28099 Commits (f16ab192922b6532e653bde41ceb7dd33e73e04e)

 

All Branches   

Search

Author	SHA1	Message	Date
Joshua Colp	f16ab19292	odbc: Remove options that are no longer applicable. ... The pooling, shared_connection, limit, and idlecheck options are no longer used in res_odbc. ASTERISK-26389 Change-Id: I2fde7b467d01f9d1c82cc0a339bb4f7e1dd6bbe6	10 years ago
zuul	a6b05e6371	Merge "asterisk.c: Non-root users also get the astcanary after core restart." into 13	10 years ago
Joshua Colp	77fafd1534	Merge "sd_notify (systemd status notifications) support" into 13	10 years ago
zuul	177557bc00	Merge "res_pjsip_multihomed: Change Contact port to listening port." into 13	10 years ago
Tzafrir Cohen	36092ee3a0	sd_notify (systemd status notifications) support ... sd_notify() is used to notify systemd of changes to the status of the process. This allows the systemd daemon to know when the process finished loading (and thus only start another program after Asterisk has finished loading). To use this, use a systemd unit with 'Type=notify' for Asterisk. This commit also adds the function ast_sd_notify(), a wrapper around sd_notify that does nothing if not built with systemd support. Also adds support for libsystemd detection in the configure script. Change-Id: Ied6a59dafd5ef331c5c7ae8f3ccd2dfc94be7811 (cherry picked from commit 07b95f7c65)	10 years ago
Walter Doekes	9372d32100	asterisk.c: Non-root users also get the astcanary after core restart. ... Without this change, a 'core restart' would kill the astcanary forever if you're not running as root. Both with and without this patch, the scheduling priority was still SCHED_RR after restart. Additionally, the astcanary is now spawned if you start with high priority and Asterisk doesn't get a chance to lower it. For example through: `chrt -r 10 sudo -u asterisk asterisk -c` Also reap killed astcanary processes on core restart. ASTERISK-26352 #close Change-Id: Iacb49f26491a0717084ad46ed96b0bea5f627a55	10 years ago
zuul	34461b89ac	Merge "asterisk.c: When astcanary dies on linux, reset priority on all threads." into 13	10 years ago
zuul	0bd4398b8a	Merge "Fix showing of swap details when sysinfo() is available" into 13	10 years ago
zuul	9383beb073	Merge "res_config_odbc.c: Fix buffer size limitation creating invalid SQL." into 13	10 years ago
Walter Doekes	e96448e991	asterisk.c: When astcanary dies on linux, reset priority on all threads. ... Previously only the canary checking thread itself had its priority set to SCHED_OTHER. Now all threads are traversed and adjusted. ASTERISK-19867 #close Reported by: Xavier Hienne Change-Id: Ie0dd02a3ec42f66a78303e9c1aac28f7ed9aae39	10 years ago
Timo Teräs	01884a7af6	Fix showing of swap details when sysinfo() is available ... If sysinfo() is available, but not sysctl() or swapctl() the printing code for swap buffer sizes is incorrectly omitted. The above condition happens with musl c-library. Fix #if rule to consider defined(HAVE_SYSINFO). And also remove the redundant || defined(HAVE_SYSCTL) which was incorrectly there to start with. Now swap information is displayed only if an actual libc function to get it is available. This also fixes warnings previously seen with musl libc: [CC] asterisk.c -> asterisk.o asterisk.c: In function 'handle_show_sysinfo': asterisk.c:773:6: warning: variable 'totalswap' set but not used [-Wunused-but-set-variable] int totalswap = 0; ^~~~~~~~~ asterisk.c:770:11: warning: variable 'freeswap' set but not used [-Wunused-but-set-variable] uint64_t freeswap = 0; ^~~~~~~~ Change-Id: I1fb21dad8f27e416c60f138c6f2bff03fb626eca	10 years ago
Richard Mudgett	cdbad152c7	res_config_odbc.c: Fix buffer size limitation creating invalid SQL. ... Creating ODBC SQL queries resulted in queries too large to fit into the supplied buffer. The resulting truncated buffer contained an invalid SQL query. * Made SQL query generation code use a thread storage buffer that can increase in size as needed. * Fixed bad multi-line warning messages. ASTERISK-26263 #close Reported by: Jeppe Ryskov Larsen Change-Id: I23f3cdd43c2dac80bed3ded4dd77d18cb17f21ae	10 years ago
Joshua Colp	449719be00	res_pjsip_multihomed: Change Contact port to listening port. ... The res_pjsip_multihomed module determines what interface and transport a request is going out on and updates the SIP message accordingly with the address information. This currently incorrectly updates the Contact header for connectionful protocols to the ephemeral connection port, instead of the bound address for the listening socket which can actually accept the connection back. If the remote side attempts to connect back on the epehemeral port it will fail. This change makes it so the port is updated to the bound port on connectionful protocols and is maintained on UDP (as there can be multiple of those). ASTERISK-26374 #close Change-Id: I50f8dab65b9f75117d73ba5f6bbcf6c9871854ab	10 years ago
George Joseph	4d64b176eb	pjproject_bundled: Prevent SERVFAIL from marking name server bad ... A name server that returns "Server Failure" is indicating only that the server couldn't process that particular request. We should NOT assume that the name server is incapable of serving other requests. Here's the scenario we've been encountering... * 2 local name servers configured in resolv.conf. * An OPTIONS request causes a request for A and AAAA records to go out to both nameservers. * The A responses both come back successfully resolved. * Because of an issue at some upstream nameserver, the AAAA responses for that particular query come back as "SERVFAIL" from both local name servers. * Both local servers are marked as bad and no further queries can be sent until the 60 second ttl expires. Only previously cached results can be used. * In this case, 60 seconds is just enough time for another OPTIONS request to go out to the same host so the cycle repeats. We could set the bad ttl really low but that also affects REFUSED and NOTAUTH which probably DO signal a real server issue. Besides, even a really low bad ttl would be an issue on a pbx. Although we use our own resolver in 14 and master and don't have this issue there, Teluu has merged this patch upstream so it's appropriate to cherry-pick to 14 and master to keep pjproject consistent. Change-Id: Ie03ba902288e274aff23f9b9bb2786e1e8be09e0	10 years ago
zuul	11d05fc369	Merge "res_pjsip_transport_management: Convert time in log message to seconds." into 13	10 years ago
zuul	1ddaa825ec	Merge "chan_sip: Fix session timeout on retransmit of non-UDP packets" into 13	10 years ago
zuul	f0baa12538	Merge "rtp: Preserve timestamps on video frames." into 13	10 years ago
zuul	139d86c8ab	Merge "sip_to_pjsip.py: Map legacy_useroption_parsing." into 13	10 years ago
Joshua Colp	1cac856e17	rtp: Preserve timestamps on video frames. ... Currently when receiving video over RTP we store only a calculated samples on the frame. When starting the video it can take some time for this calculation to actually yield a value as it requires constant changing timestamps. As well if a video frame passes over multiple RTP packets this calculation will fail as the timestamp is the same as the previous RTP packet and the number of samples calculated will be 0. This change preserves the timestamp on the frame and allows it to pass through the core. When sending the video this timestamp is used instead of a new one being calculated. ASTERISK-26367 #close Change-Id: Iba8179fb5c14c9443aee4baf670d2185da3ecfbd	10 years ago
zuul	a5f7b07579	Merge "res_pjsip: Add ignore_uri_user_options option." into 13	10 years ago
Joshua Colp	9df4056d70	res_pjsip_transport_management: Convert time in log message to seconds. ... ASTERISK-26375 #close Change-Id: I46496af5cae41413e76d44d2068a7431279f09dc	10 years ago
Steve Davies	98e42cc662	chan_sip: Fix session timeout on retransmit of non-UDP packets ... Change-Id I1cd33453c77c56c8e1394cd60a6f17bb61c1d957 Enable Session-Timers for SIP over TCP (and TLS) also disables SIP retransmits in chan_sip for non-UDP connections, allowing the TCP layer to handle the retransmits. Unfortunately, this caused sessions to be terminated with a retransmit timeout becasue it stopped at the point of the first retrans call. This patch waits for the 64*T1 timer to expire instead. ASTERISK-19968 Change-Id: I844f26801aada10bc94e9bebe6e151f0a8443204	10 years ago
zuul	efba4a7b9c	Merge "chan_sip: Allow target refresh (Contact update) on re-INVITE." into 13	10 years ago
zuul	3d7cbaa675	Merge "res_pjsip_messaging.c: Misc cleanups and fixes." into 13	10 years ago
Richard Mudgett	0388882cdb	app_queue: Fix CLI "queue show" and AMI Queues action output truncation. ... The output of CLI "queue show" and AMI Queues action is truncated and "failed to extend from 240 to 327" messages are generated if the queue member and interface names are lengthy. * Increase the string buffer size from 240 to 512 in order to accommodate for more information fields added to the output since v1.8. ASTERISK-26360 #close Reported by: Richard Mudgett Change-Id: Id99c03cf5362453b80491a4b3b0434cb67aa966d	10 years ago
zuul	c833c1fda1	Merge "contrib: Let safe_asterisk script continue without /dev/tty9." into 13	10 years ago
Walter Doekes	da8ba990d1	chan_sip: Allow target refresh (Contact update) on re-INVITE. ... Previously, the Contact was stored only on initial INVITE and on any 18X and 200. That meant that after re-INVITEs from *us* the Contact could get updated, but after re-INVITEs from the *peer*, it did not. This changeset fixes this inconsistency, properly allowing target refreshes through re-INVITES (RFC3261, 12.2). If your strictrtp setting allows it, this change allows you to switch the source IP of a connected/calling device mid-call with a simple re-INVITE from the new IP. ASTERISK-26358 #close Change-Id: Ibb8512054ab27c8c3d2514022568fde943bf2435	10 years ago
Richard Mudgett	e9ddab4685	sip_to_pjsip.py: Map legacy_useroption_parsing. ... Map the sip.conf general section legacy_useroption_parsing to the new pjsip.conf global ignore_uri_user_options. ASTERISK-26316 Reported by: Kevin Harwell Change-Id: I78108a31995db19d41f4e1a07b3324692c5363fc	10 years ago
Richard Mudgett	30af92e78d	res_pjsip: Add ignore_uri_user_options option. ... This implements the chan_sip legacy_useroption_parsing option but with a better name. * Made the caller-id number and redirecting number strings obtained from incoming SIP URI user fields always truncated at the first semicolon. People don't care about anything after the semicolon showing up on their displays even though the RFC allows the semicolon. ASTERISK-26316 #close Reported by: Kevin Harwell Change-Id: Ib42b0e940dd34d84c7b14bc2e90d1ba392624f62	10 years ago
Walter Doekes	7ed5dc2c58	contrib: Let safe_asterisk script continue without /dev/tty9. ... If you use the safe_asterisk script, it uses hardcoded defaults before running configurable values from /etc/asterisk/startup.d. The hardcoded default has TTY=9. Some containerized environments don't have such a TTY, and safe_asterisk would stop. The custom configuration from /etc/asterisk/startup.d/* isn't read until after it stopped, so changing TTY in a custom config did not help. This changeset changes safe_asterisk to continue if the TTY setting was untouched and /dev/tty9 and /dev/vc/9 aren't found. Change-Id: I2c7cdba549b77f418a0af4cb1227e8e6fe4148fc	10 years ago
Joshua Colp	7580a736bb	res_pjsip: Only invoke unidentified endpoint logic when unidentified. ... The code was incorrectly invoking the unidentified logic when an endpoint had actually been identified, causing log messages to be output. ASTERISK-26349 #close Change-Id: Id8104fc9e3d138d5e8b6f6977ecc08765fd17d4f	10 years ago
Joshua Colp	efcfc4c1ee	chan_sip: Don't allocate new RTP instances on top of old ones. ... In some scenarios dialog_initialize_rtp can be called multiple times on the same dialog. This can cause RTP instances to be leaked along with multiple file descriptors for each instance. This change makes it so the existing RTP instances are destroyed and not overwritten, stopping the memory leak. ASTERISK-26272 #close patches: ASTERISK-26272-13.patch submitted by Corey Farrell (license 5909) Change-Id: Id529de1184c68f2f4d254ab41a1f458dafdb5f73	10 years ago
Mark Michelson	f1ffc22933	res_pjsip: Do not crash on ACKs from unknown endpoints. ... The endpoint identification PJSIP module is intended to identify which endpoint an incoming request is from. If an endpoint is not identified, then an artificial endpoint is used in its place when proceeding. The problem is that the ACK request type is an exception to the rule. The artificial endpoint is not used when processing an ACK. This results in the possibility of having a NULL endpoint being used further on. The reason ACK is an exception is an attempt not to spam security logs with unidentified requests. Presumably, you've already logged the unidentified request on the preceeding INVITE. Up until Asterisk 13.10, retrieving a NULL endpoint in this fashion didn't cause an issue. A new change in 13.10 added endpoint ACL checking shortly after endpoint identification. Because we are accessing a NULL endpoint, this ACL check resulted in a crash. The fix here is to be sure to retrieve the artificial endpoint for all request types. ACKs still do not generate unidentified request security events. ASTERISK-26264 #close Reported by nappsoft AST-2016-006 Change-Id: Ie0c795ae2d72273decb972dd74b6a1489fb6b703	10 years ago
zuul	fdb29f1b4e	Merge "res_pjsip: Allow global headers to be overridden." into 13	10 years ago
zuul	02ff55626e	Merge "ConfBridge: Make some announcements asynchronous." into 13	10 years ago
zuul	7180de3f16	Merge "followme: initialize all config items on reload" into 13	10 years ago
Richard Mudgett	23d6ec7417	res_pjsip_messaging.c: Misc cleanups and fixes. ... * Eliminated RAII_VAR in get_outbound_endpoint(). * Simplify update_to() coding. However, this function can only be a NoOp because the To string can only be a URI and not a name-address formatted string. * Simplify update_from() coding. Also fixed a code path modifying the from string when the caller could still want to use the original string. * Fixed msg_data_create() incompletely removing the "pjsip:" to then add back the "sip:" string if needed. The code didn't handle the "pjsip:sip:" case because it left the colon after pjsip in the string. Change-Id: I68a09a665f6d4daa9eaa59069045ab69122e28db	10 years ago
Joshua Colp	5f19657710	res_pjsip: Allow global headers to be overridden. ... Currently when you add global headers from the dialplan both the header in the dialplan and the globally configured header are added to the resulting SIP INVITE. This change makes it so the headers in the dialplan take precedence and are the only ones added. Change-Id: I36f864298f38db3632ad503edc11267cb8ffb3ad	10 years ago
zuul	249a733c17	Merge "apps/app_dial: Fix crash on non-connect call paths for Privacy/Screening option" into 13	10 years ago
zuul	8925367291	Merge "res_pjsip_session: segfault on already disconnected session" into 13	10 years ago
zuul	4b66c74c94	Merge "apps/app_dial: Set the DIALSTATUS to NOANSWER on privacy option 5" into 13	10 years ago
Joshua Colp	89f7cd8182	Merge "build: Add download capability for external packages" into 13	10 years ago
Tzafrir Cohen	206d4f57dc	followme: initialize all config items on reload ... Some configuration directives were not initialized on reload, and hence were not reset to default if they were removed from followme.conf. ASTERISK-26288 #close Change-Id: Ief829e16374ad1e0ecfd63e6ee4923b5a1d1c150	10 years ago
zuul	6d56b87642	Merge "chan_sip: Don't refuse calls with "optional crypto"; fall back to RTP." into 13	10 years ago
zuul	9e874d2cc8	Merge "res_pjsip_registrar.c: Reduce stack usage in find_aor_name()." into 13	10 years ago
zuul	6de392eb17	Merge "pjsip_configuration.c: Ignore repeated identify by methods." into 13	10 years ago
zuul	5a63bfc8fc	Merge "config_global.c: Comments and a default expression adjustment." into 13	10 years ago
zuul	52335c3fe7	Merge "sip_to_pjsip.py: Map canreinvite as directmedia alias." into 13	10 years ago
zuul	e3f549c2f6	Merge "sip_to_pjsip.py: Fix typo converting outboundproxy registration." into 13	10 years ago
zuul	c8c83bcb37	Merge "sip_to_pjsip.py: Fix comment typo and tabs." into 13	10 years ago