asterisk

Commit Graph

Author	SHA1	Message	Date
Joshua Colp	2684ffd9cd	odbc: Remove options that are no longer applicable.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	7f8d570c8d	Merge "asterisk.c: Non-root users also get the astcanary after core restart." into 14	10 years ago
zuul	381cafe335	Merge "sd_notify (systemd status notifications) support" into 14	10 years ago
zuul	2c8b8a9ecb	Merge "res_pjsip_multihomed: Change Contact port to listening port." into 14	10 years ago
zuul	5205b0196b	Merge "rtp: Only accept the first payload for a format in SDP." into 14	10 years ago
Tzafrir Cohen	29e096cd13	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	2b75559e6c	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	b56717b396	Merge "asterisk.c: When astcanary dies on linux, reset priority on all threads." into 14	10 years ago
zuul	7340a63adf	Merge "Fix showing of swap details when sysinfo() is available" into 14	10 years ago
zuul	2acd3e38ae	Merge "res_config_odbc.c: Fix buffer size limitation creating invalid SQL." into 14	10 years ago
Walter Doekes	bd39d2c0ab	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
Joshua Colp	64d2ea44d8	Merge "pjproject_bundled: Prevent SERVFAIL from marking name server bad" into 14	10 years ago
Richard Mudgett	4c61d9f629	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
Timo Teräs	1e9ae453fd	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
Joshua Colp	696824df89	rtp: Only accept the first payload for a format in SDP. When receiving an SDP offer with multiple payloads for the same format we would generate an answer with the first payload, but during the payload crossover operation (to set the payloads for receiving) we would remove all payloads but the last. This would result in incoming traffic being matched against the wrong format and outgoing traffic being sent using the wrong payload. This change makes it so that once a format has a payload number put into the mapping all subsequent ones are ignored. This ensures there is only ever one payload in the mapping and that it is the payload placed into the answer SDP. ASTERISK-26365 #close Change-Id: I1e8150860a3518cab36d00b1fab50f9352b64e60	10 years ago
Steve Davies	58f41ebf93	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 (cherry picked from commit `98e42cc662`)	10 years ago
Joshua Colp	f1c06322ba	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	6a5683cc27	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
Joshua Colp	92deae3bb1	Merge "chan_sip: Enable Session-Timers for SIP over TCP (and TLS)." into 14	10 years ago
zuul	10078b4b3c	Merge "res_pjsip_transport_management: Convert time in log message to seconds." into 14	10 years ago
Joshua Colp	7d1bcf2f96	Merge "rtp: Preserve timestamps on video frames." into 14	10 years ago
zuul	a085874d59	Merge "sip_to_pjsip.py: Map legacy_useroption_parsing." into 14	10 years ago
zuul	324a24bf10	Merge "res_pjsip: Add ignore_uri_user_options option." into 14	10 years ago
Joshua Colp	34e82d16b4	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
Joshua Colp	70e6963c5f	res_pjsip_transport_management: Convert time in log message to seconds. ASTERISK-26375 #close Change-Id: I46496af5cae41413e76d44d2068a7431279f09dc	10 years ago
zuul	d6c9e72c93	Merge "res_pjsip: Don't assume a request will have any addresses." into 14	10 years ago
Alexander Traud	b3802e68b8	chan_sip: Enable Session-Timers for SIP over TCP (and TLS). Asterisk defaults to timers=accept/refresher=uas. In that scenario, only in that scenario, Sessions-Timers (RFC 4028) had no effect via TCP. This change enables Session-Timers for SIP over TCP (and for SIP over TLS). However with longer international calls via TCP, the SIP channel might break, because all hops on the Internet route must stay online (have not a single power outage, for example). Therefore with Session-Timers enabled (which are enabled at default), you might see dropped calls. Consequently even with this change, you might be better-off going for session-timers=refuse in your sip.conf. ASTERISK-19968 #close Change-Id: I1cd33453c77c56c8e1394cd60a6f17bb61c1d957 (cherry picked from commit `66c9dfb272`)	10 years ago
zuul	5ddf0862b4	Merge "chan_sip: Allow target refresh (Contact update) on re-INVITE." into 14	10 years ago
zuul	31bb30ddfb	Merge "res_pjsip_messaging.c: Misc cleanups and fixes." into 14	10 years ago
Joshua Colp	44644c8563	Merge "contrib: Let safe_asterisk script continue without /dev/tty9." into 14	10 years ago
Joshua Colp	b6207cab24	res_pjsip: Don't assume a request will have any addresses. When performing DNS resolution the failover code present in res_pjsip currently assumes that a request will always have at least one viable address. In practice this is not true. A domain may be used that has no records. The code now checks that at least one address exists on the request which prevents looping. ASTERISK-26364 #close Change-Id: Ic0761b0264864acd85915c94d878a81624940f4c	10 years ago
Richard Mudgett	d910a51b33	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
Walter Doekes	52e6d03401	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	fd82304aa9	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	b7ec070cd7	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	d980085d38	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	236761a5b2	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	d7d22a2be5	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	f3e79d5870	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	e1c8e7bdd1	Merge "res_pjsip: Allow global headers to be overridden." into 14	10 years ago
zuul	c396565672	Merge "ConfBridge: Make some announcements asynchronous." into 14	10 years ago
zuul	9b70c3cd30	Merge "res/res_stasis_playback: Cancel the entire playlist when a stop occurs" into 14	10 years ago
zuul	71e36845bd	Merge "apps/app_dial: Fix crash on non-connect call paths for Privacy/Screening option" into 14	10 years ago
Richard Mudgett	057cfe0cbb	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	f401187282	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	b3e25d7de6	Merge "res_resolver_unbound: Fix config documentation." into 14	10 years ago
zuul	ebae949815	Merge "res_pjsip_session: segfault on already disconnected session" into 14	10 years ago
zuul	46d5691aa4	Merge "apps/app_dial: Set the DIALSTATUS to NOANSWER on privacy option 5" into 14	10 years ago
Mark Michelson	999b34301b	ConfBridge: Make some announcements asynchronous. Confbridge announcements tend to block a channel while they are being played. In some circumstances, this is warranted since you want that particular channel not to hear the announcement (Example: "John Doe has entered the conference"). For others it makes less sense. This change first introduces methods for playing sounds asynchronously into the conference. This is very similar to how synchronous sounds are played, except the channel initiating the playback does not wait for the sound to complete before moving on. Asynchronous announcements are used for two circumstances: * Sounds played for a user after they have left the bridge * Sounds that play first to a single user and then the rest of the conference (if the channel and conference use the same language) ASTERISK-26289 #close Reported by Mark Michelson Change-Id: Ie486bb3de1646d50894489030326a423e594ab0a	10 years ago
zuul	6c4ab0203e	Merge "build: Add download capability for external packages" into 14	10 years ago

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