1749 Commits (75c2c859620edc9bfc03fada26bb83e90013b33e)

Author	SHA1	Message	Date
Joshua Colp	6e60f5d317	Recorded merge of revisions 417677 from http://svn.asterisk.org/svn/asterisk/branches/11 ... ........ res_rtp_asterisk: Add SHA-256 support for DTLS and perform DTLS negotiation on RTCP. This change fixes up DTLS support in res_rtp_asterisk so it can accept and provide a SHA-256 fingerprint, so it occurs on RTCP, and so it occurs after ICE negotiation completes. Configuration options to chan_sip and chan_pjsip have also been added to allow behavior to be tweaked (such as forcing the AVP type media transports in SDP). ASTERISK-22961 #close Reported by: Jay Jideliov Review: https://reviewboard.asterisk.org/r/3679/ Review: https://reviewboard.asterisk.org/r/3686/ ........ Merged revisions 417678 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417679 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	365ae7523b	res_http_websocket: Close websocket correctly and use careful fwrite ... When a client takes a long time to process information received from Asterisk, a write operation using fwrite may fail to write all information. This causes the underlying file stream to be in an unknown state, such that the socket must be disconnected. Unfortunately, there are two problems with this in Asterisk's existing websocket code: 1. Periodically, during the read loop, Asterisk must write to the connected websocket to respond to pings. As such, Asterisk maintains a reference to the session during the loop. When ast_http_websocket_write fails, it may cause the session to decrement its ref count, but this in and of itself does not break the read loop. The read loop's write, on the other hand, does not break the loop if it fails. This causes the socket to get in a 'stuck' state, preventing the client from reconnecting to the server. 2. More importantly, however, is that the fwrite in ast_http_websocket_write fails with a large volume of data when the client takes awhile to process the information. When it does fail, it fails writing only a portion of the bytes. With some debugging, it was shown that this was failing in a similar fashion to ASTERISK-12767. Switching this over to ast_careful_fwrite with a long enough timeout solved the problem. Note that this version of the patch, unlike r417310 in Asterisk 11, exposes configuration options beyond just chan_sip's sip.conf. Configuration options to configure the write timeout have also been added to pjsip.conf and ari.conf. #ASTERISK-23917 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3624/ ........ Merged revisions 417310 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 417311 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417317 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
George Joseph	cc4178bf25	Update extensions.lua.sample with naming conflict guidance. ... The sample extensions.lua was causing pbx_lua to fail to load when parsing 'app.goto("default", "s", 1)' because in Lua 5.2, 'goto' is now a reserved word. This patch adds guidance to extensions.lua.sample and changed 'app.goto("default", "s", 1)' to 'app.['goto']("default", "s", 1)'. ASTERISK-23844 #close Reported by: rnewton Tested by: gtjoseph Review: https://reviewboard.asterisk.org/r/3627/ ........ Merged revisions 416581 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 416582 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416589 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	0c896d8b9b	chan_dahdi: Adds support for major update to libss7. ... * SS7 support now requires libss7 v2.0 or later. The new libss7 is not backwards compatible. * Added SS7 support for connected line and redirecting. * Most SS7 CLI commands are reworked as well as new SS7 commands added. See online CLI help. * Added several SS7 config option parameters described in chan_dahdi.conf.sample. * ISUP timer support reworked and now requires explicit configuration. See ss7.timers.sample. Special thanks to Kaloyan Kovachev for his support and persistence in getting the original patch by adomjan updated and ready for release. SS7-27 #close Reported by: adomjan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416416 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	4ca5745dbe	AST-2014-007: Fix DOS by consuming the number of allowed HTTP connections. ... Simply establishing a TCP connection and never sending anything to the configured HTTP port in http.conf will tie up a HTTP connection. Since there is a maximum number of open HTTP sessions allowed at a time you can block legitimate connections. A similar problem exists if a HTTP request is started but never finished. * Added http.conf session_inactivity timer option to close HTTP connections that aren't doing anything. Defaults to 30000 ms. * Removed the undocumented manager.conf block-sockets option. It interferes with TCP/TLS inactivity timeouts. * AMI and SIP TLS connections now have better authentication timeout protection. Though I didn't remove the bizzare TLS timeout polling code from chan_sip. * chan_sip can now handle SSL certificate renegotiations in the middle of a session. It couldn't do that before because the socket was non-blocking and the SSL calls were not restarted as documented by the OpenSSL documentation. * Fixed an off nominal leak of the ssl struct in handle_tcptls_connection() if the FILE stream failed to open and the SSL certificate negotiations failed. The patch creates a custom FILE stream handler to give the created FILE streams inactivity timeout and timeout after a specific moment in time capability. This approach eliminates the need for code using the FILE stream to be redesigned to deal with the timeouts. This patch indirectly fixes most of ASTERISK-18345 by fixing the usage of the SSL_read/SSL_write operations. ASTERISK-23673 #close Reported by: Richard Mudgett ........ Merged revisions 415841 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 415854 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 415896 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415907 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Rusty Newton	4e292ea3af	configs/cli_aliases.conf: Two new aliases, plus enhancements for context names. ... Changed naming of included alias templates to avoid confusion between version names. For example, asterisk12 was for asterisk 1.2, so I changed it to asterisk_1dot2, so that later we can use asterisk_12 for Asterisk 12. Added alias for "features reload" to the template for Asterisk 11 style syntax template, as features reload was removed in 12, but you can still do "module reload features" Added alias for "pjsip reload" to the friendly template. It is shorter than "module reload res_pjsip.so" and if some are like me; I constantly forget that reloading chan_pjsip doesn't parse config. Remembering "pjsip reload" is just easier. ASTERISK-23654 #close ASTERISK-23654 #comment Fixed by adding two new aliases and enhancements for context names. Review: https://reviewboard.asterisk.org/r/3572/ ........ Merged revisions 415301 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415302 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Rusty Newton	812f33d222	pjsip.conf: privkey_file should be priv_key_file, mediaencryption=yes should be mediaencryption=sdes ... privkey_file was missed in the snake case update. An example included an invalid value for the mediaencryption option. ........ Merged revisions 414780 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414781 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Igor Goncharovskiy	d3433771c9	Introducing changes proposed to chan_unistim driver: ...	12 years ago
Jonathan Rose	ae21162a69	chan_sip: Add sendrpid trust options ... In r411189, some behavior was changed which made sendrpid behavior act in a more trusting manner by sending full user data for peers set with private caller presence in P-Asserted-Identity headers. Since this changed long time expected behaviors, we decided to pull that patch when that was pointed out by the community. Instead, this patch provides a trust_id_outbound setting which will expose the data per RFC-3325 if set to 'yes' and simply not send the PAI/RPID headers at all if set to 'no'. By default trust_id_outbound will be set to 'legacy' which will preserve the behavior prior to these patches. Extra special thanks to Walter Doekes for providing advice and feedback. (closes issue AST-1301) (closes issue ASTERISK-19465) Reported by: Krzysztof Chmielewski Review: https://reviewboard.asterisk.org/r/3447/ ........ Merged revisions 412744 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 412746 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 412747 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412759 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	cc4a0a7fc9	Reverting r411189 so that it can be put up for public review ... --- r411189 | jrose | 2014-03-26 10:50:48 -0500 (Wed, 26 Mar 2014) | 12 lines chan_sip: Send real CallerID information with P-Assserted-Identity (RFC-3325) Prior to this patch, the P-Asserted-Identity header would include anonymous caller id information which seems to go against the point of the P-Asserted-Identity header. Now the real caller ID information will be included in this header. Also, no privacy header would be included. This patch adds 'Privacy: id' to outgoing SIP messages that include the P-Asserted-Identity header. (closes issue AST-1301) --- ........ Merged revisions 412328 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 412329 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 412330 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412331 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	03beadb6e9	internal_timing: Remove the option and always make it enabled if a timing module is loaded. ... The masquerade supertest frequently fails because either the local channel chain doesn't completely optimize out or the DTMF handshake doesn't completely get accross. Local channel optimization requires frames flowing to trigger when optimization can happen. When optimization happens the media frame that triggered the optimization is dropped. Sending DTMF requires frames to flow in the other direction for timing purposes while sending nothing. If internal timing is not enabled when MOH is playing, Asterisk switches to received timing when an audio frame is received. With optimization dropping media frames and MOH not sending frames unless it receives frames, occasionaly there are no more frames being passed and the test fails. * The asterisk command line -I option and the asterisk.conf internal_timing option are removed. Asterisk now always uses internal timing when needed if any timing module is loaded. The issue ASTERISK-14861 did this quite awhile ago in v1.4 but effectively is broken if other internal timing modules besides DAHDI are used. The ast_read_generator_actions() now only does received timing if it has no choice for frame generators like MOH, silence, and playback streaming. * Cleaned up some code dealing with frame generators in ast_deactivate_generator(), generator_write_format_change(), ast_activate_generator(), and ast_channel_stop_silence_generator(). * Removed ast_internal_timing_enabled(), AST_OPT_FLAG_INTERNAL_TIMING, and ast_opt_internal_timing. ASTERISK-22846 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3414/ ........ Merged revisions 411715 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411716 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411717 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411724 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	eefcb79bfb	Prevent duplicate sorcery wizards from being applied to sorcery object types. ... This commit contains several changes to sorcery: 1) Application of sorcery configuration based on module name is automatically performed when sorcery is opened for a module. 2) Sorcery will not attempt to apply the same wizard to an object type more than once. 3) Sorcery gives more exact results when attempting to apply a wizard, whether as the default or based on configuration. Sorcery unit tests still pass for me after making these changes. Review: https://reviewboard.asterisk.org/r/3326 ........ Merged revisions 411159 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411656 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	ef0c9fe4d8	res_hep/res_hep_pjsip: Add a HEPv3 capture agent module and a logger for PJSIP ... This patch adds the following: (1) A new module, res_hep, which implements a generic packet capture agent for the Homer Encapsulation Protocol (HEP) version 3. Note that this code is based on a patch provided by Alexandr Dubovikov; I basically just wrapped it up, added configuration via the configuration framework, and threw in a taskprocessor. (2) A new module, res_hep_pjsip, which forwards all SIP message traffic that passes through the res_pjsip stack over to res_hep for encapsulation and transmission to a HEPv3 capture server. Much thanks to Alexandr for his Asterisk patch for this code and for a *lot* of patience waiting for me to port it to 12/trunk. Due to some dithering on my part, this has taken the better part of a year to port forward (I still blame CDRs for the delay). ASTERISK-23557 #close Review: https://reviewboard.asterisk.org/r/3207/ ........ Merged revisions 411534 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411556 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	fa3a2f8eca	chan_sip: Send real CallerID information with P-Assserted-Identity (RFC-3325) ... Prior too this patch, the P-Asserted-Identity header would include anonymous caller id information which seems to go against the point of the P-Asserted-Identity header. Now the real caller ID information will be included in this header. Also, no privacy header would be included. This patch adds 'Privacy: id' to outgoing SIP messages that include the P-Asserted-Identity header. (closes issue AST-1301) ........ Merged revisions 411189 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411190 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411193 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411194 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	eba91d2a98	Revert changes to sorcery that accidentally got committed. ... These changes were still up for review and have not been approved yet. I must have had the changes in my working copy when making a different change. ........ Merged revisions 410696 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410699 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	d44aefeef4	Fix stuck channel in ARI through the introduction of synchronous bridge actions. ... Playing back a file to a channel in an ARI bridge would attempt to wait until the playback concluded before returning. The method used involved signaling the waiting thread in the ARI custom playback function. The problem with this is that there were some corner cases that were not accounted for: * If a bridge channel could not be found, then we never would attempt the playback but would still attempt to wait for the playback to complete. * If the bridge playfile action failed to queue, we would still attempt to wait for the playback to complete. * If the bridge playfile action were queued but some circumstance caused the playback not to occur (the bridge dies, the channel is removed from the bridge), then we would never be notified. The solution to this is to move the waiting logic into the bridge code. A new bridge API function is added to queue a synchronous action on a bridge. The waiting thread is notified when the queued frame has been freed, either due to an error occurring or due to successful playback. As a failsafe, the waiting thread has a 10 minute timeout just in case there is a frame leak somewhere. Review: https://reviewboard.asterisk.org/r/3338 ........ Merged revisions 410673 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410684 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Corey Farrell	3ee8cf6efb	res_fax: Comment out default settings from res_fax.conf. ... Comment out many settings in res_fax.conf.sample. The defaults are set in res_fax.c, so setting the same value in sample config does nothing but make the sample config more fragile. (closes issue ASTERISK-23231) Reported by: David Brillert Review: https://reviewboard.asterisk.org/r/3261/ ........ Merged revisions 409052 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 409053 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 409054 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409055 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Rusty Newton	23b142d5c8	configs/voicemail.conf.sample - Make mailcmd sample text more explicit ... Made the wording a bit more explicit. Didn't really change the meaning. ........ Merged revisions 408876 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 408877 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 408878 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408879 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	75edef52e0	ConfBridge: Correct prompt playback target ... Currently, when the first marked user enters the conference that contains waitmarked users, a prompt is played indicating that the user is being placed into the conference. Unfortunately, this prompt is played to the marked user and not the waitmarked users which is not very helpful. This patch changes that behavior to play a prompt stating "The conference will now begin" to the entire conference after adding and unmuting the waitmarked users since the design of confbridge is not conducive to playing a prompt to a subset of users in a conference in an asynchronous manner. (closes issue PQ-1396) Review: https://reviewboard.asterisk.org/r/3155/ Reported by: Steve Pitts ........ Merged revisions 407857 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 407858 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407859 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	6f38887cb7	chan_iax2: Block unnecessary control frames to/from the wire. ... Establishing an IAX2 call between Asterisk v1.4 and v1.8 (or later) results in an unexpected call disconnect. The problem happens because newer values in the enum ast_control_frame_type are not consistent between the branch versions of Asterisk. For example: 1) v1.4 calls v1.8 (or later) using IAX2 2) v1.8 answers and sends a connected line update control frame. (on v1.8 AST_CONTROL_CONNECTED_LINE = 22) 3) v1.4 receives the control frame as an end-of-q (on v1.4 AST_CONTROL_END_OF_Q = 22) 4) v1.4 disconnects the call once the receive queue becomes empty. Several things are done by this patch to fix the problem and attempt to prevent it from happening again in the future: * Added a warning at the definition of enum ast_control_frame_type about how to add new control frame values. * Made block sending and receiving control frames that have no reason to go over the wire. * Extended the connectedline iax.conf parameter to also include the redirecting information updates. * Updated the connectedline iax.conf parameter documentation to include a notice that the parameter must be "no" when the peer is an Asterisk v1.4 instance. (closes issue AST-1302) Review: https://reviewboard.asterisk.org/r/3174/ ........ Merged revisions 407678 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 407727 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 407729 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407731 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Tzafrir Cohen	1d2d01e08d	indications.conf: add stutter tone; end properly ... * If the "stutter" (voicemail indication) tone is indeed a stutter tone, and it ends with a constant tone, make sure that it is the dial tone. This was done for India (in), Mexico (mx) and the Philippines (ph). * If no "stutter" tone exists for a country, provide one. This was done for Spain (es), Malaysia (my) and Venezuela (ve). Review: https://reviewboard.asterisk.org/r/3158/ ........ Merged revisions 407622 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 407623 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 407624 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407625 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Rusty Newton	b196e9c117	configs/pjsip.conf.sample: Configuration section naming in pjsip.conf.sample needs a little clarification ... There is a bit of nuance to how you name things in pjsip.conf. This is a documentation patch to at least clear it up a little for users. Review: https://reviewboard.asterisk.org/r/3180/ ........ Merged revisions 407587 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407588 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	12668b6659	tcptls.c: Made TLS handle a certificate chain file. ... Thanks to Guillaume Martres for doing the necessary research to validate the change. (closes issue ASTERISK-17727) Reported by: LN Patches: use_certificate_chain.patch (license #5864) patch uploaded by st documente_certificate_chain.patch (license #6576) patch uploaded by Guillaume Martres ........ Merged revisions 407272 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 407273 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 407274 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407275 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	10e38fb10c	res_pjsip: Config option to enable PJSIP logger at load time. ... Added a "debug" configuration option for res_pjsip that when set to "yes" enables SIP messages to be logged. It is specified under the "system" type. Also added an alembic script to add the option to realtime. (closes issue ASTERISK-23038) Reported by: Rusty Newton Review: https://reviewboard.asterisk.org/r/3148/ ........ Merged revisions 407036 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407037 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Russell Bryant	e7b20b1c91	queues.conf.sample Fix documented default for persistentmembers ... Closes issue ASTERISK-22662 ........ Merged revisions 406860 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 406861 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406862 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406863 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Walter Doekes	cc42229f26	manager: The eventfilter= option now takes an extended regex. ... In pre-trunk versions (...12) it accepts a basic regex, which is confusing because all other regexes in asterisk are of the extended kind. Review: https://reviewboard.asterisk.org/r/3147/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406618 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Walter Doekes	9a88cc33f8	manager: Clarify eventfilter documentation. Textual changes only. ... Review: https://reviewboard.asterisk.org/r/3133/ ........ Merged revisions 406079 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 406080 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406081 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406082 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Rusty Newton	f6647d2362	Documentation: doc fixes across various parts of the code for ASTERISK issues 23061,23028,23046,23027 ... Fixes typos of "transfered" instead of "transferred" in various code. Fixes incorrect gosub param help text for app_queue. Fixes Asterisk man pages containing unquoted minus signs. Adds note about the "textsupport" option in sip.conf.sample. (issue ASTERISK-23061) (issue ASTERISK-23028) (issue ASTERISK-23046) (issue ASTERISK-23027) (closes issue ASTERISK-23061) (closes issue ASTERISK-23028) (closes issue ASTERISK-23046) (closes issue ASTERISK-23027) Reported by: Eugene, Jeremy Laine, Denis Pantsyrev Patches: transferred.patch uploaded by Jeremy Laine (license 6561) hyphen.patch uploaded by Jeremy Laine (license 6561) sip.conf.sample.patch uploaded by Eugene (license 6360) ........ Merged revisions 405791 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 405792 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405829 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405830 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	a48798ce95	res_fax: check_modem_rate() returned incorrect rate for V.27 ... According to the new standard for V.27 and V.32 they are able to transmit at a bit rate of 4,800 or 9,600. The check_mode_rate function needed to be updated to reflect this. Also, because of this change the default 'minrate' value was updated to be 4800. (closes issue ASTERISK-22790) Reported by: Paolo Compagnini Patches: res_fax.txt uploaded by looserouting (license 6548) ........ Merged revisions 405656 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 405693 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405694 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405695 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	828f339a9c	verbosity: Fix performance of console verbose messages. ... The per console verbose level feature as previously implemented caused a large performance penalty. The fix required some minor incompatibilities if the new rasterisk is used to connect to an earlier version. If the new rasterisk connects to an older Asterisk version then the root console verbose level is always affected by the "core set verbose" command of the remote console even though it may appear to only affect the current console. If an older version of rasterisk connects to the new version then the "core set verbose" command will have no effect. * Fixed the verbose performance by not generating a verbose message if nothing is going to use it and then filtered any generated verbose messages before actually sending them to the remote consoles. * Split the "core set debug" and "core set verbose" CLI commands to remove the per module verbose support that cannot work with the per console verbose level. * Added a silent option to the "core set verbose" command. * Fixed "core set debug off" tab completion. * Made "core show settings" list the current console verbosity in addition to the root console verbosity. * Changed the default verbose level of the 'verbose' setting in the logger.conf [logfiles] section. The default is now to once again follow the current root console level. As a result, using the AMI Command action with "core set verbose" could again set the root console verbose level and affect the verbose level logged. (closes issue AST-1252) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/3114/ ........ Merged revisions 405431 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405432 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405436 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	9fa171e547	External MWI core support. ... * The core external MWI resource provides for MWI message counts persistence using sorcery. With sorcery, the user is able to configure which sorcery wizzard backend to use if the default astdb is not desired. * The core external MWI resoruce provides some debugging CLI commands enabled by defining MWI_DEBUG_CLI. The debugging CLI commands are: "mwi delete all", "mwi delete like <regex>", "mwi delete mailbox <mailbox>", "mwi list all", "mwi list like <regex>", "mwi show mailbox <mailbox>", and "mwi update mailbox <mailbox> [<new> [<old>]]". (closes issue AFS-43) Review: https://reviewboard.asterisk.org/r/3061/ ........ Merged revisions 404952 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404953 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	821ab51381	res_pjsip: add 'set_var' support on endpoints ... Added a new 'set_var' option for ast_sip_endpoint(s). For each variable specified that variable gets set upon creation of a pjsip channel involving the endpoint. (closes issue ASTERISK-22868) Reported by: Joshua Colp Review: https://reviewboard.asterisk.org/r/3095/ ........ Merged revisions 404663 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404664 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Tzafrir Cohen	82eb03b915	chan_dahdi: enable ignore_failed_channels by default ... If ignore_failed_channels is set to "true" for a channel, the channel will continue to be configured even if configuring it has failed. This allows Asterisk to start before all the DAHDI initialization is done and thus not force the starting order dahdi -> asterisk. Review: https://reviewboard.asterisk.org/r/3063/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404542 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Rusty Newton	06b577f7dc	Documentation: Updates for info about NAT-related settings and fixes for pjsip.conf.sample ... Added another NAT example to pjsip.conf.sample. We had a few mentions of NAT configuration throughout the sample, but I added another for a little bit more clarity. Additionally many pjsip options were affected by the change to snake case, so I fixed any instances of those options in pjsip.conf. I regenerated the config option list (at the bottom of the file) from a new xml config doc dump, so all the snake case changes should be reflected there, as well as any other changes to those options. (issue ASTERISK-23004) (closes issue ASTERISK-23004) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3086/ ........ Merged revisions 404405 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404406 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	e4803bbd9e	Voicemail: Remove mailbox identifier format (box@context) assumptions in the system. ... This change is in preparation for external MWI support. Removed code from the system for normal mailbox handling that appends @default to the mailbox identifier if it does not have a context. The only exception is the legacy hasvoicemail users.conf option. The legacy option will only work for app_voicemail mailboxes. The system cannot make any assumptions about the format of the mailbox identifer used by app_voicemail. chan_sip and chan_dahdi/sig_pri had the most changes because they both tried to interpret the mailbox identifier. chan_sip just stored and compared the two components. chan_dahdi actually used the box information. The ISDN MWI support configuration options had to be reworked because chan_dahdi was parsing the box@context format to get the box number. As a result the mwi_vm_boxes chan_dahdi.conf option was added and is documented in the chan_dahdi.conf.sample file. Review: https://reviewboard.asterisk.org/r/3072/ ........ Merged revisions 404348 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404350 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Alexandr Anikin	86b5e11607	Introduce new config option 'aniasdni'. If yes then asterisk set dialed number as own id back to the caller ... on incoming h.323 calls. Option can be set globally or per user section. (closes issue ASTERISK-22020) Reported by: Ross Beer git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404211 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	27f37f6e3d	Changed the default for live_dangerously to no ... ........ Merged revisions 404006 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404009 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	744556c01d	security: Inhibit execution of privilege escalating functions ... This patch allows individual dialplan functions to be marked as 'dangerous', to inhibit their execution from external sources. A 'dangerous' function is one which results in a privilege escalation. For example, if one were to read the channel variable SHELL(rm -rf /) Bad Things(TM) could happen; even if the external source has only read permissions. Execution from external sources may be enabled by setting 'live_dangerously' to 'yes' in the [options] section of asterisk.conf. Although doing so is not recommended. Also, the ABI was changed to something more reasonable, since Asterisk 12 does not yet have a public release. (closes issue ASTERISK-22905) Review: http://reviewboard.digium.internal/r/432/ ........ Merged revisions 403913 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 403917 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 403959 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403960 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	c602b086ed	res_pjsip_messaging: send message to a default outbound endpoint ... In some cases messages need to be sent to a direct URI (sip:<ip address>). This patch adds in that support by using a default outbound endpoint. When sending messages, if no endpoint can be found then the default one is used. To facilitate this a new default_outbound_endpoint option was added to the globals section for pjsip.conf. Review: https://reviewboard.asterisk.org/r/2944/ ........ Merged revisions 403680 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403687 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	1c45a32ee8	res_pjsip: convert configuration settings names to snake case ... Renamed, where appropriate, the configuration options for chan/res_pjsip to use snake case (compound words separated by an underscore). For example, faxdetect will become fax_detect, recordofffeature will become record_off_feature, etc... Review: https://reviewboard.asterisk.org/r/3002/ ........ Merged revisions 403022 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403051 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	7950118e18	Confbridge: Add option to review the recording similar to announce_join_leave ... Review: https://reviewboard.asterisk.org/r/3008/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402854 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	bf5492abd2	security_events: Push out security events over AMI events ... Security Events will now be written to any listener of the new 'security' class Review: https://reviewboard.asterisk.org/r/2998/ ........ Merged revisions 402584 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402585 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	0721b1de83	config: Allow ConfBridge DTMF menus to have '#' as the first digit. ... ConfBridge allows custom DTMF menus to be created in the confbridge.conf file by assigning a DTMF key sequence to a sequence of actions as follows: DTMF-sequence = action,action... Unfortunately, the normal config file processing code interprets an initial '#' character as starting a directive such as #include. * Add the ability to escape the first non-blank character in a config line so the '#' character can be used without triggering the directive processing code. (closes issue AFS-2) (closes issue ASTERISK-22478) Reported by: Nicolas Tanski Patches: jira_asterisk_22478_v11.patch (license #5621) patch uploaded by rmudgett (modified) Review: https://reviewboard.asterisk.org/r/2969/ ........ Merged revisions 402407 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 402416 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402417 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	4ca92e3b8a	chan_iax2: Fix Binding To Multiple Addresses Again ... When reworking chan_iax2 for IPv6, the ability to bind to multiple addresses was removed by mistake. This patch restores this functionality and adds notes about IPv6 addresses in the sample config. (closes issue ASTERISK-22741) Reported by: Joshua Colp Tested by: Michael L. Young Patches: asterisk-22741-fix-binding-multiple-addr.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2945/ ........ Merged revisions 401488 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401489 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	2127848d6c	chan_dahdi: Add config support for hwgain settings. ... * Add hwtxgain and hwrxgain config options to chan_dahdi.conf with documentation in chan_dahdi.conf.sample. (closes issue ASTERISK-22429) Reported by: Jaco Kroon Patches: jira_asterisk_22429_hwgain_trunk.patch (license #5621) patch uploaded by rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400950 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	f87086b374	app_confbridge: Can now set the language used for announcements to the conference. ... ConfBridge now has the ability to set the language of announcements to the conference. The language can be set on a bridge profile in confbridge.conf or by the dialplan function CONFBRIDGE(bridge,language)=en. (closes issue ASTERISK-19983) Reported by: Jonathan White Patches: M19983_rev2.diff (license #5138) patch uploaded by junky (modified) Tested by: rmudgett ........ Merged revisions 400741 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 400742 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400744 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	44bd543181	chan_pjsip: Add alembic scripts for generating db tables for PJSIP ... Also updates sample configurations for sorcery and extconfig to demonstrate how to use databases created by that alembic script. (closes issue ASTERISK-22133) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2892/ ........ Merged revisions 400532 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400533 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	8fbe62f5df	configuration samples: Pull all parking related stuff out of features.conf ... This patch also adds documentation for parking from features.conf to res_parking.conf ........ Merged revisions 400205 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400206 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	2de42c2a25	Multiple revisions 399887,400138,400178,400180-400181 ... ........ r399887 | dlee | 2013-09-26 10:41:47 -0500 (Thu, 26 Sep 2013) | 1 line Minor performance bump by not allocate manager variable struct if we don't need it ........ r400138 | dlee | 2013-09-30 10:24:00 -0500 (Mon, 30 Sep 2013) | 23 lines Stasis performance improvements This patch addresses several performance problems that were found in the initial performance testing of Asterisk 12. The Stasis dispatch object was allocated as an AO2 object, even though it has a very confined lifecycle. This was replaced with a straight ast_malloc(). The Stasis message router was spending an inordinate amount of time searching hash tables. In this case, most of our routers had 6 or fewer routes in them to begin with. This was replaced with an array that's searched linearly for the route. We more heavily rely on AO2 objects in Asterisk 12, and the memset() in ao2_ref() actually became noticeable on the profile. This was #ifdef'ed to only run when AO2_DEBUG was enabled. After being misled by an erroneous comment in taskprocessor.c during profiling, the wrong comment was removed. Review: https://reviewboard.asterisk.org/r/2873/ ........ r400178 | dlee | 2013-09-30 13:26:27 -0500 (Mon, 30 Sep 2013) | 24 lines Taskprocessor optimization; switch Stasis to use taskprocessors This patch optimizes taskprocessor to use a semaphore for signaling, which the OS can do a better job at managing contention and waiting that we can with a mutex and condition. The taskprocessor execution was also slightly optimized to reduce the number of locks taken. The only observable difference in the taskprocessor implementation is that when the final reference to the taskprocessor goes away, it will execute all tasks to completion instead of discarding the unexecuted tasks. For systems where unnamed semaphores are not supported, a really simple semaphore implementation is provided. (Which gives identical performance as the original taskprocessor implementation). The way we ended up implementing Stasis caused the threadpool to be a burden instead of a boost to performance. This was switched to just use taskprocessors directly for subscriptions. Review: https://reviewboard.asterisk.org/r/2881/ ........ r400180 | dlee | 2013-09-30 13:39:34 -0500 (Mon, 30 Sep 2013) | 28 lines Optimize how Stasis forwards are dispatched This patch optimizes how forwards are dispatched in Stasis. Originally, forwards were dispatched as subscriptions that are invoked on the publishing thread. This did not account for the vast number of forwards we would end up having in the system, and the amount of work it would take to walk though the forward subscriptions. This patch modifies Stasis so that rather than walking the tree of forwards on every dispatch, when forwards and subscriptions are changed, the subscriber list for every topic in the tree is changed. This has a couple of benefits. First, this reduces the workload of dispatching messages. It also reduces contention when dispatching to different topics that happen to forward to the same aggregation topic (as happens with all of the channel, bridge and endpoint topics). Since forwards are no longer subscriptions, the bulk of this patch is simply changing stasis_subscription objects to stasis_forward objects (which, admittedly, I should have done in the first place.) Since this required me to yet again put in a growing array, I finally abstracted that out into a set of ast_vector macros in asterisk/vector.h. Review: https://reviewboard.asterisk.org/r/2883/ ........ r400181 | dlee | 2013-09-30 13:48:57 -0500 (Mon, 30 Sep 2013) | 28 lines Remove dispatch object allocation from Stasis publishing While looking for areas for performance improvement, I realized that an unused feature in Stasis was negatively impacting performance. When a message is sent to a subscriber, a dispatch object is allocated for the dispatch, containing the topic the message was published to, the subscriber the message is being sent to, and the message itself. The topic is actually unused by any subscriber in Asterisk today. And the subscriber is associated with the taskprocessor the message is being dispatched to. First, this patch removes the unused topic parameter from Stasis subscription callbacks. Second, this patch introduces the concept of taskprocessor local data, data that may be set on a taskprocessor and provided along with the data pointer when a task is pushed using the ast_taskprocessor_push_local() call. This allows the task to have both data specific to that taskprocessor, in addition to data specific to that invocation. With those two changes, the dispatch object can be removed completely, and the message is simply refcounted and sent directly to the taskprocessor. Review: https://reviewboard.asterisk.org/r/2884/ ........ Merged revisions 399887,400138,400178,400180-400181 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400186 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	b44ce141e5	chan_sip: Allow Asterisk to retry after 403 on register ... This adds a global option in chan_sip to allow it to continue attempting registration if a 403 is received, clearing the cached nonce and treating it as a non-fatal response. Normally, this would cause registration attempts to that endpoint to stop. This also adds a similar per-outbound-registration option to chan_pjsip which allows the retry interval to be altered for 403 responses to REGISTER requests. (closes issue ASTERISK-17138) Review: https://reviewboard.asterisk.org/r/2874/ Reported by: Rudi ........ Merged revisions 400137 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 400140 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 400141 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400142 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	a1d56da32a	res_pjsip_notify: Add documentation ... We forgot to add documentation for res_pjsip_notify, which would prevent it from being loaded. Whoops. This patch also updates res_pjsip_notify to use pjsip_notify.conf, which now has its own sample file in the configs directory as well. Review: https://reviewboard.asterisk.org/r/2835/ ........ Merged revisions 400121 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400122 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Sean Bright	89b8ff5d78	Remove some trailing whitespace and steal revision 400000. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400000 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Rusty Newton	7c346a31ef	Documentation fix - waitfordialtone is not boolean, it's time in milliseconds ... Changing text in chan_dahdi.conf sample to be accurate. (issue ASTERISK-22308) (closes issue ASTERISK-22308) Reported By: Malcolm Davenport ........ Merged revisions 398880 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 398881 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 398882 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398883 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Russell Bryant	9b3e0b095e	Fix typo in confbridge.conf.sample ... The denoise filter requires func_speex, not codec_speex. Fix this in the description of the denoise=yes option in confbridge.conf. ........ Merged revisions 398820 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 398821 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398822 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Rusty Newton	be219c9ec9	New pjsip.conf.sample ... (issue ASTERISK-22145) (closes issue ASTERISK-22145) Reported By: Matt Jordan Review: https://reviewboard.asterisk.org/r/2811/ ........ Merged revisions 398147 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398148 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	449afdd9e8	Revert r394939 due to (numerous) objections ... The patch from ASTERISK-21965 was committed perhaps a bit too hastily. Walter and Tzafrir have pointed out numerous issues with the approach and have propsed an alternative in r/2757. Since it's not a time critical issue and is not worth holding up the release of 12 for it, I've gone ahead and reverted r394939 from 12/trunk and re-opened ASTERISK-21965. ........ Merged revisions 397938 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397939 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	d12c79f78f	Update CEL sample config ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397514 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Walter Doekes	33ec719645	Add "autoframing" option to sip.conf.sample and h323.conf.sample. ... The autoframing option was added to chan_sip.c in r43243 (mogorman, 2006-09-19 01:32:57), but never made its way into the sample configs. Review: https://reviewboard.asterisk.org/r/2768/ ........ Merged revisions 396994 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 396995 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396996 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	0c44ee3be3	Update features.conf.sample atxferdropcall option. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396793 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	f6c7e6355e	Fix remnants of the pjsip renaming ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395851 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	735b30ad71	The large GULP->PJSIP renaming effort. ... The general gist is to have a clear boundary between old SIP stuff and new SIP stuff by having the word "SIP" for old stuff and "PJSIP" for new stuff. Here's a brief rundown of the changes: * The word "Gulp" in dialstrings, functions, and CLI commands is now "PJSIP" * chan_gulp.c is now chan_pjsip.c * Function names in chan_gulp.c that were "gulp_*" are now "chan_pjsip_*" * All files that were "res_sip*" are now "res_pjsip*" * The "res_sip" directory is now "res_pjsip" * Files in the "res_pjsip" directory that began with "sip_*" are now "pjsip_*" * The configuration file is now "pjsip.conf" instead of "res_sip.conf" * The module info for all PJSIP-related files now uses "PJSIP" instead of "SIP" * CLI and AMI commands created by Asterisk's PJSIP modules now have "pjsip" as the starting word instead of "sip" git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395764 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	d8956f690e	Rename everything Stasis-HTTP to ARI ... This renames all files and API calls from several variants of Stasis-HTTP to ARI including: * Stasis-HTTP -> ARI * STASIS_HTTP -> ARI * stasis_http -> ari (ast_ari for global symbols, file names as well) * stasis http -> ARI Review: https://reviewboard.asterisk.org/r/2706/ (closes issue ASTERISK-22136) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395603 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	bb955e37fb	Provide proper ring tone in indications.conf for Malaysia ... The ring tone provided in the sample indications.conf was incorrect. This patch modifies the sample ring tone to be what it should: ring = 425/400,0/200,425/400,0/2000 This brings it in line with the tone definition in DAHDI 2.7.0. (zonedata.c) (closes issue ASTERISK-21997) Reported by: Filip Jenicek patches: malaysia_ring.patch uploaded by phill (License 6277) ........ Merged revisions 394940 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 394941 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394942 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	54803338b4	Always install safe_asterisk; add configuration file support ... This patch modifies the behavior of safe_asterisk in two ways: (1) It modifies the Asterisk Makefile such that safe_asterisk is always installed on a 'make install'. This was done as bugfixes in the safe_asterisk script were not applied in previous version of Asterisk without first removing the old version of the script. (2) In order to keep a newly installed version of safe_asterisk from impacting local modifications, a new config file - safe_asterisk.conf.sample - has been provided. Settings that were previously modified in safe_asterisk can be set there instead. (closes issue ASTERISK-21965) Reported by: Jeremy Kister patches: safe_asterisk.patch uploaded by jkister (License 6232) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394939 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	75e83bdbab	Document connectedline parameter for chan_iax2 ... The connectedline parameter for a chan_iax2 peer was undocumented. This patch documents the options in the sample configuration file. (closes issue ASTERISK-21953) Reported by: Birger "WIMPy" Harzenetter ........ Merged revisions 394886 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 394890 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394894 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	d43b17a872	Replace chan_agent with app_agent_pool. ... The ill conceived chan_agent is no more. It is now replaced by app_agent_pool. Agents login using the AgentLogin() application as before. The AgentLogin() application no longer does any authentication. Authentication is now the responsibility of the dialplan. (Besides, the authentication done by chan_agent did not match what the voice prompts asked for.) Sample extensions.conf [login] ; Sample agent 1001 login ; Set COLP for in between calls so the agent does not see the last caller COLP. exten => 1001,1,Set(CONNECTEDLINE(all)="Agent Waiting" <1001>) ; Give the agent DTMF transfer and disconnect features when connected to a caller. same => n,Set(CHANNEL(dtmf-features)=TX) same => n,AgentLogin(1001) same => n,NoOp(AGENT_STATUS is ${AGENT_STATUS}) same => n,Hangup() [caller] ; Sample caller direct connect to agent 1001 exten => 800,1,AgentRequest(1001) same => n,NoOp(AGENT_STATUS is ${AGENT_STATUS}) same => n,Hangup() ; Sample caller going through a Queue to agent 1001 exten => 900,1,Queue(agent_q) same => n,Hangup() Sample queues.conf [agent_q] member => Local/800@caller,,SuperAgent,Agent:1001 Under the hood operation overview: 1) Logged in agents wait for callers in an agents holding bridge. 2) Caller requests an agent using AgentRequest() 3) A basic bridge is created, the agent is notified, and caller joins the basic bridge to wait for the agent. 4) The agent is either automatically connected to the caller or must ack the call to connect. 5) The agent is moved from the agents holding bridge to the basic bridge. 6) The agent and caller talk. 7) The connection is ended by either party. 8) The agent goes back to the agents holding bridge. To avoid some locking issues with the agent holding bridge, I needed to make some changes to the after bridge callback support. The after bridge callback is now a list of requested callbacks with the last to be added the only active callback. The after bridge callback for failed callbacks will always happen in the channel thread when the channel leaves the bridging system or is destroyed. (closes issue ASTERISK-21554) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2657/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394417 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	684481b74c	Change ARI user config to use a type field ... When I initially wrote the configuration support for ARI users, I determined the section type by a category prefix (i.e., [user-admin]). This is neither idiomatic Asterisk configuration, nor is it really that user friendly. This patch replaces the category prefix with a type field in the section, which is much cleaner. Review: https://reviewboard.asterisk.org/r/2664/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394076 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Russell Bryant	0bfe2d4cc4	astobj2-ify the SLA code ... The SLA code within app_meetme was written before asotbj2 had been merged into Asterisk. Worse, support for reloads did not exist at first and was added later as a bolt-on feature. I knew at the time that reloading was not safe at all while SLA was in use, so the reload would be queued up to execute when the system was idle. Unfortunately, this approach was still prone to errors beyond the fact that this was the only place in Asterisk where configuration was not reloaded instantly when requested. This patch converts various SLA objects to be reference counted objects using astobj2. This allows reloads to be processed while the system is in use. The code ensures that the objects will not disappear while one of the other threads is using them. However, they will be immediately removed from the global trunk and station containers so no new calls will use them if removed from configuration. Review: https://reviewboard.asterisk.org/r/2581/ ........ Merged revisions 393928 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 393929 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393930 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	02f55a36a0	Revert accidental overcommit. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393632 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	b4e9a3fc2f	Add BUGBUG note for ASTERISK-22009 ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393631 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	c4adaf9106	Configuration for Stasis threadpool ... The appropriate settings for the Stasis threadpool is very system specific, depending upon both workload and system configuration. This patch adds a stasis.conf file which can be used to configure the key attributes of the threadpool for the Stasis message bus. (closes issue ASTERISK-21280) Review: https://reviewboard.asterisk.org/r/2651/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393542 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	9ba976b19c	ARI authentication. ... This patch adds authentication support to ARI. Two authentication methods are supported. The first is HTTP Basic authentication, as specified in RFC 2617[1]. The second is by simply passing the username and password as an ?api_key query parameter (which allows swagger-ui[2] to authenticate more easily). ARI usernames and passwords are configured in the ari.conf file (formerly known as stasis_http.conf). The user may be set to `read_only`, which will prohibit the user from issuing POST, DELETE, etc. Also, the user's password may be specified in either plaintext, or encrypted using the crypt() function. Several other notes about the patch. * A few command line commands for seeing ARI config and status were also added. * The configuration parsing grew big enough that I extracted it to its own file. [1]: http://www.ietf.org/rfc/rfc2617.txt [2]: https://github.com/wordnik/swagger-ui (closes issue ASTERISK-21277) Review: https://reviewboard.asterisk.org/r/2649/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393530 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	f306dbd841	bridge_features: Support One touch Monitor/MixMonitor ... In addition to porting those features, they now enjoy greater feature parity with one another. Specifically, AutoMixMon now has a start and stop message that can be specified with TOUCH_MIXMONITOR_MESSAGE_START and TOUCH_MIXMONITOR_MESSAGE_STOP. (closes issue ASTERISK-21553) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2620/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393309 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	909ee4bfb9	Refactor extraneous channel events ... This change removes JitterBufStats, ChannelReload, and ChannelUpdate and refactors the following events to travel over Stasis-Core: * LocalBridge * DAHDIChannel * AlarmClear * SpanAlarmClear * Alarm * SpanAlarm * DNDState * MCID * SIPQualifyPeerDone * SessionTimeout Review: https://reviewboard.asterisk.org/r/2627/ (closes issue ASTERISK-21476) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393284 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	77002bc377	Merge in current pimp_my_sip work, including: ... 1. Security events 2. Websocket support 3. Diversion header + redirecting support 4. An anonymous endpoint identifier 5. Inbound extension state subscription support 6. PIDF notify generation 7. One touch recording support (special thanks Sean Bright!) 8. Blind and attended transfer support 9. Automatic inbound registration expiration 10. SRTP support 11. Media offer control dialplan function 12. Connected line support 13. SendText() support 14. Qualify support 15. Inband DTMF detection 16. Call and pickup groups 17. Messaging support Thanks everyone! Side note: I'm reminded of the song "How Far We've Come" by Matchbox Twenty. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392565 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	c2e29abcbf	Add announce-to-first-user option for app_queue ... In r386792, the ability to play prompts to the first caller in a call queue was added. While this is arguably a bug fix for those who expect the first caller to continue receiving prompts while the agent is dialed, it has the side effect of preventing the first caller from hearing the agent immediately upon bridging. This may not be a problem for those who really want this option, but for those who didn't care whether or not the first caller in queue heard their position, it was an issue. This patch disables the ability for the first caller in the queue to hear prompts and adds a new option, announce-to-first-user, to queues.conf. Those who the behavior can enable it by setting this value to True. Note that if we ever implement the ability to have the prompts be stopped upon bridging, this option can be removed. (closes issue ASTERISK-21782) Reported by: Remi Quezada ........ Merged revisions 391215 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 391241 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391245 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jason Parker	a2d02edca5	Make app_queue AMI events more consistent. Give Join/Leave more useful names. ... This also removes the eventwhencalled and eventmemberstatus configuration options. These events can just be filtered via manager.conf blacklists. (closes issue ASTERISK-21469) Review: https://reviewboard.asterisk.org/r/2586/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390901 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	bad8caa8c6	Reimplement bridging and DTMF features related channel variables in the bridging core. ... * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is no longer channel driver specific. If the channel variable is set on the transferrer channel, the sound will be played to the target of an attended transfer. * The channel variable BRIDGEPEER becomes a comma separated list of peers in a multi-party bridge. The BRIDGEPEER value can have a maximum of 10 peers listed. Any more peers in the bridge will not be included in the list. BRIDGEPEER is not valid in holding bridges like parking since those channels do not talk to each other even though they are in a bridge. * The channel variable BRIDGEPVTCALLID is only valid for two party bridges and will contain a value if the BRIDGEPEER's channel driver supports it. * The channel variable DYNAMIC_PEERNAME is redundant with BRIDGEPEER and is removed. The more useful DYNAMIC_WHO_ACTIVATED gives the channel name that activated the dynamic feature. * The channel variables DYNAMIC_FEATURENAME and DYNAMIC_WHO_ACTIVATED are set only on the channel executing the dynamic feature. Executing a dynamic feature on the bridge peer in a multi-party bridge will execute it on all peers of the activating channel. (closes issue ASTERISK-21555) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2582/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390771 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	3d63833bd6	Merge in the bridge_construction branch to make the system use the Bridging API. ... Breaks many things until they can be reworked. A partial list: chan_agent chan_dahdi, chan_misdn, chan_iax2 native bridging app_queue COLP updates DTMF attended transfers Protocol attended transfers git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389378 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Damien Wedhorn	01d6e8dbc9	Add call forward no answer to skinny and cleanup general callfwd handling. ... CallforwardNoAnswer uses a sched to determine when to forward the call. Defaults to 20secs but configurable in skinny.conf. Adds dialType to each subchannel structure to be used to differentiate between normal dials that result in a call being placed (default) and other uses for the skinny_dialer (such as cfwd digit collection). Restructured all cfwd handling to use this new arrangement. (closes issue ASTERISK-21292) Reported by: wedhorn Tested by: myself Patches: skinny-callfwdnoans03.diff uploaded by wedhorn (license 5019) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389097 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	946eb5ede0	Example of how to use the Stasis message bus ... In order to get people familiar with the Stasis message bus, it would be useful to have something of a tutorial. Since I'm not clever enough to think of some cool integration we could do with Twitter, I settled for something that might actually be useful. This patch adds a res_statsd.so module, which implements a basic statsd[1] client. Statsd is a very simple statistics gathering server, which can publish its results to a backend graphing engine, like Graphite[2]. There are several different Statsd server implementations[3], so you can pick what works best for your environment. The actual example of how to use the Stasis message bus is in res_chan_stats.so. This module demonstrates how to use subscriptions and the message router by monitoring messages and posting channels stats to the statsd server. A wiki page walking through res_chan_stats.so is forthcoming. [1]: https://github.com/etsy/statsd/ [2]: http://graphite.readthedocs.org/en/latest/ [3]: http://joemiller.me/2011/09/21/list-of-statsd-server-implementations/ Review: https://reviewboard.asterisk.org/r/2460/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386624 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	8c1f423cf7	Don't bind to anything in the sample configuration so we don't clash with chan_sip on a "make samples" right now. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386577 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	74f2318051	Merge the pimp_my_sip branch into trunk. ... The pimp_my_sip branch is being merged at this point because it offers basic functionality, and from an API standpoint, things are complete. SIP work is *not* feature-complete; however, with the completion of the SUBSCRIBE/NOTIFY API, all APIs (except a PUBLISH API) have been created, and thus it is possible for developers to attempt to create new SIP work. API documentation can be found in the doxygen in the code, but usability documentation is still lacking. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386540 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	1c21b8575b	This patch adds a RESTful HTTP interface to Asterisk. ... The API itself is documented using Swagger, a lightweight mechanism for documenting RESTful API's using JSON. This allows us to use swagger-ui to provide executable documentation for the API, generate client bindings in different languages, and generate a lot of the boilerplate code for implementing the RESTful bindings. The API docs live in the rest-api/ directory. The RESTful bindings are generated from the Swagger API docs using a set of Mustache templates. The code generator is written in Python, and uses Pystache. Pystache has no dependencies, and be installed easily using pip. Code generation code lives in rest-api-templates/. The generated code reduces a lot of boilerplate when it comes to handling HTTP requests. It also helps us have greater consistency in the REST API. (closes issue ASTERISK-20891) Review: https://reviewboard.asterisk.org/r/2376/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386232 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	13e2aae2ef	Fix 'pri intense debug span' alias. ... ........ Merged revisions 385313 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385314 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Rusty Newton	98f2318559	Modified the list of keys for the driver backends for sake of sample clarity ... Added a line showing the mapping of "mysql" to res_config_mysql available in add-ons. We used "mysql" as an example driver key in the sample, but didn't show what module it mapped too. Also added a subtitle above the list of keys for driver backends. ........ Merged revisions 385047 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 385048 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385049 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	6a25d49296	chan_dahdi: Change inband_on_proceeding option default to no/disabled. ... (issue ASTERISK-21151) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384711 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	79818112fd	chan_dahdi: Add inband_on_proceeding compatibility option. ... The new inband_on_proceeding option causes Asterisk to assume inband audio may be present when a PROCEEDING message is received. Q.931 Section 5.1.2 says the network cannot assume that the CPE side has attached to the B channel at this time without explicitly sending the progress indicator ie informing the CPE side to attach to the B channel for audio. However, some non-compliant ISDN switches send a PROCEEDING without the progress indicator ie indicating inband audio is available and assume that the CPE device has connected the media path for listening to ringback and other messages. ASTERISK-17834 which causes this issue was dealing with a non-compliant network switch. (closes issue ASTERISK-21151) Reported by: Gianluca Merlo Tested by: rmudgett ........ Merged revisions 384685 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 384689 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384696 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	641fc7ea54	Sample config file for stasis-core. ... (issue ASTERISK-20887) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383225 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	8d5c36c9bb	Add RFC 3327 Path header support to chan_sip ... This patch adds support for RFC 3327 "Path" headers. This can be enabled in sip.conf using the 'supportpath' setting, either on a global basis or on a peer basis. This setting enables Asterisk to route outgoing out-of-dialog requests via a set of proxies by using a pre-loaded route-set defined by the Path headers in the REGISTER request. This patch also adds Realtime support for dynamically updating the Path information for a peer. A huge thank-you to Klaus Darillion and Olle E Johansson for their efforts in writing this patch. Review: https://reviewboard.asterisk.org/r/2235/ Review: https://reviewboard.asterisk.org/r/991/ (closes issue ASTERISK-16884) Reported by: klaus3000 Tested by: klaus3000, oej, mjordan patches: path-1.8.0-patch.txt uploaded by klaus3000 (License 5054) oolong-path-support-trunk in team branch by oej (License 5267) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382440 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	33e4c6115f	Ensure that the default bridge/user profiles are always available ... ConfBridge and Page require that there always be a default bridge and user profile available. While properties of the default profiles can be overriden in the configuration file, removing them can create situations where neither application can function properly. This patch ensures that if an administrator removes the profiles from the confbridge.conf configuration file, the profiles are added upon load. Documentation clarifying this has been added to the confbridge.conf.sample file. Review: https://reviewboard.asterisk.org/r/2356/ (closes issue AST-1115) Reported by: John Bigelow Tested by: John Bigelow ........ Merged revisions 382066 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382067 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Damien Wedhorn	0d553eece8	Add serviceURL stuff to skinny. ... Patch adds all the packet and structure stuff to skinny to enable setting service URLs in skinny, such as corporate directories. This stuff is only relevant during load/unload as when activated. Also some minor changes removing duplicated counting of addons and speedials in handle_skinny_show_devices. Review: https://reviewboard.asterisk.org/r/2321/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381718 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Walter Doekes	d4d1d10307	Remove "registertrying" and add "rtp_engine" from/to sip.conf.sample ... The "registertrying" option was removed in r343220. The "rtp_engine" option was added in r186078 but erroneously named "engine" in the sample. Note that there is no global sip setting for a different engine. ........ Merged revisions 381668 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 381669 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381670 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	d04ab3c645	Add CLI configuration documentation ... This patch allows a module to define its configuration in XML in source, such that it can be parsed by the XML documentation engine. Documentation is generated in a two-pass approach: 1. The documentation is first generated from the XML pulled from the source 2. The documentation is then enhanced by the registration of configuration options that use the configuration framework This patch include configuration documentation for the following modules: * chan_motif * res_xmpp * app_confbridge * app_skel * udptl Two new CLI commands have been added: * config show help - show configuration help by module, category, and item * xmldoc dump - dump the in-memory representation of the XML documentation to a new XML file. Review: https://reviewboard.asterisk.org/r/2278 Review: https://reviewboard.asterisk.org/r/2058 patches: on review 2058 uploaded by twilson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381527 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	1a70d513f1	Call Parking: Set PARKINGLOT and PARKINGSLOT variables on all parked calls ... These two variables were previously not being set when comebacktoorigin=yes and the example configs seemed to imply that they should be. Since there is no harm in this and since calls that are sent back to origin are capable of continuing in the dialplan, this seemed like a no-brainer. Also it supports some bridging tests I've been working on. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381068 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Damien Wedhorn	44872e797c	Reset skinny vmexten and immeddial char on reload. ... Make skinny reset vmexten and immeddial to '\0' on reload to ensure that it is set to '\0' if the appropriate item is removed/commented in skinny.conf. Also small fix re immeddial char in skinny.conf and add immedial setting to skinny show settings. (closes issue ASTERISK-21037) Reported by: snuffy Tested by: snuffy, myself Patches: immed_dial_fix.diff uploaded by snuffy (license 5024) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380925 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Russell Bryant	dfdf3d9909	Add queue_log_realtime_use_gmt option to logger.conf ... Add an option that lets you specify that the timestamps going into the realtime queue log should be in GMT instead of local time. Review: https://reviewboard.asterisk.org/r/2287/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380209 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	3fa4278a31	Merge the sorcery data access layer API. ... Sorcery is a unifying data access layer which provides a pluggable mechanism to allow object creation, retrieval, updating, and deletion using different backends (or wizards). This is a fancy way of saying "one interface to rule them all" where them is configuration, realtime, and anything else that comes along. Review: https://reviewboard.asterisk.org/r/2259/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380069 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Damien Wedhorn	e9446501c9	Add force dial keys to skinny. ... Adds a dial softkey when the device is in DAFD. The softkey is greyed (unusable) until a possible dialplan match is entered. Code includes updating transmit_selectsoftkeys to allow the use of a button mask. Also add option to use # or * as a dial now button. Original patch by snuffy cleaned up by myself. Review: https://reviewboard.asterisk.org/r/2277/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380057 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	8ed2c74fe3	app_queue: Fix multiple calls to a queue member that is in only one queue. ... When ringinuse=no queue members can receive more than one call if these calls happen at nearly the same time. * Fix so a queue member does not receive more than one call from a queue. NOTE: This fix does not prevent multiple calls to a member if the member is in more than one queue. * Did some refactoring to eliminate some code redundancy. (issue ASTERISK-16115) Reported by: nik600 Patches: jira_asterisk_16115_single_q_v1.8.patch (license #5621) patch uploaded by rmudgett Modified * Revert the -r341580 and -r341599 changes adding the queues.conf check_state_unknown option as it was added in an attempt to fix this problem. The fix did not need to be optional. The fix should not have tried to explicitly set the device state. Setting the device state by something other than the device introduces a race condition. I also could not see how the change would be effective other than delaying the app_queue code long enough for the device state to propagate to app_queue. ........ Merged revisions 378663 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378683 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 378687 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378688 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago