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Author	SHA1	Message	Date
zuul	324a24bf10	Merge "res_pjsip: Add ignore_uri_user_options option." into 14	9 years ago
Joshua Colp	34e82d16b4	rtp: Preserve timestamps on video frames. ... 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	9 years ago
Joshua Colp	70e6963c5f	res_pjsip_transport_management: Convert time in log message to seconds. ... ASTERISK-26375 #close Change-Id: I46496af5cae41413e76d44d2068a7431279f09dc	9 years ago
zuul	d6c9e72c93	Merge "res_pjsip: Don't assume a request will have any addresses." into 14	9 years ago
Alexander Traud	b3802e68b8	chan_sip: Enable Session-Timers for SIP over TCP (and TLS). ... Asterisk defaults to timers=accept/refresher=uas. In that scenario, only in that scenario, Sessions-Timers (RFC 4028) had no effect via TCP. This change enables Session-Timers for SIP over TCP (and for SIP over TLS). However with longer international calls via TCP, the SIP channel might break, because all hops on the Internet route must stay online (have not a single power outage, for example). Therefore with Session-Timers enabled (which are enabled at default), you might see dropped calls. Consequently even with this change, you might be better-off going for session-timers=refuse in your sip.conf. ASTERISK-19968 #close Change-Id: I1cd33453c77c56c8e1394cd60a6f17bb61c1d957 (cherry picked from commit 66c9dfb272)	9 years ago
zuul	5ddf0862b4	Merge "chan_sip: Allow target refresh (Contact update) on re-INVITE." into 14	9 years ago
zuul	31bb30ddfb	Merge "res_pjsip_messaging.c: Misc cleanups and fixes." into 14	9 years ago
Joshua Colp	44644c8563	Merge "contrib: Let safe_asterisk script continue without /dev/tty9." into 14	9 years ago
Joshua Colp	b6207cab24	res_pjsip: Don't assume a request will have any addresses. ... When performing DNS resolution the failover code present in res_pjsip currently assumes that a request will always have at least one viable address. In practice this is not true. A domain may be used that has no records. The code now checks that at least one address exists on the request which prevents looping. ASTERISK-26364 #close Change-Id: Ic0761b0264864acd85915c94d878a81624940f4c	9 years ago
Richard Mudgett	d910a51b33	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
Walter Doekes	52e6d03401	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	fd82304aa9	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	b7ec070cd7	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	d980085d38	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	236761a5b2	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	d7d22a2be5	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	f3e79d5870	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	e1c8e7bdd1	Merge "res_pjsip: Allow global headers to be overridden." into 14	9 years ago
zuul	c396565672	Merge "ConfBridge: Make some announcements asynchronous." into 14	9 years ago
zuul	9b70c3cd30	Merge "res/res_stasis_playback: Cancel the entire playlist when a stop occurs" into 14	9 years ago
zuul	71e36845bd	Merge "apps/app_dial: Fix crash on non-connect call paths for Privacy/Screening option" into 14	9 years ago
Richard Mudgett	057cfe0cbb	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	f401187282	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	b3e25d7de6	Merge "res_resolver_unbound: Fix config documentation." into 14	9 years ago
zuul	ebae949815	Merge "res_pjsip_session: segfault on already disconnected session" into 14	9 years ago
zuul	46d5691aa4	Merge "apps/app_dial: Set the DIALSTATUS to NOANSWER on privacy option 5" into 14	9 years ago
Mark Michelson	999b34301b	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	6c4ab0203e	Merge "build: Add download capability for external packages" into 14	9 years ago
Joshua Colp	7ec7d53983	res_resolver_unbound: Fix config documentation. ... The code was referencing the config section as 'globals' instead of 'general'. This change swaps it over to 'general'. Change-Id: I9dfe7788f41c4a6754c77e103880dc1a747de7fe	9 years ago
zuul	d254f31426	Merge "chan_sip: Don't refuse calls with "optional crypto"; fall back to RTP." into 14	9 years ago
zuul	f7b94dbf17	Merge "res_pjsip_registrar.c: Reduce stack usage in find_aor_name()." into 14	9 years ago
zuul	b378cc6f43	Merge "pjsip_configuration.c: Ignore repeated identify by methods." into 14	9 years ago
Joshua Colp	ed1106c891	Merge "config_global.c: Comments and a default expression adjustment." into 14	9 years ago
zuul	1c90758904	Merge "sip_to_pjsip.py: Map canreinvite as directmedia alias." into 14	9 years ago
Matt Jordan	827b235357	res/res_stasis_playback: Cancel the entire playlist when a stop occurs ... Prior to this patch, a stop issued by a delete of a Playback resource (indicated by the control frame AST_CONTROL_STREAM_STOP) would only stop the current media URI playing. Subsequent URIs specified by a playback operation would then proceed on, even though we had just indicated to the User that the Playback was finished *and* after they had just 'deleted' the resource. Whoops. This patch corrects it by bailing out of the sequence of URIs to play if one of them is terminated with an AST_CONTROL_STREAM_STOP indication. ASTERISK-26341 #close Change-Id: I2da9ec43545ba46cdfffe287c7e4907eae7fca42	9 years ago
zuul	045d29b2c0	Merge "sip_to_pjsip.py: Fix typo converting outboundproxy registration." into 14	9 years ago
zuul	4175b5081d	Merge "sip_to_pjsip.py: Fix comment typo and tabs." into 14	9 years ago
zuul	3f32149173	Merge "Sample configs: Eliminate false multiline comment block starts." into 14	9 years ago
Joshua Colp	e455036e9e	Merge "sorcery: Create function ast_sorcery_lockable_alloc." into 14	9 years ago
Joshua Colp	dec772d46c	Merge "named_locks: Use ao2_weakproxy to deal with cleanup from container." into 14	9 years ago
Joshua Colp	ddd6c810a9	Merge "astobj2: Support using a separate object for locking." into 14	9 years ago
George Joseph	8b6f9dde14	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	74a5bbfc1b	Merge "format_cap.c: Fix CLI "core show channeltype Surrogate" crash." into 14	9 years ago
Alexei Gradinari	09e9dca0c5	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
Walter Doekes	9b59ebee2f	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	f33ccbdf86	Merge "app_mp3: Use correct buffer size and the same sample rate as the channel" into 14	9 years ago
Matt Jordan	3e054a3826	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	15105c7494	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	323b74c6ca	res_pjsip_registrar.c: Reduce stack usage in find_aor_name(). ... Change-Id: I8aebad1fdcf303bd115b59a4b57fbbd5b2267f09	9 years ago
Richard Mudgett	5bc8c91863	pjsip_configuration.c: Ignore repeated identify by methods. ... Change-Id: Ied0c06043d1dfef8fdc9c9a808cf89b118119838	9 years ago