31126 Commits (c5d2bf05f4e29c57e332a6481433838db04d7908)

 

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Author	SHA1	Message	Date
Torrey Searle	c5d2bf05f4	res/res_rtp_asterisk: ensure marker bit is correctly set on ssrc change ... Certain race conditions between changing bridge types and DTMF can cause the current FLAG_NEED_MARKER_BIT to send the marker bit before the actual first packet of native bridging. This logic keeps track of the ssrc the bridge is currently sending and will correctly ensure the marker bit is set if SSRC as changed from the previous sent packet. ASTERISK-27845 Change-Id: I01858bd0235f1e5e629e20de71b422b16f55759b	8 years ago
Joshua Colp	25764691b0	Merge "netsock2: Add ast_sockaddr_resolve_first_af to netsock2 public API"	8 years ago
Matthew Fredrickson	9f9dce05b2	netsock2: Add ast_sockaddr_resolve_first_af to netsock2 public API ... This function originally was used in chan_sip to enable some simplifying assumptions and eventually was copy and pasted into res_pjsip_logger and res_hep. Since it's replicated in three places, it's probably best to move it into the public netsock2 API for these modules to use. Change-Id: Id52e23be885601c51d70259f62de1a5e59d38d04	8 years ago
Joshua Colp	21dd609e77	Merge "app_voicemail: Fix data-type mismatch between app_voicemail and database"	8 years ago
Kevin Harwell	be9a1952d8	Merge "chan_mobile: support handling of caller-id names ("cnam")."	8 years ago
Kevin Harwell	01f3483e6b	Merge "app_voicemail: Fix incorrect msg leaving/retrieving an ODBC voicemail"	8 years ago
Kevin Harwell	357654313f	Merge "rtp_engine: Allow Media Formats with add_static_payload(-1) on egress again."	8 years ago
Nic Colledge	2ca3b6d9cc	app_voicemail: Fix data-type mismatch between app_voicemail and database ... Fix data-type mismatch between app_voicemail and database columns exposed by new version of MariaDB ASTERISK-27760 Change-Id: I8543ad480a08c98be78bde1ee870e6e6c84b2c5b	8 years ago
Nic Colledge	97f20fe5ed	app_voicemail: Fix incorrect msg leaving/retrieving an ODBC voicemail ... Correct the log warning message shown when ODBC voicemail retrieve_file is called and there is a null value in the category column. A more meaningfull message is now written at debug level. ASTERISK-27853 Change-Id: Ic36e97d5eb070a23a12ba45972f6b53e2182a3f4	8 years ago
Brian P. Martin	52ed6bcc8f	chan_mobile: support handling of caller-id names ("cnam"). ... Add support to handle caller-ID names ("cnam") in addition to caller-ID numbers. The prior code ignored the caller-ID name altogether, and used the local name for the cell phone (e.g. "my-iphone") in its place. Note: as of this writing, at least some Android phones don't pass cnam to us. This can be seen by issuing "core set debug 2" in the CLI and watching the "CLIP" record when a call comes in. If cnam isn't in the CLIP record, there's nothing we can do to provide one. We'll provide a null cnam field, so later Asterisk processes know to try other sources (e.g. cidname database, OpenCNAM, etc.). Reported by: Brian Martin Tested by: Brian Martin ASTERISK-27726 Change-Id: I89490d85fa406c36261879c50ae5e65595538ba5	8 years ago
Alexander Traud	f10fc135d4	res_pjsip_endpoint_identifier_ip: Unregister the module for headers. ... Asterisk uses Reference Counting to track whether a module can be unloaded. Every consumer who requires a module, increases the reference count. When the consumer goes, is unloaded itself, it has to decrease the reference count on all its used/required modules. That way core stop gracefully works on the command-line interface (CLI): One module after the other is unloaded. A recent change broke this for the module res_pjsip. ASTERISK-27861 Change-Id: I261abcb411d026bbb0691cc78f28300bfd3103a3	8 years ago
Joshua Colp	60ce5d0003	Merge "cli: Display correct unit for HTTP timeout in "manager show settings"."	8 years ago
Joshua Colp	195af35026	Merge "Fix GCC 8 build issues."	8 years ago
Joshua Colp	1cc4313d32	Merge "rtp_engine: Remove the double assigned RTP payload ID of H.263+."	8 years ago
Joshua Colp	c0c79997f8	Merge "git: Ignore *.orig."	8 years ago
Joshua Colp	bc4b3c535d	Merge "rtp_engine: Avoid a typo error in Doxygen for ast_rtp_codecs_find_payload_code."	8 years ago
Joshua Colp	a103221de2	Merge "pjsip: Rewrite OPTIONS support with new eyes."	8 years ago
Alexander Traud	71d1e8d8c8	rtp_engine: Remove the double assigned RTP payload ID of H.263+. ... Mantis-3709 (Commit 68ff3c3, Asterisk 1.2) added support for the video format H.263+. For this, the RTP payload ID 103 got assigned statically. Commit f1aadc8 assigned another payload ID 98 for this format in Asterisk 1.6. Change-Id: I90e35b158487f8f1f8187da6241b54cd3b74e667	8 years ago
Corey Farrell	4722a653f4	cli: Display correct unit for HTTP timeout in "manager show settings". ... HTTP timeout is in seconds, not minutes. ASTERISK-27852 #close Change-Id: Ie6640835cb07307555741f9b559c2eb876d9343e	8 years ago
Alexander Traud	263637a38d	rtp_engine: Avoid a typo error in Doxygen for ast_rtp_codecs_find_payload_code. ... Change-Id: Ica089d4507a27ddfc4ce3a88d697ffbef378de48	8 years ago
Corey Farrell	b5914d90ac	Fix GCC 8 build issues. ... This fixes build warnings found by GCC 8. In some cases format truncation is intentional so the warning is just suppressed. ASTERISK-27824 #close Change-Id: I724f146cbddba8b86619d4c4a9931ee877995c84	8 years ago
Alexander Traud	919b0eb3f2	rtp_engine: Allow Media Formats with add_static_payload(-1) on egress again. ... This issue affected only installations with rtp_use_dynamic=yes in asterisk.conf which is the default since Asterisk 15. Codec 2 and SiLK were built-in examples of media formats which were affected. ASTERISK-27850 Reported by: Dinis Brazão, Selene Feigl Change-Id: I08c1e76433a67e4350141d38cacf3a1cb5086496	8 years ago
Joshua Colp	6773ea9e39	Merge "makeopts.in: Remove unused/undefined AST_MARCH_NATIVE."	8 years ago
Joshua Colp	5437b3932d	Merge "sip_to_pjsip: Enable python3 compatibility."	8 years ago
Joshua Colp	1351f42363	Merge "res_hep: Adds hostname resolution support for capture_address"	8 years ago
Jenkins2	179c794879	Merge "app_macro: Prevent infinite loop in find_matching_priority."	8 years ago
Corey Farrell	2e37684913	git: Ignore *.orig. ... This prevents accidental commit of files created by patch. Change-Id: I68380db61f0f9d620046f719ccd978811d0e9964	8 years ago
Alexander Traud	2d81709ab1	sip_to_pjsip: Enable python3 compatibility. ... The script remains compatible with Python 2.7 but now also works with Python 3.3 and newer; to ease the migration from chan_sip to chan_pjsip. ASTERISK-27811 Change-Id: I59cc6b52a1a89777eebcf25b3023bdf93babf835	8 years ago
Corey Farrell	cea87fe7b8	makeopts.in: Remove unused/undefined AST_MARCH_NATIVE. ... Change-Id: I617a96ebb83ec99f5d3176bbbee2d2a272ccb203	8 years ago
Jaco Kroon	9f1e1d153a	manager: fix digest auth for ami/http mechanism. ... Due to a fixed size buffer the digest authentication could be incorrectly calculated if a large URI was provided, causing authentication failure. The buffer is now dynamically allocated to allow any size URI within the normal limits of the HTTP request size. ASTERISK-27841 Change-Id: I660609db13b8f9e5f9567f339dd804f4985d41b3	8 years ago
Jenkins2	d83a37f0cc	Merge "stream: Make the topology a reference counted object."	8 years ago
Corey Farrell	d855658f23	app_macro: Prevent infinite loop in find_matching_priority. ... Use AST_PBX_MAX_STACK to escape if we recurse 128 times. This will prevent crash if dialplan contains an include loop. Log an error when this occurs, at most one message per call to Macro() so we avoid logger spam. ASTERISK-26570 #close Change-Id: I6c71b76998c31434391b150de055ae9a531e31da	8 years ago
Matthew Fredrickson	8f55f7c333	res_hep: Adds hostname resolution support for capture_address ... Previously, only an IP address would be accepted for the capture_address config setting in hep.conf. This change allows capture_address to be a resolvable hostname or an IP address. ASTERISK-27796 #close Reported-By: Sebastian Gutierrez Change-Id: I33e1a37a8b86e20505dadeda760b861a9ef51f6f	8 years ago
Jenkins2	57ad488451	Merge "res_ari: Remove requirement that body exists when debug is on."	8 years ago
Jenkins2	dcaaae6cd1	Merge "iostreams: Add some documentation for the ast_iostream_* functions"	8 years ago
Joshua Colp	11f5aba43b	Merge "chan_dahdi: Configurable dialed digit timeouts"	8 years ago
Jenkins2	85f894a8e0	Merge "pbx_lua: Support displaying lua error message if no debug table exists"	8 years ago
Jenkins2	8e228fc138	Merge "res_pjsip/pjsip_distributor.c: Add missing off-nominal request response."	8 years ago
Joshua Colp	7528b86cad	stream: Make the topology a reference counted object. ... The stream topology has no lock of its own resulting in another lock protecting it in some way (for example the channel lock). If multiple channels are being juggled at the same time this can be problematic. This change makes the topology a reference counted object instead which guarantees it will remain valid even without the channel lock being held. Change-Id: I4f4d3dd856a033ed55fe218c3a4fab364afedb03	8 years ago
Jenkins2	0565a6c909	Merge "Add the ability to read the media file type from HTTP header for playback"	8 years ago
Tzafrir Cohen	6301531416	chan_dahdi: Configurable dialed digit timeouts ... Analog phones dial overlap dialing and it is chan_dahdi's job to read the numbers. It has three timeout constants that this commit converts to channel-level configuration options: * firstdigit_timeout: Default time (ms) to detect first digit * interdigit_timeout: Default time (ms) to detect following digits * matchdigit_timeout: Default time (ms) to wait in case of ambiguous match. This happens when the dialed digits match a number in the current context but are also the prefix of another number. Change-Id: Ib728fa900a4f6ae56d1ed810aba61b6593fb7213	8 years ago
Jenkins2	e538fc8e86	Merge "res_rtp_asterisk: Always update SRTP on local SSRC change."	8 years ago
Jenkins2	706b899358	Merge "pjsip: Increase maximum number of usable ciphers & other cleanups"	8 years ago
Joshua Colp	de3ca9bada	res_ari: Remove requirement that body exists when debug is on. ... The "ari set debug" code for incoming requests incorrectly assumed that all requests would contain a body. If one did not exist the request would be incorrectly rejected. The response that was sent was also incomplete as an incorrect function was used to construct the response. The code has now been changed to no longer require a request to have a body and the response updated to use the correct function. ASTERISK-27801 Change-Id: I4eef036ad54550a4368118cc348765ecac25e0f8	8 years ago
Sean Bright	069a0b7593	iostreams: Add some documentation for the ast_iostream_* functions ... Change-Id: Id71b87637f0a484eb5a1cd26c3d1c7c15c7dcf26	8 years ago
Sean Bright	239074c759	pjsip: Increase maximum number of usable ciphers & other cleanups ... * Increase maximum number of ciphers from 100 to 256 (or whatever PJ_SSL_SOCK_MAX_CIPHERS is #define'd to) * Simplify logic in cipher_name_to_id() * Make signed/unsigned comparison consistent Re: https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=897412 Reported by: Ondřej Holas Change-Id: Iea620f03915a1b873e79743154255c3148a514e7	8 years ago
Richard Mudgett	11b7de82c5	res_pjsip/pjsip_distributor.c: Add missing off-nominal request response. ... Change-Id: I389579b39c523d1d1e8ce020ef549a8bb5781c9b	8 years ago
Richard Mudgett	6cab3c836a	res_pjsip/pjsip_distributor.c: Pull some assignments out of if tests. ... Change-Id: I3d30d638b53a4bbe9bf9aad853c649d583894112	8 years ago
Joshua Colp	afdca5c68c	res_rtp_asterisk: Always update SRTP on local SSRC change. ... When the local SSRC changes we need to update the SRTP information so that the proper key is used. This is commonly done as a result of bridging two channels together. Previously we only updated the SRTP information if media had already flowed, but in practice the channel driver may have already performed SRTP negotiation and set up the previous SSRC. We now always do it on a local SSRC change. ASTERISK-27795 ASTERISK-27800 Change-Id: Ia7c8e74c28841388b5244ac0b8fd6c1dc6ee4c10	8 years ago
Gaurav Khurana	0827d5cc53	Add the ability to read the media file type from HTTP header for playback ... How it works today: media_cache tries to parse out the extension of the media file to be played from the URI provided to Asterisk while caching the file. What's expected: Better will be to have Asterisk get extension from other ways too. One of the common ways is to get the type of content from the CONTENT-TYPE header in the HTTP response for fetching the media file using the URI provided. Steps to Reproduce: Provide a URL of the form: http://host/media/1234 to Asterisk for media playback. It fails to play and logs show the following error line: [Sep 15 15:48:05] WARNING [29148] [C-00000092] file.c: File http://host/media/1234 does not exist in any format Scenario this issue is blocking: In the case where the media files are stored in some cloud object store, following can block the media being played via Asterisk: Cloud storage generally needs authenticated access to the storage. The way to do that is by using signed URIs. With the signed URIs there's no way to preserve the name of the file. In most cases Cloud storage returns a key to access the object and preserving file name is also not a thing there ASTERISK-27286 Reporter: Gaurav Khurana Change-Id: I1b14692a49b2c1ac67688f58757184122e92ba89	8 years ago