asterisk

Commit Graph

Author	SHA1	Message	Date
Sean Bright	0d62735f99	utils.c: Remove all usages of ast_gethostbyname() gethostbyname() and gethostbyname_r() are deprecated in favor of getaddrinfo() which we use in the ast_sockaddr family of functions. ASTERISK-29819 #close Change-Id: Ie277c0ef768d753b169c121ef570a71665692ab7	3 years ago
Naveen Albert	97f400100c	chan_sip: Fix crash when accessing RURI before initiating outgoing call Attempting to access ${CHANNEL(ruri)} in a pre-dial handler before initiating an outgoing call will cause Asterisk to crash. This is because a null field is accessed, resulting in an offset from null and subsequent memory access violation. Since RURI is not guaranteed to exist, we now check if the base pointer is non-null before calculating an offset. ASTERISK-29772 Change-Id: Icd3b02f07256bbe6615854af5717074087b95a83	3 years ago
Alexander Traud	cc025026b7	progdocs: Fix for Doxygen, the hidden parts. ASTERISK-29779 Change-Id: If338163488498f65fa7248b60e80299c0a928e4b	3 years ago
Alexander Traud	affe7ee879	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	3 years ago
Naveen Albert	24a04054ad	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	3 years ago
Alexander Traud	726d6dd166	channels: Fix for Doxygen. ASTERISK-29762 Change-Id: Ia8811ac12b93ff8c18164699c6fbc604cb0a23f7	3 years ago
Alexander Traud	4051434be4	chan_iax2: Fix for Doxygen. ASTERISK-29737 Change-Id: I282003cc553989fd5c19ceeac9e478fa4ee06cec	3 years ago
Josh Soref	d46ba42910	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	3 years ago
Naveen Albert	f9ba1ee7c9	sig_analog: Fix truncated buffer copy Fixes compiler warning caused by a truncated copy of the ANI2 into a buffer of size 10. This could prevent the null terminator from being copied if the copy value exceeds the size of the buffer. This increases the buffer size to 101 to ensure there is no way for truncation to occur. ASTERISK-29702 #close Change-Id: Ief9052212952840fa44de6463b8699fdb3e163d0	4 years ago
Naveen Albert	df9aeea4c8	chan_iax2: Allow both secret and outkey at dial time Historically, the dial syntax for IAX2 has held that an outkey (used only for RSA authenticated calls) and a secret (used only for plain text and MD5 authenticated calls, historically) were mutually exclusive, and thus the same position in the dial string was used for both values. Now that encryption is possible with RSA authentication, this poses a limitation, since encryption requires a secret and RSA authentication requires an outkey. Thus, the dial syntax is extended so that both a secret and an outkey can be specified. The new extended syntax is backwards compatible with the old syntax. However, a secret can now be specified after the outkey, or the outkey can be specified after the secret. This makes it possible to spawn an encrypted RSA authenticated call without a corresponding peer being predefined in iax.conf. ASTERISK-29707 #close Change-Id: I1f8149313ed760169d604afbb07720a8b07dd00e	4 years ago
Mike Bradeen	99a1a427a9	various: Fix GCC 11 compilation issues. test_voicemail_api: Use empty char* for empty_msg_ids. chan_skinny: Fix size of calledParty to be maximum extension. menuselect: Change Makefile to stop deprecated warnings. Added comments test_linkedlist: 'bogus' variable was manually allocated from a macro and the test fails if this happens but the compiler couldn't 'see' this and returns a warning. memset to all 0's after allocation. chan_ooh323: Fixed various indentation issues that triggered misleading indentation warnings. ASTERISK-29682 Reported by: George Joseph Change-Id: If4fe42222c8444dc16828a42731ee53b4ce5cbbe	4 years ago
Naveen Albert	7ff6c43760	chan_iax2: Add encryption for RSA authentication Adds support for encryption to RSA-authenticated calls. Also prevents crashes if an RSA IAX2 call is initiated to a switch requiring encryption but no secret is provided. ASTERISK-20219 Change-Id: I18f1f9d7c59b4f9cffa00f3b94a4c875846efd40	4 years ago
Naveen Albert	5b5c358e4b	res_pjsip_caller_id: Add ANI2/OLI parsing Adds parsing of ANI II digits (Originating Line Information) to PJSIP, on par with what currently exists in chan_sip. ASTERISK-29472 Change-Id: Ifc938a7a7d45ce33999ebf3656a542226f6d3847	4 years ago
Naveen Albert	3072c540bb	chan_iax2: Add ANI2/OLI information element Adds an information element for ANI2 so that Originating Line Information can be transmitted over IAX2 channels. ASTERISK-29605 #close Change-Id: Iaeacdf6ccde18eaff7f776a0f49fee87dcb549d2	4 years ago
Sarah Autumn	466eb4a52b	sig_analog: Changes to improve electromechanical signalling compatibility This changeset is intended to address compatibility issues encountered when interfacing Asterisk to electromechanical telephone switches that implement ANI-B, ANI-C, or ANI-D. In particular the behaviours that this impacts include: - FGC-CAMA did not work at all when using MF signaling. Modified the switch case block to send calls to the correct part of the signaling-handling state machine. - For FGC-CAMA operation, the delay between called number ST and second wink for ANI spill has been made configurable; previously all calls were made to wait for one full second. - After the ANI spill, previous behavior was to require a 'ST' tone to advance the call. This has been changed to allow 'STP' 'ST2P' or 'ST3P' as well, for compatibility with ANI-D. - Store ANI2 (ANI INFO) digits in the CALLERID(ANI2) channel variable. - For calls with an ANI failure, No. 1 Crossbar switches will send forward a single-digit failure code, with no calling number digits and no ST pulse to terminate the spill. I've made the ANI timeout configurable so to reduce dead air time on calls with ANI fail. - ANI info digits configurable. Modern digital switches will send 2 digits, but ANI-B sends only a single info digit. This caused the ANI reported by Asterisk to be misaligned. - Changed a confusing log message to be more informative. ASTERISK-29518 Change-Id: Ib7e27d987aee4ed9bc3663c57ef413e21b404256	4 years ago
Naveen Albert	0ca3ebe7cd	chan_alsa, chan_sip: Add replacement to moduleinfo Adds replacement modules to the moduleinfo for chan_alsa and chan_sip. ASTERISK-29601 #close Change-Id: I7a4877b0d5c0c17e088e8fa8ebbfa9a195223cbc	4 years ago
Joshua C. Colp	20b2741232	chan_vpb: Remove deprecated module. ASTERISK-29597 Change-Id: I19bb39eed0257ddfef453eb2df5646d073d50fe1	4 years ago
Joshua C. Colp	1eb2d85c99	chan_misdn: Remove deprecated module. ASTERISK-29596 Change-Id: Ibae9490c1b35cadbf7028d24610f745277c8535e	4 years ago
Joshua C. Colp	6ecc48086c	chan_nbs: Remove deprecated module. ASTERISK-29595 Change-Id: Ib5c7d43a780f2fb94cee90738e4c1af211ae4a33	4 years ago
Joshua C. Colp	6cc948f94e	chan_phone: Remove deprecated module. ASTERISK-29594 Change-Id: I79a9961cb5062fadbccb0ea93f087bdd32685316	4 years ago
Joshua C. Colp	95f3a4a9ad	chan_oss: Remove deprecated module. ASTERISK-29593 Change-Id: Ib53a42ad974c63871344b95078c61c188e43da99	4 years ago
Sean Bright	743e057bb4	mgcp: Remove dead debug code ASTERISK-20339 #close Change-Id: I36f364aaa1971241d8f3ea1a5909b463d185a2d5	4 years ago
Joshua C. Colp	93870e7bb4	policy: Deprecate modules and add versions to others. 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. app_macro was deprecated in 16, to be removed in 21. chan_sip was deprecated in 17, to be removed in 21. res_monitor was deprecated in 16, to be removed in 21. ASTERISK-29548 ASTERISK-29549 ASTERISK-29550 ASTERISK-29551 ASTERISK-29552 ASTERISK-29553 ASTERISK-29558 ASTERISK-29567 ASTERISK-29572 Change-Id: Ic3bee31a10d42c4b3bbc913d893f7b2a28a27131	4 years ago
Kevin Harwell	1b62831f2c	AST-2021-008 - chan_iax2: remote crash on unsupported media format If chan_iax2 received a packet with an unsupported media format, for example vp9, then it would set the frame's format to NULL. This could then result in a crash later when an attempt was made to access the format. This patch makes it so chan_iax2 now ignores/drops frames received with unsupported media format types. ASTERISK-29392 #close Change-Id: Ifa869a90dafe33eed8fd9463574fe6f1c0ad3eb1	4 years ago
Naveen Albert	fd40752954	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
Sean Bright	1b41629447	chan_pjsip: Correct misleading trace message ASTERISK-29358 #close Change-Id: I050daff67066873df4e8fc7f4bd977c1ca06e647	4 years ago
George Joseph	09303e8e22	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
Joshua C. Colp	149e5e5b86	xml: Embed module information into core XML documentation. This change embeds the MODULEINFO block of modules into the core XML documentation. This provides a shared mechanism for use by both menuselect and Asterisk for information and a definitive source of truth. ASTERISK-29335 Change-Id: Ifbfd5c700049cf320a3e45351ac65dd89bc99d90	4 years ago
Alexander Traud	8c461845c8	chan_iax2: System Header strings is included via asterisk.h/compat.h. The system header strings was included mistakenly with commit `3de0204`. That header is included via asterisk.h and there via the compat.h. Change-Id: I3dc49060e275295f785670c87cc65fd3c3abd24a	4 years ago
Holger Hans Peter Freyther	48ed4f670f	pjsip: Generate progress (once) when receiving a 180 with a SDP ASTERISK-29105 Change-Id: If1615fe7115fe544ef974b044d3cea5c48b94a38	4 years ago
Alexander Traud	5894535fed	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	1f77c33c02	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
George Joseph	91b0778791	chan_iax2.c: Require secret and auth method if encryption is enabled If there's no secret specified for an iax2 peer and there's no secret specified in the dial string, Asterisk will crash if the auth method requested by the peer is MD5 or plaintext. You also couldn't specify a default auth method in the [general] section of iax.conf so if you don't have static peers defined and just use the dial string, Asterisk will still crash even if you have a secret specified in the dial string. * Added logic to iax2_call() and authenticate_reply() to print a warning and hanhup the call if encryption is requested and there's no secret or auth method. This prevents the crash. * Added the ability to specify a default "auth" in the [general] section of iax.conf. ASTERISK-29624 Reported by: N A Change-Id: I5928e16137581f7d383fcc7fa04ad96c919e6254	4 years ago
Alexander Traud	620d9f4782	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
Dan Cropp	55891227e8	chan_pjsip, app_transfer: Add TRANSFERSTATUSPROTOCOL variable When a Transfer/REFER is executed, TRANSFERSTATUSPROTOCOL variable is 0 when no protocl specific error SIP example of failure, 3xx-6xx for the SIP error code received This allows applications to perform actions based on the failure reason. ASTERISK-29252 #close Reported-by: Dan Cropp Change-Id: Ia6a94784b4925628af122409cdd733c9f29abfc4	4 years ago
Alexander Traud	4aff42b274	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
Ben Ford	948ceb1228	chan_pjsip.c: Add parameters to frame in indicate. There are a couple of parameters (datalen and data) that do not get set in chan_pjsip_indicate which could cause an Invalid message to pop up for things such as fax. This patch adds them to the frame. Change-Id: Ia51be086a0708be905e73d1f433572c49c7e38f8	4 years ago
Alexander Traud	1c05667cfc	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
Ivan Poddubnyi	f2aa6c7017	chan_pjsip: Assign SIPDOMAIN after creating a channel session->channel doesn't exist until chan_pjsip creates it, so intead of setting a channel variable every new incoming call sets one and the same global variable. This patch moves the code to chan_pjsip so that SIPDOMAIN is set on a newly created channel, it also removes a misleading reference to channel->session used to fetch call pickup configuraion. ASTERISK-29240 Change-Id: I90c9bbbed01f5d8863585631a29322ae4e046755	4 years ago
Ivan Poddubnyi	134d2e729d	chan_pjsip: Stop queueing control frames twice on outgoing channels The fix for ASTERISK-27902 made chan_pjsip process SIP responses twice. This resulted in extra noise in logs (for example, "is making progress" and "is ringing" get logged twice by app_dial), as well as in noise in signalling: one incoming 183 Session Progress results in 2 outgoing 183-s. This change splits the response handler into 2 functions: - one for updating HANGUPCAUSE, which is still called twice, - another that does the rest, which is called only once as before. ASTERISK-28016 Reported-by: Alex Hermann ASTERISK-28549 Reported-by: Gant Liu ASTERISK-28185 Reported-by: Julien Change-Id: I0a1874be5bb5ed12d572d17c7f80de6e5e542940	4 years ago
Dan Cropp	ffa87ecade	chan_pjsip: Incorporate channel reference count into transfer_refer(). Add channel reference count for PJSIP REFER. The call could be terminated prior to the result of the transfer. In that scenario, when the SUBSCRIBE/NOTIFY occurred several minutes later, it would attempt to access a session which was no longer valid. Terminate event subscription if pjsip_xfer_initiate() or pjsip_xfer_send_request() fails in transfer_refer(). ASTERISK-29201 #close Reported-by: Dan Cropp Change-Id: I3fd92fd14b4e3844d3d7b0f60fe417a4df5f2435	4 years ago
Alexander Traud	80c14f74bc	codecs: Remove test-law. This was dead code, test code introduced with Asterisk 13. This was found while analyzing ASTERISK_28416 and ASTERISK_29185. This change partly fixes, not closes those two issues. Change-Id: I42d0daa37f6f334c7d86672f06f085858a3f3940	4 years ago
Richard Mudgett	058bc0d593	chan_vpb.cc: Fix compile errors. Fix the usual compile problem when someone adds a new callback to struct ast_channel_tech. Change-Id: I9bdeb8a8cc65f03b2d6e4f2eb5809af47c906c32	4 years ago
Joshua C. Colp	6475fe3dd7	pjsip: Match lifetime of INVITE session to our session. In some circumstances it was possible for an INVITE session to be destroyed while we were still using it. This occurred due to the reference on the INVITE session being released internally as a result of its state changing to DISCONNECTED. This change adds a reference to the INVITE session which is released when our own session is destroyed, ensuring that the INVITE session remains valid for the lifetime of our session. ASTERISK-29022 Change-Id: I300c6d9005ff0e6efbe1132daefc7e47ca6228c9	4 years ago
Alexander Traud	103d7da3bb	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	4 years ago
Alexander Traud	57ee79a563	Compiler fixes for GCC with -Og ASTERISK-29144 Change-Id: I2a72c072083b4492a223c6f9d73d21f4f424db62	5 years ago
Alexander Traud	28faafd1c4	Compiler fixes for GCC when printf %s is NULL ASTERISK-29146 Change-Id: Ib04bdad87d729f805f5fc620ef9952f58ea96d41	5 years ago
Alexander Traud	cd32317691	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
Sean Bright	52ca2323aa	chan_sip.c: Don't build by default ASTERISK-29083 #close Change-Id: I9ea25fba3ba8f63a47886894bd966e0f08d5e003	5 years ago
Sean Bright	5a0e1d256d	audiosocket: Fix module menuselect descriptions The module description needs to be on the same line as the AST_MODULE_INFO or it is not parsed correctly. Change-Id: I9ba11df1415369790e8656fcb527bb2749373c21	5 years ago

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8662 Commits (bc59b66de32e320a959118a7315426e02127f101)