8640 Commits (93870e7bb4d5ef2cf5f9f06c110d7d45221cf067)

Author	SHA1	Message	Date
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
George Joseph	44bb0858cb	debugging: Add enough to choke a mule ... Added to: * bridges/bridge_softmix.c * channels/chan_pjsip.c * include/asterisk/res_pjsip_session.h * main/channel.c * res/res_pjsip_session.c There NO functional changes in this commit. Change-Id: I06af034d1ff3ea1feb56596fd7bd6d7939dfdcc3	5 years ago
Kevin Harwell	3c4a1722b6	chan_pjsip: disallow PJSIP_SEND_SESSION_REFRESH pre-answer execution ... This patch makes it so if the PJSIP_SEND_SESSION_REFRESH dialplan function is called on a channel prior to answering a warning is issued and the function returns unsuccessful. ASTERISK-28878 #close Change-Id: I053f767d10cf3b2b898fa9e3e7c35ff07e23c9bb	5 years ago
Dennis Buteyn	aab666bb9d	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
George Joseph	9bd1d686a1	ACN: Add tracing to existing code ... Prior to making any modifications to the pjsip infrastructure for ACN, I've added the tracing functions to the existing code. This should make the final commit easier to review, but we can also now run a "before and after" trace. No functional changes were made with this commit. Change-Id: Ia83a1a2687ccb96f2bc8a2a3928a5214c4be775c	5 years ago
Kevin Harwell	4eba6b9eb2	PJSIP_MEDIA_OFFER: override configuration on refresh ... When using the PSJIP_MEDIA_OFFER dialplan function it was not overriding an endpoint's configured codecs on refresh unless they had a shared codec between the two. This patch makes it so whatever is set using PJSIP_MEDIA_OFFER is used when creating the SDP for a refresh no matter what. ASTERISK-28878 #close Change-Id: I0f7dc86fd0fb607c308e6f98ede303c54d1eacb6	5 years ago
George Joseph	8d1064eaaf	Streams: Add features for Advanced Codec Negotiation ... The Streams API becomes the home for the core ACN capabilities. These include... * Parsing and formatting of codec negotation preferences. * Resolving pending streams and topologies with those configured using configured preferences. * Utility functions for creating string representations of streams, topologies, and negotiation preferences. For codec negotiation preferences: * Added ast_stream_codec_prefs_parse() which takes a string representation of codec negotiation preferences, which may come from a pjsip endpoint for example, and populates a ast_stream_codec_negotiation_prefs structure. * Added ast_stream_codec_prefs_to_str() which does the reverse. * Added many functions to parse individual parameter name and value strings to their respectrive enum values, and the reverse. For streams: * Added ast_stream_create_resolved() which takes a "live" stream and resolves it with a configured stream and the negotiation preferences to create a new stream. * Added ast_stream_to_str() which create a string representation of a stream suitable for debug or display purposes. For topology: * Added ast_stream_topology_create_resolved() which takes a "live" topology and resolves it, stream by stream, with a configured topology stream and the negotiation preferences to create a new topology. * Added ast_stream_topology_to_str() which create a string representation of a topology suitable for debug or display purposes. * Renamed ast_format_caps_from_topology() to ast_stream_topology_get_formats() to be more consistent with the existing ast_stream_get_formats(). Additional changes: * A new function ast_format_cap_append_names() appends the results to the ast_str buffer instead of replacing buffer contents. Change-Id: I2df77dedd0c72c52deb6e329effe057a8e06cd56	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
Kevin Harwell	8b925fbda3	chan_pjsip: don't use PJSIP_SC_NULL as it only exists pjproject 2.8+ ... A patch made a reference to the PJSIP_SC_NULL enumeration value, which was added to pjproject 2.8 and above thus making it so Asterisk would fail to compile with prior versions of pjproject. This patch removes the reference, and instead initializes the value to '0'. ASTERISK-28886 #close Change-Id: I68491c80da1a0154b2286c9458440141c98db9d7	5 years ago
Guido Falsi	d88e230037	chan_dadhi: Fix setvar in dahdi channels ... The change to how setvar works for various channels performed in ASTERISK~23756 missed some required change in the dahdi channel, where the variables are actually set while reading configuration. This change should fix the issue. ASTERISK-28955 Change-Id: Ibfeb7f8cbdd735346dc4028de6a265f24f9df274	5 years ago
George Joseph	41f3a7da4d	res_fax: Don't start a gateway if either channel is hung up ... When fax_gateway_framehook is called and a gateway hasn't already been started, the framehook gets the t38 state for both the current channel and the peer. That call trickles down to the channel driver which determines the state. If either channel is hung up (or in the process of being hung up), the channel driver's tech_pvt is going to be NULL which, in the case of chan_pjsip, will cause a segfault. * Added a hangup check for both the channel and peer channel before starting a fax gateway. * Added a check for NULL tech_pvt to chan_pjsip_queryoption so we don't attempt to reference a tech_pvt that's already gone. ASTERISK-28923 Reported by: Yury Kirsanov Change-Id: I4e10e63b667bbb68c1c8623f977488f5d807897c	5 years ago
Joshua C. Colp	1c5e68580a	stream: Enforce formats immutability and ensure formats exist. ... Some places in Asterisk did not treat the formats on a stream as immutable when they are. The ast_stream_get_formats function is now const to enforce this and parts of Asterisk have been updated to take this into account. Some violations of this were also fixed along the way. An additional minor tweak is that streams are now allocated with an empty format capabilities structure removing the need in various places to check that one is present on the stream. ASTERISK-28846 Change-Id: I32f29715330db4ff48edd6f1f359090458a9bfbe	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
Kevin Harwell	fa3c8f94e0	chan_pjsip: digit_begin - constant DTMF tone if RTP is not setup yet ... If chan_pjsip is configured for DTMF_RFC_4733, and the core triggers a digit begin before media, or rtp has been setup then it's possible the outgoing channel will hear a constant DTMF tone upon answering. This happens because when there is no media, or rtp chan_pjsip notifies the core to initiate inband DTMF. However, upon digit end if media, and rtp become available then chan_pjsip does not notify the core to stop inband DTMF. Thus the tone continues playing. This patch makes it so chan_pjsip only notifies the core to start inband DTMF in only the required cases. Now if there is no media, or rtp availabe upon digit begin chan_pjsip does nothing, but tells the core it handled it. ASTERISK-28817 #close Change-Id: I0dbea9fff444a2595fb18c64b89653e90d2f6eb5	5 years ago
traud	b38f664250	chan_unistim: Avoid tautological warnings with clang. ... ASTERISK-28803 Change-Id: I15449621b68d0ad4d57b7c337c1167adb15135af	5 years ago
Joshua C. Colp	1b6c58896f	chan_sip: Send 403 when ACL fails. ... Change-Id: I0910c79196f2b7c7e5ad6f1db95e83800ac737a2	5 years ago
Michael Neuhauser	5562fb2ea0	chan_psip, res_pjsip_sdp_rtp: ignore rtptimeout if direct-media is active ... Do not hang up a PJSIP channel on RTP timeout if that channel is in a direct-media bridge. Also reset the time of the last received RTP packet when direct-media ends (wait full rtp_timeout period before checking first time after audio came back to Asterisk). ASTERISK-28774 Reported-by: Michael Neuhauser Change-Id: I8b62012be7685849e8fb2b1c5dd39d35313ca2d1	5 years ago
Sean Bright	49cf84578e	chan_vpb: Fix 'catching polymorphic type ... by value' error ... Fixes the following compile error: chan_vpb.cc:2688:26: error: catching polymorphic type ‘class std::exception’ by value Change-Id: Ic87bc357d72427d77626735c83200fd278a7a649	5 years ago
Paulo Vicentini	ed2a7e3eaf	chan_pjsip: Check audio frame when remote SSRC changes. ... If the SSRC of a received RTP packet differed from the previous SSRC an SSRC change control frame would be queued ahead of the media frame. In the case of audio this would result in the format of the audio frame not being checked, and if it differed or was not allowed then it could cause the call to drop due to failure to set up a translation path. The chan_pjsip module will now no longer assume the first frame will be the audio frame and instead goes through the complete list to find it. ASTERISK-28759 Change-Id: I6d854cc523f343e299a615636fc65bdbd5f809ec	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
Friendly Automation	f29ddd8925	Merge "chan_sip: Always process updated SDP on media source change"	5 years ago