asterisk

Commit Graph

Author	SHA1	Message	Date
Asterisk Development Team	b3d0754f82	Update CHANGES and UPGRADE.txt for 13.28.0	6 years ago
Walter Doekes	3449240130	sched: Don't allow ast_sched_del to deadlock ast_sched_runq from same thread When fixing ASTERISK~24212, a change was done so a scheduled callback could not be removed while it was running. The caller of ast_sched_del would have to wait. However, when the caller of ast_sched_del is the callback itself (however wrong this might be), this new check would cause a deadlock: it would wait forever for itself. This changeset introduces an additional check: if ast_sched_del is called by the callback itself, it is immediately rejected (along with an ERROR log and a backtrace). Additionally, the AST_SCHED_DEL_UNREF macro is adjusted so the after-ast_sched_del-refcall function is only run if ast_sched_del returned success. This should fix the following spurious race condition found in chan_sip: - thread 1: schedule sip_poke_peer_now (using AST_SCHED_REPLACE) - thread 2: run sip_poke_peer_now - thread 2: blank out sched-ID (too soon!) - thread 1: set sched-ID (too late!) - thread 2: try to delete the currently running sched-ID After this fix, an ERROR would be logged, but no deadlocks (in do_monitor) nor excess calls to sip_unref_peer(peer) (causing double frees of rtp_instances and other madness) should occur. (Thanks Richard Mudgett for reviewing/improving this "scary" change.) Note that this change does not fix the observed race condition: unlocked access to peer->pokeexpire (and potentially other scheduled items in chan_sip), causing AST_SCHED_DEL_UNREF to look at a changing id. But it will make the deadlock go away. And in the observed case, it will not have adverse affects (like memory leaks) because the scheduled item is removed through a different path. ASTERISK-28282 Change-Id: Ic26777fa0732725e6ca7010df17af77a012aa856	6 years ago
George Joseph	aaafe8a108	Build: Add separate header install/uninstall targets Two new Makefile targets have been added... "install-headers" and "uninstall-headers" to separately control header installation. The existing behavior has not changed so "make install" and "make uninstall" will continue to also install/uninstall the headers. The new targets were added for forward compatibility with Asterisk 17 in which the headers are no longer installed/uninstalled with the "install" and "uninstall" targets. Also corrects an issue where /usr/include/asterisk.h was never being removed at all. Change-Id: Ia7399f3a0203a4825fc4a9f43b9034dae9a2b643	6 years ago
Kevin Harwell	d2eaba4853	manager: Log AMI actions When manager debugging is turned on, this patch makes it so incoming AMI actions are now also logged. Change-Id: I8047524510e7ac97d99482b2448f8e368f29cd47	6 years ago
Joshua Colp	3cb839d111	res_rtp_asterisk: Move where DTLS MTU variable is defined. The DTLS MTU variable is not dependent on pjproject and should not exist in its block. Change-Id: I7e97d64dc192f2ac81bfe2b72b8229d321c7d026	6 years ago
Friendly Automation	a7bb02ae34	Merge "res_pjsip_messaging: Check for body in in-dialog message" into 13	6 years ago
George Joseph	41d6492de0	res_pjsip_messaging: Check for body in in-dialog message We now check that a body exists and it has a length > 0 before attempting to process it. ASTERISK-28447 Reported-by: Gil Richard Change-Id: Ic469544b22ab848734636588d4c93426cc6f4b1f	6 years ago
Francesco Castellano	1318a3a2b7	chan_sip: Handle invalid SDP answer to T.38 re-invite The chan_sip module performs a T.38 re-invite using a single media stream of udptl, and expects the SDP answer to be the same. If an SDP answer is received instead that contains an additional media stream with no joint codec a crash will occur as the code assumes that at least one joint codec will exist in this scenario. This change removes this assumption. ASTERISK-28465 Change-Id: I8b02845b53344c6babe867a3f0a5231045c7ac87	6 years ago
Tzafrir Cohen	d8bf4b1608	openr2(1/6): bugfix in configuration saving Details: - The memcpy() call copied part of "dahdi_conf" and not "dahdi_conf.mfcr2" - As a result, the memcmp() in dahdi_r2_get_link() always fails - This cause dahdi_r2_get_link() to create new link for every channel (instead of a new link for every ~30 channels) - With the fix, far less links are generated -- so we use far less threads Change-Id: I7259dd6272f5e46e8a6c7f5bf3e8c2ec01b8c132 Signed-off-by: Oron Peled <oron.peled@xorcom.com>	6 years ago
Chris-Savinovich	e206a54d59	chan_dahdi.c: crash in chan_dahdi Fixes a crash in chan_dahdi occurring on 32-bit systems. A previous patch introduced a variable of type unassigned long long which is 64-bits. Casting it as 'ast_json_int_t' along with JSON type 'I' makes it work with 32-bit systems. ASTERISK-28457 Change-Id: I9cef6b5f2d826fc5c93f2f6a1c997c4e3e6c93fe	6 years ago
George Joseph	72f001d558	Merge "pjproject_bundled: Add peer information to most SSL/TLS errors" into 13	6 years ago
Friendly Automation	6ac6218f73	Merge "tcptls.c: Add peer hostname and port to some error messages" into 13	6 years ago
Kevin Harwell	8ef52e8a19	Merge "pjproject: Update to 2.9 release" into 13	6 years ago
George Joseph	be54e94773	tcptls.c: Add peer hostname and port to some error messages Where possble, hostname and port has been added to error messages, mostly on the server side. ASTERISK-26006 Reported by: Oleksandr Natalenko Change-Id: Iff4f897277bc36ce8c5b493b71d0a4a7b74e62f0	6 years ago
George Joseph	493268a938	pjproject_bundled: Add peer information to most SSL/TLS errors Most SSL/TLS error messages coming from pjproject now have either the peer address:port or peer hostname, depending on what was available at the time and code location where the error was generated. ASTERISK-28444 Reported by: Bernhard Schmidt Change-Id: I41770e8a1ea5e96f6e16b236692c4269ce1ba91e	6 years ago
Kevin Harwell	a302d46dd8	Merge "app_amd: issue with silence suppression fixed" into 13	6 years ago
Sean Bright	ba57b004c4	pjproject: Update to 2.9 release Relies on https://github.com/asterisk/third-party/pull/4 Change-Id: Iec9cad42cb4ae109a86a3d4dae61e8bce4424ce3	6 years ago
George Joseph	94e0e26c2a	Merge "CI: New way to determnine libdir" into 13	6 years ago
George Joseph	44e65b92c1	Merge "sig_pri: Address gcc9 issues" into 13	6 years ago
George Joseph	dd20beedb9	Merge "res_pjsip_sdp_rtp: Fix ICE candidates leak." into 13	6 years ago
George Joseph	e39dbc4909	Merge "res_fax: gateway sends T.38 request to both endpoints if V.21 detected" into 13	6 years ago
George Joseph	79087b6aeb	sig_pri: Address gcc9 issues A few more format truncation issues addressed. Change-Id: I047f373169caaca0eec4889d3c0e5e10f130017a	6 years ago
Joshua Colp	9721f3908d	res_pjsip_sdp_rtp: Fix ICE candidates leak. Given the non-default configuration of enabling ICE support on an endpoint that does not result in an ICE negotiation occurring the ICE candidates would be leaked. This change makes it so that the ICE candidates are only retrieved if ICE negotiation is occurring. ASTERISK-28460 Change-Id: I7b3f76f031c41fb8a3dc3ef1a84b77e2a8cb969f	6 years ago
George Joseph	f22dedc597	Merge "app_confbridge: Attended transfer event fixup" into 13	6 years ago
George Joseph	5d45686c28	Merge "translate.c do not log WARNING on empty audio frame" into 13	6 years ago
Alexei Gradinari	d5db7473e7	res_fax: gateway sends T.38 request to both endpoints if V.21 detected According T.38 Gateway 'Use case 3' https://wiki.asterisk.org/wiki/display/AST/T.38+Gateway T.38 Gateway should send T.38 negotiation request to called endpoint if FAX preamble (using V.21 detector) generated by called endpoint. But it does not, because fax_gateway_detect_v21 constructs T.38 negotiation request, but forwards it only to other channel, not to the channel on which FAX preamble is detected. Some SIP endpoints could be improperly configured to rely on the other side to initiate T.38 re-INVITEs. With this patch the T.38 Gateway tries to negotiate with both sides by sending T.38 negotiation request to both endpoints supported T.38. Change-Id: I73bb24799bfe1a48adae9c034a2edbae54cc2a39	6 years ago
George Joseph	e4ee209bfe	CI: New way to determnine libdir We were using the presence of /usr/lib64 to determine where shared libraries should be installed. This only existed on Redhat based systems and was safe. If it existed, use it, otherwise use /usr/lib. Unfortunately, Ubuntu 19 decided to create a /usr/lib64 BUT NOT INCLUDE IT IN THE DEFAULT ld.so.conf. So if anything is installed there, it won't work. The new method, just looks for $ID in /etc/os-release and if it's centos or fedora, uses /usr/lib64 and if ubuntu, uses /usr/lib. NOTE: This applies only to the CI scripts. Normal asterisk build and install is not affected. Change-Id: Iad66374b550fd89349bedbbf2b93f8edd195a7c3	6 years ago
Alexei Gradinari	3bfe1f3a1b	translate.c do not log WARNING on empty audio frame There is WARNING "no samples for ..." on each Playtones. The function ast_playtones_start calls ast_activate_generator, which calls ast_prod. The function ast_prod calls ast_write with empty audio frame. In this case it's spam log. Change-Id: Id4ac309489d9ff281bad02abdef341cecdede660	6 years ago
George Joseph	d0f01af913	chan_dahdi: Address gcc9 issues Fixed format-truncation issues in chan_dahdi.c and sig_analog.c. Since they're related to fields provided by dahdi-tools we can't change the buffer sizes so we're just checking the return from snprintf and printing an errior if we overflow. Change-Id: Idc1f3c1565b88a7d145332a0196074b5832864e5	6 years ago
Joshua Colp	d5fed38ab5	Merge "res_rtp_asterisk: Add support for DTLS packet fragmentation." into 13	6 years ago
George Joseph	41f5d15763	app_confbridge: Attended transfer event fixup When a channel already in a conference bridge is attended transfered to another extension, or when an existing call is attended transferred into a conference bridge, we now generate ConfbridgeJoin and ConfbridgeLeave events for the entering and departing channels. Change-Id: Id7709cfbceb26fbcb828b2d0d2a6b2fbeaf028e1	6 years ago
Joshua Colp	1ea9bad34d	res_rtp_asterisk: Add support for DTLS packet fragmentation. This change adds support for larger TLS certificates by allowing OpenSSL to fragment the DTLS packets according to the configured MTU. By default this is set to 1200. This is accomplished by implementing our own BIO method that supports MTU querying. The configured MTU is returned to OpenSSL which fragments the packet accordingly. When a packet is to be sent it is done directly out the RTP instance. ASTERISK-28018 Change-Id: If2d5032019a28ffd48f43e9e93ed71dbdbf39c06	6 years ago
George Joseph	83c353c650	Merge "app_attended_transfer: new application AttendedTransfer" into 13	6 years ago
George Joseph	dd4f0c94e8	Merge "app_blind_transfer: new application BlindTransfer" into 13	6 years ago
George Joseph	0e30d17293	Merge "chan_pjsip.c: Check for channel and session to not be NULL in hangup" into 13	6 years ago
Alexei Gradinari	45a9ee4c53	app_attended_transfer: new application AttendedTransfer AttendedTransfer queues up attended transfer to the given extension. This application can be useful with Custom Dynamic Features. For example to make attended transfer to a predefined number. features.conf ;;; [applicationmap] my_atxfer => 7,self,GoSub,"my_atxfer,s,1",default ;;; extensions.conf ;;; [globals] DYNAMIC_FEATURES=my_atxfer TRANSFER_CONTEXT=my_transfer [my_atxfer] exten => s,1,AttendedTransfer(1234567890) same => n,Return() [my_transfer] include => default ;;; This application also can be used to completly redefine Attended transfer feature using dialplan. For example: features.conf ;;; [featuremap] atxfer => 7 [applicationmap] custom_atxfer => *2,self,GoSub,"custom_atxfer,s,1",default ;;; extensions.conf ;;; [globals] DYNAMIC_FEATURES=custom_atxfer TRANSFER_CONTEXT=my_transfer [custom_atxfer] exten => s,1, same => n,Playback(pbx-transfer) same => n,Read(dest,dial,10,i,3,3) same => n,AttendedTransfer(${dest}) same => n,Return() [my_transfer] include => default ;;; Change-Id: Ie5cfa455d0813cffd5c85a6fb117f07d8f0b903b	6 years ago
agupta	67841b8f55	chan_pjsip.c: Check for channel and session to not be NULL in hangup We have seen some rare case of segmentation fault in hangup function and we could notice that channel pointer was NULL. Debug log shows that there is a 200 OK answer and SIP timeout at the same time. It looks that while the SIP session was being destroyed due to timeout call hangup due to answer event lead to race condition and channel is being destroyed from two different places. The check ensures we check it not to be NULL before freeing it. ASTERISK-25371 Change-Id: I19f6566830640625e08f7b87bfe15758ad33a778	6 years ago
Alexei Gradinari	dd12e1cbd3	app_blind_transfer: new application BlindTransfer BlindTransfer redirects all channels currently bridged to the caller channel to the specified destination. This application can be useful with Custom Dynamic Features. For example to make blind transfer to a predefined number. features.conf ;;; [applicationmap] my_blindxfer => *6,self,GoSub,"my_blindxfer,s,1",default ;;; extensions.conf ;;; [globals] DYNAMIC_FEATURES=my_blindxfer [my_blindxfer] exten => s,1,BlindTransfer(1234567890,default) same => n,Return() ;;; This application also can be used to completly redefine Blind transfer feature using dialplan. For example: features.conf ;;; [featuremap] blindxfer => [applicationmap] custom_blindxfer => ##,self,GoSub,"custom_blindxfer,s,1",default ;;; extensions.conf ;;; [globals] DYNAMIC_FEATURES=custom_blindxfer [custom_blindxfer] exten => s,1, same => n,Playback(pbx-transfer) same => n,Read(dest,dial,10,i,3,3) same => n,BlindTransfer(${dest},default) same => n,Return() ;;; Change-Id: I9d55e7f69ccfd4472dec00d62771d6de8803215a	6 years ago
Chris-Savinovich	45c1159c62	cdr_pgsql: fix error in connection string Fixes an error occurring in function pgsql_reconnect() caused when value of hostname is blank. Which in turn will cause the connection string to look like this: "host= port=xx", which creates a sintax error. This fix now checks if the corresponding values for host, port, dbname, and user are blank. Note that since this is a reconnect function the database library will replace any missing value pairs with default ones. ASTERISK-28435 Change-Id: I0a921f99bbd265768be08cd492f04b30855b8423	6 years ago
Joshua Colp	1c665ae39b	Merge "res_fax: fix segfault on inactive "reserved" fax session" into 13	6 years ago
Friendly Automation	6df25921ca	Merge "app_readexten: new option 'p' to stop reading on '#' key" into 13	6 years ago
Friendly Automation	066ff4d3a8	Merge "res_fax: add channel name to CLI 'fax show session'" into 13	6 years ago
Friendly Automation	e4d91ab4b2	Merge "pjsip: replace 180 by 183 if SDP negotiation has completed" into 13	6 years ago
Alexei Gradinari	e306c62ff1	res_fax: fix segfault on inactive "reserved" fax session The change #10017 "Handle fax gateway being started more than once" introdiced a bug which leads to segfault in res_fax_spandsp. The res_fax_spandsp module does not support reserving sessions, so fax_session_reserve returns a fax session with state AST_FAX_STATE_INACTIVE. The fax_gateway_start does not create a real fax session if the fax session is already present and the state is not AST_FAX_STATE_RESERVED. But the "reserved" session created for res_fax_spandsp has state AST_FAX_STATE_INACTIVE, so fax_gateway_start not starting. Then when fax_gateway_framehook is called and gateway T.38 state is NEGOTIATED the call of gateway->s->tech->write(gateway->s, f) leads to segfault, because session tech_pvt is not set, i.e. the tech session was not initialized/started. This patch adds check also on AST_FAX_STATE_INACTIVE to the "reserved" session created for res_fax_spandsp will start. This patch also adds extra check and log ERROR if tech_pvt is not set before call tech->write. ASTERISK-27981 #close Change-Id: Ife3e65e5f18c902db2ff0538fccf7d28f88fa803	6 years ago
Asterisk Development Team	fba341af8b	Update CHANGES and UPGRADE.txt for 13.27.0	6 years ago
Friendly Automation	7dd93bfe9a	Merge "build: Fix file format in CHANGES-staging." into 13	6 years ago
Nasir Iqbal	f7527921b6	app_amd: issue with silence suppression fixed Now AMD algorithm will not ignore AST_FRAME_NULL, As I think using manual wait time instead of `framelength` is enough to fix timeout / TOOLONG issue. ASTERISK-28419 #close Change-Id: I16ea2d6295bc99b975e8c092e5f9fbd9214debdb	6 years ago
Alexei Gradinari	dfa513c565	res_fax: add channel name to CLI 'fax show session' This patch adds a channel name to output of CLI 'fax show session' and also expands the channel name field up to 30 characters on CLI 'fax show sessions' Change-Id: Id059c43ff41811f5e76712b83fb63b8f246da953	6 years ago
Ben Ford	6aeab9d5e7	build: Fix file format in CHANGES-staging. One of the change files doesn't conform to the format that the release scripts need in order to parse it. Change-Id: Ie0b634cf27e4cbc671b9fe92993b6f2ecf60254c	6 years ago
Guido Falsi	ac4921c373	chan_dahdi: add missing include. After some definitions have been moved to asterisk/mwi.h the files channels/chan_dahdi.h channels/sig_pri.c are missing this new include. ASTERISK-28427 #close Change-Id: Ia8cc595eeda653324643f40dcd9799d4c3f0ac91	6 years ago

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