asterisk

Commit Graph

Author	SHA1	Message	Date
George Joseph	ad6af63895	GCC12: Fixes for 16+ Most issues were in stringfields and had to do with comparing a pointer to an constant/interned string with NULL. Since the string was a constant, a pointer to it could never be NULL so the comparison was always "true". gcc now complains about that. There were also a few issues where determining if there was enough space for a memcpy or s(n)printf which were fixed by defining some of the involved variables as "volatile". There were also a few other miscellaneous fixes. ASTERISK-30044 Change-Id: Ia081ca1bcfb329df6487c4660aaf1944309eb570	3 years ago
Mark Petersen	bb2102a991	chan_sip.c Session timers get removed on UPDATE If Asterisk receives a SIP REFER with Session-Timers UAC maintain Session-Timers when sending UPDATE" ASTERISK-29843 Change-Id: I8e9a21c13bf757fa34d778f49ba3cf859b29ae5c	3 years ago
Mark Petersen	4f7e3d1609	chan_sip: SIP route header is missing on UPDATE if Asterisk need to send an UPDATE before answer on a channel that uses Record-Route: it will not include a Route header ASTERISK-29955 Change-Id: Id1920ecbfea7739a038b14dc94487ecfe7b57eef	3 years ago
Mark Petersen	95ee1d06d6	chan_sip.c Fix pickup on channel that are in AST_STATE_DOWN resolve issue with pickup on device that uses "183" and not "180" ASTERISK-29832 Change-Id: I4c7d223870f8ce9a7354e0f73d4e4cb2e8b58841	3 years ago
Alexander Traud	751bbf4b97	progdocs: Fix grouping for latest Doxygen. Since Doxygen 1.8.16, a special comment block is required. Otherwise (pure C comment), the group command is ignored. Additionally, several unbalanced group commands were fixed. ASTERISK-29732 Change-Id: I4687857b9d56e6f44fd440b73af156691660202e	4 years ago
Naveen Albert	bcb7aee723	documentation: Standardize examples Most examples in the XML documentation use the example tag to demonstrate examples, which gets parsed specially in the Wiki to make it easier to follow for users. This fixes a few modules to use the example tag instead of vanilla para tags to bring them in line with the standard syntax. ASTERISK-29777 #close Change-Id: I9acb6cc5faf1d220e73c6dd28592371d768d279b	4 years ago
Alexander Traud	3f86c95cf5	channels: Fix for Doxygen. ASTERISK-29762 Change-Id: Ia8811ac12b93ff8c18164699c6fbc604cb0a23f7	4 years ago
Josh Soref	c1b21bee6d	channels: Spelling fixes Correct typos of the following word families: appease permanently overriding residue silliness extension channels globally reference japanese group coordinate registry information inconvenience attempts cadence payloads presence provisioning mimics behavior width natively syslabel not owning unquelch mostly constants interesting active unequipped brodmann commanding backlogged without bitstream firmware maintain exclusive practically structs appearance range retransmission indication provisional associating always whether cyrillic distinctive components reinitialized initialized capability switches occurring happened outbound ASTERISK-29714 Change-Id: Ife52ee89cd2170b684fa651ca72b1cb911a57339	4 years ago
Joshua C. Colp	13fd0789a2	policy: Add deprecation and removal versions to modules. app_meetme is deprecated in 19, to be removed in 21. app_osplookup is deprecated in 19, to be removed in 21. chan_alsa is deprecated in 19, to be removed in 21. chan_mgcp is deprecated in 19, to be removed in 21. chan_skinny is deprecated in 19, to be removed in 21. res_pktccops is deprecated in 19, to be removed in 21. cdr_mysql was deprecated in 1.8, to be removed in 19. app_mysql was deprecated in 1.8, to be removed in 19. app_ices was deprecated in 16, to be removed in 19. app_macro was deprecated in 16, to be removed in 21. app_fax was deprecated in 16, to be removed in 19. app_url was deprecated in 16, to be removed in 19. app_image was deprecated in 16, to be removed in 19. app_nbscat was deprecated in 16, to be removed in 19. app_dahdiras was deprecated in 16, to be removed in 19. cdr_syslog was deprecated in 16, to be removed in 19. chan_oss was deprecated in 16, to be removed in 19. chan_phone was deprecated in 16, to be removed in 19. chan_sip was deprecated in 17, to be removed in 21. chan_nbs was deprecated in 16, to be removed in 19. chan_misdn was deprecated in 16, to be removed in 19. chan_vpb was deprecated in 16, to be removed in 19. res_config_sqlite was deprecated in 16, to be removed in 19. res_monitor was deprecated in 16, to be removed in 21. conf2ael was deprecated in 16, to be removed in 19. muted was deprecated in 16, to be removed in 19. ASTERISK-29548 ASTERISK-29549 ASTERISK-29550 ASTERISK-29551 ASTERISK-29552 ASTERISK-29553 ASTERISK-29554 ASTERISK-29555 ASTERISK-29557 ASTERISK-29558 ASTERISK-29559 ASTERISK-29560 ASTERISK-29561 ASTERISK-29562 ASTERISK-29563 ASTERISK-29564 ASTERISK-29565 ASTERISK-29566 ASTERISK-29567 ASTERISK-29568 ASTERISK-29569 ASTERISK-29570 ASTERISK-29571 ASTERISK-29572 ASTERISK-29573 ASTERISK-29574 Change-Id: Ic3bee31a10d42c4b3bbc913d893f7b2a28a27131	4 years ago
Naveen Albert	7b82587dd6	chan_sip: Expand hook flash recognition. Some ATAs send hook flash events as application/hook-flash, rather than a DTMF event. Now, we also recognize hook-flash as a flash event. ASTERISK-29370 Change-Id: I1c3b82a040dff3affcd94bad8ce33edc90c04725	4 years ago
George Joseph	40bdfff73b	Updates for the MessageSend Dialplan App Enhancements: * The MessageSend dialplan application now takes an optional third argument that can set the message's "To" field on outgoing messages. It's an alternative to using the MESSAGE(to) dialplan function. NOTE: No channel driver currently implements this field. A follow-on commit for res_pjsip_messaging will implement it for the chan_pjsip channel driver. * To prevent confusion with the first argument, currently named "to", it's been renamed to "destination". Its function, creating the request URI, hasn't changed. * The documentation for MessageSend was updated to be more clear about the parameters and how they interact the MESSAGE() dialplan function. * With the rename of MessageSend's first parameter, and the fact that message.c references <info> elements in chan_sip.c, res_pjsip_messaging.c and res_xmpp, they each needed documentation updates to use MessageDestinationInfo instead of MessageToInfo. * appdocsxml.dtd was updated to include a missing element declaration for "dataType". This was showing up as an error in Eclipse's dtd editor. * Despite the changes in this commit, there should be no impact to current users of MessageSend. Change-Id: I6fb5b569657a02866a66ea352fd53d30d8ac965a	4 years ago
Alexander Traud	1adf9368ee	chan_sip: Filter pass-through audio/video formats away, again. Instead of looking for pass-through formats in the list of transcodable formats (which is going to find nothing), go through the result which is going to be the jointcaps of the tech_pvt of the channel. Finally, only with that list, ast_format_cap_remove(.) is going to succeed. This restores the behaviour of Asterisk 1.8. However, it does not fix ASTERISK_29282 because that issue report is about chan_sip and PJSIP. Here, only chan_sip is fixed because PJSIP does not even call ast_rtp_instance_available_formats -> ast_translate_available_format. Change-Id: Icade2366ac2b82935b95a9981678c987da2e8c34	4 years ago
Alexander Traud	45e48e387c	chan_sip: Allow [peer] without audio (text+video). Two previous commits, `620d9f4` and `6d980de`, allow to set up a call without audio, again. That was introduced originally with commit `f04d5fb` but changed and broke over time. The original commit missed one scenario: A [peer] section in sip.conf, which does not allow audio at all. In that case, chan_sip rejected the call, although even when the requester offered no audio. Now, chan_sip does not check whether there is no audio format but checks whether there is no format in general. In other words, if there is at least one format to offer, the call succeeds. However, to prevent calls with no-audio, chan_sip still rejects calls when both call parties (caller = requester of the call and callee = [peer] section in sip.conf) included audio. In such a case, it is expected that the call should have audio. ASTERISK-29280 Change-Id: I0fb74faf51ef22a60c10b467df6a4d1c1943b73e	4 years ago
Alexander Traud	87ad1138ff	chan_sip: Set up calls without audio (text+video), again. The previous commit `6d980de` fixed this issue in the core of Asterisk. With that, each channel technology can be used without audio theoretically. Practically, the channel-technology driver chan_sip turned out to have an invalid check preventing that. chan_sip tested whether there is at least one audio format. However, chan_sip has to test whether there is at least one format. More cannot be tested while requesting chan_sip because only the [general] capabilities but not the [peer] caps are known yet. And the [peer] caps might not be a subset or show any intersection with the [general] caps. This change here fixes this. The original commit `f04d5fb`, thirteen years ago, contained a software bug as it passed ANY audio capability to the channel-technology driver. Instead, it should have passed NO audio format. Therefore, this addressed issue here was not noticed in Asterisk 1.6.x and Asterisk 1.8. Then, Asterisk 10 changed that from ANY to NO, but nobody reported since then. ASTERISK-29265 Change-Id: Ic16a3bf13cd1b5c4fc4041ed74961177d96b600f	4 years ago
Alexander Traud	4c154f3431	chan_sip: SDP: Reject audio streams correctly. This completes the fix for ASTERISK_24543. Only when the call is an outgoing call, consult and append the configured format capabilities (p->caps). When all audio formats got rejected the negotiated format capabilities (p->jointcaps) contain no audio formats for incoming calls. This is required when there are other accepted media streams. ASTERISK-29258 Change-Id: I8bab31c7f3f3700dce204b429ad238a524efebb9	4 years ago
Alexander Traud	ad606d4ad1	chan_sip: SDP: Sidestep stream parsing when its media is disabled. Previously, chan_sip parsed all known media streams in an SDP offer like video (and text) even when videosupport=no (and textsupport=no). This wasted processor power. Furthermore, chan_sip accepted SDP offers, including no audio but just video (or text) streams although videosupport=no (or textsupport=no). Finally, chan_sip denied the whole offer instead of individual streams when they had encryption (SDES-sRTP) unexpectedly enabled. ASTERISK-29238 ASTERISK-29237 ASTERISK-29222 Change-Id: Ie49e4e2a11f0265f914b684738348ba8c0f89755	4 years ago
Alexander Traud	e884d935f6	chan_sip: Remove unused sip_socket->port. 12 years ago, with ASTERISK_12115 the last four get/uses of socket.port vanished. However, the struct member itself and all seven set/uses remained as dead code. ASTERISK-28798 Change-Id: Ib90516a49eca3d724a70191278aaf2144fb58c59	5 years ago
Alexander Traud	f86af1fbd0	Compiler fixes for GCC when printf %s is NULL ASTERISK-29146 Change-Id: Ib04bdad87d729f805f5fc620ef9952f58ea96d41	5 years ago
Alexander Traud	5b25c75d7b	chan_sip: On authentication, pick MD5 for sure. RFC 8760 added new digest-access-authentication schemes. Testing revealed that chan_sip does not pick MD5 if several schemes are offered by the User Agent Server (UAS). This change does not implement any of the new schemes like SHA-256. This change makes sure, MD5 is picked so UAS with SHA-2 enabled, like the service www.linphone.org/freesip, can still be used. This should have worked since day one because SIP/2.0 already envisioned several schemes (see RFC 3261 and its augmented BNF for 'algorithm' which includes 'token' as third alternative; note: if 'algorithm' was not present, MD5 is still assumed even in RFC 7616). Change-Id: I61ca0b1f74b5ec2b5f3062c2d661cafeaf597fcd	5 years ago
Dennis Buteyn	9058d9e591	chan_sip: Clear ToHost property on peer when changing to dynamic host The ToHost parameter was not cleared when a peer's host value was changed to dynamic. This causes invites to be sent to the original host. ASTERISK-29011 #close Change-Id: I9678d512741f71baca8f131a65b7523020b07d5c	5 years ago
Frederic LE FOLL	a423f935c9	chan_sip: chan_sip does not process 400 response to an INVITE. chan_sip handle_response() function, for a 400 response to an INVITE, calls handle_response_invite() and does not generate ACK. handle_response_invite() does not recognize 400 response and has no default response processing for unexpected responses, thus it does not generate ACK either. The ACK on response repetition comes from handle_response() mechanism "We must re-send ACKs to re-transmitted final responses". According to code history, 400 response specific processing was introduced with commit "channels/chan_sip: Add improved support for 4xx error codes" This commit added support for : - 400/414/493 in handle_response_subscribe() handle_response_register() and handle_response(). - 414/493 only in handle_response_invite(). This fix adds 400 response support in handle_response_invite(). ASTERISK-28957 Change-Id: Ic71a087e5398dfc7273946b9ec6f9a36960218ad	5 years ago
Alexander Traud	52f07176b6	chan_sip: externhost/externaddr with non-default TCP/TLS ports. ASTERISK-28372 Reported by: Anton Satskiy ASTERISK-24428 Reported by: sstream Change-Id: I2b7432a9bf3b09dc8515297ff955636db7a6224c	5 years ago
Alexander Traud	4d0ab620be	chan_sip: DiffServ/ToS not only on UDP but also on TCP and TLS sockets. ASTERISK-27195 Reported by: Joshua Roys Change-Id: I6e72ecb874200dec7a3865c7babaf5ac0d3101de	5 years ago
traud	da9554d925	chan_sip: TCP/TLS client without server. It is possible to configure a TCP/TLS client without having a TCP/TLS server. In that case, no error or warning was printed but the headers Contact and Via in SIP REGISTER were "(null)". ASTERISK-28798 Change-Id: I387ca5cb6a65f1eb675a29c5e41df8ec6c242ab2	5 years ago
Joshua C. Colp	1b6c58896f	chan_sip: Send 403 when ACL fails. Change-Id: I0910c79196f2b7c7e5ad6f1db95e83800ac737a2	5 years ago
Walter Doekes	43620cbf6c	chan_sip: Return 503 if we're out of RTP ports If you're for some reason out of RTP ports, chan_sip would previously responde to an INVITE with a 403, which will fail the call. Now, it returns a 503, allowing the device/proxy to retry the call on a different machine. ASTERISK-28718 Change-Id: I968dcf6c1e30ecddcce397dcda36db727c83ca90	5 years ago
Walter Doekes	711a3fed56	chan_sip: Always process updated SDP on media source change Fixes no-audio issues when the media source is changed and strictrtp is enabled (default). If the peer media source changes, the SDP session version also changes. If it is lower than the one we had stored, chan_sip would ignore it. This changeset keeps track of the remote media origin identifier, comparing that as well. If it changes, the session version needn't be higher for us to accept the SDP. Common scenario where this would've caused problems: a separate media gateway that informs the caller about premium rates before handing off the call to the final destination. (An alternative fix would be to set ignoresdpversion=yes on the peer.) ASTERISK-28686 Change-Id: I88fdbc5aeb777b583e7738c084254c482a7776ee	5 years ago
Sean Bright	f309b86e36	chan_sip.c: Stop handling continuation lines after reading headers lws2sws() does not stop trying to handle header continuation lines even after all headers have been found. This is problematic if the first character of a SIP message body is a space or tab character, so we update to recognize the end of the message header. ASTERISK-28693 #close Reported by: Frank Matano Change-Id: Idec8fa58545cd3fd898cbe0075d76c223f8d33df	5 years ago
Friendly Automation	2a8f759374	Merge "chan_sip: voice frames are no longer transmitted after emitting a COLP"	5 years ago
Friendly Automation	c3cf0e330c	Merge "chan_sip: in case of tcp/tls, be less annoying about tx errors."	6 years ago
Jaco Kroon	365d007eb6	chan_sip: in case of tcp/tls, be less annoying about tx errors. chan_sip.c:3782 __sip_xmit: sip_xmit of 0x7f1478069230 (len 600) to 213.150.203.60:1492 returned -2: Interrupted system call returned -2 implies this wasn't actually an OS error, so errno makes no sense either. Internal error was already logged higher up, and -2 generally means that either there isn't a valid connection available, or the pipe notification failed, and that is already correctly logged. ASTERISK-28651 #close Change-Id: I46eb82924beeff9dfd86fa6c7eb87d2651b950f2 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	6 years ago
Jean Aunis	9c9296c635	chan_sip: voice frames are no longer transmitted after emitting a COLP The SIP transaction state was reset when emitting an UPDATE or a re-INVITE related to a COLP, preventing RTP packets to be emitted. ASTERISK-28647 Change-Id: Ie7a30fa7a97f711e7ba6cc17f221a0993d48bd8b	6 years ago
Frederic LE FOLL	7624cbb155	chan_sip+native_bridge_rtp: no directmedia for ptime other than default ptime. During capabilities selection (joint capabilities of us and peer, configured capability for this peer, or general configured capabilities), if sip_new() does not keep framing information, then directmedia activation will fail for any framing different from default framing. ASTERISK-28637 Change-Id: I99257502788653c2816fc991cac7946453082466	6 years ago
Ben Ford	4a1cadeadb	chan_sip.c: Prevent address change on unauthenticated SIP request. If the name of a peer is known and a SIP request is sent using that peer's name, the address of the peer will change even if the request fails the authentication challenge. This means that an endpoint can be altered and even rendered unusuable, even if it was in a working state previously. This can only occur when the nat option is set to the default, or auto_force_rport. This change checks the result of authentication first to ensure it is successful before setting the address and the nat option. ASTERISK-28589 #close Change-Id: I581c5ed1da60ca89f590bd70872de2b660de02df	6 years ago
Sean Bright	32ce6e9a06	channels: Allow updating variable value When modifying an already defined variable in some channel drivers they add a new variable with the same name to the list, but that value is never used, only the first one found. Introduce ast_variable_list_replace() and use it where appropriate. ASTERISK-23756 #close Patches: setvar-multiplie.patch submitted by Michael Goryainov Change-Id: Ie1897a96c82b8945e752733612ee963686f32839	6 years ago
George Joseph	5fb9b23105	chan_sip: Update links referenced in deprecation notice The links in the deprecation notice were the shortened variety but it makes better sense to show the unshortened links as they're more descriptive. I.E. wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip rather than wiki.asterisk.org/wiki/x/tAHOAQ Change-Id: If2da5d5243e2d4a6f193b15691d23e7e5a7c57a9	6 years ago
Kevin Harwell	857ee76f4b	Merge "MWI: Update modules that subscribe to MWI to use new API calls"	6 years ago
Francesco Castellano	8438d19b81	chan_sip: Handle invalid SDP answer to T.38 re-invite The chan_sip module performs a T.38 re-invite using a single media stream of udptl, and expects the SDP answer to be the same. If an SDP answer is received instead that contains an additional media stream with no joint codec a crash will occur as the code assumes that at least one joint codec will exist in this scenario. This change removes this assumption. ASTERISK-28465 Change-Id: I8b02845b53344c6babe867a3f0a5231045c7ac87	6 years ago
Kevin Harwell	9637e1dfdc	MWI: Update modules that subscribe to MWI to use new API calls The MWI core recently got some new API calls that make tracking MWI state lifetime more reliable. This patch updates those modules that subscribe to specific MWI topics to use the new API. Specifically, these modules now subscribe to both MWI topics and MWI state. ASTERISK-28442 Change-Id: I32bef880b647246823dbccdf44a98d384fcabfbd	6 years ago
Kevin Harwell	ff0d0ac23a	mwi core: Move core MWI functionality into its own files There is enough MWI functionality to warrant it having its own 'c' and header files. This patch moves all current core MWI data structures, and functions into the following files: main/mwi.h main/mwi.c Note, code was simply moved, and not modified. However, this patch is also in preparation for core MWI changes, and additions to come. Change-Id: I9dde8bfae1e7ec254fa63166e090f77e4d3097e0	6 years ago
Ben Ford	dd1cc7791c	build: Fix compiler warnings/errors. The compiler complained about a couple of variables that weren't initialized but were being used. Initializing them to NULL resolves the warnings/errors. ASTERISK-28362 #close Change-Id: I6243afc5459b416edff6bbf571b0489f6b852e4b	6 years ago
Sean Bright	1499640da9	chan_sip: Ensure 'qualifygap' isn't negative Passing negative intervals to the scheduler rips a hole in the space-time continuum. ASTERISK-25792 #close Reported by: Paul Sandys Change-Id: Ie706f21cee05f76ffb6f7d89e9c867930ee7bcd7	6 years ago
Sean Bright	2473b791b9	Replace calls to strtok() with strtok_r() strtok() uses a static buffer, making it not thread safe. Also add a #define to cause a compile failure if strtok is used. Change-Id: Icce265153e1e65adafa8849334438ab6d190e541	6 years ago
Giuseppe Sucameli	0bde3751a0	chan_sip: Fix leak using contact ACL Free old peer's contactacl before overwrite it within build_peer. ASTERISK-28194 Change-Id: Ie580db6494e50cee0e2a44b38e568e34116ff54c	7 years ago
Corey Farrell	021ce938ca	astobj2: Remove legacy ao2_container_alloc routine. Replace usage of ao2_container_alloc with ao2_container_alloc_hash or ao2_container_alloc_list. Remove ao2_container_alloc macro. Change-Id: I0907d78bc66efc775672df37c8faad00f2f6c088	7 years ago
Joshua Colp	3077ad0c24	stasis: Add internal filtering of messages. This change adds the ability for subscriptions to indicate which message types they are interested in accepting. By doing so the filtering is done before being dispatched to the subscriber, reducing the amount of work that has to be done. This is optional and if a subscriber does not add message types they wish to accept and set the subscription to selective filtering the previous behavior is preserved and they receive all messages. There is also the ability to explicitly force the reception of all messages for cases such as AMI or ARI where a large number of messages are expected that are then generically converted into a different format. ASTERISK-28103 Change-Id: I99bee23895baa0a117985d51683f7963b77aa190	7 years ago
Jasper Hafkenscheid	2cf5079205	chan_sip: Attempt ast_do_pickup in handle_invite_replaces When a call pickup is performed using and invite with replaces header the ast_do_pickup method is attempted and a PICKUP stasis message is sent. ASTERISK-28081 #close Reported-by: Luit van Drongelen Change-Id: Ieb1442027a3ce6ae55faca47bc095e53972f947a	7 years ago
Corey Farrell	90a11c4ae7	chan_sip deprecation. This officially deprecates chan_sip in Asterisk 17+. A warning is printed upon startup or module load to tell users that they should consider migrating. chan_sip is still built by default but the default modules.conf skips loading it at startup. Very important to note we are not scheduling a time where chan_sip will be removed. The goal of this change is to accurately inform end users of the current state of chan_sip and encourage movement to the fully supported chan_pjsip. Change-Id: Icebd8848f63feab94ef882d36b2e99d73155af93	7 years ago
Corey Farrell	687ab7aeee	astobj2: Eliminate legacy container allocation macros. These macros have been documented as legacy for a long time but are still used in new code because they exist. Remove all references to: * ao2_container_alloc_options * ao2_t_container_alloc_options * ao2_t_container_alloc These macro's are also removed. Only ao2_container_alloc remains due to it's use in over 100 places. Change-Id: I1a26258b5bf3deb081aaeed11a0baa175c933c7a	7 years ago
pk16208	6627c56b3d	chan_sip: SipNotify on Chan_Sip vi AMI behave different to CLI With tls and udp enabled asterisk generates a warning about sending message via udp instead of tls. sip notify command via cli works as expected and without warning. asterisk has to set the connection information accordingly to connection and not on presumption ASTERISK-28057 #close Change-Id: Ib43315aba1f2c14ba077b52d8c5b00be0006656e	7 years ago

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3897 Commits (1ddfb7551a42fce6165726b0d5834668c3fdbcf6)