8363 Commits (c8d53a1638e751b19c4eea51101341dd111a4380)

Author	SHA1	Message	Date
Alexander Traud	4fc22c7673	chan_sip: Session Timers required but refused wrongly. ... SIP user-agents indicate which protocol extensions are allowed in headers like Supported and Required. Such protocol extensions are Session Timers (RFC 4028) for example. Session Timers are supported since Mantis-10665. Since ASTERISK-21721, not only the first but multiple Supported/Required headers in a message are parsed. In that change, an existing variable was re-used within a newly added do-loop. Currently, at the end of that loop, that variable is an empty string always. Previously, that variable was used within log output. However, the log output was not changed. ASTERISK-26915 #close Change-Id: I09315f31b4d78fb214bb2a9fb6c0f5e143eae990	8 years ago
Richard Mudgett	e8b1bb3041	chan_vpb.cc: Fix compiler error. ... Added missing channel technology read/write stream callback initialization. Change-Id: I829043a327d987e0d964485dd3d27964bebbd623	8 years ago
Sean Bright	fc71c18a9b	thread safety: Don't use getprotobyname() ... POSIX does not require getprotobyname() to be thread safe and some implementations use static memory which causes issues when multiple threads are used. Further, our usage of it today is just to ultimately get IPPROTO_TCP for calls to setsockopt(). So instead we just use IPPROTO_TCP directly. Change-Id: I2e14e58674808f7ce99b2f5e900d0f90d0d8da48	8 years ago
Sean Bright	15aa3c0a23	chan_sip: Add rtcp-mux support ... ASTERISK-26846 #close Change-Id: I541a1602ff55ab73684e9f8002edb9e0e745d639	8 years ago
Joshua Colp	aec2a087ed	Merge "chan_iax2: Reload of iax peer results in loss of host address/port"	8 years ago
zuul	3fe1d8afba	Merge "core: Add stream topology changing primitives with tests."	8 years ago
Richard Begg	0dc007e94d	chan_iax2: Reload of iax peer results in loss of host address/port ... When using a non-dynamic peer address, build_peer() invalidates the peer address structure by setting the address family to unspecified. However, if dnsmgr is enabled, the subsequent call to ast_dnsmgr_lookup() will not amend the peer address if the cache is still valid, resulting in peer connectivity failures. To fix this, we call ast_dnsmgr_refresh() instead. ASTERISK-26865 Change-Id: Id8a89a2f771ebbaf32255a35fe596a6dcb97a082	8 years ago
zuul	2b611a8d93	Merge "chan_pjsip: Don't assume a session will have a channel."	8 years ago
Joshua Colp	578bc33f6f	Merge "chan_sip: Call not cancelled after receiving a 422 response"	8 years ago
Joshua Colp	12460b05c1	chan_pjsip: Don't assume a session will have a channel. ... When querying for PJSIP specific information using the dialplan function CHANNEL() it is possible that the underlying session will no longer have a channel associated with it. This is most likely to occur when the RTCP HEP module attempts to get the channel name. If this happens then a crash will occur. This change just adds a check that the channel exists on the session before querying it. ASTERISK-26857 Change-Id: I113479cffff6ae64cf8ed089e9e1565223426f01	8 years ago
Daniel Journo	36fed72614	pjsip/cli_commands: pjsip show channelstats shows wrong codec ... * cli_commands.c Fixed CLI output ASTERISK-26822 #close Change-Id: I3889ef6a8f6738fc312fab42db5efacd6e452b01	8 years ago
Jean Aunis	d51ca4b406	chan_sip: Call not cancelled after receiving a 422 response ... When receiving a 422 response, the invitestate variable must be reset to INV_CALLING. ASTERISK-26841 Change-Id: Ia0502d6b02192664cefa4e75bafdd2645ce56099	8 years ago
Joshua Colp	3ed05badb9	core: Add stream topology changing primitives with tests. ... This change adds a few things to facilitate stream topology changing: 1. Control frame types have been added for use by the channel driver to notify the application that the channel wants to change the stream topology or that a stream topology change has been accepted. They are also used by the indicate interface to the channel that the application uses to indicate it wants to do the same. 2. Legacy behavior has been adopted in ast_read() such that if a channel requests a stream topology change it is denied automatically and the current stream topology is preserved if the application is not capable of handling streams. Tests have also been written which confirm the multistream and non-multistream behavior. ASTERISK-26839 Change-Id: Ia68ef22bca8e8457265ca4f0f9de600cbcc10bc9	8 years ago
Richard Mudgett	c9296b23d1	core: Cleanup ast_get_hint() usage. ... * manager.c:manager_state_cb() Fix potential use of uninitialized hint[] if a hint does not exist for the requested extension. Ran into this when developing a testsuite test. The AMI event ExtensionStatus came out with the hint header value containing garbage. The AMI event PresenceStatus also had the same issue. * manager.c:action_extensionstate() no need to completely initialize the hint[]. Only initialize the first element. * pbx.c:ast_add_hint() Remove unnecessary assignment. * chan_sip.c: Eliminate an unneeded hint[] local variable. We only care about the return value of ast_get_hint() there. Change-Id: Ia9a8786f01f93f1f917200f0a50bead0319af97b	8 years ago
Igor Goncharovsky	135bea931c	chan_unistim: fix char type to have consistent behavior on ARM ... There is difference exists in behaviour of char type on x86 and ARM. On x86 by default char variable type means signed char, but in ARM unsigned char used. This make binary calculations and negative values works wrong on ARM. This patch change type of char variables used for store negative values and binary calculations to signed char. ASTERISK-26714 Change-Id: Id78716dee9568a58419d4ef63c038affc3dfc7ab	8 years ago
zuul	182c737353	Merge "cli: Fix various CLI documentation and completion issues"	8 years ago
Joshua Colp	72845bd4b5	Merge "core: Cleanup some channel snapshot staging anomalies."	8 years ago
Sean Bright	3f94373778	cli: Fix various CLI documentation and completion issues ... * app_minivm: Use built-in completion facilities to complete optional arguments. * app_voicemail: Use built-in completion facilities to complete optional arguments. * app_confbridge: Add missing colons after 'Usage' text. * chan_alsa: Use built-in completion facilities to complete optional arguments. * chan_sip: Use built-in completion facilities to complete optional arguments. Add completions for 'load' for 'sip show user', 'sip show peer', and 'sip qualify peer.' * chan_skinny: Correct and extend completions for 'skinny reset' and 'skinny show line.' * func_odbc: Correct completions for 'odbc read' and 'odbc write' * main/astmm: Use built-in completion facilities to complete arguments for 'memory' commands. * main/bridge: Correct completions for 'bridge kick.' * main/ccss: Use built-in completion facilities to complete arguments for 'cc cancel' command. * main/cli: Add 'all' completion for 'channel request hangup.' Correct completions for 'core set debug channel.' Correct completions for 'core show calls.' * main/pbx_app: Remove redundant completions for 'core show applications.' * main/pbx_hangup_handler: Remove unused completions for 'core show hanguphandlers all.' * res_sorcery_memory_cache: Add completion for 'reload' argument of 'sorcery memory cache stale' and properly implement. Change-Id: Iee58c7392f6fec34ad9d596109117af87697bbca	8 years ago
Norbert Varga	75f8167e66	chan_pjsip: Multidomain endpoint finding on call ... When PJSIP tries to call an endpoint with a domain (e.g. 1000@test.com), the user part is stripped down as it would be a trunk with a specified user, and only the host part is called as a PJSIP endpoint and can't be found. This is not correct in the case of a multidomain SIP account, so the stripping after the @ sign is done only if the whole endpoint (in multidomain case 1000@test.com) can't be found. ASTERISK-26248 Change-Id: I3a2dd6f57f3bd042df46b961eccd81d31ab202e6	8 years ago
Richard Mudgett	16fdb11bc3	core: Cleanup some channel snapshot staging anomalies. ... We shouldn't unlock the channel after starting a snapshot staging because another thread may interfere and do its own snapshot staging. * app_dial.c:dial_exec_full() made hold the channel lock while setting up the outgoing channel staging. Made hold the channel lock after the called party answers while updating the caller channel staging. * chan_sip.c:sip_new() completed the channel staging on off-nominal exit. Also we need to use ast_hangup() instead of ast_channel_unref() at that location. * channel.c:__ast_channel_alloc_ap() added a comment about not needing to complete the channel snapshot staging on off-nominal exit paths. * rtp_engine.c:ast_rtp_instance_set_stats_vars() made hold the channel locks while staging the channels for the stats channel variables. Change-Id: Iefb6336893163f6447bad65568722ad5d5d8212a	8 years ago
George Joseph	ef4deb8ecd	debug_utilities: Add ast_logescalator ... The escalator works by creating a set of startup commands in cli.conf that set up logger channels and issue the debug commands for the subsystems specified. If asterisk is running when it is executed, the same commands will be issued to the running instance. The original cli.conf is saved before any changes are made and can be restored by executing '$prog --reset'. The log output will be stored in... $astlogdir/message.$uniqueid $astlogdir/debug.$uniqueid $astlogdir/dtmf.$uniqueid $astlogdir/fax.$uniqueid $astlogdir/security.$uniqueid $astlogdir/pjsip_history.$uniqueid $astlogdir/sip_history.$uniqueid Some minor tweaks were made to chan_sip, and res_pjsip_history so their history output could be send to a log channel as packets are captured. A minor tweak was also made to manager so events are output to verbose when "manager set debug on" is issued. Change-Id: I799f8e5013b86dc5282961b27383d134bf09e543	8 years ago
Richard Mudgett	d87f81ddb1	chan_oss.c: Fix format ref leak in oss_read(). ... Change-Id: I0a5d56c7dcf327d60f86a4c25a23571733709fd0	8 years ago
zuul	acc7f3c243	Merge "chan_sip: Remember SDP negotiation on SIP_CODEC_INBOUND."	8 years ago
Joshua Colp	38b4189643	Merge changes from topic 'ASTERISK-26672' ... * changes: res_rtp_asterisk.c: Fix uninitialized memory crash. chan_rtp.c: Fix uninitialized memory crash. res_rtp_asterisk.c: Initialize ourip passed to ast_find_ourip().	8 years ago
Alexander Traud	e220c11bec	chan_sip: Remember SDP negotiation on SIP_CODEC_INBOUND. ... After a SIP_CODEC_INBOUND in the dialplan, do not continue with cached formats but remember the joint format. Cached formats contain default parameters, often create an empty fmtp line. However, a joint format might have passed format_get_joint(.) in a res_format_attr_* module (like Opus Codec) and contain the resulting format parameters from a SDP negotiation. ASTERISK-26691 #close Change-Id: I35712d98a793d4c3efdd156cec57deab9014b1dc	8 years ago
Joshua Colp	ae57652983	chan_pjsip: Use session for retrieving CHANNEL() information. ... The CHANNEL() dialplan function implementation for PJSIP allows querying of PJSIP specific information. This used the channel passed in to get the PJSIP session and associated information. It is possible for this channel to be masqueraded and end up as a different channel type by the time the information request is actually acted upon. This change retrieves the PJSIP session safely and accesses data from it (including channel). This provides a guarantee that the session and channel will not be altered when the request is being acted upon. ASTERISK-26673 Change-Id: I335e12b89e1820cafdd92b3e7526b8ba649eb7e6	8 years ago
Richard Mudgett	67b47191e9	chan_rtp.c: Fix uninitialized memory crash. ... unicast_rtp_request() could pass an uninitialized 'us' parameter to ast_ouraddrfor(). If ast_ouraddrfor() returns an error then the 'us' parameter may not get initialized. Thus when the code tries to save the 'us' parameter to the local address we could try to copy a ridiculous sized memory buffer and segfault. * Made pass an initialized 'us' parameter to ast_ouraddrfor() and abort the UnicastRTP channel request if it fails. ASTERISK-26672 Change-Id: I1ef7a7c09f4da4f15dcb6de660d2bcac5f2a95c0	8 years ago
Joshua Colp	1943ece514	Merge "chan_dahdi.c: Fix bounds check regression."	8 years ago
Corey Farrell	8fbb384ea2	chan_sip: Reorder unload_module to deal with stuck TCP threads. ... In some situations TCP threads may become frozen. This creates the possibility that Asterisk could segfault if they become unfrozen after chan_sip has been dlclose'd. This reorders the unload_module process to allow abort if threads do not exit within 5 seconds. High level order as follows: 1) Unregister from the core to stop new requests. 2) Signal threads to stop 3) Clear config based tables (but do not free the table itself). 4) Verify that threads have shutdown, cancel unload if not. 5) Clean all remaining resources. ASTERISK-26586 Change-Id: Ie23692041d838fbd35ece61868f4c640960ff882	8 years ago
Richard Mudgett	9404efa6f4	chan_dahdi.c: Fix bounds check regression. ... Caused by ASTERISK-25494 Change-Id: I1fc408c1a083745ff59da5c4113041bbfce54bcb	8 years ago
Joshua Colp	963735dfce	Merge "Fix typo in chan_sip"	9 years ago
Joshua Colp	b8a0770d74	Merge "chan_sip: Delete unneeded check"	9 years ago
Badalyan Vyacheslav	4c6ba1dbba	Fix typo in chan_sip ... The conditional expressions of the 'if' operators situated alongside each other are identical. Change-Id: I652b6dcddb3be007e669a6aa8107edb31a1ddafb	9 years ago
Badalyan Vyacheslav	51118e7d70	chan_sip: Delete unneeded check ... P is always true. We check it before Change-Id: Iee61cda002a9f61aee26b9f66c5f9b59e3389efb	9 years ago
Badalyan Vyacheslav	fe5be81821	Small code cleanup in chan_sip ... The conditional expressions of the 'if' operators situated alongside each other are identical. Change-Id: I2cf7c317b106ec14440c7f1b5dcfbf03639f748a	9 years ago
Walter Doekes	c796f00c35	chan_sip: Do not allow non-SP/HTAB between header key and colon. ... RFC says SIP headers look like: HCOLON = *( SP / HTAB ) ":" SWS SWS = [LWS] ; sep whitespace LWS = [*WSP CRLF] 1*WSP ; linear whitespace WSP = SP / HTAB ; from rfc2234 chan_sip implemented this: HCOLON = *( LOWCTL / SP ) ":" SWS LOWCTL = %x00-1F ; CTL without DEL This discrepancy meant that SIP proxies in front of Asterisk with chan_sip could pass on unknown headers with \x00-\x1F in them, which would be treated by Asterisk as a different (known) header. For example, the "To\x01:" header would gladly be forwarded by some proxies as irrelevant, but chan_sip would treat it as the relevant "To:" header. Those relying on a SIP proxy to scrub certain headers could mistakenly get unexpected and unvalidated data fed to Asterisk. This change fixes so chan_sip only considers SP/HTAB as valid tokens before the colon, making it agree on the headers with other speakers of SIP. ASTERISK-26433 #close AST-2016-009 Change-Id: I78086fbc524ac733b8f7f78cb423c91075fd489b	9 years ago
Joshua Colp	2a415187c5	Merge "res_pjsip/chan_sip: Advertise 'ws' in the SIP URI transport parameter"	9 years ago
zuul	a0c0b1c9cb	Merge "chan_pjsip: fix switching sending codec when asymmetric_rtp_codec=no"	9 years ago
Alexei Gradinari	e5e887be53	chan_pjsip: fix switching sending codec when asymmetric_rtp_codec=no ... The sending codec is switched to the receiving codec and then is switched back to the best native codec on EVERY receiving RTP packets. This is because after call of ast_channel_set_rawwriteformat there is call of ast_set_write_format which calls set_format which sets rawwriteformat to the best native format. This patch adds a new function ast_set_write_format_path which set specific write path on channel and uses this function to switch the sending codec. ASTERISK-26603 #close Change-Id: I5b7d098f8b254ce8f45546e6c36e5d324737f71d	9 years ago
Matt Jordan	0e15760795	res_pjsip/chan_sip: Advertise 'ws' in the SIP URI transport parameter ... Per RFC 7118 5.2, the SIP URI 'transport' parameter should advertise 'ws' when WebSockets are to be used as the transport. This applies to both secure and insecure WebSockets. There were two bugs in Asterisk with respect to this: (1) The most egregious occurs in res_pjsip. There, we advertise 'ws' for insecure websockets and 'wss' for secure websockets. While this would seem to make sense - since 'WS' and 'WSS' are used for the Via Transport parameter - this is not the case for the SIP URI. This patch corrects that by registering the secure websockets with pjproject using the shorthand 'WS', and by returning 'ws' when asked for the transport parameter. Note that in pjproject, it is perfectly valid to have multiple transports use the same shorthand. (2) In chan_sip, we return an upper-case version of the transport 'WS' instead of 'ws'. Since we should be strict in what we send and liberal in what we accept (within reason), this patch lower-cases the transport before appending it to the parameter. ASTERISK-24330 #close Reported by: cervajs, Inaki Baz Castillo Change-Id: Iff77b645f8cc3b7cd35168a6676c26b147f22f42	9 years ago
Michael Kuron	0b588778c0	chan_sip: Fix segfault during module unload ... If a TCP/TLS connection was pending (not accepted and not timed out) during unload of chan_sip, Asterisk would segfault when trying to send a signal to a thread whose thread ID hadn't been recorded yet. This commit fixes that by recording the thread ID before calling the blocking connect() syscall. This was a regression introduced by 776a14386a. The above wasn't enough to fix the segfault, which was now delayed to the point where connect() timed out. Therefore, it was necessary to also remove the SA_RESTART flag from the SIGURG sigaction so that pthread_kill() could be used to interruput the connect() syscall. This was a regression introduced by 5d313f51b9. ASTERISK-26586 #close Change-Id: I76fd9d47d56e4264e2629bce8ec15fecba673e7b	9 years ago
Joshua Colp	d3dba74017	Merge "Implement internal abstraction for iostreams"	9 years ago
Timo Teräs	070a51bf7c	Implement internal abstraction for iostreams ... fopencookie/funclose is a non-standard API and should not be used in portable software. Additionally, the way FILE's fd is used in non-blocking mode is undefined behaviour and cannot be relied on. This introduces internal abstraction for io streams, that allows implementing the desired virtualization of read/write operations with necessary timeout handling. ASTERISK-24515 #close ASTERISK-24517 #close Change-Id: Id916aef418b665ced6a7489aef74908b6e376e85	9 years ago
Igor Goncharovskiy	dfb951817f	Fix closing rtp ports after call finished in chan_unistim. ... Fix ASTERISK-26565 by adding ast_rtp_instance_stop before rtp instance destroy for chan_unistim. Also several fixes for displayed text translation. Change-Id: If42a03eea09bd1633471406bdc829cf98bf6affc	9 years ago
C.J. Collier	73524bde9c	chan_sip: Fix typo and re-wrap surrounding docs ... Correct typo of end-pints to end-points Re-wrap session timer parameter docs to max 80 chars wide; this eases reading on terminals with lower resolution, commonly the case for those with visual impairments. ASTERISK-26573 Change-Id: I22c94459f4bb6b8a2f6713cfd22e87c32f204e6b Signed-off-by: C.J. Collier <cjcollier@linuxfoundation.org>	9 years ago
Kevin Harwell	bf01ff53f8	Revert "chan_sip: Fix lastrtprx always updated" ... This reverts commit 93332cb1d0. Unfortunately, the aforementioned commit caused a regression (incoming calls would eventually disconnect). Thus it is being removed. ASTERISK-26523 #close ASTERISK-25270 Change-Id: Ibf5586adc303073a8eac667a4cbfdb6be184a64d	9 years ago
Joshua Colp	a1bbdabb8e	Merge "chan_sip: add missing account code"	9 years ago
zuul	673964d330	Merge "chan_dahdi: remove by_name support"	9 years ago
Sebastian Gutierrez	0904c1f4cc	chan_sip: add missing account code ... Added missing account to AMI event of sip show peers ASTERISK-26176 #close Change-Id: Ieb6c2c80a838a1b59c82103eba4c63ba238dc482	9 years ago
Grachev Sergey	2526dff94d	chan_sip: Incorrect display option Outbound reg. retry 403 ... If in sip.conf (general section) set option register_retry_403=no, the command "sip show settings" return value: Outbound reg. retry 403:0 If in sip.conf (general section) set option register_retry_403=yes, the command "sip show settings" return value: Outbound reg. retry 403:-1 * In static char "sip show settings" for "Outbound.reg. retry 403" option use AST_CLI_YESNO ASTERISK-26476 #close Change-Id: I3c14272f05f1067bd2aeaa8b3ef9cf8fcb12dcf9	9 years ago