L. Young Patches: res_jabber-segfault-pubsub-not-connected2.patch (license #5026) patch uploaded by Michael L. Young Tested by: Jonathan Rose ........ ................ 2011-08-25 19:01 +0000 [r333159-333269] Jason Parker * Makefile, /: Merged revisions 333268 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r333268 | qwell | 2011-08-25 14:01:18 -0500 (Thu, 25 Aug 2011) | 9 lines Merged revisions 333267 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333267 | qwell | 2011-08-25 14:00:55 -0500 (Thu, 25 Aug 2011) | 2 lines Fix for DESTDIR spaces patch. ........ ................ * Makefile, build_tools/mkpkgconfig, /, configure, configure.ac, makeopts.in, sounds/Makefile: Merged revisions 333203 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r333203 | qwell | 2011-08-25 10:29:56 -0500 (Thu, 25 Aug 2011) | 15 lines Merged revisions 333201 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333201 | qwell | 2011-08-25 10:27:06 -0500 (Thu, 25 Aug 2011) | 8 lines Fix installation into directories containing spaces. This also vastly simplifies the logic in sounds/Makefile (Closes issue ASTERISK-18290) Reported by: Paul Belanger Review: https://reviewboard.asterisk.org/r/1379/ ........ ................ * channels/chan_local.c: Fix typo from r333070 2011-08-24 16:52 +0000 [r333117] Matthew Nicholson * /, res/res_fax.c: Merged revisions 333115 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r333115 | mnicholson | 2011-08-24 11:51:42 -0500 (Wed, 24 Aug 2011) | 4 lines Changed the "timeout" option to "gwtimeout". ASTERISK-18219 ........ 2011-08-24 09:17 +0000 [r333070-333075] Olle Johansson * channels/chan_local.c: Formatting changes - Removing some red white space and adding some curly brackets. * CHANGES: Add documentation for new manager event in chan_local AST-17623 * channels/chan_local.c: Add manager event for local channel semi-bridge (issue AST-17623) Review: https://reviewboard.asterisk.org/r/1154 2011-08-23 18:17 +0000 [r332881-333014] Richard Mudgett * /, apps/app_queue.c: Merged revisions 333011 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r333011 | rmudgett | 2011-08-23 13:15:49 -0500 (Tue, 23 Aug 2011) | 19 lines Merged revisions 333010 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333010 | rmudgett | 2011-08-23 13:14:01 -0500 (Tue, 23 Aug 2011) | 12 lines Memory Leak in app_queue The patch that was committed in the 1.6.x versions of Asterisk for ASTERISK-15862 actually fixed two issues. One was not applicable to 1.8 but the other is. queue_leak.patch fixes the portion applicable to 1.8. (closes issue ASTERISK-18265) Reported by: Fred Schroeder Patches: queue_leak.patch (license #5049) patch uploaded by mmichelson Tested by: Thomas Arimont ........ ................ * /, main/config.c: Merged revisions 332940 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332940 | rmudgett | 2011-08-22 16:23:40 -0500 (Mon, 22 Aug 2011) | 14 lines Merged revisions 332939 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332939 | rmudgett | 2011-08-22 16:22:24 -0500 (Mon, 22 Aug 2011) | 7 lines Minor code optimizations. * Simplify ast_category_browse() logic for easier understanding. * Remove dead code in ast_variable_delete() and simplify some of its logic. ........ ................ * /, apps/app_queue.c: Merged revisions 332875,332878 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332875 | rmudgett | 2011-08-22 14:41:03 -0500 (Mon, 22 Aug 2011) | 1 line Fix merge property. ................ r332878 | rmudgett | 2011-08-22 14:46:25 -0500 (Mon, 22 Aug 2011) | 25 lines Merged revisions 332874 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332874 | rmudgett | 2011-08-22 14:32:19 -0500 (Mon, 22 Aug 2011) | 18 lines Reference leaks in app_queue. * Fixed load_realtime_queue() leaking a queue reference when it overwrites q when processing a realtime queue. (issue ASTERISK-18265) * Make join_queue() unreference the queue returned by load_realtime_queue() when it is done with the pointer. The load_realtime_queue() returns a reference to the just loaded realtime queue. * Fixed queues container reference leak in queues_data_provider_get(). * queue_unref() should not return q that was just unreferenced. * Made logic in __queues_show() and queues_data_provider_get() when calling load_realtime_queue() easier to understand. ........ ................ 2011-08-22 19:56 +0000 [r332880] Paul Belanger * /, channels/chan_gtalk.c: Merged revisions 332877 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332877 | pabelanger | 2011-08-22 15:43:33 -0400 (Mon, 22 Aug 2011) | 13 lines Merged revisions 332876 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332876 | pabelanger | 2011-08-22 15:41:24 -0400 (Mon, 22 Aug 2011) | 6 lines Revert previous commit It seems google is still making changes to the protocol. (issue ASTERISK-18301) ........ ................ 2011-08-22 19:52 +0000 [r332879] Richard Mudgett * /: Fix merge 10 branch merge properties. 2011-08-22 19:19 +0000 [r332844] Matthew Jordan * include/asterisk/test.h, main/manager.c, /, main/file.c, main/test.c, main/app.c, configs/manager.conf.sample, include/asterisk/manager.h, apps/app_voicemail.c: Merged revisions 332817 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332817 | mjordan | 2011-08-22 13:15:51 -0500 (Mon, 22 Aug 2011) | 4 lines Review: https://reviewboard.asterisk.org/r/1364/ This update adds a new AMI event, TestEvent, which is enabled when the TEST_FRAMEWORK compiler flag is defined. It also adds initial usage of this event to app_voicemail. The TestEvent AMI event is used extensively by the voicemail tests in the Asterisk Test Suite. ........ 2011-08-22 18:33 +0000 [r332762-332831] Richard Mudgett * res/res_config_pgsql.c, res/res_config_odbc.c, /: Merged revisions 332830 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332830 | rmudgett | 2011-08-22 13:32:09 -0500 (Mon, 22 Aug 2011) | 15 lines Merged revisions 332816 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332816 | rmudgett | 2011-08-22 13:14:59 -0500 (Mon, 22 Aug 2011) | 8 lines Memory leaks in realtime_multi_xxx() when database access returns error. * Fix realtime_multi_pgsql() configuration memory leak when the database access returns an error. * Fix realtime_multi_odbc() configuration category use after free when the database access returns an error. ........ ................ * /, main/config.c: Merged revisions 332761 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332761 | rmudgett | 2011-08-22 12:05:35 -0500 (Mon, 22 Aug 2011) | 22 lines Merged revisions 332759 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332759 | rmudgett | 2011-08-22 12:00:03 -0500 (Mon, 22 Aug 2011) | 15 lines Memory leak reading realtime database variable list. Calling ast_load_realtime() can leak the last list node if the read list only contains empty variable value items. * Fixed list filter loop in ast_load_realtime() to delete the list node immediately instead of the next time through the loop. The next time through the loop may not happen if the node to delete is the last in the list. (issue ASTERISK-18277) (issue ASTERISK-18265) Patches: jira_asterisk_18265_v1.8_config.patch (license #5621) patch uploaded by rmudgett ........ ................ 2011-08-22 17:05 +0000 [r332760] Jonathan Rose * main/cdr.c, main/pbx.c, configs/cdr.conf.sample, include/asterisk/cdr.h, CHANGES: Add option for logging congested calls as CONGESTION instead of NO_ANSWER in CDR This patch adds a CDR option to cdr.conf that will allow CDR files to log calls ending with congestion in a way that is unique from other unanswered calls. (closes issue ASTERISK-14842) Reported by: Alec Davis Patches: cdr_congestion.diff.txt (License #5546) patch uploaded by Alec Davis 2011-08-22 16:31 +0000 [r332757] Matthew Nicholson * /, res/res_fax.c, include/asterisk/res_fax.h: Merged revisions 332756 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r332756 | mnicholson | 2011-08-22 11:29:45 -0500 (Mon, 22 Aug 2011) | 4 lines add a way to disable and/or modify the gateway timeout ASTERISK-18219 ........ 2011-08-21 14:34 +0000 [r332701] Paul Belanger * /, channels/chan_gtalk.c: Merged revisions 332700 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332700 | pabelanger | 2011-08-21 10:33:23 -0400 (Sun, 21 Aug 2011) | 12 lines Merged revisions 332699 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332699 | pabelanger | 2011-08-21 10:31:31 -0400 (Sun, 21 Aug 2011) | 5 lines Fix outgoing calls in chan_gtalk (closes issue ASTERISK-18301) Reported by: az1324 ........ ................ 2011-08-19 20:00 +0000 [r332655] Kinsey Moore * /, apps/app_confbridge.c: Merged revisions 332654 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r332654 | kmoore | 2011-08-19 14:59:34 -0500 (Fri, 19 Aug 2011) | 8 lines Make CONFBRIDGE_INFO behave more nicely CONFBRIDGE_INFO doesn't behave as well in edge cases as MEETME_INFO. With this patch, CONFBRIDGE_INFO should behave in a much more reasonable manner when presented with invalid conferences and keywords. Review: https://reviewboard.asterisk.org/r/1359/ ........ 2011-08-19 17:24 +0000 [r332615] Richard Mudgett * res/res_config_ldap.c: Fix infinite loop releasing the same memory in ldap_loadentry(). * Fixed memory leak of vars in ldap_loadentry(). * Fixed potential NULL ptr dereference of vars in ldap_loadentry(). 2011-08-18 21:39 +0000 [r332561] Terry Wilson * main/netsock2.c, /: Merged revisions 332560 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332560 | twilson | 2011-08-18 16:34:04 -0500 (Thu, 18 Aug 2011) | 12 lines Merged revisions 332559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332559 | twilson | 2011-08-18 16:26:01 -0500 (Thu, 18 Aug 2011) | 5 lines Fix possible error on stringification of IPv4-mapped addrs The FreeBSD netsock2 test has been failing for a while. We were pasing sa->len to getnameinfo instead of sa_tmp->len. ASTERISK-18289 ........ ................ 2011-08-18 19:30 +0000 [r332505] Kinsey Moore * channels/chan_dahdi.c, /: Merged revisions 332504 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332504 | kmoore | 2011-08-18 14:29:15 -0500 (Thu, 18 Aug 2011) | 15 lines Merged revisions 332503 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332503 | kmoore | 2011-08-18 14:28:00 -0500 (Thu, 18 Aug 2011) | 8 lines CRC4 in "dahdi show status" gives wrong impression to T1 users Change CRC4 to CRC in the output of "dahdi show status" so that it can apply in more situations without confusing users, especially since T1 lines use CRC6 instead of CRC4. (closes issue AST-471) ........ ................ 2011-08-18 14:49 +0000 [r332388-332448] Tilghman Lesher * build_tools/cflags.xml, build_tools/cflags-devmode.xml, /: Merged revisions 332447 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332447 | tilghman | 2011-08-18 09:48:40 -0500 (Thu, 18 Aug 2011) | 9 lines Merged revisions 332446 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332446 | tilghman | 2011-08-18 09:46:54 -0500 (Thu, 18 Aug 2011) | 2 lines Move BETTER_BACKTRACES out of development mode, as it's useful when DEBUG_THREADS is enabled. ........ ................ * Makefile, agi/Makefile, utils/Makefile, /, configure, include/asterisk/autoconfig.h.in, configure.ac, Makefile.moddir_rules, makeopts.in, sounds/Makefile: Merged revisions 332369 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332369 | tilghman | 2011-08-17 14:24:59 -0500 (Wed, 17 Aug 2011) | 17 lines Merged revisions 332355 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332355 | tilghman | 2011-08-17 14:21:36 -0500 (Wed, 17 Aug 2011) | 10 lines Re-add support for spaces in pathnames, including now spaces in DESTDIR. This was initially added to 1.8 prior to release, primarily to support the standard paths on Mac OS X, but was partially reverted recently in Subversion, due to the lack of support for spaces in DESTDIR. This commit restores support for the standard paths on Mac OS X, and also includes support for spaces in DESTDIR. (closes issue ASTERISK-18290) Reported by: pabelanger Review: https://reviewboard.asterisk.org/r/1326/ ........ ................ 2011-08-17 18:31 +0000 [r332337] Terry Wilson * /, res/res_timing_timerfd.c: Merged revisions 332321 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332321 | twilson | 2011-08-17 13:09:49 -0500 (Wed, 17 Aug 2011) | 17 lines Merged revisions 332320 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332320 | twilson | 2011-08-17 12:35:27 -0500 (Wed, 17 Aug 2011) | 10 lines Don't read from a disarmed or invalid timerfd Numerous isues have been reported for deadlocks that are caused by a blocking read in res_timing_timerfd on a file descriptor that will never be written to. This patch adds some checks to make sure that the timerfd is both valid and armed before calling read(). Should fix: ASTERISK-18142, ASTERISK-18166, ASTERISK-18197, AST-486, AST-495, AST-507 and possibly others. Review: https://reviewboard.asterisk.org/r/1361/ ........ ................ 2011-08-17 16:18 +0000 [r332270] Richard Mudgett * channels/sig_pri.h, channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /, configure, include/asterisk/autoconfig.h.in, configure.ac, channels/sig_pri.c: Merged revisions 332265 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332265 | rmudgett | 2011-08-17 11:01:29 -0500 (Wed, 17 Aug 2011) | 33 lines Merged revisions 332264 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332264 | rmudgett | 2011-08-17 10:51:08 -0500 (Wed, 17 Aug 2011) | 26 lines Outgoing BRI calls fail when using Asterisk 1.8 with HA8, HB8, and B410P cards. France Telecom brings layer 2 and layer 1 down on BRI lines when the line is idle. When layer 1 goes down Asterisk cannot make outgoing calls and the HA8 and HB8 cards also get IRQ misses. The inability to make outgoing calls is because the line is in red alarm and Asterisk will not make calls over a line it considers unavailable. The IRQ misses for the HA8 and HB8 card are because the hardware is switching clock sources from the line which just brought layer 1 down to internal timing. There is a DAHDI option for the B410P card to not tell Asterisk that layer 1 went down so Asterisk will allow outgoing calls: "modprobe wcb4xxp teignored=1". There is a similar DAHDI option for the HA8 and HB8 cards: "modprobe wctdm24xxp bri_teignored=1". Unfortunately that will not clear up the IRQ misses when the telco brings layer 1 down. * Add layer 2 persistence option to customize the layer 2 behavior on BRI PTMP lines. The new option has three settings: 1) Use libpri default layer 2 setting. 2) Keep layer 2 up. Bring layer 2 back up when the peer brings it down. 3) Leave layer 2 down when the peer brings it down. Layer 2 will be brought up as needed for outgoing calls. JIRA AST-598 ........ ................ 2011-08-16 20:15 +0000 [r332178] Paul Belanger * tests/test_substitution.c, tests/test_heap.c, /, tests/test_expr.c, tests/test_ast_format_str_reduce.c, tests/test_logger.c, tests/test_gosub.c, tests/test_app.c, tests/test_dlinklists.c, tests/test_event.c, tests/test_db.c, tests/test_linkedlists.c, tests/test_sched.c, tests/test_netsock2.c, tests/test_strings.c, tests/test_pbx.c, tests/test_func_file.c, tests/test_security_events.c, tests/test_stringfields.c, tests/test_time.c, tests/test_skel.c, tests/test_acl.c, tests/test_locale.c, tests/test_utils.c, tests/test_devicestate.c, tests/test_aoc.c, tests/test_astobj2.c, tests/test_poll.c, tests/test_amihooks.c: Merged revisions 332177 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332177 | pabelanger | 2011-08-16 16:11:49 -0400 (Tue, 16 Aug 2011) | 11 lines Merged revisions 332176 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332176 | pabelanger | 2011-08-16 16:10:13 -0400 (Tue, 16 Aug 2011) | 4 lines Flag test modules as 'core' Review: https://reviewboard.asterisk.org/r/1369/ ........ ................ 2011-08-16 17:53 +0000 [r332120] Jonathan Rose * /, channels/chan_sip.c: Merged revisions 332119 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332119 | jrose | 2011-08-16 12:45:38 -0500 (Tue, 16 Aug 2011) | 23 lines Merged revisions 332118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332118 | jrose | 2011-08-16 12:38:19 -0500 (Tue, 16 Aug 2011) | 16 lines ASTERISK-18067 ASTERISK-15479 - White Space affects mailbox value, multiple MWI subs Before, having multiple subscriptions to mailboxes on a sip peer set via the mailbox setting in sip.conf would only result in updates being sent on whichever mailbox triggered the mwi event. Now all of them get counted regardless. Also fixes a bug involving parsing of the mailbox option in sip.conf so that trailing and leading spaces before/after commas are trimmed. (closes issue ASTERISK-18067) Reported by: aragon (closes issue ASTERISK-15479) Reported by: Ben Winslow Patches: chan_sip.c-mwi_multi_mailbox_fix-1.6.2.13.diff (License #5288) patch uploaded by Ben Winslow ........ ................ 2011-08-16 17:23 +0000 [r332117] Richard Mudgett * /, main/features.c, CHANGES, configs/features.conf.sample, main/asterisk.c: Merged revisions 332101 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332101 | rmudgett | 2011-08-16 12:17:28 -0500 (Tue, 16 Aug 2011) | 140 lines Merged revisions 332100 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332100 | rmudgett | 2011-08-16 11:31:36 -0500 (Tue, 16 Aug 2011) | 133 lines Fix multiple parking issues. JIRA ASTERISK-17183 Multi-parkinglot directs calls to wrong parkinglot. JIRA ASTERISK-17870 Cannot retrieve parked calls. JIRA ASTERISK-17430 ParkedCall() with no extension should pickup first available call and does not. JIRA AST-576 Issues with parking lots * Removed searching for parking lots by extension. Parking lots can only be found by the parking lot name since parking lot access extensions and spaces are not guaranteed to be unique. * Added parking_lot_name option to the Park and ParkedCall applications. Updated documentation for Park and ParkedCall applications. * Add parkext_exclusive configuration option to make parking entry extensions specify which parking lot they access. (closes issue ASTERISK-17183) Reported by: David Cabrejos Tested by: rmudgett, David Cabrejos (closes issue ASTERISK-17870) Reported by: Remi Quezada (closes issue ASTERISK-17430) Reported by: Philippe Lindheimer JIRA ASTERISK-17452 Parking_offset not used JIRA AST-624 'next' setting for findslot does nothing * Reimplemented since findslot feature option broken by -r114655. (closes issue ASTERISK-17452) Reported by: David Woolley Tested by: rmudgett JIRA ASTERISK-15792 Dialplan continues execution after transfer to park. This happens for DTMF attended transfer, DTMF blind transfer, and DTMF one-touch-parking if the party initiating these features also initiated the call. * Fixed the return code from the affected builtin features when parking a call. (closes issue ASTERISK-15792) Reported by: Mat Murdock Tested by: rmudgett, twilson JIRA AST-607 The courtesytone is not playing to the expected call when picking up a parked call. This is mostly a documentation problem. However, the option is not reset to the default when features.conf is reloaded. * Updated features.conf.sample documentation for courtesytone and parkedplay options. * Reset the parkedplay option to default when features.conf is reloaded. JIRA AST-615 AMI Park action followed by features reload results in orphaned channels in parking lot. * Reloading features.conf will not touch parking lots that have calls still parked in them. Reload again at a later time. Misc additional fixes: * Added unit test for parking lot dialplan usage checking. * Made update connected line when a parked call is retrieved from a parking lot. * Made retrieved parked call stop ringing or MOH depending upon how the call was waiting in the parking lot. * Made CLI "features show" indicate if the parking lot is enabled for use. * Added PARKINGDYNEXTEN channel variable to allow dynamic parking lots to specify the parking lot access extension. * Made AMI ParkedCalls action ParkedCall events have a Parkinglot header. * Made AMI ParkedCalls action ParkedCallsComplete event have a Total header. * Fixed potential deadlock from AMI Park action holding channel locks while calling masq_park_call(). * Fixed several places where ast_strdupa() were used inside of loops. (Mostly fixed by refactoring the loop body into its own function.) * Fixed copy_parkinglot() copying too much from the source parking lot. Extracted the parking lot configuration settings into struct parkinglot_cfg. * Refactored courtesytone playing code to put the channel not playing the tone in autoservice. * Fix when pbx-parkingfailed is played that the other channel is put in autoservice if it exists. * Fixed parkinglot reference leak in parked_call_exec() error paths. * Fixed parkinglot_unref() use of parkinglot after it was unreffed. * Made destroy the struct ast_parkinglot parkings lock when done. * Refactored the features.conf parking lot configuration code to eliminate redundancy. * Fixed feature reload to better protect parking lots. * Fixed parking lot container reference leak in handle_parkedcalls(). * Fixed the total count in handle_parkedcalls(). Review: https://reviewboard.asterisk.org/r/1358/ ........ ................ 2011-08-16 15:21 +0000 [r332028-332044] Matthew Nicholson * /, channels/sip/include/sip.h: Merged revisions 332042 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r332042 | mnicholson | 2011-08-16 10:20:48 -0500 (Tue, 16 Aug 2011) | 2 lines fix a code comment AST-580 ........ * /, UPGRADE.txt, CHANGES: Merged revisions 332029 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r332029 | mnicholson | 2011-08-16 10:17:16 -0500 (Tue, 16 Aug 2011) | 2 lines Moved notes about 'storesipcause' to UPGRADE.txt from CHANGES AST-580 ........ * /, channels/chan_sip.c, channels/sip/include/sip.h: Merged revisions 332027 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332027 | mnicholson | 2011-08-16 10:08:40 -0500 (Tue, 16 Aug 2011) | 9 lines Merged revisions 332026 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332026 | mnicholson | 2011-08-16 10:06:31 -0500 (Tue, 16 Aug 2011) | 2 lines use DEFAULT_STORE_SIP_CAUSE to set the default value for the 'storesipcause' option AST-580 ........ ................ 2011-08-16 14:47 +0000 [r332024] Olle Johansson * channels/chan_local.c: Formatting changes while working with DTMF... 2011-08-16 14:41 +0000 [r332023] Matthew Nicholson * /, channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Merged revisions 332022 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332022 | mnicholson | 2011-08-16 09:40:37 -0500 (Tue, 16 Aug 2011) | 16 lines In 10 and trunk this option is disabled by default. Merged revisions 332021 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332021 | mnicholson | 2011-08-16 09:20:43 -0500 (Tue, 16 Aug 2011) | 7 lines Added the 'storesipcause' option to sip.conf to allow the user to disable the setting of HASH(SIP_CAUSE,) on the channel. Having chan_sip set HASH(SIP_CAUSE,) on the channel carries a significant performance penalty because of the usage of the MASTER_CHANNEL() dialplan function. AST-580 ........ ................ 2011-08-15 17:36 +0000 [r331957] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 331956 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331956 | rmudgett | 2011-08-15 12:35:03 -0500 (Mon, 15 Aug 2011) | 20 lines Merged revisions 331955 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331955 | rmudgett | 2011-08-15 12:24:08 -0500 (Mon, 15 Aug 2011) | 13 lines Fix some minor chan_dahdi config load issues. * Address chan_dahdi.conf dahdichan option todo item about needing line number. * Make ignore_failed_channels option also apply to dahdichan option. * Don't attempt to create a default pseudo channel if the chan_dahdi.conf channel/channels option is not allowed. * Add a similar check for dahdichan in normal chan_dahdi.conf sections as is done in users.conf. ........ ................ 2011-08-15 15:24 +0000 [r331903] Paul Belanger * main/rtp_engine.c, /: Merged revisions 331894 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331894 | pabelanger | 2011-08-15 11:22:45 -0400 (Mon, 15 Aug 2011) | 12 lines Merged revisions 331886 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331886 | pabelanger | 2011-08-15 11:21:16 -0400 (Mon, 15 Aug 2011) | 5 lines Fix noisy message when briding channels (closes issue ASTERISK-18270) Reported by: Federico Alves ........ ................ 2011-08-15 15:15 +0000 [r331869] David Vossel * /, channels/chan_sip.c: Merged revisions 331868 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331868 | dvossel | 2011-08-15 10:14:13 -0500 (Mon, 15 Aug 2011) | 12 lines Merged revisions 331867 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331867 | dvossel | 2011-08-15 10:12:16 -0500 (Mon, 15 Aug 2011) | 6 lines Fixes locking inversion issues present in the handling of the sip REFER method. (closes issue ASTERISK-18082) Reported by: James Van Vleet ........ ................ 2011-08-15 13:27 +0000 [r331830] Olle Johansson * channels/chan_sip.c: Formatting guideline fixes 2011-08-12 19:06 +0000 [r331776] Matthew Nicholson * /, apps/app_queue.c: Merged revisions 331775 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331775 | mnicholson | 2011-08-12 14:03:31 -0500 (Fri, 12 Aug 2011) | 17 lines Merged revisions 331774 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331774 | mnicholson | 2011-08-12 14:01:27 -0500 (Fri, 12 Aug 2011) | 11 lines Unlock the channel before calling update_queue. Holding the channel lock when calling update_queue which attempts to lock the queue lock can cause a deadlock. This deadlock involves the following chain: 1. hold chan lock -> wait queue lock 2. hold queue lock -> wait agent list lock 3. hold agent list lock -> wait chan list lock 4. hold chan list lock -> wait chan lock ........ ................ 2011-08-12 19:01 +0000 [r331773] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 331772 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331772 | rmudgett | 2011-08-12 13:59:45 -0500 (Fri, 12 Aug 2011) | 15 lines Merged revisions 331771 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331771 | rmudgett | 2011-08-12 13:58:40 -0500 (Fri, 12 Aug 2011) | 8 lines Suppress warning message when using DAHDITransfer or DAHDIHangup. * The fake event should only be processed by the channel that currently owns the private and not the associated call waiting or 3-way channel. JIRA AST-620 JIRA SWP-3616 ........ ................ 2011-08-12 18:03 +0000 [r331717] Jonathan Rose * apps/app_dial.c, /, apps/app_meetme.c: Merged revisions 331644 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331644 | jrose | 2011-08-12 11:18:57 -0500 (Fri, 12 Aug 2011) | 9 lines Merged revisions 331635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331635 | jrose | 2011-08-12 10:49:17 -0500 (Fri, 12 Aug 2011) | 1 line Fixes 32bit compilation warnings brought on by 331634 in app_dial and app_meetme ........ ................ 2011-08-12 17:56 +0000 [r331716] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 331715 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331715 | rmudgett | 2011-08-12 12:54:47 -0500 (Fri, 12 Aug 2011) | 29 lines Merged revisions 331714 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331714 | rmudgett | 2011-08-12 12:47:57 -0500 (Fri, 12 Aug 2011) | 22 lines AMI actions DAHDIHangup and DAHDITransfer have no effect. The AMI actions DAHDIHangup and DAHDITransfer have no effect on a DAHDI channel. These two AMI actions are highly specialized to analog channels and appear to make the channel behave like a jack port for headsets. * Made the faked DAHDI event get processed before a normal media stream read in dahdi_read() instead of trying to trigger an exception read by setting the AST_FLAG_EXCEPTION flag. Apparently a change was made long ago that changed how AST_FLAG_EXCEPTION is processed in the core. Unfortunately, the faked DAHDI events no longer worked when that happened. * Updated the DAHDI AMI action documentation for the following actions: DAHDITransfer, DAHDIHangup, DAHDIDialOffhook, DAHDIDNDon, DAHDIDNDoff, DAHDIShowChannels, and DAHDIRestart. * Made use sscanf() instead of atoi() for better error checking of the DAHDIChannel header string. JIRA AST-620 JIRA SWP-3616 ........ ................ 2011-08-12 16:32 +0000 [r331660] Terry Wilson * /, tests/test_netsock2.c: Merged revisions 331659 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331659 | twilson | 2011-08-12 11:31:21 -0500 (Fri, 12 Aug 2011) | 11 lines Merged revisions 331658 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331658 | twilson | 2011-08-12 11:30:26 -0500 (Fri, 12 Aug 2011) | 4 lines Fix netsock2 multiple zero-expansion test Remove erroneous single bracket. ........ ................ 2011-08-12 16:22 +0000 [r331657] Kinsey Moore * /, main/logger.c: Merged revisions 331654 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331654 | kmoore | 2011-08-12 11:21:37 -0500 (Fri, 12 Aug 2011) | 19 lines Merged revisions 331649 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331649 | kmoore | 2011-08-12 11:20:25 -0500 (Fri, 12 Aug 2011) | 12 lines Logger does not warn of failure to open logging channels Currently, logger only prints an error message to stderr when it fails to open a logger channel where many users will not see it because the logger lock is held. The alternative provided by this patch is to log the error to all attached consoles in the hopes that it will be easier to see. Additionally, this patch prevents the failed logger channel from being added to the list where it would silently fail on each call to the Asterisk logger. (closes issue ASTERISK-16231) Review: https://reviewboard.asterisk.org/r/1338 ........ ................ 2011-08-11 21:55 +0000 [r331580] Jason Parker * apps/app_dial.c, /, apps/app_meetme.c: Merged revisions 331579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331579 | qwell | 2011-08-11 16:54:54 -0500 (Thu, 11 Aug 2011) | 13 lines Merged revisions 331578 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331578 | qwell | 2011-08-11 16:46:39 -0500 (Thu, 11 Aug 2011) | 6 lines Use proper values for 64-bit option flags. Also, reusing bits es no bueno, so change the value of a duplicate. (issue ASTERISK-18239) ........ ................ 2011-08-11 21:44 +0000 [r331577] Richard Mudgett * /, funcs/func_shell.c: Merged revisions 331576 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331576 | rmudgett | 2011-08-11 16:42:21 -0500 (Thu, 11 Aug 2011) | 16 lines Merged revisions 331575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331575 | rmudgett | 2011-08-11 16:39:58 -0500 (Thu, 11 Aug 2011) | 9 lines Segfault in shell_helper in func_shell.c. The return value of popen() was not checked for failure to open. (closes issue ASTERISK-18109) JIRA SWP-3633 Reported by: Michael Myles Tested by: rmudgett ........ ................ 2011-08-10 22:24 +0000 [r331519] Kinsey Moore * /, channels/chan_sip.c: Merged revisions 331518 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331518 | kmoore | 2011-08-10 17:23:49 -0500 (Wed, 10 Aug 2011) | 17 lines Merged revisions 331517 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331517 | kmoore | 2011-08-10 17:23:08 -0500 (Wed, 10 Aug 2011) | 10 lines SIP Notify via AMI or CLI leaks SIP PVTs Any SIP notify sent via AMI or CLI leaks a SIP PVT with ref count +2. Removing the additional ref just before the invite and adding an unref following it corrects the issue as seen via REF_DEBUG. The unref existed in a distant revision and it appears as though the wrong ref operation was removed. (closes issue ASTERISK-18091) Review: https://reviewboard.asterisk.org/r/1332/ ........ ................ 2011-08-10 20:51 +0000 [r331419-331463] Richard Mudgett * /, main/logger.c: Merged revisions 331462 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331462 | rmudgett | 2011-08-10 15:41:35 -0500 (Wed, 10 Aug 2011) | 37 lines Merged revisions 331461 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331461 | rmudgett | 2011-08-10 15:29:59 -0500 (Wed, 10 Aug 2011) | 30 lines Output of queue log not started until logger reloaded. ASTERISK-15863 caused a regression with queue logging. The output of the queue log is not started until the logger configuration is reloaded. * Queue log initialization is completely delayed until the first message is posted to the queue log system. Including the initial opening of the queue log file. * Fixed rotate_file() ROTATE strategy to give the file just rotated out to the configured exec function after rotate. Just like the other strategies. * Fixed logger reload to always post the queue reload entry instead of just if there is a queue log file. * Refactored some code to eliminate some redundancy and to reduce stack utilization. (closes issue ASTERISK-17036) JIRA SWP-2952 Reported by: Juan Carlos Valero Patches: jira_asterisk_17036_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett (closes issue ASTERISK-18208) Reported by: Christian Pinedo Review: https://reviewboard.asterisk.org/r/1333/ ........ ................ * /, main/features.c: Merged revisions 331420 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r331420 | rmudgett | 2011-08-10 14:07:53 -0500 (Wed, 10 Aug 2011) | 2 lines Make sure feature_request_and_dial() initializes outstate if passed in. ........ * /, main/features.c, CHANGES: Merged revisions 331418 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r331418 | rmudgett | 2011-08-10 13:25:08 -0500 (Wed, 10 Aug 2011) | 6 lines Revert -r318141. It was a band-aid that only partially fixed parking. A better fix is on reviewboard review 1358. (issue ASTERISK-17374) ........ 2011-08-10 15:45 +0000 [r331371] Jonathan Rose * channels/chan_sip.c, CHANGES: SIP display-name needed to be empty for Avaya IP500 In order to address a compatability issue with certain features on certain devices which rely on display name content to change behavior, initreqprep in chan_sip.c has been changed to no longer substitute cid_number into the display name when cid_name isn't present. Instead, it will send no display name in that case. (closes issue ASTERISK-16198) Reported by: Walter Doekes Review: https://reviewboard.asterisk.org/r/1341/ 2011-08-10 13:49 +0000 [r331317] Kinsey Moore * main/manager.c, /: Merged revisions 331316 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331316 | kmoore | 2011-08-10 08:48:41 -0500 (Wed, 10 Aug 2011) | 15 lines Merged revisions 331315 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331315 | kmoore | 2011-08-10 08:47:46 -0500 (Wed, 10 Aug 2011) | 8 lines AMI action ModuleReload returns Error if Module: missing or empty An empty string was not being checked for properly causing identification of the module to be reloaded to fail and return an Error with message "No such module." (closes issue AST-616) ........ ................ 2011-08-09 23:17 +0000 [r331266] Richard Mudgett * main/pbx.c, /, channels/chan_sip.c, main/features.c, channels/chan_iax2.c, apps/app_parkandannounce.c: Merged revisions 331265 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331265 | rmudgett | 2011-08-09 18:12:49 -0500 (Tue, 09 Aug 2011) | 22 lines Merged revisions 331248 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331248 | rmudgett | 2011-08-09 17:12:59 -0500 (Tue, 09 Aug 2011) | 15 lines Misc minor items found in code. * Add some reentrancy protection in pbx.c when creating the contexts_table hash table. * Fix inverted test in chan_sip.c conditional code. * Fix uninitialized variable and use of the wrong variable in chan_iax2.c. * Fix test of return value in app_parkandannounce.c. Explicitly testing for -1 is bad if the function does not actually return that value when it fails. * Fixup some comments and add some curly braces in features.c. ........ ................ 2011-08-09 17:12 +0000 [r331202] Alexandr Anikin * addons/ooh323c/src/ooGkClient.c, addons/chan_ooh323.c, /, addons/ooh323c/src/ooLogChan.c, addons/ooh323c/src/ooq931.c: Merged revisions 331147,331200 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331147 | may | 2011-08-09 20:16:55 +0400 (Tue, 09 Aug 2011) | 11 lines Merged revisions 331146 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331146 | may | 2011-08-09 20:13:09 +0400 (Tue, 09 Aug 2011) | 4 lines move ast_cond_signal for admitted call after all data filled/freed clear all log channels by pointed number not only first free allocated callToken in ooh323_answer ........ ................ r331200 | may | 2011-08-09 20:36:39 +0400 (Tue, 09 Aug 2011) | 9 lines Setup IP proto version for call in GK mode Added additional check for IP semantics before parse destination by ast_parse_args due to it can parse numeric as IP. (closes issue ASTERISK-18218) Reported by: slesru Patch: ASTERISK-18218.patch ................ 2011-08-09 17:08 +0000 [r331201] Kinsey Moore * funcs/func_enum.c, UPGRADE.txt, main/enum.c: Allow ENUM query functions to report lookup errors The ENUM dialplan functions do not report DNS query errors properly. It is useful to differentiate between failed query (e.g. non-existent domain) vs. no data records of the appropriate type. This is required to make overlapped dialing work. (closes issue ASTERISK-13769) Review: https://reviewboard.asterisk.org/r/1355/ Patch-by: Timo Teras 2011-08-09 16:02 +0000 [r331140-331144] Jason Parker * /, doc/asterisk.8: Merged revisions 331143 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331143 | qwell | 2011-08-09 10:59:54 -0500 (Tue, 09 Aug 2011) | 9 lines Merged revisions 331142 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331142 | qwell | 2011-08-09 10:58:16 -0500 (Tue, 09 Aug 2011) | 1 line Regenerate asterisk man page from sgml. ........ ................ * /, doc/asterisk.8, configs/asterisk.conf.sample, configs/voicemail.conf.sample, doc/asterisk.sgml: Merged revisions 331139 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331139 | qwell | 2011-08-09 10:50:07 -0500 (Tue, 09 Aug 2011) | 19 lines Merged revisions 306999 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r306999 | lathama | 2011-02-08 14:22:35 -0600 (Tue, 08 Feb 2011) | 12 lines Documentation Updates Note default polling setting in voicemail.conf Add missing config to asterisk.conf Update manpage (issue #16505) Reported by: tzafrir Patches: asterisk_sgml_fixes_demo.diff uploaded by tzafrir (license 46) Tested by: lathama, tzafrir ........ ................ * doc/asterisk.8, configs/asterisk.conf.sample, configs/voicemail.conf.sample, doc/asterisk.sgml: Merged revisions 331138 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r331138 | qwell | 2011-08-09 10:47:20 -0500 (Tue, 09 Aug 2011) | 1 line Revert merge of r306999, due to merge conflict. ........ 2011-08-08 22:59 +0000 [r331042-331098] Terry Wilson * /, UPGRADE.txt, CHANGES, include/asterisk/manager.h: Merged revisions 331097 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r331097 | twilson | 2011-08-08 17:59:01 -0500 (Mon, 08 Aug 2011) | 5 lines Bump the AMI protocol version to 1.2 As a result of converting Unlink events that were missed in the AMI 1.1 update to Bridge events, the AMI protocol version is being incremented. ........ * main/channel.c, /, CHANGES: Merged revisions 331041 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r331041 | twilson | 2011-08-08 16:12:51 -0500 (Mon, 08 Aug 2011) | 6 lines Replace AMI Unlink events with Bridge events A previous update converted some of the Link and Unlink events to Bridge events, but a couple of Unlink events were missed. This patch rectifies the situation. (closes issues ASTERISK-17455) ........ 2011-08-08 20:54 +0000 [r331000-331040] Kinsey Moore * /, res/res_musiconhold.c: Merged revisions 331039 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331039 | kmoore | 2011-08-08 15:53:30 -0500 (Mon, 08 Aug 2011) | 18 lines Merged revisions 331038 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331038 | kmoore | 2011-08-08 15:52:45 -0500 (Mon, 08 Aug 2011) | 11 lines In-queue MOH stops after a periodic announcement If the seek value is past the end of file when resuming G.722 MOH, MOH will cease to function for the duration of the MOH session through all starts and stops until saved state is cleared. Adjusting the code to guarantee a single valid read (which is already assumed) fixes the bug. (closes issue ASTERISK-18077) Review: https://reviewboard.asterisk.org/r/1328/ Tested-by: Jonathan Rose ........ ................ * configs/queues.conf.sample, apps/app_queue.c: Log queue member name when state_interface is set for ADDMEMBER and REMOVEMEMBER events app_queue logs the events ADDMEMBER and REMOVEMEMBER with the agent field set to the interface value rather than the membername value when a member is added with a state_interface value set. However all other member related queue events are logged with the membername when a state_interface is set. This patch makes these fields optionally more consistent and correct. (closes issue ASTERISK-14769) Review: https://reviewboard.asterisk.org/r/1286 Patch-by: Jamuel Starkey Tested-by: Kinsey Moore * apps/app_queue.c: app_queue: Add StateInterface to output of "queue show" and "QueueStatus" This patch adds the state_interface of the queue member struct to the output of "queue show" (CLI command) and "QueueStatus" (AMI action) when displaying relevant queue member information. For the AMI event message the variable StateInterface has been added. (closes issue ASTERISK-18071) Review: https://reviewboard.asterisk.org/r/1300/ Patch-by: Jamuel Starkey 2011-08-05 15:57 +0000 [r330941] David Vossel * /, codecs/codec_resample.c: Merged revisions 330940 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r330940 | dvossel | 2011-08-05 10:53:49 -0500 (Fri, 05 Aug 2011) | 2 lines The slin resampler is no longer dependent on an external library, but the dependency was not removed correctly. ........ 2011-08-05 08:47 +0000 [r330903] Alexandr Anikin * addons/ooh323c/src/ooGkClient.c, /, addons/ooh323c/src/ooCmdChannel.c: Merged revisions 330899 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r330899 | may | 2011-08-05 11:38:28 +0400 (Fri, 05 Aug 2011) | 11 lines Merged revisions 330827 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330827 | may | 2011-08-04 23:37:16 +0400 (Thu, 04 Aug 2011) | 4 lines change gk client behaivour on rrq/grq failures to setup timers and next tries after timeout instead of complete failure in the ooh323 stack ........ ................ 2011-08-04 20:53 +0000 [r330845] Terry Wilson * /, configure, configure.ac: Merged revisions 330844 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r330844 | twilson | 2011-08-04 15:51:23 -0500 (Thu, 04 Aug 2011) | 11 lines Merged revisions 330843 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330843 | twilson | 2011-08-04 15:29:19 -0500 (Thu, 04 Aug 2011) | 4 lines Make libsrtp instructions more explicit when linking fails (closes issue ASTERISK-18139) ........ ................ 2011-08-03 15:16 +0000 [r330707-330764] Kinsey Moore * /, main/Makefile: Merged revisions 330763 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r330763 | kmoore | 2011-08-03 10:15:26 -0500 (Wed, 03 Aug 2011) | 16 lines Merged revisions 330762 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330762 | kmoore | 2011-08-03 10:14:36 -0500 (Wed, 03 Aug 2011) | 9 lines editing files in main/editline does not ensure rebuild of libedit.a When editing a source file in main/editline, the build system does not rebuild libedit.a and uses the already existing one instead. Adding a PHONY to CHECK_SUBDIR fixes this problem. (closes issue ASTERISK-16221) Patch-by: Walter Doekes ........ ................ * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions 330706 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r330706 | kmoore | 2011-08-03 08:39:06 -0500 (Wed, 03 Aug 2011) | 17 lines Merged revisions 330705 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330705 | kmoore | 2011-08-03 08:38:17 -0500 (Wed, 03 Aug 2011) | 10 lines Call pickup broken for DAHDI channels when beginning with # The call pickup feature did not work on DAHDI devices for anything other than feature codes beginning with * since all feature codes in chan_dahdi were originally hard-coded to begin with *. This patch is also applied to chan_dahdi.c to fix this bug with radio modes. (closes issue AST-621) Review: https://reviewboard.asterisk.org/r/1336/ ........ ................ 2011-08-02 20:54 +0000 [r330650] Kevin P. Fleming * /, res/res_jabber.c: Merged revisions 330649 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r330649 | kpfleming | 2011-08-02 15:52:44 -0500 (Tue, 02 Aug 2011) | 9 lines Merged revisions 330648 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330648 | kpfleming | 2011-08-02 15:51:56 -0500 (Tue, 02 Aug 2011) | 2 lines Convert an error message to actually be helpful. ........ ................ 2011-08-02 16:19 +0000 [r330577-330593] David Vossel * /, channels/chan_iax2.c: Merged revisions 330586 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r330586 | dvossel | 2011-08-02 11:17:59 -0500 (Tue, 02 Aug 2011) | 15 lines Merged revisions 330581 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330581 | dvossel | 2011-08-02 11:15:08 -0500 (Tue, 02 Aug 2011) | 8 lines Fixes crash in chan_iax2. Fixes crash in chan_iax2 resulting from an edge case in the way control frames are queued during calltoken negotiation is complete. (closes issue ASTERISK-17610) Reported by: mgrobecker ........ ................ * /, channels/chan_sip.c: Merged revisions 330579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r330579 | dvossel | 2011-08-02 11:08:57 -0500 (Tue, 02 Aug 2011) | 9 lines Merged revisions 330578 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330578 | dvossel | 2011-08-02 11:07:02 -0500 (Tue, 02 Aug 2011) | 2 lines Optimization to buffer initialization fix. ........ ................ * /, channels/chan_sip.c: Merged revisions 330576 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r330576 | dvossel | 2011-08-02 10:55:36 -0500 (Tue, 02 Aug 2011) | 12 lines Merged revisions 330575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330575 | dvossel | 2011-08-02 10:53:21 -0500 (Tue, 02 Aug 2011) | 5 lines Fixes uninitialized string buffer in log message. (closes issue ASTERISK-17200) Reported by: lmadsen ........ ................ 2011-08-01 15:24 +0000 [r330435] Kinsey Moore * /, main/say.c: Merged revisions 330434 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r330434 | kmoore | 2011-08-01 10:23:29 -0500 (Mon, 01 Aug 2011) | 16 lines Merged revisions 330433 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330433 | kmoore | 2011-08-01 10:22:10 -0500 (Mon, 01 Aug 2011) | 9 lines Incorrect playback for Spanish in some circumstances When you say the time in spanish and it is 01:00 - 01:59 or 13:00 - 13:59 you must use female pronunciation "1F". The function "say_date_with_format_es" does not take this in account. (closes ASTERISK-15016) Patch-by: Luis Jimenez ........ ................ 2011-07-31 00:19 +0000 [r330370-330379] Richard Mudgett * main/astobj2.c: Fixed compiler warning and a couple prototype mismatches. * main/channel.c, /: Merged revisions 330369 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r330369 | rmudgett | 2011-07-30 18:57:56 -0500 (Sat, 30 Jul 2011) | 11 lines Merged revisions 330368 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330368 | rmudgett | 2011-07-30 18:56:29 -0500 (Sat, 30 Jul 2011) | 4 lines Remove some redundant locking code in ast_do_masquerade(). Also updated some comments. ........ ................ 2011-07-30 15:54 +0000 [r330313] Gregory Nietsky * main/channel.c, /: Merged revisions 330312 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r330312 | irroot | 2011-07-30 17:34:41 +0200 (Sat, 30 Jul 2011) | 15 lines Merged revisions 330311 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330311 | irroot | 2011-07-30 17:25:16 +0200 (Sat, 30 Jul 2011) | 9 lines prevent double masqurading channels when one is been hung up and deadlock avoidance is used. There is a race condition in ast_do_masquerade / ast_hangup (at least) Reported by me signed off by schmidts with input from David Vossel Review: https://reviewboard.asterisk.org/r/1323/ ........ ................ 2011-07-29 19:34 +0000 [r330273] Russell Bryant * include/asterisk/astobj2.h, tests/test_astobj2.c, channels/chan_iax2.c, main/astobj2.c: astobj2: Avoid using temporary objects + ao2_find() with OBJ_POINTER. There is a fairly common pattern making its way through the code base where we put a temporary object on the stack so we can call ao2_find() with OBJ_POINTER. The purpose is so that it can be passed into the object hash function. However, this really seems like a hack and potentially error prone. This patch is a first stab at approach to avoid having to do that. It adds a new flag, OBJ_KEY, which can be used instead of OBJ_POINTER in these situations. Then, the hash function can know whether it was given an object or some custom data to hash. The patch also changes some uses of ao2_find() for iax2_user and iax2_peer objects to reflect how OBJ_KEY would be used. So long, and thanks for all the fish. Review: https://reviewboard.asterisk.org/r/1184/ 2011-07-29 17:20 +0000 [r330205-330221] Sean Bright * /, formats/format_wav.c: Merged revisions 330217 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r330217 | seanbright | 2011-07-29 13:19:42 -0400 (Fri, 29 Jul 2011) | 9 lines Merged revisions 330213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330213 | seanbright | 2011-07-29 13:18:56 -0400 (Fri, 29 Jul 2011) | 2 lines Correct the check for O_RDONLY. ........ ................ * /, formats/format_wav.c: Merged revisions 330204 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r330204 | seanbright | 2011-07-29 12:58:40 -0400 (Fri, 29 Jul 2011) | 9 lines Merged revisions 330203 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330203 | seanbright | 2011-07-29 12:58:08 -0400 (Fri, 29 Jul 2011) | 2 lines Only write to wav files that were opened to be written to. ........ ................ 2011-07-29 05:27 +0000 [r330163] Paul Belanger * /, apps/app_confbridge.c: Merged revisions 330162 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r330162 | pabelanger | 2011-07-29 01:25:18 -0400 (Fri, 29 Jul 2011) | 4 lines Fix typo pointed out on #asterisk Thanks notten ........ 2011-07-28 21:46 +0000 [r330109] Terry Wilson * /, main/term.c: Merged revisions 330108 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r330108 | twilson | 2011-07-28 16:44:31 -0500 (Thu, 28 Jul 2011) | 9 lines Merged revisions 330107 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330107 | twilson | 2011-07-28 16:42:41 -0500 (Thu, 28 Jul 2011) | 2 lines Make console colors work for TERM=xterm-256color ........ ................ 2011-07-28 17:16 +0000 [r330052] Richard Mudgett * /, channels/sig_pri.c: Merged revisions 330051 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r330051 | rmudgett | 2011-07-28 12:10:37 -0500 (Thu, 28 Jul 2011) | 29 lines Merged revisions 330050 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r330050 | rmudgett | 2011-07-28 12:04:24 -0500 (Thu, 28 Jul 2011) | 22 lines Merged revisions 330033 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r330033 | rmudgett | 2011-07-28 11:26:38 -0500 (Thu, 28 Jul 2011) | 15 lines Datacalls with B410P fail. Incoming and outgoing call legs of a data call are using different formats: a-law, u-law. When the call is bridged, the media stream is run through translation to convert the media formats. The translation is bad for data calls. * Make incoming call that does not explicitly specify u-law or a-law use the DAHDI channel's default law. The outgoing call always uses the default law from the DAHDI channel. (closes issue ABE-2800) Patches: jira_abe_2800_companding.patch (license #5621) patch uploaded by rmudgett .......... ................ ................ 2011-07-28 15:46 +0000 [r329996] Jason Parker * /, channels/chan_sip.c: Merged revisions 329995 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r329995 | qwell | 2011-07-28 10:45:49 -0500 (Thu, 28 Jul 2011) | 13 lines Merged revisions 329994 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329994 | qwell | 2011-07-28 10:45:24 -0500 (Thu, 28 Jul 2011) | 6 lines Fix a SIP transfer deadlock. The locking in this function is very scary. There are like 6 structs involved. (closes issue AST-470) ........ ................ 2011-07-28 15:30 +0000 [r329993] Matthew Nicholson * /, res/res_fax.c: Merged revisions 329992 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r329992 | mnicholson | 2011-07-28 10:28:21 -0500 (Thu, 28 Jul 2011) | 13 lines Merged revisions 329991 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329991 | mnicholson | 2011-07-28 10:26:56 -0500 (Thu, 28 Jul 2011) | 6 lines check for CONFIG_STATUS_FILE_INVALID when loading the res_fax config file Patch by: tzafrir Reported by: tzafrir (closes issue ASTERISK-18161) ........ ................ 2011-07-28 13:04 +0000 [r329897-329953] Sean Bright * configs/confbridge.conf.sample, /: Merged revisions 329952 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r329952 | seanbright | 2011-07-28 09:03:58 -0400 (Thu, 28 Jul 2011) | 4 lines The default conf-usermenu says that '8' can be used to leave the conference, so put that in the sample user menu. '5' is supposed to extend the conference, but there doesn't appear to be a concept of that in the menu actions. ........ * /, apps/app_confbridge.c: Merged revisions 329950 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r329950 | seanbright | 2011-07-28 08:43:55 -0400 (Thu, 28 Jul 2011) | 1 line Correct the spelling of 'conference.' ........ * /, channels/chan_sip.c: Merged revisions 329896 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r329896 | seanbright | 2011-07-28 07:35:27 -0400 (Thu, 28 Jul 2011) | 9 lines Merged revisions 329895 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329895 | seanbright | 2011-07-28 07:34:33 -0400 (Thu, 28 Jul 2011) | 2 lines Make the output of Externhost in 'sip show settings' more consistent. ........ ................ 2011-07-27 21:22 +0000 [r329835-329856] Jonathan Rose * main/cdr.c, main/pbx.c, include/asterisk/cdr.h, CHANGES: reverting 329840 due to failing tests. Going to change this feature to be purely optional. * main/cdr.c, main/pbx.c, include/asterisk/cdr.h, CHANGES: Adds cdr logging of calls resulting in CONGESTION Applies a patch made a long time ago by alecdavis which adds a CDR feature for logging calls that failed due to congestion. (closes issue #15907) Reported by: alecdavis Patches: cdr_congestion.diff.txt uploaded by alecdavis (license #5546) Review: https://reviewboard.asterisk.org/r/454/ 2011-07-27 19:19 +0000 [r329775] Sean Bright * /, Makefile.moddir_rules: Merged revisions 329771 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r329771 | seanbright | 2011-07-27 15:18:47 -0400 (Wed, 27 Jul 2011) | 15 lines Merged revisions 329767 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329767 | seanbright | 2011-07-27 15:17:46 -0400 (Wed, 27 Jul 2011) | 8 lines Explicitly sort the module list so that the menuselect lists are sorted. (closes ASTERISK-18141) Reported by: Richard Miller Patches: sort-order.diff uploaded by seanbright (License #5060) Tested by: leifmadsen ........ ................ 2011-07-27 18:12 +0000 [r329711] Jonathan Rose * /, configs/indications.conf.sample: Merged revisions 329710 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r329710 | jrose | 2011-07-27 13:11:07 -0500 (Wed, 27 Jul 2011) | 14 lines Merged revisions 329709 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329709 | jrose | 2011-07-27 13:10:30 -0500 (Wed, 27 Jul 2011) | 8 lines Fix New Zealand indications profile based on http://www.telepermit.co.nz/TNA102.pdf (closes issue ASTERISK-16263) Reported by: richardf Patches: nz-indications.patch uploaded by richardf (License #6015) ........ ................ 2011-07-27 15:26 +0000 [r329671] Sean Bright * /, main/loader.c: Merged revisions 329670 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r329670 | seanbright | 2011-07-27 11:25:53 -0400 (Wed, 27 Jul 2011) | 2 lines Sort the module list so that 'module show' is alphabetical. ........ 2011-07-27 04:27 +0000 [r329615] Tilghman Lesher * /, cdr/cdr_odbc.c: Merged revisions 329614 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r329614 | tilghman | 2011-07-26 23:25:26 -0500 (Tue, 26 Jul 2011) | 13 lines Merged revisions 329613 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329613 | tilghman | 2011-07-26 23:23:46 -0500 (Tue, 26 Jul 2011) | 6 lines Duration and billsec are swapped in high resolution time. Closes ASTERISK-18024 Patches: 20110726__ASTERISK-18024.diff by Tilghman Lesher (License 5003) ........ ................ 2011-07-26 14:27 +0000 [r329530-329564] Jonathan Rose * /, apps/app_voicemail.c: Merged revisions 329538 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r329538 | jrose | 2011-07-26 09:19:34 -0500 (Tue, 26 Jul 2011) | 11 lines Merged revisions 329529 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329529 | jrose | 2011-07-26 09:04:55 -0500 (Tue, 26 Jul 2011) | 5 lines Changes sound file for prepend "then-press-pound" to "vm-then-pound" which is the same prompt, only it turned out "then-press-pound" was part of extra sounds. Also, vm is more appropriate anyway. ........ ................ * include/asterisk/app.h, /, configs/voicemail.conf.sample, main/app.c, apps/app_voicemail.c: Merged revisions 329528 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r329528 | jrose | 2011-07-26 08:52:34 -0500 (Tue, 26 Jul 2011) | 24 lines Merged revisions 329527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329527 | jrose | 2011-07-26 08:25:35 -0500 (Tue, 26 Jul 2011) | 17 lines Fixes some voicemail forwarding behavior based around prepend mode. Formerly, prepend forwarding would have the user record a message with no useful prompt and an expectation for the user to push a button on the phone when finished recording. If a length of silence was detected instead, the recording would be canceled and the user would re-enter the voicemail forwarding menu. Subsequent time-outs in prepend recording would also bug out in the sense that they would write over the original message and get sent to the recipient regardless of whether they timed out or were accepted. This patch fixes this issue and adds a prompt which will be played after a timeout informing the user that they needed to press a button. Currently, the sound files that we have are somewhat inadquate for this, so after the call we simply have Allison say "Please try again. Then press pound." which actually relies on two separate sound files. Just one would be more appropriate. reporter: Vlad Povorozniuc Review: https://reviewboard.asterisk.org/r/1327/ ........ ................ 2011-07-25 19:57 +0000 [r329473] Paul Belanger * /, main/enum.c: Merged revisions 329472 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r329472 | pabelanger | 2011-07-25 15:55:33 -0400 (Mon, 25 Jul 2011) | 9 lines Merged revisions 329471 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329471 | pabelanger | 2011-07-25 15:49:40 -0400 (Mon, 25 Jul 2011) | 2 lines Decrease verbose messages to debug, to help clean up CLI. ........ ................ 2011-07-25 14:07 +0000 [r329391-329432] Gregory Nietsky * include/asterisk/dsp.h, main/dsp.c: dsp_process was enhanced to work with alaw and ulaw in addition to slin. noticed that some functions could be refactored here it is. Reported by: irroot Tested by: irroot, mnicholson Review: https://reviewboard.asterisk.org/r/1304/ * channels/chan_sip.c, channels/sip/include/sip.h: Remove lastmsgssent from sip it has not been working since 1.6 Clean up the return values to be consistant not currently used Add doxygen returns MWI Event is sent on Register (closes issue ASTERISK-17866) Reported by: one47 Tested by: irroot, mvanbaak Review: https://reviewboard.asterisk.org/r/1172/ 2011-07-22 21:15 +0000 [r329332-329335] Richard Mudgett * main/pbx.c, /: Merged revisions 329334 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r329334 | rmudgett | 2011-07-22 16:14:22 -0500 (Fri, 22 Jul 2011) | 1 line Make use less redundant loop construct for iterating over hints. ........ * main/pbx.c, /: Merged revisions 329331 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r329331 | rmudgett | 2011-07-22 15:43:07 -0500 (Fri, 22 Jul 2011) | 55 lines Merged revisions 329299 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329299 | rmudgett | 2011-07-22 10:44:58 -0500 (Fri, 22 Jul 2011) | 48 lines Deadlocks dealing with dialplan hints during reload. There are two remaining different deadlocks reported dealing with dialplan hints. The deadlock in ASTERISK-17666 is caused by invalid locking order in ast_remove_hint(). The hints container must be locked before the hint object. The deadlock in ASTERISK-17760 is caused by a catch-22 situation in handle_statechange(). The deadlock is caused by not having the conlock before calling the watcher callbacks. Unfortunately, having that lock causes a different deadlock as reported in ASTERISK-16961. * Fixed ast_remove_hint() locking order. * Made handle_statechange() no longer call the watcher callbacks holding any locks that matter. * Made hint ao2 destructor do the watcher callbacks for extension deactivation to guarantee that they get called. * Fixed hint reference leak in ast_add_hint() if the callback container constructor failed. * Fixed hint reference leak in complete_core_show_hint() for every hint it found for CLI tab completion. * Adjusted locking in ast_merge_contexts_and_delete() for safety. * Added context_merge_lock to prevent ast_merge_contexts_and_delete() and handle_statechange() from interfering with each other. * Fixed ast_change_hint() not taking into account that the extension is used for the hash key. (closes issue ASTERISK-17666) Reported by: irroot Tested by: irroot JIRA SWP-3318 (closes issue ASTERISK-17760) Reported by: Byron Clark Tested by: irroot JIRA SWP-3393 Review: https://reviewboard.asterisk.org/r/1313/ ........ ................ 2011-07-21 20:26 +0000 [r329258] Russell Bryant * channels/chan_dahdi.c, /, main/features.c, include/asterisk/netsock2.h, CHANGES, channels/sig_pri.c, include/asterisk/rtp_engine.h: Merged revisions 329257 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r329257 | russell | 2011-07-21 15:22:36 -0500 (Thu, 21 Jul 2011) | 2 lines s/1.10/10.0/ ........ 2011-07-21 18:06 +0000 [r329146-329205] Richard Mudgett * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged revisions 329204 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r329204 | rmudgett | 2011-07-21 13:05:18 -0500 (Thu, 21 Jul 2011) | 13 lines Merged revisions 329203 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329203 | rmudgett | 2011-07-21 13:04:09 -0500 (Thu, 21 Jul 2011) | 6 lines Document parkinglot in chan_dahdi.conf.sample. * Document existing feature in chan_dahdi.conf.sample. * Remove some dead code related to the parkinglot option. ........ ................ * /, apps/app_directed_pickup.c: Merged revisions 329200 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r329200 | rmudgett | 2011-07-21 12:32:02 -0500 (Thu, 21 Jul 2011) | 24 lines Merged revisions 329199 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329199 | rmudgett | 2011-07-21 12:30:57 -0500 (Thu, 21 Jul 2011) | 17 lines Update PickupChan documentation. The PickupChan uses the ampersand as the argument separator. Was documented as: PickupChan(channel[,channel2[,...][,options]]) Fixed documentation to: PickupChan(Technology/Resource[&Technology2/Resource2[&...]][,options]) This is a continuation of ASTERISK-17494 for v1.8 and later. (closes issue ASTERISK-18144) Reported by: Erik Smith Patches: pickupchan_ducumentation-v2.patch (License #6263) patch uploaded by Erik Smith Tested by: Erik Smith ........ ................ * /, main/features.c: Merged revisions 329145 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r329145 | rmudgett | 2011-07-21 11:52:17 -0500 (Thu, 21 Jul 2011) | 16 lines Merged revisions 329144 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329144 | rmudgett | 2011-07-21 11:46:21 -0500 (Thu, 21 Jul 2011) | 9 lines Dialplan bridge() app mutex 'current_dest_chan' freed more times than we've locked! This appears to be a leftover from when ast_channel was converted to ao2 objects. Simply removed the extraneous unlock. (closes issue ASTERISK-17772) ........ ................ 2011-07-21 16:22 +0000 [r329106-329130] Jason Parker * UPGRADE-1.10.txt (removed), UPGRADE-10.txt (added), UPGRADE.txt: Fix UPGRADE.txt files for Asterisk 10. * /: Remove another 2.0 property. 2011-07-21 16:05 +0000 [r329105] Russell Bryant * /: Fix merge properties to reflect Asterisk 10 branch