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Author	SHA1	Message	Date
Joshua Colp	9df4056d70	res_pjsip_transport_management: Convert time in log message to seconds. ... ASTERISK-26375 #close Change-Id: I46496af5cae41413e76d44d2068a7431279f09dc	9 years ago
Steve Davies	98e42cc662	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	9 years ago
zuul	efba4a7b9c	Merge "chan_sip: Allow target refresh (Contact update) on re-INVITE." into 13	9 years ago
zuul	3d7cbaa675	Merge "res_pjsip_messaging.c: Misc cleanups and fixes." into 13	9 years ago
Richard Mudgett	0388882cdb	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	9 years ago
zuul	c833c1fda1	Merge "contrib: Let safe_asterisk script continue without /dev/tty9." into 13	9 years ago
Walter Doekes	da8ba990d1	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	9 years ago
Richard Mudgett	e9ddab4685	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	9 years ago
Richard Mudgett	30af92e78d	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	9 years ago
Walter Doekes	7ed5dc2c58	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	9 years ago
Joshua Colp	7580a736bb	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	9 years ago
Joshua Colp	efcfc4c1ee	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	9 years ago
Mark Michelson	f1ffc22933	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	9 years ago
zuul	fdb29f1b4e	Merge "res_pjsip: Allow global headers to be overridden." into 13	9 years ago
zuul	02ff55626e	Merge "ConfBridge: Make some announcements asynchronous." into 13	9 years ago
zuul	7180de3f16	Merge "followme: initialize all config items on reload" into 13	9 years ago
Richard Mudgett	23d6ec7417	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	9 years ago
Joshua Colp	5f19657710	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	9 years ago
zuul	249a733c17	Merge "apps/app_dial: Fix crash on non-connect call paths for Privacy/Screening option" into 13	9 years ago
zuul	8925367291	Merge "res_pjsip_session: segfault on already disconnected session" into 13	9 years ago
zuul	4b66c74c94	Merge "apps/app_dial: Set the DIALSTATUS to NOANSWER on privacy option 5" into 13	9 years ago
Joshua Colp	89f7cd8182	Merge "build: Add download capability for external packages" into 13	9 years ago
Tzafrir Cohen	206d4f57dc	followme: initialize all config items on reload ... 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	9 years ago
zuul	6d56b87642	Merge "chan_sip: Don't refuse calls with "optional crypto"; fall back to RTP." into 13	9 years ago
zuul	9e874d2cc8	Merge "res_pjsip_registrar.c: Reduce stack usage in find_aor_name()." into 13	9 years ago
zuul	6de392eb17	Merge "pjsip_configuration.c: Ignore repeated identify by methods." into 13	9 years ago
zuul	5a63bfc8fc	Merge "config_global.c: Comments and a default expression adjustment." into 13	9 years ago
zuul	52335c3fe7	Merge "sip_to_pjsip.py: Map canreinvite as directmedia alias." into 13	9 years ago
zuul	e3f549c2f6	Merge "sip_to_pjsip.py: Fix typo converting outboundproxy registration." into 13	9 years ago
zuul	c8c83bcb37	Merge "sip_to_pjsip.py: Fix comment typo and tabs." into 13	9 years ago
zuul	899385d47b	Merge "Sample configs: Eliminate false multiline comment block starts." into 13	9 years ago
George Joseph	117a7741c8	build: Add download capability for external packages ... 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	9 years ago
zuul	1b752842b9	Merge "format_cap.c: Fix CLI "core show channeltype Surrogate" crash." into 13	9 years ago
Walter Doekes	d04ae7d1d8	chan_sip: Don't refuse calls with "optional crypto"; fall back to RTP. ... 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	9 years ago
zuul	9470848fba	Merge "app_mp3: Use correct buffer size and the same sample rate as the channel" into 13	9 years ago
Matt Jordan	df3d0188e4	apps/app_dial: Fix crash on non-connect call paths for Privacy/Screening option ... 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	9 years ago
Matt Jordan	a64063cc97	apps/app_dial: Set the DIALSTATUS to NOANSWER on privacy option 5 ... If the callee selects option '5' using the Dial application's privacy (P) option, the DIALSTATUS is erroneously set to ANSWER. This option reflects the callee sending the caller to VoiceMail one time; the call is definitely *not* ANSWERed in such a scenario. With this patch, the DIALSTATUS is instead set to NOANSWER, which is the same DIALSTATUS that is set when the 'send to VoiceMail every time' option is set. ASTERISK-25691 Change-Id: Iaf0c9f0fa00545e7366443875e2bb7d9a89a1358	9 years ago
Richard Mudgett	03fc438f6e	res_pjsip_registrar.c: Reduce stack usage in find_aor_name(). ... Change-Id: I8aebad1fdcf303bd115b59a4b57fbbd5b2267f09	9 years ago
Richard Mudgett	b5e753227d	pjsip_configuration.c: Ignore repeated identify by methods. ... Change-Id: Ied0c06043d1dfef8fdc9c9a808cf89b118119838	9 years ago
Richard Mudgett	9b7501b6ad	config_global.c: Comments and a default expression adjustment. ... Change-Id: Ia6a58f8c73a30da6874b3f94364dce162d6f1ad3	9 years ago
Richard Mudgett	3314e1cec2	sip_to_pjsip.py: Map canreinvite as directmedia alias. ... Change-Id: I48b8e150f96a3d2a24d8fc25fbe4f5aff9f4a6b2	9 years ago
Richard Mudgett	6372f40ba0	sip_to_pjsip.py: Fix typo converting outboundproxy registration. ... Change-Id: I6f30e5f9fcf8469ba0079fbf884047d54c2c0b15	9 years ago
Richard Mudgett	11eb1afd2d	sip_to_pjsip.py: Fix comment typo and tabs. ... Change-Id: If35174614545727817d329c60ba4456c028941b5	9 years ago
Richard Mudgett	0f9b144c1a	Sample configs: Eliminate false multiline comment block starts. ... Change-Id: Ie627def9604ae30abd80754f9e6f09874825aec6	9 years ago
Richard Mudgett	8d1c535bd6	format_cap.c: Fix CLI "core show channeltype Surrogate" crash. ... * Make ast_format_cap_get_names() NULL tolerant. ASTERISK-26331 #close Reported by: CGI.NET Change-Id: Id67e93936dc8ec2a33a9d33655843d43b59285a3	9 years ago
Alexei Gradinari	9bca895469	res_pjsip_session: segfault on already disconnected session ... On heavy loaded system the TCP/TLS incoming calls could be disconnected by pjproject while these calls are being processed by asterisk which could use the session's memory pools. If the session in the disconnected state then the session memory pools were already freed, so we get segfault. This patch adds a lifetime control on an INVITE session to pjproject. The lifetime of the session is manipulated by calling pjsip_inv_add_ref/pjsip_inv_dec_ref. This patch uses these functions to inform pjproject that the session is in use. This patch adds check if the session state is not disconnected and also checks if the memory pool is not NULL. This patch also places tasks 'session_end' and 'session_end_completion' into session's serializer to avoid race condition. ASTERISK-26291 #close Change-Id: I4d28b1fb3b91f0492a911d110049d670fdc3c8d7	9 years ago
Mark Michelson	63feffa126	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	9 years ago
zuul	1bd571ef75	Merge "res_pjsip: qualify/unqualify added/deleted realtime endpoints" into 13	9 years ago
zuul	84b7bda139	Merge "sip_to_pjsip: Migrate IPv4/IPv6 (Dual Stack) configurations." into 13	9 years ago
Michael Kuron	a002a4d2db	app_mp3: Use correct buffer size and the same sample rate as the channel ... Previously, the buffer used for MP3 streamed from HTTP servers had a size of 1 MB. For 8 kHz mono audio at 16 bit resolution, such a buffer covers about 1 minute. Only when the buffer is full does audio start to play. For MP3 files streamed from a server, that is usually not a big deal as long as the connection to the server is fast enough to supply that much data within a second or two. For MP3 live streams however, it takes 1 minute to download 1 minute of audio, so without this change, app_mp3 wasn't really usable for MP3 live streams. This commit changes the buffer size so that it covers 6 seconds of an MP3 file streamed from a server and 0.5 seconds of an MP3 live stream. The latter is identified by the use of a .m3u file extension. app_mp3 so far only supported 8 kHz audio. Now it always runs at the sample rate of the channel. ASTERISK-26085 #close Change-Id: Id1ee274733cd804a0edecf7450329b72f1235af0	9 years ago