1977 Commits (ff4371e45d6d7d57f33a47ce71fe1a569d7de7f4)

Author	SHA1	Message	Date
Joshua Colp	ff4371e45d	res_rtp_asterisk: Fix a myriad of TURN client issues. ... 1. The number of file descriptors an ioqueue instance can handle is fixed, so we now spawn the required number to handle the load. 2. Our transport identifiers were exceeding the range supported by pjnath. 3. The TURN client did not set up client binding causing needless bandwidth usage. 4. The code no longer updates address information on each packet. 5. STUN traffic was getting looped back to Asterisk instead of going through the TURN server. 6. Synchronization now ensures things are completely setup or destroyed. 7. Logging now reflects the target the TURN server is sending to/receiving from on our behalf. ASTERISK-23577 #close Reported by: Jay Jideliov ASTERISK-23634 #close Reported by: Roman Skvirsky Review: https://reviewboard.asterisk.org/r/3982/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@423150 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	b3572f3828	res_musiconhold: Fix MOH restarting where it left off from the last hold. ... Restore code removed by https://reviewboard.asterisk.org/r/3536/ that introduced a regression that prevents MOH from restarting were it left off the last time. ASTERISK-24019 #close Reported by: Jason Richards Patches: jira_asterisk_24019_v1.8.patch (license #5621) patch uploaded by rmudgett Review: https://reviewboard.asterisk.org/r/3928/ ........ Merged revisions 421976 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@421977 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	6f9257db92	res_musiconhold.c: Remove obsolete REF_DEBUG code. ... Remove unneeded code that writes to the wrong file location in an obsolete format. ........ Merged revisions 421799 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@421800 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	3045568e47	res_musiconhold: Fix reference leaks caused when reloading with REF_DEBUG set ... Due to a faulty function for debugging reference decrementing, it was possible to reduce the refcount on the wrong object if two moh classes of the same name were in the moh class container. (closes issue ASTERISK-22252) Reported by: Walter Doekes Patches: 18_moh_debug_ref_patch.diff Uploaded by Jonathan Rose (license 6182) ........ Merged revisions 398937 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@421777 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	22b9d0ddff	Fix more dev-mode build issues ... ........ Merged revisions 419129 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@419162 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	915de454f8	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 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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@417677 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	5d3078bd48	res_http_websocket: Export symbol for ast_websocket_set_timeout ... Thanks to Sean Bright for pointing out that this was missed in #asterisk-dev. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@417419 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	32d684ee25	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. ASTERISK-23917 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3624/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@417310 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	3e642e66d1	res_rtp_asterisk: Return the length of data written when sending via ICE instead of 0. ... ASTERISK-23834 #close Reported by: Richard Kenner git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@417141 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	15d2f541b4	MoH: Don't restart stream on repeated start calls ... Currently, music on hold will stop and then start again from the beginning if ast_moh_start() is called multiple times. This can happen if a call is put on hold repeatedly (the channel receives multiple HOLD control frames) and can be triggered from ARI by starting MoH on a channel multiple times. This is fairly jarring/annoying to users. This change prevents MoH from being restarted if the requested music class is the same as the one currently playing. This includes an extra check to prevent the errors previously experienced in the testsuite and has 100+ test runs behind it. Review: https://reviewboard.asterisk.org/r/3615/ ........ Merged revisions 416439 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@416440 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Igor Goncharovskiy	07a0838e2a	We have faced situation when using CDR and CEL by sqlite3 modules. With system having high load (~100 concurrent calls created by sipp) we found many cdr and cel records missed. There is special finction in sqlite3, that make able to fix this situation - sqlite3_wait_timeout, that also can replace awful code cdr_sqlite3 ad cel_sqlite3 modules. Also this function can be used for aastdb and res_config_sqlite3 to avoid missed writes to sqlite db. ...	11 years ago
Matthew Jordan	8d848c048d	MoH: Undo commit r416150 (1.8) ... This patch reverts r416150. When the comparison between mohclass->name and state->class->name is made, you are not guaranteed that (a) state->class is non-NULL or that state or state->class are in a safe state. Crashes caught by the bridges/transfer_capabilities test. ........ Merged revisions 416251 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@416252 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	71c9c7612e	MoH: Don't restart stream on repeated start calls ... Currently, music on hold will stop and then start again from the beginning if ast_moh_start() is called multiple times. This can happen if a call is put on hold repeatedly (the channel receives multiple HOLD control frames) and can be triggered from ARI by starting MoH on a channel multiple times. This is fairly jarring/annoying to users. This change prevents MoH from being restarted if the requested music class is the same as the one currently playing. Review: https://reviewboard.asterisk.org/r/3615/ ........ Merged revisions 416150 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@416151 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	df686c50d8	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@415854 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	80237dcf5b	res_config_odbc: Use dynamically sized buffers to store row data so values do not get truncated. ... ASTERISK-23582 #close ASTERISk-23582 #comment Reported by: Walter Doekes Review: https://reviewboard.asterisk.org/r/3557/ ........ Merged revisions 414693 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@414694 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	7041eee5e5	res_config_odbc: Fix old and new ast_string_field memory leaks. ... The ODBC realtime driver uses ^NN parameter encoding to cope with the special meaning of the semi-colon. A semi-colon in a field is interpreted as if the key was supplied twice, something which isn't otherwise possible with fixed database columns. E.g. allow=alaw;ulaw is parsed as allow=alaw and allow=ulaw. A literal semi-colon is rewritten to ^3B when stored in the database. The module uses a stringfield to efficiently store the encoded parameters. However, this stringfield wasn't always freed in some off-nominal cases. Commit r413241 fixed initialization so the encoding for INSERT and DELETE queries wouldn't crash. (Only SELECTs and UPDATEs worked apparently.) But that commit forgot the frees. This change cleans that up. Review: https://reviewboard.asterisk.org/r/3555/ ........ Merged revisions 414564 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@414565 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	655e69954c	res_musiconhold: Minor cleanup. ... Fix a few free()'s that should be ast_free()'s. Reverted an old workaround that isn't necessary. Reorder a tiny bit of code. Remove a bit of commented-out code. Review: https://reviewboard.asterisk.org/r/3536/ ........ Merged revisions 413894 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413895 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	8ade79ebe3	h264: Fix H264 SDP payload format. ... https://tools.ietf.org/html/rfc3984#section-8.1 says profile-level-id takes 3 bytes in base16 (6 hex digits). This fixes video setup in certain cases. ASTERISK-23664 #close ASTERISK-23664 #comment Patch r3530.patch uploaded by Guillaume Maudoux. Review: https://reviewboard.asterisk.org/r/3530/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413791 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	abac3330cf	Allow Asterisk to compile under GCC 4.10 ... This resolves a large number of compiler warnings from GCC 4.10 which cause the build to fail under dev mode. The vast majority are signed/unsigned mismatches in printf-style format strings. ........ Merged revisions 413586 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413587 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	0becabfd1b	Fix encoding of custom prepare extra data. ... Patches: res_config_odbc-take2.patch by John Hardin (License #6512) ........ Merged revisions 413396 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413397 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	01731d5566	Ensure that all parts of SQL UPDATEs and DELETEs are encoded. ... Patches: res_config_odbc.patch by John Hardin (License #6512) ........ Merged revisions 413304 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413305 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	b57c1dd870	Prevent crashes in res_config_odbc due to uninitialized string fields. ... Patches: odbc-crash.patch by John Hardin (License #6512) ........ Merged revisions 413241 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413251 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	96cb273948	Return the number of rows affected by a SQL insert, rather than an object ID. ... The realtime API specifies that the store callback is supposed to return the number of rows affected. res_config_pgsql was instead returning an Oid cast as an int, which during any nominal execution would be cast to 0. Returning 0 when more than 0 rows were inserted causes problems to the function's callers. To give an idea of how strange code can be, this is the necessary code change to fix a device state issue reported against chan_pjsip in Asterisk 12+. The issue was that the registrar would attempt to insert contacts into the database. Because of the 0 return from res_config_pgsql, the registrar would think that the contact was not successfully inserted, even though it actually was. As such, even though the contact was query-able and it was possible to call the endpoint, Asterisk would "think" the endpoint was unregistered, meaning it would report the device state as UNAVAILABLE instead of NOT_INUSE. The necessary fix applies to all versions of Asterisk, so even though the bug reported only applies to Asterisk 12+, the code correction is being inserted into 1.8+. Closes issue ASTERISK-23707 Reported by Mark Michelson ........ Merged revisions 413224 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413225 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	4fce454645	Websocket: Add session locking and delay close ... This resolves a race condition where data could be written to a NULL FILE pointer causing a crash as a websocket connection was in the process of shutting down by adding locking to websocket session writes and by deferring session teardown until session destruction. (closes issue ASTERISK-23605) Review: https://reviewboard.asterisk.org/r/3481/ Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413123 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	374f32fcc9	res_rtp_asterisk: Add support for DTLS handshake retransmissions ... On congested networks, it is possible for the DTLS handshake messages to get lost. This patch adds a timer to res_rtp_asterisk that will periodically check to see if the handshake has succeeded. If not, it will retransmit the DTLS handshake. Review: https://reviewboard.asterisk.org/r/3337 ASTERISK-23649 #close Reported by: Nitesh Bansal patches: dtls_retransmission.patch uploaded by Nitesh Bansal (License 6418) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413008 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	45fd6e01bd	res_config_odbc/res_odbc: Fix handling of non-text columns updates with empty values. ... This patch fixes setting nullable integer columns to NULL instead of an empty string, which fails for PostgreSQL, for example. The current code is supposed to do so, but the check is broken. The patch also allows the first column in the list to be a nullable integer. This patch also adds a compatibility setting in res_odbc.conf, allow_empty_string_in_nontext. It is enabled by default. It should be disabled for database backends (such as PostgreSQL) that require NULL instead of an empty string for Integer columns. Review: https://reviewboard.asterisk.org/r/3375 (issue ASTERISK-23459) Reported by: zvision patches: res_config_odbc.diff uploaded by zvision (License 5755) ........ Merged revisions 411399 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@411408 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	8fe29356ac	Fix dialplan function NULL channel safety issues ... (closes issue ASTERISK-23391) Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/3386/ ........ Merged revisions 411313 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@411314 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Sean Bright	7199c911d9	res_fax_spandsp: Use g711_free() when available. ... Per Johann Steinwendtner on the asterisk-dev mailing list: http://lists.digium.com/pipermail/asterisk-dev/2014-March/066102.html g711_free() was introduced in spandsp 0.0.6pre4 and g711_release() became a noop. I opted not to remove the call to g711_release() since it is harmless and to call g711_free() if we have a sufficiently recent version of spandsp. (issue ASTERISK-20149) Reported by: Alexandr Gordeev git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@410829 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Russell Bryant	fa1d3b0941	moh: fix a refcount error with realtime MOH ... I observed a crash in res_musiconhold on an Asterisk 11 system using realtime MOH. Investigation of the backtrace showed a corrupt mohclass, implying that it got destroyed before the code expected it to. I went looking for reference counting errors that could have caused this crash and this patch this result. It contains 2 changes. 1) Remove a usless block of code that was impossible to reach. There was even a comment indicating that it was impossible to reach. The conditional includes "!ast_test_flag(global_flags, MOH_CACHERTCLASSES)" and it's inside of an if block with the opposite check "ast_test_flag(global_flags, MOH_CACHERTCLASSES)". There's no good reason to keep it around. 2) A similar block to #1 contained a reference counting error. It stores state->class in the local variable mohclass without increasing its reference count. The reference count on mohclass is decremented at the end of the function. This block of code probably very rarely runs, which would help explain why this system was working fine for many months before experiencing a crash. Review: https://reviewboard.asterisk.org/r/3282/ ........ Merged revisions 410043 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@410044 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	69192903af	res_fax_spandsp: Fix crash when passing ulaw/alaw data to spandsp ... When acting as a T.38 fax gateway, res_fax_spandsp would at times cause a crash in libspandsp. This would occur when, during fax tone detection, a ulaw/alaw frame would be passed to modem_connect_tones_rx. That particular routine expects the data to be in slin format. This patch looks at the frame type and, if the data is ulaw/alaw, converts the format to slin before passing it to modem_connect_tones_rx. Review: https://reviewboard.asterisk.org/r/3296 (closes issue ASTERISK-20149) Reported by: Alexandr Gordeev Tested by: Michal Rybarik patches: spandsp_g711decode.diff uploaded by Michal Rybarik (license 6578) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@409990 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Moises Silva	7f1450e32c	Fix res/res_http_websocket.c build failure in 32bit due to incorrect print format for uint64_t ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@409703 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Moises Silva	9792c46a09	Fix WebRTC over WSS not working ... Several fixes for the WebSockets implementation in res/res_http_websocket.c * Flush the websocket session FILE* as fwrite() may not actually guarantee sending the data to the network. If we do not flush, it seems that buffering on the SSL socket for outbound messages causes issues * Refactored ast_websocket_read to take into account that SSL file descriptors may be ready to read via fread() but poll() will not actually say so because the data was already read from the network buffers and is now in the libc buffers (closes issue ASTERISK-23099) (closes issue ASTERISK-21930) Review: https://reviewboard.asterisk.org/r/3248/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@409681 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	7e083b9781	res_rtp_asterisk: Fix one way audio problems with hold/unhold when using ICE ... ICE sessions will now be restarted if sessions are changed to use new sets of remote candidates. (closes issue ASTERISK-22911) Reported by: Vytis Valentinavičius Review: https://reviewboard.asterisk.org/r/3275/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@409565 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	c382714769	res_rtp_asterisk: correct build error from r409129 ... Accidentally placed a declaration below functional code (issue ASTERISK-23213) Reported by: Andrea Suisani Review: https://reviewboard.asterisk.org/r/3256/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@409130 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	8a7ef07f94	res_rtp_asterisk: Fix checklist creating problems in ICE sessions ... Prior to this patch, local candidate lists including SRFLX would fail to start properly when building ICE candidate check lists. This patch fixes that problem by making sure that each SRFLX candidate is associated with the proper base address so that the check list can create matches properly. This patch was written by jcolp. The issue will be left open to await testing by the issue participants. (issue ASTERISK-23213) Reported by: Andrea Suisani Review: https://reviewboard.asterisk.org/r/3256/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@409129 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	78d560157a	res_fax: Warn that minrate=2400 is not valid for V.27 instead of failing load. ... Change minrate from 2400 to 4800 on config reload in response to changes from ASTERISK-22790 only. Any config with minrate of 2400 that would fail before r405693 will still fail. 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@409053 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	5a7221decb	Remove extra defines of AST_PBX_MAX_STACK. ... * Ensure AST_PBX_MAX_STACK is only defined in extconf.h and pbx.h. * Fix incorrect function parameters in utils/extconf.c. (closes issue ASTERISK-23141) Reported by: Maxim Review: https://reviewboard.asterisk.org/r/3241/ ........ Merged revisions 408785 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@408786 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	05710c7ea5	res_config_pgsql: Fix ast_update2_realtime calls. ... Fix so multiple updates from a single call works (add missing ','). Remove bogus ast_free's that weren't supposed to be there. Moved a few spaces for readability. Review: https://reviewboard.asterisk.org/r/3194/ ........ Merged revisions 407873 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@407874 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	65e9857f09	res_clialiases: Fix crash when reloading and re-aliasing an alias that is in use. ... The code assumed that unregistering the alias would always succeed while in practice this is not actually true. A common case is the "reload" command itself. If the cli_aliases.conf configuration file was changed and reload executed the command would fail to unregister and ultimately point to freed memory. The reload process now checks whether unregistering succeeded or not and if not the old CLI alias is retained. (closes issue ASTERISK-19773) Reported by: Joel Vandal (closes issue ASTERISK-22757) Reported by: Gareth Blades ........ Merged revisions 407205 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@407210 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Corey Farrell	1968d29447	res_rtp_asterisk & udptl: fix port selection to work with SELinux restrictions ... ast_bind to a port reserved for another program by SELinux causes errno == EACCES. This caused random failures when binding rtp or udptl sockets. Treat EACCES as a non-fatal error, try next port. (closes issue ASTERISK-23134) Reported by: Corey Farrell ........ Merged revisions 406933 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@406934 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	ab81490ebc	res_config_pgsql: Fix a memory leak and use RAII_VAR for cleanup when practical ... Review: https://reviewboard.asterisk.org/r/3141/ ........ Merged revisions 406360 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@406361 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	4314a33ae2	chan_mgcp: Enforce locking for oseq ... This restricts direct usage of global oseq so that all accesses are locked and threads are not racing to get oseq values that they did not claim. This also fixes a build error in res_pktccops under dev mode. (closes issue ASTERISK-23100) Reported by: adomjan Patch by: adomjan ........ Merged revisions 406037 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@406038 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	b83fb704c9	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@405693 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	6d73ecd6e9	res/Makefile: alias dist-clean to distclean ... A 'make distclean' is equivalent to 'make dist-clean' in the top most Makefile. This patch updates the res/Makefile to recognize both distclean and dist-clean. Note that this is needed for removing build.mak, which can run into problems if the source file of Asterisk or its path is changed after build.mak is generated. (issue ASTERISK-22480) Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@405362 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	dee9ce4a0b	res_rtp_asterisk: Fails to resume WebRTC call from hold ... In ast_rtp_ice_start if the ice session create check list failed, start check was never initiated and ice_started was never set to true. Upon re-entering the function (for instance, [un]hold) it would try to create the check list again with duplicate remote candidates. Fixed so that if the create check list fails the necessary data structures are properly re-initialized for any subsequent retries. Note, it was decided to not stop ice support (by calling ast_rtp_ice_stop) on a check list failure because it possible things might still work. However, a debug message was added to help with any future troubleshooting. (closes issue ASTERISK-22911) Reported by: Vytis Valentinavičius Patches: works_on_my_machine.patch uploaded by xytis (license 6558) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@405234 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	1f1ac57bbe	res_fax.c: crash on framehook with no dsp in fax detect ... In fax_detect_framehook() a null pointer reference can occur where a voice frame is processed but no dsp is attached to the fax detection structure. The code block that rejects frames that detection cannot be processed on is checking for dsp but falls through when it should instead return, as this change implements. (closes issue ASTERISK-22942) Reported by: adomjan Review: https://reviewboard.asterisk.org/r/3076/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@404351 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	d9b7ce0599	res_calendar: Protect channel when adding datastore. ... This change adds a missing channel lock when adding a datastore to a channel. ........ Merged revisions 404135 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@404136 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	e7d81f02e5	res_fax_spandsp: Always init T.38 session to avoid crashes during state change ... Prior to this patch, res_fax_spandsp was conservative with how it initialized the spandsp T.38 context. It would only initialize it if the driver thought the current state was a T.38 fax. While this works fine in nominal situations, in certain off nominal situations, res_fax_spandsp can believe that a T.38 fax will not occur when in fact one has started. In particular, this was discovered when res_fax would fall back to audio after timing out on a T.38 upgrade. The SIP channel driver would continue to retry the re-INVITE and - if the remote end responded after res_fax timed out with a 200 OK - a T.38 frame would be delivered to the res_fax stack when it no longer expected it. As it turns out, there does not appear to be any downside to always initializing the T.38 context, other than the actual memory allocation. Since that avoids this off nominal situation (and others which are equally likely hard to predict), this is the safest way to avoid this problem. Much thanks to Torrey as well for providing a scenario that reproduces this issue. (closes issue ASTERISK-21242) Reported by: Ashley Winters Tested by: Torrey Searle patches: always-init-t38.patch uploaded by awinters (License 6477) A_PARTY.xml uploaded by tsearle (License 5334) ........ Merged revisions 403449 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@403450 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	2d5debf45c	chan_sip: Fix RTCP port for SRFLX ICE candidates ... This corrects one-way audio between Asterisk and Chrome/jssip as a result of Asterisk inserting the incorrect RTCP port into RTCP SRFLX ICE candidates. This also exposes an ICE component enumeration to extract further details from candidates. (closes issue ASTERISK-21383) Reported by: Shaun Clark Review: https://reviewboard.asterisk.org/r/2967/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@402345 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	4241f7b6ea	res_rtp_asterisk: Address jittery DTMF events in RTP streams ... (closes issue ASTERISK-21170) Reported by: NITESH BANSAL Patches: dtmf-timestamp.patch uploaded by NITESH BANSAL (license 6418) Review: https://reviewboard.asterisk.org/r/2938/ ........ Merged revisions 401619 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@401620 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago