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2016-09-20logger: Fix default console settings.Corey Farrell
When logger.conf is missing or invalid we should be printing notices, warnings and errors to the console. The logmask was incorrectly calculated. Change-Id: Ibaa9465a8682854bc1a5e9ba07079bea1bfb6bb3
2016-09-19Merge "asterisk.c: When astcanary dies on linux, reset priority on all ↵zuul
threads." into 13
2016-09-19Merge "Fix showing of swap details when sysinfo() is available" into 13zuul
2016-09-19Merge "res_config_odbc.c: Fix buffer size limitation creating invalid SQL." ↵zuul
into 13
2016-09-19asterisk.c: When astcanary dies on linux, reset priority on all threads.Walter Doekes
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
2016-09-16Fix showing of swap details when sysinfo() is availableTimo Teräs
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
2016-09-15res_config_odbc.c: Fix buffer size limitation creating invalid SQL.Richard Mudgett
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
2016-09-15pjproject_bundled: Prevent SERVFAIL from marking name server badGeorge Joseph
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
2016-09-14Merge "res_pjsip_transport_management: Convert time in log message to ↵zuul
seconds." into 13
2016-09-14Merge "chan_sip: Fix session timeout on retransmit of non-UDP packets" into 13zuul
2016-09-14Merge "rtp: Preserve timestamps on video frames." into 13zuul
2016-09-14Merge "sip_to_pjsip.py: Map legacy_useroption_parsing." into 13zuul
2016-09-14rtp: Preserve timestamps on video frames.Joshua Colp
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
2016-09-14Merge "res_pjsip: Add ignore_uri_user_options option." into 13zuul
2016-09-14res_pjsip_transport_management: Convert time in log message to seconds.Joshua Colp
ASTERISK-26375 #close Change-Id: I46496af5cae41413e76d44d2068a7431279f09dc
2016-09-13chan_sip: Fix session timeout on retransmit of non-UDP packetsSteve Davies
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
2016-09-13Merge "chan_sip: Allow target refresh (Contact update) on re-INVITE." into 13zuul
2016-09-13Merge "res_pjsip_messaging.c: Misc cleanups and fixes." into 13zuul
2016-09-12app_queue: Fix CLI "queue show" and AMI Queues action output truncation.Richard Mudgett
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
2016-09-12Merge "contrib: Let safe_asterisk script continue without /dev/tty9." into 13zuul
2016-09-12chan_sip: Allow target refresh (Contact update) on re-INVITE.Walter Doekes
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
2016-09-09sip_to_pjsip.py: Map legacy_useroption_parsing.Richard Mudgett
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
2016-09-09res_pjsip: Add ignore_uri_user_options option.Richard Mudgett
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
2016-09-09contrib: Let safe_asterisk script continue without /dev/tty9.Walter Doekes
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
2016-09-09res_pjsip: Only invoke unidentified endpoint logic when unidentified.Joshua Colp
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
2016-09-09chan_sip: Don't allocate new RTP instances on top of old ones.Joshua Colp
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
2016-09-09res_pjsip: Do not crash on ACKs from unknown endpoints.Mark Michelson
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
2016-09-08Merge "res_pjsip: Allow global headers to be overridden." into 13zuul
2016-09-07Merge "ConfBridge: Make some announcements asynchronous." into 13zuul
2016-09-07Merge "followme: initialize all config items on reload" into 13zuul
2016-09-07res_pjsip_messaging.c: Misc cleanups and fixes.Richard Mudgett
* 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
2016-09-07res_pjsip: Allow global headers to be overridden.Joshua Colp
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
2016-09-07Merge "apps/app_dial: Fix crash on non-connect call paths for ↵zuul
Privacy/Screening option" into 13
2016-09-07Merge "res_pjsip_session: segfault on already disconnected session" into 13zuul
2016-09-07Merge "apps/app_dial: Set the DIALSTATUS to NOANSWER on privacy option 5" ↵zuul
into 13
2016-09-07Merge "build: Add download capability for external packages" into 13Joshua Colp
2016-09-07followme: initialize all config items on reloadTzafrir Cohen
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
2016-09-06Merge "chan_sip: Don't refuse calls with "optional crypto"; fall back to ↵zuul
RTP." into 13
2016-09-06Merge "res_pjsip_registrar.c: Reduce stack usage in find_aor_name()." into 13zuul
2016-09-06Merge "pjsip_configuration.c: Ignore repeated identify by methods." into 13zuul
2016-09-06Merge "config_global.c: Comments and a default expression adjustment." into 13zuul
2016-09-06Merge "sip_to_pjsip.py: Map canreinvite as directmedia alias." into 13zuul
2016-09-06Merge "sip_to_pjsip.py: Fix typo converting outboundproxy registration." into 13zuul
2016-09-06Merge "sip_to_pjsip.py: Fix comment typo and tabs." into 13zuul
2016-09-06Merge "Sample configs: Eliminate false multiline comment block starts." into 13zuul
2016-09-06build: Add download capability for external packagesGeorge Joseph
The DPMA and g729a, silk, siren7 and siren14 codecs hosted at http://downloads.digium.com/pub/telephony/ are now listed in the "External" sections of the "Resource Modules" and "Codec Translators" pages in menuselect. Any that are selected will automatically be downloaded and installed when "make install" is run. Their LICENSE and README (if avaialble) files will be installed to ASTVARLIBDIR/documentation/thirdparty/<product_name>. Example use with codecs: The codecs/codecs.xml file is a menuselect style xml file that lists the codecs to be included. Their support levels are 'external', which triggers the download and install, and defaultenabled is no. Also because codec_g729a is actually in a directory named codec_g729 on the download server, the newly added 'member_data' element is used to override the default of the directory name being the package name. You can use the 'directory_name' attribute to keep default base URL (http://downloads.digium.com/pub/telephony/) but use the new directory, or you use the 'remote_url' attribute to specify a full URL to the download directory. In this case, you must still follow the same subdirectory naming conventions as that used for the packages located at 'http://downloads.digium.com/pub/telephony'. A new configure option '--with-externals-cache' was added and like '--with-sounds-cache' it allows the installer to cache tarballs so they're not downloaded every time. To assist with the download and install process, each external package now has a manifest.xml file that, among other things, contains a package version and checksums for each file in the tarball. The manifest is saved to both the cache directory and ASTMODDIR and together with the manifest.xml on the downloads site, tells the install scripts whether a download and/or update is needed. bash and xmlstarlet are required for downloader operation. If they're not installed, the external items in menuselect will be unavailable. Change-Id: Id3dcf1289ffd3cb0bbd7dfab3cafbb87be60323a
2016-09-06Merge "format_cap.c: Fix CLI "core show channeltype Surrogate" crash." into 13zuul
2016-09-06chan_sip: Don't refuse calls with "optional crypto"; fall back to RTP.Walter Doekes
Certain SNOM phones send so-called "optional crypto" in their SDP body. Regular SRTP setup looks like this: m=audio 64620 RTP/SAVP 8 0 9 99 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:... SNOM-style "optional crypto" looks like this: m=audio 61438 RTP/AVP 8 0 9 99 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:... A crypto line is supplied, but the m-line does not have SAVP. When res_srtp.so is *not* loaded, then chan_sip.so treats the optional crypto as regular RTP, but when res_srtp.so *is* loaded, it refuses the incoming call with the following message: WARNING: process_sdp: Failed to receive SDP offer/answer with required SRTP crypto attributes for audio For platforms that want to start providing SRTP this presents a compatibility problem. This changeset lets chan_sip handle the SDP as if no crypto-line was supplied: i.e. accept the call as regular RTP, just like it did before res_srtp was loaded. Now you'll get this informative warning instead: WARNING: Ignoring crypto attribute in SDP because RTP transport is insecure ASTERISK-23989 #close Reported by: Olle Johansson Change-Id: I91a15ae05a0296e398d6b65f53bb11afde1d80e2
2016-09-04Merge "app_mp3: Use correct buffer size and the same sample rate as the ↵zuul
channel" into 13
2016-09-03apps/app_dial: Fix crash on non-connect call paths for Privacy/Screening optionMatt Jordan
In any scenario in which the callee is not connected to the caller, the current code in app_dial will crash due to raising a Dial End Stasis Message after the callee channel has been hung up. This patch corrects the error by simply moving the explicit hangup of the callee (peer) channel until after the dial end message. ASTERISK-25691 #close Change-Id: I816a414014424d0d8c80e2a3cbef13ef8c63798d
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