asterisk

Commit Graph

Author	SHA1	Message	Date
Pirmin Walthert	84da8622f6	res_pjsip_logger: use the correct pointer when logging tx_messages to pcap When writing tx messages to pcap files, Asterisk is using the wrong pointer resulting in lots of wasted space. This patch fixes it to use the correct pointer. ASTERISK-28932 #close Change-Id: I5b8253dd59a083a2ca2c81f232f1d14d33c6fd23	5 years ago
George Joseph	edf9471e05	Scope Tracing: Add stubs to allow easier cherry-picking Scope tracing isn't supported in Asterisk 13 due to changes made to logging between 13 and 16 but since the scope tracing macros may be present in the 16, 17 and master branches, those macros are defined here as NOOPs so cherry-picking changes downward to 13 can still be seamless. Change-Id: I0390ce5651374d8f3e06d7620050acb22c5440a2	5 years ago
Joshua C. Colp	e7f7c8c9de	res_rtp_asterisk: Re-order RTP destruction. The destructor for RTP deallocated transport resources before terminating the ICE support. This could result in a crash as the thread handling ICE would access already freed parts of the RTP data. This change re-orders the destruction so that ICE is stopped before destroying things. ASTERISK-28885 Change-Id: Ie71add549f12b6cdea7e9dbf976d1bd1d2fc0bdc	5 years ago
Pirmin Walthert	81b86702fb	res_pjsip_logger.c: correct the return value checks when writing to pcap files fwrite() does return the number of elements written and not the number of bytes. However asterisk is currently comparing the return value to the size of the written element what means that asterisk logs five WARNING messages on every packet written to the pcap file. This patch changes the code to check for the correct value, which will always be 1. ASTERISK-28921 #close Change-Id: I2455032d9cb4c5a500692923f9e2a22e68b08fc2	5 years ago
Asterisk Development Team	07bf7890dc	Update CHANGES and UPGRADE.txt for 13.34.0	5 years ago
Joshua C. Colp	c8cabb08aa	res_pjsip: Use correct pool for storing the contact_user value. When replacing the user portion of the Contact URI the code was using the ephemeral pool instead of the tdata pool. This could cause the Contact user value to become invalid after a period of time. The code will now use the tdata pool which persists for the lifetime of the message instead. ASTERISK-28794 Change-Id: I31e7b958e397cbdaeedd0ebb70bcf8dd2ed3c4d5	5 years ago
Pirmin Walthert	4b7293f44e	res_pjsip_nat.c: remove x-ast-orig-host from request URI and To header While asterisk is filtering out the x-ast-orig-host parameter from the contact on response messages, it is not filtering it out from the request URI and the to header on SIP requests (for example INVITE). ASTERISK-28884 #close Change-Id: Id032b33098a1befea9b243ca994184baecccc59e	5 years ago
Joshua C. Colp	0d86f49f7f	res_sorcery_config: Always reload configuration on errors. When a configuration file in Asterisk is loaded information about it is stored such that on a reload it is not reloaded if nothing has changed. This can be problematic when an error exists in a configuration file in PJSIP since the error will be output at start and not subsequently on reload if the file is unchanged. This change makes it so that if an error is encountered when res_sorcery_config is loading a configuration file a reload will always read in the configuration file, allowing the error to be seen easier. Change-Id: If2e05a017570f1f5f4f49120da09601e9ecdf9ed	5 years ago
Joshua C. Colp	17a878dd99	bridge_softmix: Always remove audio from mixed frame. When receiving audio from a channel we determine if it is talking or silence based on a threshold value. If this threshold is met we always mix the audio into the conference bridge. If this threshold is not met we also mix the audio into the conference bridge UNLESS the drop silence option is enabled. The code that removed the audio from the mixed frame assumed that it was always not present if it did not meet the threshold to be considered talking. This is incorrect. If it has been stated that the audio was mixed into the mixed frame then it has been mixed into the mixed frame. By not removing audio that was considered non-talking it was possible for a channel to receive a slight echo of audio of itself at times. This change ensures that the audio is always removed from the mixed frame going back to the channel so it no longer receives the slight echo. ASTERISK-28898 Change-Id: I7b1b582cc1bcdb318ecc60c9d2e3d87ae31d55cb	5 years ago
Joshua C. Colp	4fba0f678b	res_pjsip_logger: Expand functionality to improve logging. The PJSIP packet logger now has the following CLI commands: pjsip set logger pcap <filename> When used this will create a pcap file containing the incoming and outgoing SIP packets, in unencrypted form. pjsip set logger verbose <on / off> This allows you to toggle logging to verbose on and off. pjsip set logger host <IP/subnet mask> add This allows you to add an additional IP address or subnet mask to logging, allowing you to log multiple instead of just a single IP address or all traffic. The normal "pjsip set logger host" CLI command has also been expanded to allow subnet masks as well. ASTERISK-28895 Change-Id: If5859161a72b0d7dd2d1f92d45bed88e0cd07d0e	5 years ago
sungtae kim	5de3f22ee0	res_rtp_asterisk.c: Fixed memory leak Added freeifaddrs() for memory releasing. ASTERISK-28904 Change-Id: I109403866e85a30659351946903a679de9727a8f	5 years ago
Guido Falsi	683f0b269e	pjproject: Fix race condition when building with parallel make Pjproject makefiles miss some dependencies which can cause race conditions when building with parallel make processes. This patch adds such dependencies correctly. ASTERISK-28879 #close Reported-by: Dmitry Wagin <dmitry.wagin@ya.ru> Change-Id: Ie1b0dc365dafe4a84c5248097fe8d73804043c22	5 years ago
Roger James	3df1b65dd0	res_pjsip_history.c: Fix to stop SIGSEGV when IPv6 addresses are encountered. Changed source and destination address fields in struct pjsip_history_entry so that they are long enough to hold an IPv6 address. ASTERISK-28854 Change-Id: Id65bb9aa961e9ecbcb500815e18170f774e34d3e	5 years ago
traud	377f5b6992	tcptls: Fix notice when TLS is enabled but not supported. ASTERISK-28797 Change-Id: Iab364a2c2519fd9d11d1c28293fda43d61b64c28	5 years ago
traud	103b789e56	app_osplookup: Avoid a format truncation. Ensure that output buffers for the osp_convert_inout function have sufficient space for additional data such as brackets and ports. ASTERISK-28804 Change-Id: Ie54c8241ff0cc653910539c2db00ff2a4869750b	5 years ago
Pirmin Walthert	146e7c54f1	app.c: make sure that no non-async-signal-safe syscalls are used after fork before exec Posix does only allow async-signal-safe syscalls after fork before exec. As asterisk ignores this, functions like TrySystem or System sometimes end up in a deadlocked child process. The patch prevents the use of non-async-signal-safe syscalls. ASTERISK-28776 Change-Id: Idc76365c0592ee3f3b3bd72a4f48f7a098978e8e	5 years ago
Nathan Bruning	10292dc663	app_queue: track masquerades in app_queue to avoid leaked stasis subscriptions Add a new "masquarade" channel event, and use it in app_queue to track unique id's. Testcase is submitted as https://gerrit.asterisk.org/c/testsuite/+/14210 ASTERISK-28829 #close ASTERISK-25844 #close Change-Id: Ifc5f9f9fd70903f3c6e49738d3bc632b085d2df6	5 years ago
Guido Falsi	77837059ac	pjproject: Remove bashism from configure.m4 script The configure.m4 script for pjproject contains some += syntax, which is specific to bash, replacing it with string substitutions makes the script compatible with traditional Bourne shells. ASTERISK-28866 #close Reported-by: Christoph Moench-Tegeder <cmt@FreeBSD.org> Change-Id: I382a78160e028044598b7da83ec7e1ff42b91c05	5 years ago
Jaco Kroon	f5a1a91b03	Remove #include <sys/cdefs.h> These are not provided by standards, and as a result causes failure to compile on musl. https://wiki.musl-libc.org/faq.html#Q:-When-compiling-something-against-musl,-I-get-error-messages-about-%3Ccode%3Esys/cdefs.h%3C/code%3E Change-Id: I6a357cefd106c72cfecafd898638f6b5692c2e05	5 years ago
George Joseph	2d1713395b	app_voicemail: Add workaround for a gcc 10 issue with -Wrestrict The gcc 10 -Wrestrict option was causing "overlap" errors when snprintf was copying one char[256] structure member to another char[256] member in the same structure. Using ast_alloca instead of declaring the structure inline solves the issue. Here's a link to the "enhancement": https://gcc.gnu.org/legacy-ml/gcc-patches/2019-10/msg00570.html We may follow up with a gcc bug report. Change-Id: Ie0099adcb0a9727bd9aa99e024dd912a67eaf534	5 years ago
Alexander Traud	18cc090761	core_local: Local calls are always secure. In a Dialplan, the channel drivers 'chan_sip' and 'chan_iax2' support the channel items 'secure_bridge_media' and 'secure_bridge_signaling'. That way, a channel can be forced to use encryption even if not specified in its configuration. However, when the Local Proxy kicks in, for example, in case of a forwarding (SIP status 302), Local Proxy stated it does not know those items. Consequently, such a call could not be proxied how clever your Dialplan was. Because local calls within Asterisk are always secure, Local Proxy accepts such a request now. ASTERISK-22920 Change-Id: I4c143bb70f686790953cc04c5a4b810bbb03636c	5 years ago
Joshua C. Colp	35f7e24ebc	pjsip: Increase maximum ICE candidate count. In practice it has been seen that some users come close to our maximum ICE candidate count of 32. In case people have gone over this increases the count to 64, giving ample room. ASTERISK-28859 Change-Id: I35cd68948ec0ada86c14eb53092cdaf8b62996cf	5 years ago
Guido Falsi	5c74f22977	res_rtp_asterisk: Protect access to nochecksums with #ifdef Recently code accessing nochecksums variable has been added without including #ifdef SO_NO_CHECK protection, while the variable is created only when such constant is defined. ASTERISK-28852 #close Change-Id: I381718893b80599ab8635f2b594a10c1000d595e	5 years ago
Peter Turczak	7714bbfeaa	chan_mobile: Add smoother to make SIP/RTP endpoints happy. In contrast to RFC 3551, section 4.2, several SIP/RTP clients misbehave severly (up to crashing). This patch adds another smoother for the audio received via bt. Therefore the audio frames sent to the core will be CHANNEL_FRAME_SIZE. ASTERISK-28832 #close Change-Id: Ic5f9e2f35868ae59cc9356afbd1388b779a1267f	5 years ago
Asterisk Development Team	074d766fb3	Update CHANGES and UPGRADE.txt for 13.33.0	5 years ago
sungtae kim	f2d9d31ef5	res_ari_channels: Fixed endpoint 80 characters limit Fixed it to copy the entire string from the requested endpoint body except tech-prefix. ASTERISK-28847 Change-Id: I91b5f6708a1200363f3267b847dd6a0915222c25	5 years ago
DanielYK	e4508fcc6e	res_pjsip: Fixed format of IPv6 addresses for external media addresses ASTERISK-28835 Change-Id: I66289afd164c5cdd6c5caa39e79d629a467e7a26	5 years ago
Alexander Traud	91ce91dbc6	chan_sip: externhost/externaddr with non-default TCP/TLS ports. ASTERISK-28372 Reported by: Anton Satskiy ASTERISK-24428 Reported by: sstream Change-Id: I2b7432a9bf3b09dc8515297ff955636db7a6224c	5 years ago
Alexander Traud	f91dc188dd	curl: Add build-time dependency. ASTERISK-28838 Change-Id: I34724e799e1ffaf723eb2c358abe8934dbadcd52	5 years ago
Alexander Traud	bb5c0f30c4	res_pjsip_refer: Add build-time dependency. ASTERISK-28838 Change-Id: Ic693c3f464e35ec0db242afdb0a1415806af4e25	5 years ago
Alexander Traud	7124850d3c	app_getcpeid: Add build-time dependency. ASTERISK-28838 Change-Id: I68b78e7e4718be82507247433127ce3992a5ba96	5 years ago
Alexander Traud	59f2ad2019	res_pjsip: Add build-time dependency. ASTERISK-28838 Change-Id: Icb08304744ae3f34dce6ccb76f94379b8382a074	5 years ago
Alexander Traud	1edf0047a1	pjproject_bundled: Honor --without-pjproject. ASTERISK-28837 Change-Id: Id057324912a3cfe6f50af372675626bb515907d9	5 years ago
Alexander Traud	dac20ce461	chan_sip: DiffServ/ToS not only on UDP but also on TCP and TLS sockets. ASTERISK-27195 Reported by: Joshua Roys Change-Id: I6e72ecb874200dec7a3865c7babaf5ac0d3101de	5 years ago
Alexander Traud	65598af3ff	install_prereq: Add libcap for high bits in DiffServ/ToS. works automatically; see Mantis 7047 (now ASTERISK-6863) Change-Id: I27d2c109180bd857b6757fd532de48eddb78aee6	5 years ago
Alexander Traud	47e887dbf6	BuildSystem: Only if found LibPRI, check its optional parts. Change-Id: If8445f899ee4ce0c606c484943d4ce0c8e43b5da	5 years ago
Alexander Traud	007a6ab05a	BuildSystem: Only if found external PJProject, check its optional parts. Change-Id: I11d5693d25c166c99d8cebffc16184d58f6362de	5 years ago
bernard merindol	43dfd0d300	res_rtp_asterisk.c: Check for first DTMF having timestamp set to 0 When the first DTMF receive in RF2833 codec have TimeStamp at 0 is not processed. ASTERISK-28812 Change-Id: I3196803a062dd2daee4938c9a778c3810cb7e504	5 years ago
Jean Aunis	947a6e8674	func_volume: Accept decimal number as argument Allow voice volume to be multiplied or divided by a floating point number. ASTERISK-28813 Change-Id: I5b42b890ec4e1f6b0b3400cb44ff16522b021c8c	5 years ago
Jaco Kroon	ce1213e72f	res_rtp_asterisk: iterate all local addresses looking to populate ICE. By using pjproject to give us a list of candidates, and then filtering, if the host has >32 addresses configured, then it is possible that we end up filtering out all 32 of those, and ending up with no candidates at all. Instead, get getifaddrs (which pjsip is using underlying anyway) to retrieve all local addresses, and iterate those, adding the first 32 addresses not excluded by the ICE ACL. In our setup at any point in time We've got between 6 and 328 addresses on any given system. The lower limit is the lower limit but the upper limit is growing on a near daily basis currently. Change-Id: I109eaffc3e2b432f00bf958e3caa0f38cacb4edb Signed-off-by: Jaco Kroon <jaco@uls.co.za>	5 years ago
Alexander Traud	64427c667b	pjproject_bundled: Repair ./configure --with-ssl without ARG. ASTERISK-28758 Reported by: Patrick Wakano Reported by: Dmitriy Serov Change-Id: Ifb6b85c559d116739af00bc48d1f547caa85efac	5 years ago
Jaco Kroon	a142a1455c	res_pjsip: document legal dtls_verify endpoint options. Change-Id: I7fa7c5c8a7ddb0bd525982f58bff3264ebbd9a1b	5 years ago
Alexander Traud	23211669a7	BuildSystem: Search for Python/C API when possibly needed only. The Python/C API is used only if the Test Framework was enabled in Asterisk 'make menuselect'. The Test Framework is available only if the Developer Mode was enabled in Asterisk './configure --enable-dev-mode'. And that Python/C API is used only if the PJProject was found and not disabled in Asterisk; the user did not go for './configure --without-pjproject'. Furthermore, because version 2 of that Python/C API is required (currently) and because some platforms do not offer a generic version 2, the script searches for 2.7 explicitly as well. To avoid version mismatch between the Python/C API and the Python environment, the script searches for the latter in the same versions, in the same the order as well. Because this Python/C API is just for (some) Asterisk contributors, the script also goes for the Python 3 environment as a last resort for all other Asterisk users. This allows 'make full' even on minimal installations of Ubuntu 18.04 LTS and newer. Because the Python/C API is Asterisk contributor specific, the Python packages are removed from the script './contrib/scripts/install_prereq' as this script is intended for Asterisk users. Asterisk contributors have to install much more packages in any case, like: sudo apt install autoconf automake git git-review python2.7-dev ASTERISK-28824 ASTERISK-27717 Change-Id: Id46d357e18869f64dcc217b8fdba821b63eeb876	5 years ago
traud	2bcc3df533	chan_sip: TCP/TLS client without server. It is possible to configure a TCP/TLS client without having a TCP/TLS server. In that case, no error or warning was printed but the headers Contact and Via in SIP REGISTER were "(null)". ASTERISK-28798 Change-Id: I387ca5cb6a65f1eb675a29c5e41df8ec6c242ab2	5 years ago
Alexander Traud	a0d5f7ccff	_pjsua: Build even with Clang. Change-Id: Iebf7687613aa0295ea3c82256460b337f1595be2	5 years ago
Alexander Traud	92efcc9047	res_rtp_asterisk: Build without PJProject. Change-Id: Ifc5059cd867e77b9c92ed9f4b895a9a91200d3ec	5 years ago
Kevin Harwell	b2e7fc6972	chan_pjsip: digit_begin - constant DTMF tone if RTP is not setup yet If chan_pjsip is configured for DTMF_RFC_4733, and the core triggers a digit begin before media, or rtp has been setup then it's possible the outgoing channel will hear a constant DTMF tone upon answering. This happens because when there is no media, or rtp chan_pjsip notifies the core to initiate inband DTMF. However, upon digit end if media, and rtp become available then chan_pjsip does not notify the core to stop inband DTMF. Thus the tone continues playing. This patch makes it so chan_pjsip only notifies the core to start inband DTMF in only the required cases. Now if there is no media, or rtp availabe upon digit begin chan_pjsip does nothing, but tells the core it handled it. ASTERISK-28817 #close Change-Id: I0dbea9fff444a2595fb18c64b89653e90d2f6eb5	5 years ago
traud	9b259c6514	BuildSystem: Remove doc/tex and doc/pdf leftovers. Furthermore, the nowhere used compress is removed. ASTERISK-28816 Change-Id: I77daab80cfabb56d51c3ea6b1d14bd9b9fbc577c	5 years ago
Alexander Traud	785225c5a0	BuildSystem: Allow space in path. ASTERISK-28818 Change-Id: Ib7f246896457d9e3b14d7f5199136d6545ce0b6f	5 years ago
traud	9082f90217	stasis: Avoid always true warnings with clang. ASTERISK-28801 Change-Id: I63ba125226b9fe8a018bd28825c877603eb8f398	5 years ago

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