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Author	SHA1	Message	Date
Vitezslav Novy	597d1f8951	chan_sip: Do not change IP address in SDP origin line (o=) in SIP reINVITE ... If directmedia=yes is configured, when call is answered, Asterisk sends reINVITE to both parties to set up media path directly between the endpoints. In this reINVITE msg SDP origin line (o=) contains IP address of endpoint instead of IP of asterisk. This behavior violates RFC3264, sec 8: "When issuing an offer that modifies the session, the "o=" line of the new SDP MUST be identical to that in the previous SDP, except that the version in the origin field MUST increment by one from the previous SDP." This patch assures IP address of Asterisk is always sent in SDP origin line. ASTERISK-17540 Reported by: saghul Change-Id: I533a047490c43dcff32eeca8378b2ba02345b64e	8 years ago
Jenkins2	b55bab3237	Merge "formats: Restore previous fread() behavior" into 13	8 years ago
George Joseph	15ddc9acb3	alembic: Fix enum creation for dtls_fingerprint ... Change-Id: Ic061c5066a146616a68376881c7e4cf6d6e7e7db	8 years ago
Jenkins2	f9b28dd990	Merge "res_pjsip_t38: Make t38_reinvite_response_cb tolerant of NULL channel" into 13	8 years ago
Florian Floimair	2370469645	alembic: fix erroneous commit for add_prune_on_boot ... Added include for postgresql ENUM type and redefined values in the same way as in the other migration scripts. ASTERISK-27254 #close Change-Id: Id667304cdf3891b1c2f7d35fab3e2a84026159fa	8 years ago
Alexander Traud	13aa1241c3	res_srtp: Add support for libsrtp2.1. ... Asterisk is able to use libSRTP 2.0.x. However since libSRTP 2.1.x, the macro SRTP_AES_ICM got renamed to SRTP_AES_ICM_128. Beside to still compile with previous versions of libSRTP, this change allows libSRTP 2.1.x as well. ASTERISK-27253 #close Change-Id: I2e6eb3c3bc844fee8a624060a2eb6f182dc70315	8 years ago
Ben Ford	598a18ffee	chan_pjsip: Suppress frame warnings. ... When rtp_keepalive is on for a PJSIP endpoint dialing to another Asterisk instance also using PJSIP, Asterisk will continue to print warning messages about not being able to send frames of a certain type. This suppresses that warning message. Change-Id: I0332a05519d7bda9cacfa26d433909ff1909be67	8 years ago
Richard Mudgett	6c922b3157	res_rtp_asterisk.c: Check RTP packet version earlier. ... Change-Id: Ic6493a7d79683f3e5845dff1cee49445fd5a0adf	8 years ago
Sean Bright	3f7d0b63fc	formats: Restore previous fread() behavior ... Some formats are able to handle short reads while others are not, so restore the previous behavior for the format modules so that we don't have spurious errors when playing back files. ASTERISK-27232 #close Reported by: Jens T. Change-Id: Iab7f52b25a394f277566c8a2a4b15a692280a300	8 years ago
Walter Doekes	45744fc53d	res/res_pjsip: Standardize/fix localnet checks across pjsip. ... In 2dee95cc (ASTERISK-27024) and 776ffd77 (ASTERISK-26879) there was confusion about whether the transport_state->localnet ACL has ALLOW or DENY semantics. For the record: the localnet has DENY semantics, meaning that "not in the list" means ALLOW, and the local nets are in the list. Therefore, checks like this look wrong, but are right: /* See if where we are sending this request is local or not, and if not that we can get a Contact URI to modify */ if (ast_apply_ha(transport_state->localnet, &addr) != AST_SENSE_ALLOW) { ast_debug(5, "Request is being sent to local address, " "skipping NAT manipulation\n"); (In the list == localnet == DENY == skip NAT manipulation.) And conversely, other checks that looked right, were wrong. This change adds two macro's to reduce the confusion and uses those instead: ast_sip_transport_is_nonlocal(transport_state, addr) ast_sip_transport_is_local(transport_state, addr) ASTERISK-27248 #close Change-Id: Ie7767519eb5a822c4848e531a53c0fd054fae934	8 years ago
Joshua Colp	7b240d1734	Merge "app_directory: Handle a NULL mailbox without crashing" into 13	8 years ago
Jenkins2	6b9d18be7a	Merge "chan_ooh323: Fix confusing indentation warning" into 13	8 years ago
George Joseph	786c4791f9	res_pjsip_t38: Make t38_reinvite_response_cb tolerant of NULL channel ... t38_reinvite_response_cb can get called by res_pjsip_session's session_inv_on_tsx_state_changed in situations where session->channel is NULL. If it is, the ast_log warning segfaults because it tries to get the channel name from a NULL channel. * Check session->channel and print "unknown channel" when it's NULL. ASTERISK-27236 Reported by: Ross Beer Change-Id: I4326e288d36327f6c79ab52226d54905cdc87dc7	8 years ago
Sean Bright	55f30c29fd	rtp_engine: Prevent possible double free with DTLS config ... ASTERISK-27225 #close Reported by: Richard Kenner Change-Id: I097b81734ef730f8603c0b972909d212a3a5cf89	8 years ago
Sean Bright	f36db2dbdc	chan_ooh323: Fix confusing indentation warning ... ASTERISK-27177 #close Reported by: Tzafrir Cohen Change-Id: I40311c404edb2302a7543ad5ca7a06b2a38f2d97	8 years ago
Sean Bright	5f4863d4f9	app_directory: Handle a NULL mailbox without crashing ... ASTERISK-27241 #close Reported by: David Moore Change-Id: Ibbbca85517b04c315406ebfe3b6f7e0763daedc6	8 years ago
Jenkins2	07577fc997	Merge "chan_pjsip: Add tag info in CHANNEL function" into 13	8 years ago
Jenkins2	b0064245b3	Merge "pjsip_message_ip_updater: Fix issue handling "tel" URIs" into 13	8 years ago
Jenkins2	c4254e237c	Merge "AST-2017-006: Fix app_minivm application MinivmNotify command injection" into 13	8 years ago
George Joseph	990b017668	pjsip_message_ip_updater: Fix issue handling "tel" URIs ... sanitize_tdata was assuming all URIs were SIP URIs so when a non SIP uri was in the From, To or Contact headers, the unconditional cast of a non-pjsip_sip_uri structure to pjsip_sip_uri caused a segfault when trying to access uri->other_param. * Added PJSIP_URI_SCHEME_IS_SIP(uri) || PJSIP_URI_SCHEME_IS_SIPS(uri) checks before attempting to cast or use the returned uri. ASTERISK-27152 Reported-by: Ross Beer Change-Id: Id380df790e6622c8058a96035f8b8f4aa0b8551f	8 years ago
Corey Farrell	04ee3eb774	AST-2017-006: Fix app_minivm application MinivmNotify command injection ... An admin can configure app_minivm with an externnotify program to be run when a voicemail is received. The app_minivm application MinivmNotify uses ast_safe_system() for this purpose which is vulnerable to command injection since the Caller-ID name and number values given to externnotify can come from an external untrusted source. * Add ast_safe_execvp() function. This gives modules the ability to run external commands with greater safety compared to ast_safe_system(). Specifically when some parameters are filled by untrusted sources the new function does not allow malicious input to break argument encoding. This may be of particular concern where CALLERID(name) or CALLERID(num) may be used as a parameter to a script run by ast_safe_system() which could potentially allow arbitrary command execution. * Changed app_minivm.c:run_externnotify() to use the new ast_safe_execvp() instead of ast_safe_system() to avoid command injection. * Document code injection potential from untrusted data sources for other shell commands that are under user control. ASTERISK-27103 Change-Id: I7552472247a84cde24e1358aaf64af160107aef1	8 years ago
Joshua Colp	1a022285dd	res_rtp_asterisk: Only learn a new source in learn state. ... This change moves the logic which learns a new source address for RTP so it only occurs in the learning state. The learning state is entered on initial allocation of RTP or if we are told that the remote address for the media has changed. While in the learning state if we continue to receive media from the original source we restart the learning process. It is only once we receive a sufficient number of RTP packets from the new source that we will switch to it. Once this is done the closed state is entered where all packets that do not originate from the expected source are dropped. The learning process has also been improved to take into account the time between received packets so a flood of them while in the learning state does not cause media to be switched. Finally RTCP now drops packets which are not for the learned SSRC if strict RTP is enabled. ASTERISK-27013 Change-Id: I56a96e993700906355e79bc880ad9d4ad3ab129c	8 years ago
Jenkins2	844f70f301	Merge "bridge_native_rtp.c: Fixup native_rtp_framehook()" into 13	8 years ago
Richard Mudgett	4aaccb7795	bridge_native_rtp.c: Fixup native_rtp_framehook() ... * Fix framehook to test frame type for control frame. * Made framehook exit early if frame type is not a control frame. * Eliminated RAII_VAR in framehook. * Use switch instead of else-if ladder for control frame handling. Change-Id: Ia555fc3600bd85470e3c0141147dbe3ad07c1d18	8 years ago
Sean Bright	d2ace23248	confbridge: Handle user hangup during name recording ... This prevents orphaned CBAnn channels from getting stuck in the bridge. ASTERISK-26994 #close Reported by: James Terhune Change-Id: I5e43e832a9507ec3f2c59752cd900b41dab80457	8 years ago
Jenkins2	2d69137bf4	Merge "app_record: Resolve some absolute vs. relative filename bugs" into 13	8 years ago
Jenkins2	378e0f7764	Merge "voicemail: Fix various abuses of mkstemp" into 13	8 years ago
Jenkins2	71cc740828	Merge "main/app: Only look to end of file if ':end' is specified, and not just ':'" into 13	8 years ago
Jenkins2	06e7d6e750	Merge "res/res_pjsip_session: allow SDP answer to be regenerated" into 13	8 years ago
Jenkins2	a3b36b5043	Merge "alembic: Add dtls_fingerprint column in ps_endpoints table" into 13	8 years ago
Andre Nazario	a45af32983	chan_pjsip: Add tag info in CHANNEL function ... Create local_tag and remote_tag in CHANNEL info to get tag from From and To headers of a SIP dialog. ASTERISK-27220 Change-Id: I59b16c4b928896fcbde02ad88f0e98922b15d524	8 years ago
Sean Bright	9e6efcace5	voicemail: Fix various abuses of mkstemp ... mkstemp() returns a unique filename, but appending an extension to that filename does not guarantee uniqueness. Instead, use mkdtemp() and we can put whatever extension we want on the files that we create inside the directory. In the case of app_minivm, we also now properly clean up any temporary files that we create. ASTERISK-20858 #close Reported by: Walter Doekes Change-Id: I30ad04f0e115f0b11693ff678ba5184d8b938e43	8 years ago
Sean Bright	01b5913ce0	app_record: Resolve some absolute vs. relative filename bugs ... If the Record() application is called with a relative filename that includes directories, we were not properly creating the intermediate directories and Record() would fail. Secondarily, updated the documentation for RECORDED_FILE to mention that it does not include a filename extension. Finally, rewrote the '%d' functionality to be a bit more straight forward and less noisy. ASTERISK-16777 #close Reported by: klaus3000 Change-Id: Ibc2640cba3a8c7f17d97b02f76b7608b1e7ffde2	8 years ago
Jenkins2	28857047da	Merge "app_queue: Evaluate realtime queues when running dialplan functions" into 13	8 years ago
Jenkins2	c0251ae004	Merge "app_voicemail: Honor escape digits in "greeting only" mode" into 13	8 years ago
Florian Floimair	bf178a0f4f	alembic: Add dtls_fingerprint column in ps_endpoints table ... The ps_endpoints table was missing the dtls_fingerprint column introduced with commit adba2a8d7f. ASTERISK-27168 #close Change-Id: I9cb5006f7f50718b5239919562773adabb334cfd	8 years ago
Matt Jordan	fff2f68616	main/app: Only look to end of file if ':end' is specified, and not just ':' ... There is a little known feature in app_controlplayback that will cause the specified offset to be used relative to the end of a file if a ':end' is detected within the filename. This feature is pretty bad, but okay. However, a bug exists in this code where a ':' detected in the filename will cause the end pointer to be non-NULL, even if the full ':end' isn't specified. This causes us to treat an unspecified offset (0) as being "start playing from the end of the file", resulting in no file playback occurring. This patch fixes this bug by resetting the end pointer if ':end' is not found in the filename. ASTERISK-23608 #close Reported by: Jonathan White Change-Id: Ib4c7b1b45283e4effd622a970055c51146892f35 (cherry picked from commit 13efea24f7)	8 years ago
Sean Bright	579d4593ac	app_queue: Evaluate realtime queues when running dialplan functions ... ASTERISK-19103 #close Reported by: Jim Van Meggelen Change-Id: I4bd32a9d1fcebb8ac56bff0e084d4f53e31b692b	8 years ago
Sean Bright	0af145de2d	app_voicemail: Honor escape digits in "greeting only" mode ... ASTERISK-21241 #close Reported by: Eelco Brolman Patches: Patch uploaded by Eelco Brolman (License 6442) Change-Id: Icbe39b5c82a49b46cf1d168dc17766f3d84f54fe	8 years ago
Sean Bright	d251a961ac	res_smdi: Clean up memory leak ... Change-Id: I1e33290929e1aa7c5b9cb513f8254f2884974de8	8 years ago
Jenkins2	e8cfd5a80f	Merge "bridge_softmix.c: Remove always true test." into 13	8 years ago
Richard Mudgett	3f22b53349	bridge_softmix.c: Remove always true test. ... Change-Id: I26238df2ff0d0f6dfe95c3aa35da588f1ee71727	8 years ago
Sungtae Kim	b88c3a4209	app_queue: Fix initial hold time queue statistic ... Fixed to use correct initial value and fixed to use the correct queue info to check the first value. ASTERISK-27204 Change-Id: Ia9e36c828e566e1cc25c66f73307566e4acb8e73	8 years ago
Jenkins2	71753d67f4	Merge "res_xmpp: fix inverted return code check in OAuth" into 13	8 years ago
Torrey Searle	8e99969000	res/res_pjsip_session: allow SDP answer to be regenerated ... If an SDP answer hasn't been sent yet, it's legal to change it. This is required for PJSIP_DTMF_MODE to work correctly, and can also have use in the future for updating codecs too. ASTERISK-27209 #close Change-Id: Idbbfb7cb3f72fbd96c94d10d93540f69bd51e7a1	8 years ago
Jenkins2	3fffa7ae86	Merge "res_calendar_icalendar: Properly handle recurring events" into 13	8 years ago
Michael Kuron	4faf77feec	res_xmpp: fix inverted return code check in OAuth ... fetch_access_token calls func_curl via ast_func_read. The latter returns 0 upon success and -1 if the function is not available. This commit inverts the return code check so that an error is printed if the module is not loaded and not if it is loaded. ASTERISK-27207 #close Change-Id: I9ef903f80702d1218e8701f65a4e5e918e6548fb	8 years ago
Sean Bright	a6251ec373	res_calendar_icalendar: Properly handle recurring events ... When looking for recurring events, use the correct end time based on the configured 'timeframe.' ASTERISK-27174 #close Reported by: Mark Thompson Change-Id: Id90c3cfc79d561a5521d79be176683e225f2edef	8 years ago
George Joseph	572b5307e0	Fix downloader not working with curl ... The codec/dpma downloader wasn't handling curl correctly. The logic that transforms makeopts into a bash-sourceable file wasn't handling the make 'or' command in DOWNLOAD_TIMEOUT so bash was looking for an 'or' command. That logic has been eliminated. Instead of trying to transform and source makeopts, the downloader now calls a make scriptlet to print the value of a specific variable. This way, make handles the ors (or any other make construct that happens to creep into that file). ASTERISK-27202 Reported by: Sean McCord Change-Id: Iadfb6693528e4d4da7b8bb201fa66da2c71c7f99	8 years ago
Jenkins2	b461fe8bcd	Merge "configure: Check cache for valid pjproject tarball before downloading." into 13	8 years ago