asterisk

Commit Graph

Author	SHA1	Message	Date
Joshua Colp	a2343aa200	Merge "pbx_lua: On configuration errors report module load failure instead of decline."	9 years ago
Joshua Colp	2679f80d3c	Merge "res_rtp: Fix regression when IPv6 is not available."	9 years ago
Joshua Colp	29596f1538	Merge "res_calendar_caldav: Add support reading gmail calendar"	9 years ago
Joshua Colp	02588a1aab	Merge "Frame deferral: Re-queue deferred frames one-at-a-time."	9 years ago
zuul	68fc035795	Merge "OpenSSL 1.1.0 support"	9 years ago
Tzafrir Cohen	26c8552fff	OpenSSL 1.1.0 support OpenSSL 1.1.0 includes some major changes in the interface. See https://wiki.openssl.org/index.php/1.1_API_Changes . Status: Right now there are still a few deprecation notes with OpenSSL 1.1.0. But it's a start. Changes: * CRYPTO_LOCK is no longer available. Replace it with its value for now. I don't completely understand what it is used for there. * Remove several functions from libasteriskssl that seem to no longer be needed. * Structures have become opaque and are accesses with accessors. * ERR_remove_thread_state() no longer needed. * SSLv2 code now could no longer be used in 1.1. ASTERISK-26109 #close Change-Id: I5e29d477d486ca29b6aae0dc2f5dff960c1cb82b	9 years ago
Guido Falsi	75230f4c01	res_rtp: Fix regression when IPv6 is not available. The latest Release candidate fails to create RTP streams when IPv6 is not available. Due to the changes made in September the ast_sockaddr structure passed around to create these streams is always of AF_INET6 type, causing failure when used for IPv4. This patch adds a utility function to check for availability of IPv6 and applies such check at startup to determine how to create the ast_sockaddr structures. ASTERISK-26617 #close Change-Id: I627a4e91795e821111e1cda523f083a40d0e0c3e	9 years ago
Richard Mudgett	1dfa11b65c	PJPROJECT logging: Made easier to get available logging levels. Use of the new logging is as simple as issuing the new CLI command or setting the new pjproject.conf option. Other options that can affect the logging are how you have the pjproject log levels mapped to Asterisk log types in pjproject.conf and if you have configured Asterisk to log the DEBUG type messages. Altering the pjproject.conf level mapping shouldn't be necessary for most installations as the default mapping is sensible. Configuring Asterisk to log the DEBUG message type is standard practice for collecting debug information. * Added CLI "pjproject set log level" command to dynamically adjust the maximum pjproject log message level. * Added CLI "pjproject show log level" command to see the currently set maximum pjproject log message level. * Added pjproject.conf startup section "log_level" option to set the initial maximum pjproject log message level so all messages could be captured from initialization. * Set PJ_LOG_MAX_LEVEL to 6 to compile in all defined logging levels into bundled pjproject. Pjproject will use the currently set run time log level to determine if a log message is generated just like Asterisk verbose and debug logging levels. * In log_forwarder(), made always log enabled and mapped pjproject log messages. DEBUG mapped log messages are no longer gated by the current Asterisk debug logging level. * Removed RAII_VAR() from res_pjproject.c:get_log_level(). ASTERISK-26630 #close Change-Id: I6dca12979f482ffb0450aaf58db0fe0f6d2e5389	9 years ago
Mark Michelson	621d886ca7	Frame deferral: Re-queue deferred frames one-at-a-time. The recent change that made frame deferral into an API had a behavior change to it. When frame deferral was completed, we would take all of the deferred frames and queue them all onto the channel in one call to ast_queue_frame_head(). Before frame deferral was API-ized, places that performed manual frame deferral would actually take each deferred frame and queue them onto the channel. This change in behavior caused the confbridge_recording test to start failing consistently. Without going too crazily deep into the details, a channel was getting "stuck" in an ast_safe_sleep(). An AMI redirect was attempting to break it out of the sleep, but because there were more frames in the channel read queue than expected, the channel ended up being unable to break from its sleep loop. By restoring the behavior of individual frame queuing after deferral, the test starts passing again. Note, this points to a potential underlying issue pointing to an "unbalance" that can occur when queuing multiple frames at once, and so a follow-up issue is being created to investigate that possibility. Change-Id: Ied5dacacda06d343dea751ed5814a03364fe5a7d	9 years ago
zuul	a0c0b1c9cb	Merge "chan_pjsip: fix switching sending codec when asymmetric_rtp_codec=no"	9 years ago
Joshua Colp	bd20127e64	Merge "chan_sip: Fix segfault during module unload"	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
David Kerr	ddc951060a	app_originate: Add option to execute gosub prior to dial Issue/patch ASTERISK-26587 was inspired by issue ASTERISK-22992 that requested ability to add callerid into app_originate. Comments in that issue suggested that it was better solved by adding an option to gosub prior to originating the call. The attached patch implements this much like app_dial with two options one to gosub on the originating channel and one to gosub on the newly created channel and behaves just like app_dial. I have tested this patch by adding callerid info to the new channel and also SIPAddHeader (to e.g. add header to force auto answer) and confirmed it works. Have also tested both 'exten' and 'app' versions of app_originate. Opened by: dkerr Patch by: dkerr Change-Id: I36abc39b58567ffcab4a636ea196ef48be234c57	9 years ago
Eduardo S. Libardi	0e214c4932	res_calendar_caldav: Add support reading gmail calendar The response from gmail calendar includes the string name "caldav:calendar-data". res_calendar_caldav implements the example included in RFC 4791: string "C:calendar-data". When reading the calendar, res_calendar_caldav compare the string and if does not match just discards the event. This commit compares the response to both strings, successfully loading gmail calendar events. Writing to gmail calendar is working prior to this fix. ASTERISK-26624 Reported by: Eduardo S. Libardi Change-Id: Ia1eef10552ae616efb645d390f5ffe81260d7d4a	9 years ago
Joshua Colp	fdf4355bd0	Merge "res/res_pjsip: Fix documentation whitespace issues"	9 years ago
zuul	350a9a6bd4	Merge "build_tools: Fix download_externals to handle certified branches"	9 years ago
Matt Jordan	a3f48be0da	res/res_pjsip: Fix documentation whitespace issues Tabs > Spaces. Change-Id: If1e43a71822615a898e958e0f8b2e882606f0bd0	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
Joshua Colp	606ab90c25	Merge "autoconf: more variants for OSARCH linux-gnu"	9 years ago
George Joseph	8a68289766	build_tools: Fix download_externals to handle certified branches download_externals wasn't handling the "certified/13.x" version correctly. Change-Id: I124d195bb117ca36fd7bf1150c630f3b474a9d9a	9 years ago
Joshua Colp	acdae08cf7	Merge "codec_dahdi: Fix poll.h include."	9 years ago
Joshua Colp	c9cc64b911	Merge "ast_format: Adds an identifier for interleaved audio formats to the ast_format"	9 years ago
Joshua Colp	e3dae763ee	iostream: Move include of asterisk.h The asterisk.h header file needs to be included first or else some things go awry, such as: implicit declaration of function 'vasprintf' Change-Id: I981dc2a77a1ba791888e4f1726644d4656c0407c	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
Dennis Guse	ead773f801	pbx_lua: On configuration errors report module load failure instead of decline. Switched from AST_MODULE_LOAD_DECLINE to AST_MODULE_LOAD_FAILURE. Therefore, if pbx_lua fails to load and pbx_lua is marked as required, Asterisk exits as expected. If extensions.lua cannot be opened, AST_MODULE_LOAD_DECLINE is reported. Change-Id: I8e5a0037e69b41743db60c568541ebb2f52a7a8f	9 years ago
gestoip2	d9b24cce0a	res_rtp_asterisk: RTT miscalculation in RTCP When retrieving RTCP stats for PJSIP channels, RTT values are unreliable. RTT calculation is correct, but the data representation isn't. RTT is represented by a 32-bit fixed-point number with the integer part in the first 16 bits and the fractional part in the last 16 bits. In order to get the RTT value, the fractional part is miscalculated, there is an unnecessary 16 bit shift that causes overflow. Besides this there is another mistake, when transforming the integer value to the fixed point fractional part via bitwise operation, that loses precision. * RTT fractional part is no longer shifted, avoiding overflow. * RTT fractional part is transformed to its fixed-point value more precisely. * Fixed timeval2ntp() and ntp2timeval() second fraction conversions. * Fixed NTP timestamp report logging. The usec was inexplicably multiplied by 4096. ASTERISK-26566 #close Reported by Hector Royo Concepcion Change-Id: Ie09bdabfee75afb3f1b8ddfd963e5219ada3b96f	9 years ago
Michael Kuron	635b0a0a55	tcptls: Use new certificate upon sip reload Previously, a TLS server socket would only be restarted upon sip reload if the bind address had changed. This commit adds checking for changes to TLS parameters like certificate, ciphers, etc. so they get picked up without requiring a reload of the entire chan_sip module. This does not affect open connections in any way, but new connections will use the new TLS parameters. The changes also apply to HTTP and Manager. ASTERISK-26604 #close Change-Id: I169e86cefc6dcd627c915134015a6a1ab1aadbe6	9 years ago
Joshua Colp	038158bf7b	Merge "addons/chan_mobile: do not use strerror_r"	9 years ago
Joshua Colp	b1f7cc4223	Merge "Add support for older name resolving version libraries like openBSD"	9 years ago
George Joseph	abae3dc36e	pjproject_bundled: Use $(LIB_RT) for link of libasteriskpj libasteriskpj was hard coded to use -lrt but librt is linux specific so we now use the LIB_RT variable which gets set by configure. Change-Id: I41148884517e3031f7675a413d524c86e8614694	9 years ago
zuul	aa9d91c290	Merge "pjproject_bundled: Improve reliability of pjproject download"	9 years ago
Joshua Colp	84e508c999	Merge "main/app.c: Transmit Silence on ControlPlayback pause"	9 years ago
zuul	120a4999f0	Merge "Add support for building RADIUS with radcli"	9 years ago
snuffy	b546497fe0	Add support for older name resolving version libraries like openBSD Fix support of OS's like openBSD that use an older nameser.h, this change reverts the defines to the older style which on other systems is found in nameser_compat.h Tested on openBSD 6.0, Debian 8 ASTERISK-26608 #close Change-Id: Iffb36caab8c5aa9dece0ce2d009041f7b56cc86a	9 years ago
Mark Michelson	7a8d6bc81b	Bump ARI version to 2.0.0 In order to not have version number overlap between different versions of Asterisk, each new major version of Asterisk will mean we also bump the ARI major version number. This particular change does NOT introduce any known breaking changes to ARI. For discussion relating to this topice, see: http://lists.digium.com/pipermail/asterisk-dev/2016-November/075964.html Change-Id: I712ee0df177a8fe1252da2bc029705268b97b665	9 years ago
zuul	782985631e	Merge "build: Various OpenBSD issues"	9 years ago
George Joseph	d3f070c7a2	pjproject_bundled: Improve reliability of pjproject download The download process now has a timeout which will cause wget to retry if it stops retrieving data for 5 seconds and fetch and curl to timeout if the whole retrieval take smore than 30 seconds. If the tarball retrieval works, the MD5SUM file is retrieved from the downloads site and the md5 checksum is verified. If either the tarball retrieval or MD5SUM retrieval fails, or the checksums don't match, the entire process is retried once. If it fails again, any incomplete tarball is deleted. .DELETE_ON_ERROR: was also added to the Makefile. Not only does this delete the tarball on failure, it till also delete corrupted library files from the pjproject source directory should they fail to build correctly. Tested all the way back to FreeBSD 9, CentOS 6, Debian 6 and Ubuntu 14. Change-Id: Iea7d33b96a31622ab1b6e54baebaf271959514e1	9 years ago
Joshua Colp	98b3b500dc	Merge "manager: update minor version"	9 years ago
misha	e822a50f86	main/app.c: Transmit Silence on ControlPlayback pause ASTERISK-26562 #close Change-Id: Ie6cb0ffc2b8c775639ce7784fe96f4ea00cfa2f8	9 years ago
Joshua Colp	d3dba74017	Merge "Implement internal abstraction for iostreams"	9 years ago
Mark Michelson	d670ea6297	manager: update minor version Based on bridge video AMI event changes, bump the minor version of AMI. Change-Id: Idf84507354170400813cda780906c94c9f1b60b4	9 years ago
Timo Teräs	349e08cb48	codec_dahdi: Fix poll.h include. POSIX defines poll.h. sys/poll.h should not be used as it is c-library internal header which may or may not exist. Notably in musl including sys/poll.h generates warning of being incorrect. Change-Id: Ib318c1c7142a737bcf3caa4d8d72560bebe39252	9 years ago
Joshua Colp	ed143a3b7c	Merge "res_pjsip_outbound_authenticator_digest.c: Fix memory pool leak."	9 years ago
Joshua Colp	09d1958448	Merge "codec_opus: Fix warning when Opus negotiated but codec_opus not loaded."	9 years ago
zuul	8fbdedb36e	Merge "res_format_attr_opus: Fix fmtp generation."	9 years ago
George Joseph	935f5d003b	build: Various OpenBSD issues OpenBSD's 'find' doesn't take the -delete argument so you have to pipe through 'xargs rm -rf'. 'echo -e' doesn't like \t starting a line. It just prints 't' which causes the libasteriskpj.exports file to be garbage. They were just cosmetic so they were removed. librt doesn't exist so the link of libasteriskpj.so fails. It's not actually needed for linux anyway so -lrt was removed from the link. res_rtp_asterisk was failing to load because of an undefined DTLS_method. '\|\| defined(LIBRESSL_VERSION_NUMBER)' was added to the #if so DTLSv1_method is used instead. ASTERISK-26608 Change-Id: I926ec95b0b69633231e3ad1d6e803b977272c49c	9 years ago
George Joseph	97b2ba472d	Merge "channel: Fix issues in hangup scenarios caused by frame deferral"	9 years ago
George Joseph	b8e91bb9cc	Merge "Revert "Revert "channel: Use frame deferral API for safe sleep."""	9 years ago
George Joseph	99e97154bb	Merge "Revert "Revert "autoservice: Use frame deferral API"""	9 years ago
George Joseph	013e7dd4a6	Merge "Revert "Revert "AGI: Only defer frames when in an interception routine."""	9 years ago

1 2 3 4 5 ...

28847 Commits (740ca862e42e6dfbd747257c84f22ff47eb8ff14) All Branches Search

28847 Commits (740ca862e42e6dfbd747257c84f22ff47eb8ff14)

All Branches