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Author	SHA1	Message	Date
Joshua Colp	de2a202e56	Merge "alembic: Turn off execute bit on non-executable python scripts" into 13	8 years ago
Alexander Traud	ef19db9261	srtp: Allow zero as tag value for a sRTP Crypto Suite. ... ASTERISK-25490 #close Change-Id: I1c5fc0942c33c96d62b24203aad0f1e1a1a0131f	8 years ago
Joshua Colp	95c66234ba	Merge "res_musiconhold: Don't chdir() when scanning MoH files" into 13	8 years ago
Joshua Colp	b0a1e40542	Merge changes from topic 'ASTERISK-24712' into 13 ... * changes: res_xmpp: Use incremental backoff when a read error occurs res_xmpp: Try to provide useful errors messages from OpenSSL res_xmpp: Correctly check return value of SSL_connect	8 years ago
zuul	92517fc6a7	Merge "res_xmpp: Fix ref counting issue" into 13	8 years ago
George Joseph	a827892ff7	res_pjsip_config_wizard: Add 2 new parameters to help with proxy config ... Two new parameters have been added to the pjsip config wizard. * Setting 'sends_line_with_registrations' to true will cause the wizard to skip the creation of an identify object to match incoming request to the endpoint and instead add the line and endpoint parameters to the outbound registration object. * Setting 'outbound_proxy' is a shortcut for adding individual endpoint/outbound_proxy, aor/outbound_proxy and registration/outbound_proxy parameters. Change-Id: I678e5f80765734c056620528a6d40d82736ceeb0	8 years ago
Sean Bright	864dda07f3	alembic: Turn off execute bit on non-executable python scripts ... Change-Id: I744c986da4a38aeff8c00837eb89de7841fbc86c	8 years ago
Richard Mudgett	a9529cbb21	Add DTLS sanity check. ... Change-Id: Ib32612cf6c7ce9213a11b9cba82f630f8cd3564b	8 years ago
Josh Roberson	bb68f57a03	cel_pgsql.c: Fix buffer overflow calling libpq ... PQEscapeStringConn() expects the buffer passed in to be an adequitely sized buffer to write out the escaped SQL value string into. It is possible, for large values (such as large values to Dial with a lot of devices) to have more than our 512+1 byte allocation and thus cause libpq to create a buffer overrun. glibc will nicely ABRT asterisk for you, citing a stack smash. Let's only allocate it to be as large as needed: If we have a value, then (strlen(value) * 2) + 1 (as recommended by libpq), and if we have none, just one byte to hold our null will do. ASTERISK-26896 #close Change-Id: If611c734292618ed68dde17816d09dd16667dea2	8 years ago
Sean Bright	79a2c26c03	core: Remove embedded module support ... This has not worked for some time and is no longer actively maintained. Change-Id: I5110b0db69c152761b58fa025cb0a53b0e544d99	8 years ago
Sean Bright	2c28f7a922	res_musiconhold: Document the 'format' option ... ASTERISK-26086 #close Reported by: Jens Bürger Change-Id: I6aab666c0bf01fd0c64d7a5bcb22fa7f5d41335e	8 years ago
Sean Bright	61fd70c250	res_musiconhold: Don't chdir() when scanning MoH files ... There doesn't appear to be any reason that we are chdir'ing in moh_scan_files, and in the event of an Asterisk crash, the core files may not get written because we have changed into a read-only directory. ASTERISK-23996 #close Reported by: Walter Doekes Change-Id: Iac806dce01b3335963fbd62d4b4da9a65c614354	8 years ago
Sean Bright	73bb08fd6a	res_xmpp: Use incremental backoff when a read error occurs ... If a read error occurs, we immediately attempt a reconnect without any delay. Instead, let's sleep and backoff up to 60 seconds before we try again. ASTERISK-24712 #close Reported by: Matthias Urlichs Change-Id: I6fe10ef4734837727437beab715e336777f13f48	8 years ago
Sean Bright	55693383e2	res_xmpp: Fix ref counting issue ... The only remaining reference to the endpoint is in the endpoints container, and because it is unlinked in ast_endpoint_shutdown, we don't have to explicitly cleanup the endpoint ourselves. Change-Id: I912a2692e52d3e2ed445b32d8ae3f9004bc2f2e8	8 years ago
Sean Bright	1966265562	res_xmpp: Try to provide useful errors messages from OpenSSL ... If any errors occur during the TLS connection setup, we currently dump a fairly generic error message. So instead we try to pull in something useful from OpenSSL to report instead. ASTERISK-24712 Reported by: Matthias Urlichs Change-Id: I288500991a9681f447d92913b11fedaf426087f4	8 years ago
Sean Bright	03b99ae3d2	res_xmpp: Correctly check return value of SSL_connect ... SSL_connect returns non-zero for both success and some error conditions so simply negating is inadequate. Change-Id: Ifbf882896e598703b6c615407fa456d3199f95b1	8 years ago
zuul	39fff2ebf9	Merge "res_pjsip_sdp_rtp: Set hangup cause for RTP timeouts" into 13	8 years ago
Joshua Colp	ddbff36c12	Merge "res_xmpp: Correct implementation of JABBER_STATUS & JabberStatus" into 13	8 years ago
zuul	30f5409eef	Merge "cdr: Allow setting of user field from 'h' extension" into 13	8 years ago
zuul	1fccedf0fd	Merge "pjproject_bundled: raise timeout value used when downloading" into 13	8 years ago
zuul	2f2811c2dc	Merge "res_xmpp: Don't crash when trying to send a message without a connection" into 13	8 years ago
zuul	b7b171c06f	Merge "res_xmpp: Include client name in connection related error messages" into 13	8 years ago
Sean Bright	d9d2beba1c	res_pjsip_sdp_rtp: Set hangup cause for RTP timeouts ... chan_sip sets the hangup cause code to AST_CAUSE_REQUESTED_CHAN_UNAVAIL (44) when a channel is hung up due to an RTP timeout. So do the same when it happens with PJSIP for parity. Change-Id: I3546ebbde6460c22a27c9da1bf321711b5961ab8	8 years ago
zuul	68d523a1af	Merge "audiohook.c: Lost RTP packets lead to out-of-sync MixMonitor." into 13	8 years ago
Kevin Harwell	552cf009c0	Update for 13.15.0-rc1	8 years ago
Kevin Harwell	f1b34e6eb4	AMI: Updated version ... Updated the AMI version for the following reason (see CHANGES for more details): The 'PJSIPShowEndpoint' command's response event of 'IdentifyDetail' now contains a new optional parameter, 'MatchHeader'. Change-Id: I9aeac4decc89f9b464b3f026e97c7ef1acc79242	8 years ago
Kevin Harwell	e6aeeabddf	pjproject_bundled: raise timeout value used when downloading ... After configuring Asterisk with '--with-pjproject-bundled' the configure/build process attempts to download pjproject from its download site. Currently, a timeout of 10 seconds is used that will stop the download process if pjproject has not been fully downloaded in that time. For some systems this was not enough time and the process was timing out too early. This patch raises the download timeout value to '60'. Also, this patch fixes another bug where the DOWNLOAD_TIMEOUT variable was not being properly exported due to a naming error. DOWNLOAD_MAX_TIMEOUT is now properly renamed to DOWNLOAD_TIMEOUT. ASTERISK-26814 #close Change-Id: Ia56e4e8a3d39db76bc8a1852b2cf07ec10b39842	8 years ago
Sean Bright	0939a19cff	res_xmpp: Correct implementation of JABBER_STATUS & JabberStatus ... The documentation for JABBER_STATUS (and the deprecated JabberStatus app) indicate that a return value of 7 indicates that the specified buddy was not in the roster. It also indicates that you can specify a "bare" JID (one without a resource). Unfortunately the actual behavior does not match the documented behavior. Assuming that our roster includes the buddy online and available "valid@example.org/Valid" and does *not* include the buddy "invalid@example.org", the JABBER_STATUS() function returns the following before this patch: +------------------------------+------------+--------------------------+ | Buddy | Status | Result | +------------------------------+------------+--------------------------+ | valid@example.org | Online | 7 (Not in roster) | | valid@example.org/Valid | Online | 1 (Online) | | valid@example.org/Invalid | N/A | 7 (Not in roster) | | invalid@example.org | N/A | Error logged, no return | | invalid@example.org/Valid | N/A | Error logged, no return | +------------------------------+------------+--------------------------+ And after this patch: +------------------------------+------------+--------------------------+ | Buddy | Status | Result | +------------------------------+------------+--------------------------+ | valid@example.org | Online | 1 (Online) | | valid@example.org/Valid | Online | 1 (Online) | | valid@example.org/Invalid | N/A | 6 (Offline) | | invalid@example.org | N/A | 7 (Not in roster) | | invalid@example.org/Valid | N/A | 7 (Not in roster) | +------------------------------+------------+--------------------------+ This brings the behavior in line with the documentation. ASTERISK-23510 #close Reported by: Anthony Critelli Change-Id: I9c3241035363ef4a6bdc21fabfd8ffcd9ec657bf	8 years ago
Sean Bright	a487f6fb97	res_xmpp: Don't crash when trying to send a message without a connection ... If we never establish a connection to our Jabber server, iksemel never sets up its internal transport pointer, so attempting to send a message dereferences a NULL pointer and causes a crash. ASTERISK-21855 #close Reported by: Jeremy Kister Change-Id: I204a568894e4a53ab929783ecc594a000f04d79c	8 years ago
Sean Bright	90fb1fca41	res_xmpp: Include client name in connection related error messages ... ASTERISK-25622 #close Reported by: Sean Darcy Change-Id: I8472cb7bfb58d411a3cfbd482da98cae2d94d1e9	8 years ago
zuul	4fcb8d807e	Merge "CHANNEL(callid): Give dialplan access to the callid." into 13	8 years ago
zuul	b79e67ba47	Merge "res_pjsip_session: Enable RFC3578 overlap dialing support." into 13	8 years ago
zuul	5f75cc8279	Merge "res_pjsip_messaging: Check URI type before dereferencing" into 13	8 years ago
zuul	d7c52125e9	Merge "Revert "app_queue: Handle the caller being redirected out of a queue bridge"" into 13	8 years ago
Joshua Colp	f762dfaf2a	Merge "app_queue: Member stuck as pending after forwarding previous call from queue" into 13	8 years ago
Sebastian Gutierrez	e196190f11	cdr: Allow setting of user field from 'h' extension ... The CDR code previously did not allow the user field to be set from the 'h' extension in the dialplan. This change removes that limitation and allows it to be set. ASTERISK-26818 Change-Id: I0fed8a79b5e408bac4e30542b8f33a61c5ed9aa6	8 years ago
zuul	14a9a6fc09	Merge "pjsip: prevent memory corruption on creation of xml bodies" into 13	8 years ago
Richard Begg	398e5ec16c	res_pjsip_session: Enable RFC3578 overlap dialing support. ... Support for RFC3578 overlap dialling (i.e. 484 Response to partially matched destinations) as currently provided by chan_sip is missing from res_pjsip. This patch adds a new endpoint attribute (allow_overlap) [defaults to yes] which when set to yes enables 484 responses to partial destination matches rather than the current 404. ASTERISK-26864 Change-Id: Iea444da3ee7c7d4f1fde1d01d138a3d7b0fe40f6	8 years ago
zuul	d7ba743329	Merge "autochan/mixmonitor/chanspy: Fix unsafe channel locking and references." into 13	8 years ago
zuul	72ae513f15	Merge "res_hep: Capture actual transport type in use" into 13	8 years ago
Sean Bright	218f618095	res_hep: Capture actual transport type in use ... Rather than hard-coding UDP, allow consumers of the HEP API to specify which protocol is in use. Update the PJSIP provider to pass in the current protocol type. ASTERISK-26850 #close Change-Id: I54bbb0a001cfe4c6a87ad4b6f2014af233349978	8 years ago
Sean Bright	1c8b81a2a4	Revert "app_queue: Handle the caller being redirected out of a queue bridge" ... This reverts commit 163e9e53dc. Change-Id: Ief28479c77a298879dfe2c56be7ee92dc465da4b	8 years ago
Sean Bright	b3cc20799b	res_pjsip_messaging: Check URI type before dereferencing ... We aren't validating that the URI we just parsed is a SIP/SIPS one before trying to access the user, host, and port members of a possibly uninitialized structure. Also update the MessageSend documentation to indicate what 'from' formats are accepted. ASTERISK-26484 #close Reported by: Vinod Dharashive Change-Id: I476b5cc5f18a7713d0ee945374f2a1c164857d30	8 years ago
Joshua Elson	91c97b5da5	pjsip: prevent memory corruption on creation of xml bodies ... ASTERISK-26776 #close Change-Id: I884b6f4e8233a355d0be687ec78d41bc0e4d3fd2	8 years ago
Sean Bright	7f34c11b6a	bridge_softmix: Ignore non-voice frames from translator ... Some codecs - codec_speex specifically - take voice frames and return other types of frames, like CNG. If we subsequently treat those as voice frames, we'll run into trouble when destroying the frame because of the requirement that each voice frame have an associated format. ASTERISK-26880 #close Reported by: Kirsty Tyerman Change-Id: I43f8450c48fb276ad8b99db8512be82949c1ca7c	8 years ago
zuul	fdea369852	Merge "res/res_pjsip_session: Only check localnet if it is defined" into 13	8 years ago
Aaron An	d5b480afca	audiohook.c: Lost RTP packets lead to out-of-sync MixMonitor. ... Fixed a bug in function "ast_audiohook_write_frame" that checked the variable other_factory_samples and only flushed the factories, so they would be in sync, when other_factory_samples > 0. When there is not any rtp incoming the variable other_factory_samples will be 0, and although the result of "our_factory_ms - other_factory_ms" may be very large, this led to the record file not syncing. ASTERISK-26875 #close Reported-by: Aaron An Tested-by: Aaron An Change-Id: Ia4d890fb8fc1636a7188502bab35f555685aea22	8 years ago
Joshua Colp	9613391868	Merge "thread safety: Don't use getprotobyname()" into 13	8 years ago
Sean Bright	38cebc73a3	thread safety: Don't use getprotobyname() ... POSIX does not require getprotobyname() to be thread safe and some implementations use static memory which causes issues when multiple threads are used. Further, our usage of it today is just to ultimately get IPPROTO_TCP for calls to setsockopt(). So instead we just use IPPROTO_TCP directly. Change-Id: I2e14e58674808f7ce99b2f5e900d0f90d0d8da48	8 years ago
Sean Bright	265455bc2d	res_rtp_asterisk: Pass correct data length to ast_rtcp_interpret ... We are currently passing in the capacity of the read buffer instead of the number of bytes that we actually read off the wire. Change-Id: I60465049727d955c7f9a5e529e6f2aaff04cda36	8 years ago