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Author	SHA1	Message	Date
Sean Bright	a5d55fc9e1	sip_to_pjsip.py: Handle #include globs and other fixes ... * Wildcards in #includes are now properly expanded * Implement operators for Section class to allow sorting ASTERISK-29142 #close Change-Id: I9b9cd95f4cbe5c24506b75d17173c5aa1a83e5df	5 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	914aecb8d8	Compiler fixes for GCC with -Os ... ASTERISK-29145 Change-Id: I9af705f2b9725c53141aef5d0ff512a1800f073c	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
Walter Doekes	1650d50e91	main/say: Work around gcc 9 format-truncation false positive ... Version: gcc (Ubuntu 9.3.0-10ubuntu2) 9.3.0 Warning: say.c:2371:24: error: ‘%d’ directive output may be truncated writing between 1 and 11 bytes into a region of size 10 [-Werror=format-truncation=] 2371 | snprintf(buf, 10, "%d", num); say.c:2371:23: note: directive argument in the range [-2147483648, 9] That's not possible though, as the if() starts out checking for (num < 0), making this Warning a false positive. (Also replaced some else<TAB>if with else<SP>if while in the vicinity.) Change-Id: Ic7a70120188c9aa525a6d70289385bfce878438a	5 years ago
Kevin Harwell	c62193c5de	res_pjsip, res_pjsip_session: initialize local variables ... This patch initializes a couple of local variables to some default values. Interestingly, in the 'pj_status_t dlg_status' case the value not being initialized caused memory to grow, and not be recovered, in the off nominal path (at least on my machine). Change-Id: I22ee65e1e1bff8efacea8a167c6c8428898523f7	5 years ago
Alexander Traud	f3452c85e5	install_prereq: Add GMime 3.0. ... Ubuntu 20.10 does not come with GMime 2.6. Ubuntu 16.04 LTS does not come with GMime 3.0. aptitude ignores any missing package. Therefore, it installs the correct package(s). However, in Ubuntu 18.04 LTS and Ubuntu 20.04 LTS, both versions are installed alongside although only one is really needed. Change-Id: Ic58aa9f2e131d94671f286f17dbd61e1ccbabcb7	5 years ago
Alexander Traud	db4320a6a0	BuildSystem: Enable Lua 5.4. ... Note to maintainers: Lua 5.4, Lua 5.3, and Lua 5.2 have not been tested at runtime with pbx_lua. Until then, use the lowest available version of Lua, if you enabled the module pbx_lua at all. Change-Id: Ie5270448b11fcb4e2a53d899e4fe7fea793ce7e0	5 years ago
Nick French	bd98e153d1	res_pjsip_session: Restore calls to ast_sip_message_apply_transport() ... Commit 44bb0858cb ("debugging: Add enough to choke a mule") accidentally removed calls to ast_sip_message_apply_transport when it was attempting to just add debugging code. The kiss of death was saying that there were no functional changes in the commit comment. This makes outbound calls that use the 'flow' transport mechanism fail, since this call is used to relay headers into the outbound INVITE requests. ASTERISK-29124 #close Change-Id: I0f3e32c2e8ac415e30b1d29966d75a1546f0526a	5 years ago
Sean Bright	8f33e23dfb	features.conf.sample: Sample sound files incorrectly quoted ... ASTERISK-29136 #close Change-Id: I3186536d65a50014c8da4780c9224919caa81440	5 years ago
Andrew Siplas	0190e706b8	logger.conf.sample: add missing comment mark ... Add missing comment mark from stock configuration. ASTERISK-29123 #close Change-Id: I4f94eb4544166bca8af4c17fd11edee3c6980620	5 years ago
Joshua C. Colp	dcd2ed69a3	res_pjsip: Adjust outgoing offer call pref. ... This changes the outgoing offer call preference default option to match the behavior of previous versions of Asterisk. The additional advanced codec negotiation options have also been removed from the sample configuration and marked as reserved for future functionality in XML documentation. The codec preference options have also been fixed to enforce local codec configuration. ASTERISK-29109 Change-Id: Iad19347bd5f3d89900c15ecddfebf5e20950a1c2	5 years ago
Sean Bright	fa023cbfa0	tcptls.c: Don't close TCP client file descriptors more than once ... ASTERISK-28430 #close Change-Id: Ib556b0a0c95cca939e956886214ec8d828d89606	5 years ago
Jean Aunis	61116d5dbc	resource_endpoints.c: memory leak when providing a 404 response ... When handling a send_message request to a non-existing endpoint, the response's body is overriden and not properly freed. ASTERISK-29108 Change-Id: Ie1d3d70065f80793445b60f5e4a7eb31b4b9c5c8	5 years ago
Kevin Harwell	56028426de	Logging: Add debug logging categories ... Added debug logging categories that allow a user to output debug information based on a specified category. This lets the user limit, and filter debug output to data relevant to a particular context, or topic. For instance the following categories are now available for debug logging purposes: dtls, dtls_packet, ice, rtcp, rtcp_packet, rtp, rtp_packet, stun, stun_packet These debug categories can be enable/disable via an Asterisk CLI command. While this overrides, and outputs debug data, core system debugging is not affected by this patch. Statements still output at their appropriate debug level. As well backwards compatibility has been maintained with past debug groups that could be enabled using the CLI (e.g. rtpdebug, stundebug, etc.). ASTERISK-29054 #close Change-Id: I6e6cb247bb1f01dbf34750b2cd98e5b5b41a1849	5 years ago
Sean Bright	51cba591e3	pbx.c: On error, ast_add_extension2_lockopt should always free 'data' ... In the event that the desired extension already exists, ast_add_extension2_lockopt() will free the 'data' it is passed before returning an error, so we should not be freeing it ourselves. Additionally, there were two places where ast_add_extension2_lockopt() could return an error without also freeing the 'data' pointer, so we add that. ASTERISK-29097 #close Change-Id: I904707aae55169feda050a5ed7c6793b53fe6eae	5 years ago
George Joseph	773f424c7f	app_confbridge/bridge_softmix: Add ability to force estimated bitrate ... app_confbridge now has the ability to set the estimated bitrate on an SFU bridge. To use it, set a bridge profile's remb_behavior to "force" and set remb_estimated_bitrate to a rate in bits per second. The remb_estimated_bitrate parameter is ignored if remb_behavior is something other than "force". Change-Id: Idce6464ff014a37ea3b82944452e56cc4d75ab0a	5 years ago
Sean Bright	4b5ed817bd	app_voicemail.c: Document VMSayName interruption behavior ... ASTERISK-26424 #close Change-Id: I797ad0ed302d0b3d2c90543eff5b7207ed08ecf0	5 years ago
Holger Hans Peter Freyther	9c0ded6e76	res_pjsip_sdp_rtp: Fix accidentally native bridging calls ... Stop advertising RFC2833 support on the rtp_engine when DTMF mode is auto but no tel_event was found inside SDP file. On an incoming call create_rtp will be called and when session->dtmf is set to AST_SIP_DTMF_AUTO, the AST_RTP_PROPERTY_DTMF will be set without looking at the SDP file. Once get_codecs gets called we move the DTMF mode from RFC2833 to INBAND but continued to advertise RFC2833 support. This meant the native_rtp bridge would falsely consider the two channels as compatible. In addition to changing the DTMF mode we now set or remove the AST_RTP_PROPERTY_DTMF. The property is checked in ast_rtp_dtmf_compatible and called by native_rtp_bridge_compatible. ASTERISK-29051 #close Change-Id: I1e0c1e324598a437932c0b7836bcb626aba8e287	5 years ago
lvl	990c72bbcf	res_musiconhold: Load all realtime entries, not just the first ... ASTERISK-29099 Change-Id: I45636679c0fb5a5f59114c8741626631a604e8a6	5 years ago
Jasper van der Neut	e831952eba	channels: Don't dereference NULL pointer ... Check result of ast_translator_build_path against NULL before dereferencing. ASTERISK-29091 Change-Id: Ia3538ea190bd371f70c9dd49984b021765691b29	5 years ago
Torrey Searle	e7bd97e2e5	res_pjsip_diversion: fix double 181 ... Arming response to both AST_SIP_SESSION_BEFORE_REDIRECTING and AST_SIP_SESSION_BEFORE_MEDIA causes 302 to to be handled twice, resulting in to 181 being generated. Change-Id: I866e5461564644ffb8a5e12b6f1330b50a7b63ab	5 years ago
Sean Bright	505211551a	res_musiconhold: Clarify that playlist mode only supports HTTP(S) URLs ... Change-Id: I41e77a04e4a523f4ed61a7a20b738ffd42be441e	5 years ago
Sean Bright	16dfe8f03f	dsp.c: Update calls to ast_format_cmp to check result properly ... ASTERISK-28311 #close Change-Id: Ib1ce8fc1a8752751f5bf3615c59245532dfd9aa2	5 years ago
Joshua C. Colp	23e427bbd2	res_pjsip_session: Fix stream name memory leak. ... When constructing a stream name based on the media type and position the allocated name was not being freed causing a leak. Change-Id: I52510863b24a2f531f0a55b440bb2c81844029de	5 years ago
Sean Bright	b11b49945b	func_curl.c: Prevent crash when using CURLOPT(httpheader) ... Because we use shared thread-local cURL instances, we need to ensure that the state of the cURL instance is correct before each invocation. In the case of custom headers, we were not resetting cURL's internal HTTP header pointer which could result in a crash if subsequent requests do not configure custom headers. ASTERISK-29085 #close Change-Id: I8b4ab34038156dfba613030a45f10e932d2e992d	5 years ago
Sean Bright	0aaf9aa6de	res_musiconhold: Start playlist after initial announcement ... Only track our sample offset if we are playing a non-announcement file, otherwise we will skip that number of samples when we start playing the first MoH file. ASTERISK-24329 #close Change-Id: Ib6b3c84fcaa1063889ab38ba7e7fc50050a3ccfc	5 years ago
Joshua C. Colp	f67f5676b7	res_pjsip_session: Fix session reference leak. ... The ast_sip_dialog_get_session function returns the session with reference count increased. This was not taken into account and was causing sessions to remain around when they should not be. ASTERISK-29089 Change-Id: I430fa721b0a824311a59effec6056e9ec528e3e8	5 years ago
Michal Hajek	b4ab0dd41a	res_stasis.c: Add compare function for bridges moh container ... Sometimes not play MOH on bridge. ASTERISK-29081 Reported-by: Michal Hajek <michal.hajek@daktela.com> Change-Id: I760c73e0c9be1d340303b5d1c18a00c4759e8232	5 years ago
George Joseph	923d95cc84	logger.h: Fix ast_trace to respect scope_level ... ast_trace() was always emitting messages when it's level was set to -1 because it was ignoring scope_level. Change-Id: I849c8f4f4613899c37f82be0202024e7d117e506	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	39bb45cdfc	bridge_softmix/sfu_topologies_on_join: Ignore topology change failures ... When a channel joins a bridge, we do topology change requests on all existing channels to add the new participant to them. However the announcer channel will return an error because it doesn't support topology in the first place. Unfortunately, there doesn't seem to be a reliable way to tell if the error is expected or not so the error is ignored for all channels. If the request fails on a "real" channel, that channel just won't get the new participant's video. Change-Id: Ic95db4683f27d224c1869fe887795d6b9fdea4f0	5 years ago
Sean Bright	bc038e6191	res_pjsip_session.c: Fix build when TEST_FRAMEWORK is not defined ... Change-Id: Id4852c26e9c412af8e37b5dd3c15da9453ad3276	5 years ago
Torrey Searle	888090ab18	res_pjsip_diversion: implement support for History-Info ... Implemention of History-Info capable of interworking with Diversion Header following RFC7544 ASTERISK-29027 #close Change-Id: I2296369582d4b295c5ea1e60bec391dd1d318fa6	5 years ago
Sean Bright	30e08ce1bb	format_cap: Perform codec lookups by pointer instead of name ... ASTERISK-28416 #close Change-Id: I069420875ebdbcaada52d92599a5f7de3cb2cdf4	5 years ago
George Joseph	53910b1f25	res_pjsip_session: Fix issue with COLP and 491 ... The recent 491 changes introduced a check to determine if the active and pending topologies were equal and to suppress the re-invite if they were. When a re-invite is sent for a COLP-only change, the pending topology is NULL so that check doesn't happen and the re-invite is correctly sent. Of course, sending the re-invite sets the pending topology. If a 491 is received, when we resend the re-invite, the pending topology is set and since we didn't request a change to the topology in the first place, pending and active topologies are equal so the topologies-equal check causes the re-invite to be erroneously suppressed. This change checks if the topologies are equal before we run the media state resolver (which recreates the pending topology) so that when we do the final topologies-equal check we know if this was a topology change request. If it wasn't a change request, we don't suppress the re-invite even though the topologies are equal. ASTERISK-29014 Change-Id: Iffd7dd0500301156a566119ebde528d1a9573314	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
George Joseph	86f1bce186	res_pjsip_session: Handle multi-stream re-invites better ... When both Asterisk and a UA send re-invites at the same time, both send 491 "Transaction in progress" responses to each other and back off a specified amount of time before retrying. When Asterisk prepares to send its re-invite, it sets up the session's pending media state with the new topology it wants, then sends the re-invite. Unfortunately, when it received the re-invite from the UA, it partially processed the media in the re-invite and reset the pending media state before sending the 491 losing the state it set in its own re-invite. Asterisk also was not tracking re-invites received while an existing re-invite was queued resulting in sending stale SDP with missing or duplicated streams, or no re-invite at all because we erroneously determined that a re-invite wasn't needed. There was also an issue in bridge_softmix where we were using a stream from the wrong topology to determine if a stream was added. This also caused us to erroneously determine that a re-invite wasn't needed. Regardless of how the delayed re-invite was triggered, we need to reconcile the topology that was active at the time the delayed request was queued, the pending topology of the queued request, and the topology currently active on the session. To do this we need a topology resolver AND we need to make stream named unique so we can accurately tell what a stream has been added or removed and if we can re-use a slot in the topology. Summary of changes: * bridge_softmix: * We no longer reset the stream name to "removed" in remove_all_original_streams(). That was causing multiple streams to have the same name and wrecked the checks for duplicate streams. * softmix_bridge_stream_sources_update() was checking the old_stream to see if it had the softmix prefix and not considering the stream as "new" if it did. If the stream in that slot has something in it because another re-invite happened, then that slot in old might have a softmix stream but the same stream in new might actually be a new one. Now we check the new_stream's name instead of the old_stream's. * stream: * Instead of using plain media type name ("audio", "video", etc) as the default stream name, we now append the stream position to it to make it unique. We need to do this so we can distinguish multiple streams of the same type from each other. * When we set a stream's state to REMOVED, we no longer reset its name to "removed" or destroy its metadata. Again, we need to do this so we can distinguish multiple streams of the same type from each other. * res_pjsip_session: * Added resolve_refresh_media_states() that takes in 3 media states and creates an up-to-date pending media state that includes the changes that might have happened while a delayed session refresh was in the delayed queue. * Added is_media_state_valid() that checks the consistency of a media state and returns a true/false value. A valid state has: * The same number of stream entries as media session entries. Some media session entries can be NULL however. * No duplicate streams. * A valid stream for each non-NULL media session. * A stream that matches each media session's stream_num and media type. * Updated handle_incoming_sdp() to set the stream name to include the stream position number in the name to make it unique. * Updated the ast_sip_session_delayed_request structure to include both the pending and active media states and updated the associated delay functions to process them. * Updated sip_session_refresh() to accept both the pending and active media states that were in effect when the request was originally queued and to pass them on should the request need to be delayed again. * Updated sip_session_refresh() to call resolve_refresh_media_states() and substitute its results for the pending state passed in. * Updated sip_session_refresh() with additional debugging. * Updated session_reinvite_on_rx_request() to simply return PJ_FALSE to pjproject if a transaction is in progress. This stops us from creating a partial pending media state that would be invalid later on. * Updated reschedule_reinvite() to clone both the current pending and active media states and pass them to delay_request() so the resolver can tell what the original intention of the re-invite was. * Added a large unit test for the resolver. ASTERISK-29014 Change-Id: Id3440972943c611a15f652c6c569fa0e4536bfcb	5 years ago
Sungtae Kim	9052e448ec	realtime: Increased reg_server character size ... Currently, the ps_contacts table's reg_server column in realtime database type is varchar(20). This is fine for normal cases, but if the hostname is longer than 20, it returns error and then failed to register the contact address of the peer. Normally, 20 characters limitation for the hostname is fine, but with the cloud env. So, increased the size to 255. ASTERISK-29056 Change-Id: Iac52c8c35030303cfa551bb39f410b33bffc507d	5 years ago
Sungtae Kim	aae0904c7d	res_stasis.c: Added video_single option for bridge creation ... Currently, it was not possible to create bridge with video_mode single. This made hard to put the bridge in a vidoe_single mode. So, added video_single option for Bridge creation using the ARI. This allows create a bridge with video_mode single. ASTERISK-29055 Change-Id: I43e720e5c83fc75fafe10fe22808ae7f055da2ae	5 years ago
Ben Ford	80a609fcce	Bridging: Use a ref to bridge_channel's channel to prevent crash. ... There's a race condition with bridging where a bridge can be torn down causing the bridge_channel's ast_channel to become NULL when it's still needed. This particular case happened with attended transfers, but the crash occurred when trying to publish a stasis message. Now, the bridge_channel is locked, a ref to the ast_channel is obtained, and that ref is passed down the chain. Change-Id: Ic48715c0c041615d17d286790ae3e8c61bb28814	5 years ago
Patrick Verzele	f8fe20eb9f	res_pjsip_session: Deferred re-INVITE without SDP send a=sendrecv instead of a=sendonly ... Building on ASTERISK-25854. When the device requests hold by sending SDP with attribute recvonly, asterisk places the session in sendonly mode. When the device later requests to resume the call by using a re-INVITE excluding SDP, asterisk needs to change the sendonly mode to sendrecv again. Change-Id: I60341ce3d87f95869f3bc6dc358bd3e8286477a6	5 years ago
Kevin Harwell	1a5597741f	conversions: Add string to signed integer conversion functions ... Change-Id: Id603b0b03b78eb84c7fca030a08b343c0d5973f9	6 years ago
Kfir Itzhak	c3a3ab8628	app_queue: Fix leave-empty not recording a call as abandoned ... This fixes a bug introduced mistakenly in ASTERISK-25665: If leave-empty is enabled, a call may sometimes be removed from a queue without recording it as abandoned. This causes Asterisk to not generate an abandon event for that call, and for the queue abandoned counter to be incorrect. ASTERISK-29043 #close Change-Id: I1a71b81df78adff59af587f1d8483cf57df430c7	6 years ago
George Joseph	5989e0de0f	ast_coredumper: Fix issues with naming ... If you run ast_coredumper --tarball-coredumps in the same directory as the actual coredump, tar can fail because the link to the actual coredump becomes recursive. The resulting tarball will have everything _except_ the coredump (which is usually what you need) There's also an issue that the directory name in the tarball is the same as the coredump so if you extract the tarball the directory it creates will overwrite the coredump. So: * Made the link to the coredump use the absolute path to the file instead of a relative one. This prevents the recursive link and allows tar to add the coredump. * The tarballed directory is now named <coredump>.output instead of just <coredump> so if you expand the tarball it won't overwrite the coredump. Change-Id: I8b3eeb26e09a577c702ff966924bb0a2f9a759ea	6 years ago
Joshua C. Colp	c4bed96742	parking: Copy parker UUID as well. ... When fixing issues uncovered by GCC10 a copy of the parker UUID was removed accidentally. This change restores it so that the subscription has the data it needs. ASTERISK-29042 Change-Id: I7d396a14ea648bd26d3c363dd78e78bd386b544a	6 years ago
Alexander Traud	f225e9bf35	sip_nat_settings: Update script for latest Linux. ... With the latest Linux, 'ifconfig' is not installed on default anymore. Furthermore, the output of the current net-tools 'ifconfig' changed. Therefore, parsing failed. This update uses 'ip addr show' instead. Finally, the service for the external IP changed. Change-Id: I9b1a7c3f457e3553b50a3e9a55524e40d70245a0	6 years ago
Alexander Traud	8907a9f0b9	samples: Fix keep_alive_interval default in pjsip.conf. ... Since ASTERISK_27978 the default is not off but 90 seconds. That change happened because ASTERISK_27347 disabled the keep-alives in the bundled PJProject and Asterisk should behave the same as before. Change-Id: Ie63dc558ade6a5a2b969c30a4bd492d63730dc46	6 years ago